From c0454d31e05062b1c7df7eef21855ba1f56c5158 Mon Sep 17 00:00:00 2001 From: Samuel Holland Date: Sat, 2 Mar 2024 15:00:35 +0100 Subject: [PATCH 001/603] ASoC: sun50i-codec-analog: Move suspend/resume to set_bias_level MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With idle_bias_on and suspend_bias_off, there are bias level transitions that match the suspend/resume callbacks. However, there are also transitions during probe (OFF => STANDBY) and removal (STANDBY => OFF). By using the set_bias_level hook, the driver can have one copy of code that would otherwise be duplicated between the probe/resume and suspend/remove hooks. Signed-off-by: Samuel Holland Signed-off-by: Ondřej Jirman Link: https://msgid.link/r/20240302140042.1990256-2-megi@xff.cz Signed-off-by: Mark Brown --- sound/soc/sunxi/sun50i-codec-analog.c | 29 +++++++++++++++++---------- 1 file changed, 18 insertions(+), 11 deletions(-) diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index 8a32d05e23e1a..cedd4de42d1a3 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -471,17 +471,23 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { { "EARPIECE", NULL, "Earpiece Amp" }, }; -static int sun50i_a64_codec_suspend(struct snd_soc_component *component) +static int sun50i_a64_codec_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) { - return regmap_update_bits(component->regmap, SUN50I_ADDA_HP_CTRL, - BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE), - BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE)); -} + switch (level) { + case SND_SOC_BIAS_OFF: + regmap_set_bits(component->regmap, SUN50I_ADDA_HP_CTRL, + BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE)); + break; + case SND_SOC_BIAS_STANDBY: + regmap_clear_bits(component->regmap, SUN50I_ADDA_HP_CTRL, + BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE)); + break; + default: + break; + } -static int sun50i_a64_codec_resume(struct snd_soc_component *component) -{ - return regmap_update_bits(component->regmap, SUN50I_ADDA_HP_CTRL, - BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE), 0); + return 0; } static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { @@ -491,8 +497,9 @@ static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { .num_dapm_widgets = ARRAY_SIZE(sun50i_a64_codec_widgets), .dapm_routes = sun50i_a64_codec_routes, .num_dapm_routes = ARRAY_SIZE(sun50i_a64_codec_routes), - .suspend = sun50i_a64_codec_suspend, - .resume = sun50i_a64_codec_resume, + .set_bias_level = sun50i_a64_codec_set_bias_level, + .idle_bias_on = true, + .suspend_bias_off = true, }; static const struct of_device_id sun50i_codec_analog_of_match[] = { From c556814b926c3f16fdca3d18cac793ccf0d14c44 Mon Sep 17 00:00:00 2001 From: Samuel Holland Date: Sat, 2 Mar 2024 15:00:36 +0100 Subject: [PATCH 002/603] ASoC: sun8i-codec: Enable bus clock at STANDBY and higher bias MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For codec variants that have a bus clock, that clock must be running to receive interrupts. Since jack and mic accessory detection should work even when no audio is playing, that means the bus clock should be enabled any time the system is on. Accomplish that by tying the bus clock to the runtime PM state, which is then tied to the bias level not being OFF. Since the codec sets idle_bias_on, bias will generally never be OFF. However, we can set suspend_bias_off to maintain the power savings of gating the bus clock during suspend, when we don't expect jack/accessory detection to work. Signed-off-by: Samuel Holland Signed-off-by: Ondřej Jirman Link: https://msgid.link/r/20240302140042.1990256-3-megi@xff.cz Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 41 ++++++++++++++++++++++++++++------- 1 file changed, 33 insertions(+), 8 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 7b45ddffe9900..2a46b96b03cce 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -177,12 +177,14 @@ struct sun8i_codec_aif { }; struct sun8i_codec_quirks { - bool legacy_widgets : 1; - bool lrck_inversion : 1; + bool bus_clock : 1; + bool legacy_widgets : 1; + bool lrck_inversion : 1; }; struct sun8i_codec { struct regmap *regmap; + struct clk *clk_bus; struct clk *clk_module; const struct sun8i_codec_quirks *quirks; struct sun8i_codec_aif aifs[SUN8I_CODEC_NAIFS]; @@ -197,6 +199,14 @@ static int sun8i_codec_runtime_resume(struct device *dev) struct sun8i_codec *scodec = dev_get_drvdata(dev); int ret; + if (scodec->clk_bus) { + ret = clk_prepare_enable(scodec->clk_bus); + if (ret) { + dev_err(dev, "Failed to enable the bus clock\n"); + return ret; + } + } + regcache_cache_only(scodec->regmap, false); ret = regcache_sync(scodec->regmap); @@ -215,6 +225,9 @@ static int sun8i_codec_runtime_suspend(struct device *dev) regcache_cache_only(scodec->regmap, true); regcache_mark_dirty(scodec->regmap); + if (scodec->clk_bus) + clk_disable_unprepare(scodec->clk_bus); + return 0; } @@ -1277,6 +1290,7 @@ static const struct snd_soc_component_driver sun8i_soc_component = { .num_dapm_routes = ARRAY_SIZE(sun8i_codec_dapm_routes), .probe = sun8i_codec_component_probe, .idle_bias_on = 1, + .suspend_bias_off = 1, .endianness = 1, }; @@ -1299,6 +1313,18 @@ static int sun8i_codec_probe(struct platform_device *pdev) if (!scodec) return -ENOMEM; + scodec->quirks = of_device_get_match_data(&pdev->dev); + + platform_set_drvdata(pdev, scodec); + + if (scodec->quirks->bus_clock) { + scodec->clk_bus = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(scodec->clk_bus)) { + dev_err(&pdev->dev, "Failed to get the bus clock\n"); + return PTR_ERR(scodec->clk_bus); + } + } + scodec->clk_module = devm_clk_get(&pdev->dev, "mod"); if (IS_ERR(scodec->clk_module)) { dev_err(&pdev->dev, "Failed to get the module clock\n"); @@ -1311,17 +1337,14 @@ static int sun8i_codec_probe(struct platform_device *pdev) return PTR_ERR(base); } - scodec->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "bus", base, - &sun8i_codec_regmap_config); + scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun8i_codec_regmap_config); if (IS_ERR(scodec->regmap)) { dev_err(&pdev->dev, "Failed to create our regmap\n"); return PTR_ERR(scodec->regmap); } - scodec->quirks = of_device_get_match_data(&pdev->dev); - - platform_set_drvdata(pdev, scodec); - + regcache_cache_only(scodec->regmap, true); pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { ret = sun8i_codec_runtime_resume(&pdev->dev); @@ -1357,11 +1380,13 @@ static void sun8i_codec_remove(struct platform_device *pdev) } static const struct sun8i_codec_quirks sun8i_a33_quirks = { + .bus_clock = true, .legacy_widgets = true, .lrck_inversion = true, }; static const struct sun8i_codec_quirks sun50i_a64_quirks = { + .bus_clock = true, }; static const struct of_device_id sun8i_codec_of_match[] = { From d5961e43b28668088087befbf4f7a043bd0ae65c Mon Sep 17 00:00:00 2001 From: Arnaud Ferraris Date: Sat, 2 Mar 2024 15:00:37 +0100 Subject: [PATCH 003/603] ASoC: sun50i-codec-analog: Enable jack detection on startup MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This commit adds the necessary setup to enable jack detection on startup as well as the callback function enabling the microphone ADC when headset bias is enabled. The microphone ADC is also disabled in suspend. Signed-off-by: Arnaud Ferraris [Samuel: Moved MICADCEN setup to HBIAS event, added bias hooks] Signed-off-by: Samuel Holland Signed-off-by: Ondřej Jirman Link: https://msgid.link/r/20240302140042.1990256-4-megi@xff.cz Signed-off-by: Mark Brown --- sound/soc/sunxi/sun50i-codec-analog.c | 32 ++++++++++++++++++++++++++- 1 file changed, 31 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index cedd4de42d1a3..2081721a8ff24 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -116,8 +116,10 @@ #define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7 #define SUN50I_ADDA_JACK_MIC_CTRL 0x1d +#define SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN 7 #define SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN 6 #define SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN 5 +#define SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN 4 /* mixer controls */ static const struct snd_kcontrol_new sun50i_a64_codec_mixer_controls[] = { @@ -296,6 +298,19 @@ static const struct snd_kcontrol_new sun50i_codec_earpiece_switch[] = { SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE, 1, 0), }; +static int sun50i_codec_hbias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + u32 value = !!SND_SOC_DAPM_EVENT_ON(event); + + regmap_update_bits(component->regmap, SUN50I_ADDA_JACK_MIC_CTRL, + BIT(SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN), + value << SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN); + + return 0; +} + static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { /* DAC */ SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, @@ -367,7 +382,8 @@ static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { /* Microphone Bias */ SND_SOC_DAPM_SUPPLY("HBIAS", SUN50I_ADDA_JACK_MIC_CTRL, SUN50I_ADDA_JACK_MIC_CTRL_HMICBIASEN, - 0, NULL, 0), + 0, sun50i_codec_hbias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), /* Mic input path */ SND_SOC_DAPM_PGA("Mic2 Amplifier", SUN50I_ADDA_MIC2_CTRL, @@ -474,14 +490,28 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { static int sun50i_a64_codec_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + int hbias; + switch (level) { case SND_SOC_BIAS_OFF: + regmap_clear_bits(component->regmap, SUN50I_ADDA_JACK_MIC_CTRL, + BIT(SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN) | + BIT(SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN)); + regmap_set_bits(component->regmap, SUN50I_ADDA_HP_CTRL, BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE)); break; case SND_SOC_BIAS_STANDBY: regmap_clear_bits(component->regmap, SUN50I_ADDA_HP_CTRL, BIT(SUN50I_ADDA_HP_CTRL_PA_CLK_GATE)); + + hbias = snd_soc_dapm_get_pin_status(dapm, "HBIAS"); + regmap_update_bits(component->regmap, SUN50I_ADDA_JACK_MIC_CTRL, + BIT(SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN) | + BIT(SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN), + BIT(SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN) | + hbias << SUN50I_ADDA_JACK_MIC_CTRL_MICADCEN); break; default: break; From 21fa98f4197bb3365dda1417708b318f403c13c1 Mon Sep 17 00:00:00 2001 From: Arnaud Ferraris Date: Sat, 2 Mar 2024 15:00:38 +0100 Subject: [PATCH 004/603] ASoC: sun8i-codec: Implement jack and accessory detection Add support for the jack detection functionality in the A64 variant, which uses a pair of IRQs; and microphone accessory (button) detection, which uses an ADC with an IRQ trigger. IRQs will only be triggered if the JACKDETEN, HMICBIASEN, and MICADCEN bits are set appropriately in the analog codec component (sun50i-codec-analog), but there is no direct software dependency between the two components. Setup ADC so that it samples with period of 16ms, disable smoothing and enable MDATA threshold (should be below idle voltage/HMIC_DATA value). Also enable HMIC_N, which makes sure we get HMIC_N samples after HMIC_DATA crosses the threshold. This allows us to perform steady state detection of HMIC_DATA, by comparing current and previous ADC samples, to detect end of the transient when the user de-presses the button. Otherwise ADC could sample anywhere within the transient, and the driver may mis-issue key-press events for other buttons attached to the resistor ladder. [Ondrej: Almost complete rewrite of the patch, change to use set_jack API. Better de-bounce, fix mic button handling, better interrupt processing.] Signed-off-by: Arnaud Ferraris [Samuel: Decouple from analog codec, fixes] Co-developed-by: Samuel Holland Signed-off-by: Samuel Holland Co-developed-by: Ondrej Jirman Signed-off-by: Ondrej Jirman Link: https://msgid.link/r/20240302140042.1990256-5-megi@xff.cz Signed-off-by: Mark Brown --- sound/soc/sunxi/sun50i-codec-analog.c | 12 + sound/soc/sunxi/sun8i-codec.c | 305 ++++++++++++++++++++++++++ 2 files changed, 317 insertions(+) diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index 2081721a8ff24..2dcdf113b66e5 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -115,6 +115,11 @@ #define SUN50I_ADDA_HS_MBIAS_CTRL 0x0e #define SUN50I_ADDA_HS_MBIAS_CTRL_MMICBIASEN 7 +#define SUN50I_ADDA_MDET_CTRL 0x1c +#define SUN50I_ADDA_MDET_CTRL_SELDETADC_FS 4 +#define SUN50I_ADDA_MDET_CTRL_SELDETADC_DB 2 +#define SUN50I_ADDA_MDET_CTRL_SELDETADC_BF 0 + #define SUN50I_ADDA_JACK_MIC_CTRL 0x1d #define SUN50I_ADDA_JACK_MIC_CTRL_JACKDETEN 7 #define SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN 6 @@ -564,6 +569,13 @@ static int sun50i_codec_analog_probe(struct platform_device *pdev) BIT(SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN), enable << SUN50I_ADDA_JACK_MIC_CTRL_INNERRESEN); + /* Select sample interval of the ADC sample to 16ms */ + regmap_update_bits(regmap, SUN50I_ADDA_MDET_CTRL, + 0x7 << SUN50I_ADDA_MDET_CTRL_SELDETADC_FS | + 0x3 << SUN50I_ADDA_MDET_CTRL_SELDETADC_BF, + 0x3 << SUN50I_ADDA_MDET_CTRL_SELDETADC_FS | + 0x3 << SUN50I_ADDA_MDET_CTRL_SELDETADC_BF); + return devm_snd_soc_register_component(&pdev->dev, &sun50i_codec_analog_cmpnt_drv, NULL, 0); diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 2a46b96b03cce..43795362fed0a 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -12,12 +12,16 @@ #include #include #include +#include #include +#include +#include #include #include #include #include +#include #include #include #include @@ -118,6 +122,23 @@ #define SUN8I_ADC_VOL_CTRL 0x104 #define SUN8I_ADC_VOL_CTRL_ADCL_VOL 8 #define SUN8I_ADC_VOL_CTRL_ADCR_VOL 0 +#define SUN8I_HMIC_CTRL1 0x110 +#define SUN8I_HMIC_CTRL1_HMIC_M 12 +#define SUN8I_HMIC_CTRL1_HMIC_N 8 +#define SUN8I_HMIC_CTRL1_MDATA_THRESHOLD_DB 5 +#define SUN8I_HMIC_CTRL1_JACK_OUT_IRQ_EN 4 +#define SUN8I_HMIC_CTRL1_JACK_IN_IRQ_EN 3 +#define SUN8I_HMIC_CTRL1_HMIC_DATA_IRQ_EN 0 +#define SUN8I_HMIC_CTRL2 0x114 +#define SUN8I_HMIC_CTRL2_HMIC_SAMPLE 14 +#define SUN8I_HMIC_CTRL2_HMIC_MDATA_THRESHOLD 8 +#define SUN8I_HMIC_CTRL2_HMIC_SF 6 +#define SUN8I_HMIC_STS 0x118 +#define SUN8I_HMIC_STS_MDATA_DISCARD 13 +#define SUN8I_HMIC_STS_HMIC_DATA 8 +#define SUN8I_HMIC_STS_JACK_OUT_IRQ_ST 4 +#define SUN8I_HMIC_STS_JACK_IN_IRQ_ST 3 +#define SUN8I_HMIC_STS_HMIC_DATA_IRQ_ST 0 #define SUN8I_DAC_DIG_CTRL 0x120 #define SUN8I_DAC_DIG_CTRL_ENDA 15 #define SUN8I_DAC_VOL_CTRL 0x124 @@ -143,6 +164,17 @@ #define SUN8I_AIF_CLK_CTRL_WORD_SIZ_MASK GENMASK(5, 4) #define SUN8I_AIF_CLK_CTRL_DATA_FMT_MASK GENMASK(3, 2) #define SUN8I_AIF3_CLK_CTRL_AIF3_CLK_SRC_MASK GENMASK(1, 0) +#define SUN8I_HMIC_CTRL1_HMIC_M_MASK GENMASK(15, 12) +#define SUN8I_HMIC_CTRL1_HMIC_N_MASK GENMASK(11, 8) +#define SUN8I_HMIC_CTRL1_MDATA_THRESHOLD_DB_MASK GENMASK(6, 5) +#define SUN8I_HMIC_CTRL2_HMIC_SAMPLE_MASK GENMASK(15, 14) +#define SUN8I_HMIC_CTRL2_HMIC_SF_MASK GENMASK(7, 6) +#define SUN8I_HMIC_STS_HMIC_DATA_MASK GENMASK(12, 8) + +#define SUN8I_CODEC_BUTTONS (SND_JACK_BTN_0|\ + SND_JACK_BTN_1|\ + SND_JACK_BTN_2|\ + SND_JACK_BTN_3) #define SUN8I_CODEC_PASSTHROUGH_SAMPLE_RATE 48000 @@ -178,16 +210,32 @@ struct sun8i_codec_aif { struct sun8i_codec_quirks { bool bus_clock : 1; + bool jack_detection : 1; bool legacy_widgets : 1; bool lrck_inversion : 1; }; +enum { + SUN8I_JACK_STATUS_DISCONNECTED, + SUN8I_JACK_STATUS_WAITING_HBIAS, + SUN8I_JACK_STATUS_CONNECTED, +}; + struct sun8i_codec { + struct snd_soc_component *component; struct regmap *regmap; struct clk *clk_bus; struct clk *clk_module; const struct sun8i_codec_quirks *quirks; struct sun8i_codec_aif aifs[SUN8I_CODEC_NAIFS]; + struct snd_soc_jack *jack; + struct delayed_work jack_work; + int jack_irq; + int jack_status; + int jack_last_sample; + ktime_t jack_hbias_ready; + struct mutex jack_mutex; + int last_hmic_irq; unsigned int sysclk_rate; int sysclk_refcnt; }; @@ -1245,6 +1293,8 @@ static int sun8i_codec_component_probe(struct snd_soc_component *component) struct sun8i_codec *scodec = snd_soc_component_get_drvdata(component); int ret; + scodec->component = component; + /* Add widgets for backward compatibility with old device trees. */ if (scodec->quirks->legacy_widgets) { ret = snd_soc_dapm_new_controls(dapm, sun8i_codec_legacy_widgets, @@ -1281,6 +1331,251 @@ static int sun8i_codec_component_probe(struct snd_soc_component *component) return 0; } +static void sun8i_codec_set_hmic_bias(struct sun8i_codec *scodec, bool enable) +{ + struct snd_soc_dapm_context *dapm = &scodec->component->card->dapm; + int irq_mask = BIT(SUN8I_HMIC_CTRL1_HMIC_DATA_IRQ_EN); + + if (enable) + snd_soc_dapm_force_enable_pin(dapm, "HBIAS"); + else + snd_soc_dapm_disable_pin(dapm, "HBIAS"); + + snd_soc_dapm_sync(dapm); + + regmap_update_bits(scodec->regmap, SUN8I_HMIC_CTRL1, + irq_mask, enable ? irq_mask : 0); +} + +static void sun8i_codec_jack_work(struct work_struct *work) +{ + struct sun8i_codec *scodec = container_of(work, struct sun8i_codec, + jack_work.work); + unsigned int mdata; + int type_mask = scodec->jack->jack->type; + int type; + + guard(mutex)(&scodec->jack_mutex); + + if (scodec->jack_status == SUN8I_JACK_STATUS_DISCONNECTED) { + if (scodec->last_hmic_irq != SUN8I_HMIC_STS_JACK_IN_IRQ_ST) + return; + + scodec->jack_last_sample = -1; + + if (type_mask & SND_JACK_MICROPHONE) { + /* + * If we were in disconnected state, we enable HBIAS and + * wait 600ms before reading initial HDATA value. + */ + scodec->jack_hbias_ready = ktime_add_ms(ktime_get(), 600); + sun8i_codec_set_hmic_bias(scodec, true); + queue_delayed_work(system_power_efficient_wq, + &scodec->jack_work, + msecs_to_jiffies(610)); + scodec->jack_status = SUN8I_JACK_STATUS_WAITING_HBIAS; + } else { + snd_soc_jack_report(scodec->jack, SND_JACK_HEADPHONE, + type_mask); + scodec->jack_status = SUN8I_JACK_STATUS_CONNECTED; + } + } else if (scodec->jack_status == SUN8I_JACK_STATUS_WAITING_HBIAS) { + /* + * If we're waiting for HBIAS to stabilize, and we get plug-out + * interrupt and nothing more for > 100ms, just cancel the + * initialization. + */ + if (scodec->last_hmic_irq == SUN8I_HMIC_STS_JACK_OUT_IRQ_ST) { + scodec->jack_status = SUN8I_JACK_STATUS_DISCONNECTED; + sun8i_codec_set_hmic_bias(scodec, false); + return; + } + + /* + * If we're not done waiting for HBIAS to stabilize, wait more. + */ + if (!ktime_after(ktime_get(), scodec->jack_hbias_ready)) { + s64 msecs = ktime_ms_delta(scodec->jack_hbias_ready, + ktime_get()); + + queue_delayed_work(system_power_efficient_wq, + &scodec->jack_work, + msecs_to_jiffies(msecs + 10)); + return; + } + + /* + * Everything is stabilized, determine jack type and report it. + */ + regmap_read(scodec->regmap, SUN8I_HMIC_STS, &mdata); + mdata &= SUN8I_HMIC_STS_HMIC_DATA_MASK; + mdata >>= SUN8I_HMIC_STS_HMIC_DATA; + + regmap_write(scodec->regmap, SUN8I_HMIC_STS, 0); + + type = mdata < 16 ? SND_JACK_HEADPHONE : SND_JACK_HEADSET; + if (type == SND_JACK_HEADPHONE) + sun8i_codec_set_hmic_bias(scodec, false); + + snd_soc_jack_report(scodec->jack, type, type_mask); + scodec->jack_status = SUN8I_JACK_STATUS_CONNECTED; + } else if (scodec->jack_status == SUN8I_JACK_STATUS_CONNECTED) { + if (scodec->last_hmic_irq != SUN8I_HMIC_STS_JACK_OUT_IRQ_ST) + return; + + scodec->jack_status = SUN8I_JACK_STATUS_DISCONNECTED; + if (type_mask & SND_JACK_MICROPHONE) + sun8i_codec_set_hmic_bias(scodec, false); + + snd_soc_jack_report(scodec->jack, 0, type_mask); + } +} + +static irqreturn_t sun8i_codec_jack_irq(int irq, void *dev_id) +{ + struct sun8i_codec *scodec = dev_id; + int type = SND_JACK_HEADSET; + unsigned int status, value; + + guard(mutex)(&scodec->jack_mutex); + + regmap_read(scodec->regmap, SUN8I_HMIC_STS, &status); + regmap_write(scodec->regmap, SUN8I_HMIC_STS, status); + + /* + * De-bounce in/out interrupts via a delayed work re-scheduling to + * 100ms after each interrupt.. + */ + if (status & BIT(SUN8I_HMIC_STS_JACK_OUT_IRQ_ST)) { + /* + * Out interrupt has priority over in interrupt so that if + * we get both, we assume the disconnected state, which is + * safer. + */ + scodec->last_hmic_irq = SUN8I_HMIC_STS_JACK_OUT_IRQ_ST; + mod_delayed_work(system_power_efficient_wq, &scodec->jack_work, + msecs_to_jiffies(100)); + } else if (status & BIT(SUN8I_HMIC_STS_JACK_IN_IRQ_ST)) { + scodec->last_hmic_irq = SUN8I_HMIC_STS_JACK_IN_IRQ_ST; + mod_delayed_work(system_power_efficient_wq, &scodec->jack_work, + msecs_to_jiffies(100)); + } else if (status & BIT(SUN8I_HMIC_STS_HMIC_DATA_IRQ_ST)) { + /* + * Ignore data interrupts until jack status turns to connected + * state, which is after HMIC enable stabilization is completed. + * Until then tha data are bogus. + */ + if (scodec->jack_status != SUN8I_JACK_STATUS_CONNECTED) + return IRQ_HANDLED; + + value = (status & SUN8I_HMIC_STS_HMIC_DATA_MASK) >> + SUN8I_HMIC_STS_HMIC_DATA; + + /* + * Assumes 60 mV per ADC LSB increment, 2V bias voltage, 2.2kOhm + * bias resistor. + */ + if (value == 0) + type |= SND_JACK_BTN_0; + else if (value == 1) + type |= SND_JACK_BTN_3; + else if (value <= 3) + type |= SND_JACK_BTN_1; + else if (value <= 8) + type |= SND_JACK_BTN_2; + + /* + * De-bounce. Only report button after two consecutive A/D + * samples are identical. + */ + if (scodec->jack_last_sample >= 0 && + scodec->jack_last_sample == value) + snd_soc_jack_report(scodec->jack, type, + scodec->jack->jack->type); + + scodec->jack_last_sample = value; + } + + return IRQ_HANDLED; +} + +static int sun8i_codec_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct sun8i_codec *scodec = snd_soc_component_get_drvdata(component); + struct platform_device *pdev = to_platform_device(component->dev); + int ret; + + if (!scodec->quirks->jack_detection) + return 0; + + scodec->jack = jack; + + scodec->jack_irq = platform_get_irq(pdev, 0); + if (scodec->jack_irq < 0) + return scodec->jack_irq; + + /* Reserved value required for jack IRQs to trigger. */ + regmap_write(scodec->regmap, SUN8I_HMIC_CTRL1, + 0xf << SUN8I_HMIC_CTRL1_HMIC_N | + 0x0 << SUN8I_HMIC_CTRL1_MDATA_THRESHOLD_DB | + 0x4 << SUN8I_HMIC_CTRL1_HMIC_M); + + /* Sample the ADC at 128 Hz; bypass smooth filter. */ + regmap_write(scodec->regmap, SUN8I_HMIC_CTRL2, + 0x0 << SUN8I_HMIC_CTRL2_HMIC_SAMPLE | + 0x17 << SUN8I_HMIC_CTRL2_HMIC_MDATA_THRESHOLD | + 0x0 << SUN8I_HMIC_CTRL2_HMIC_SF); + + /* Do not discard any MDATA, enable user written MDATA threshold. */ + regmap_write(scodec->regmap, SUN8I_HMIC_STS, 0); + + regmap_set_bits(scodec->regmap, SUN8I_HMIC_CTRL1, + BIT(SUN8I_HMIC_CTRL1_JACK_OUT_IRQ_EN) | + BIT(SUN8I_HMIC_CTRL1_JACK_IN_IRQ_EN)); + + ret = devm_request_threaded_irq(&pdev->dev, scodec->jack_irq, + NULL, sun8i_codec_jack_irq, + IRQF_ONESHOT, + dev_name(&pdev->dev), scodec); + if (ret) + return ret; + + return 0; +} + +static void sun8i_codec_disable_jack_detect(struct snd_soc_component *component) +{ + struct sun8i_codec *scodec = snd_soc_component_get_drvdata(component); + + if (!scodec->quirks->jack_detection) + return; + + devm_free_irq(component->dev, scodec->jack_irq, scodec); + + cancel_delayed_work_sync(&scodec->jack_work); + + regmap_clear_bits(scodec->regmap, SUN8I_HMIC_CTRL1, + BIT(SUN8I_HMIC_CTRL1_JACK_OUT_IRQ_EN) | + BIT(SUN8I_HMIC_CTRL1_JACK_IN_IRQ_EN) | + BIT(SUN8I_HMIC_CTRL1_HMIC_DATA_IRQ_EN)); + + scodec->jack = NULL; +} + +static int sun8i_codec_component_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + int ret = 0; + + if (jack) + ret = sun8i_codec_enable_jack_detect(component, jack, data); + else + sun8i_codec_disable_jack_detect(component); + + return ret; +} + static const struct snd_soc_component_driver sun8i_soc_component = { .controls = sun8i_codec_controls, .num_controls = ARRAY_SIZE(sun8i_codec_controls), @@ -1288,16 +1583,23 @@ static const struct snd_soc_component_driver sun8i_soc_component = { .num_dapm_widgets = ARRAY_SIZE(sun8i_codec_dapm_widgets), .dapm_routes = sun8i_codec_dapm_routes, .num_dapm_routes = ARRAY_SIZE(sun8i_codec_dapm_routes), + .set_jack = sun8i_codec_component_set_jack, .probe = sun8i_codec_component_probe, .idle_bias_on = 1, .suspend_bias_off = 1, .endianness = 1, }; +static bool sun8i_codec_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == SUN8I_HMIC_STS; +} + static const struct regmap_config sun8i_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, + .volatile_reg = sun8i_codec_volatile_reg, .max_register = SUN8I_DAC_MXR_SRC, .cache_type = REGCACHE_FLAT, @@ -1314,6 +1616,8 @@ static int sun8i_codec_probe(struct platform_device *pdev) return -ENOMEM; scodec->quirks = of_device_get_match_data(&pdev->dev); + INIT_DELAYED_WORK(&scodec->jack_work, sun8i_codec_jack_work); + mutex_init(&scodec->jack_mutex); platform_set_drvdata(pdev, scodec); @@ -1387,6 +1691,7 @@ static const struct sun8i_codec_quirks sun8i_a33_quirks = { static const struct sun8i_codec_quirks sun50i_a64_quirks = { .bus_clock = true, + .jack_detection = true, }; static const struct of_device_id sun8i_codec_of_match[] = { From 0cb3b7fd530b8c107443218ce6db5cb6e7b5dbe1 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 8 Mar 2024 10:04:58 +0100 Subject: [PATCH 005/603] ASoC: Intel: Disable route checks for Skylake boards Topology files that are propagated to the world and utilized by the skylake-driver carry shortcomings in their SectionGraphs. Since commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") route checks are no longer permissive. Probe failures for Intel boards have been partially addressed by commit a22ae72b86a4 ("ASoC: soc-core: disable route checks for legacy devices") and its follow up but only skl_nau88l25_ssm4567.c is patched. Fix the problem for the rest of the boards. Link: https://lore.kernel.org/all/20200309192744.18380-1-pierre-louis.bossart@linux.intel.com/ Fixes: daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240308090502.2136760-2-cezary.rojewski@intel.com Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 ++ sound/soc/intel/boards/kbl_da7219_max98357a.c | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 4 ++++ sound/soc/intel/boards/kbl_rt5660.c | 1 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 ++ sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 ++ sound/soc/intel/boards/skl_nau88l25_max98357a.c | 1 + sound/soc/intel/boards/skl_rt286.c | 1 + 11 files changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 540f7a29310a9..3fe3f38c6cb69 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -768,6 +768,7 @@ static struct snd_soc_card broxton_audio_card = { .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = bxt_card_late_probe, }; diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index c0eb65c14aa97..afc499be8db26 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -574,6 +574,7 @@ static struct snd_soc_card broxton_rt298 = { .dapm_routes = broxton_rt298_map, .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = bxt_card_late_probe, }; diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index 657e4658234ce..4098b2d32f9bc 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -649,6 +649,8 @@ static int geminilake_audio_probe(struct platform_device *pdev) card = &glk_audio_card_rt5682_m98357a; card->dev = &pdev->dev; snd_soc_card_set_drvdata(card, ctx); + if (!snd_soc_acpi_sof_parent(&pdev->dev)) + card->disable_route_checks = true; /* override platform name, if required */ mach = pdev->dev.platform_data; diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index a5d8965303a88..9dbc15f9d1c9b 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -639,6 +639,7 @@ static struct snd_soc_card kabylake_audio_card_da7219_m98357a = { .dapm_routes = kabylake_map, .num_dapm_routes = ARRAY_SIZE(kabylake_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 98c11ec0adc01..e662da5af83b5 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -1036,6 +1036,7 @@ static struct snd_soc_card kbl_audio_card_da7219_m98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1054,6 +1055,7 @@ static struct snd_soc_card kbl_audio_card_max98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1071,6 +1073,7 @@ static struct snd_soc_card kbl_audio_card_da7219_m98373 = { .codec_conf = max98373_codec_conf, .num_configs = ARRAY_SIZE(max98373_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -1088,6 +1091,7 @@ static struct snd_soc_card kbl_audio_card_max98373 = { .codec_conf = max98373_codec_conf, .num_configs = ARRAY_SIZE(max98373_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 30e0aca161cd5..894d127c482a3 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -518,6 +518,7 @@ static struct snd_soc_card kabylake_audio_card_rt5660 = { .dapm_routes = kabylake_rt5660_map, .num_dapm_routes = ARRAY_SIZE(kabylake_rt5660_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 9071b1f1cbd00..646e8ff8e9619 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -966,6 +966,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663_m98927 = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; @@ -982,6 +983,7 @@ static struct snd_soc_card kabylake_audio_card_rt5663 = { .dapm_routes = kabylake_5663_map, .num_dapm_routes = ARRAY_SIZE(kabylake_5663_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 178fe9c37df62..924d5d1de03ac 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -791,6 +791,7 @@ static struct snd_soc_card kabylake_audio_card = { .codec_conf = max98927_codec_conf, .num_configs = ARRAY_SIZE(max98927_codec_conf), .fully_routed = true, + .disable_route_checks = true, .late_probe = kabylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 6e172719c9795..4aa7fd2a05e46 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -227,6 +227,8 @@ static int skl_hda_audio_probe(struct platform_device *pdev) ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; hda_soc_card.dev = &pdev->dev; + if (!snd_soc_acpi_sof_parent(&pdev->dev)) + hda_soc_card.disable_route_checks = true; if (mach->mach_params.dmic_num > 0) { snprintf(hda_soc_components, sizeof(hda_soc_components), diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 0e7025834594a..e4630c33176e2 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -654,6 +654,7 @@ static struct snd_soc_card skylake_audio_card = { .dapm_routes = skylake_map, .num_dapm_routes = ARRAY_SIZE(skylake_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = skylake_card_late_probe, }; diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index c59c60e280916..9a80442749081 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -523,6 +523,7 @@ static struct snd_soc_card skylake_rt286 = { .dapm_routes = skylake_rt286_map, .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), .fully_routed = true, + .disable_route_checks = true, .late_probe = skylake_card_late_probe, }; From 6974857c2b2c7e2d2db59c0e23cc42b0efc58cd8 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 8 Mar 2024 10:04:59 +0100 Subject: [PATCH 006/603] ASoC: topology: Do not ignore route checks when parsing graphs One of the framework responsibilities is to ensure that the enumerated DPCMs are valid i.e.: a valid BE is connected to a valid FE DAI. While the are checks in soc-core.c and soc-pcm.c that verify this, a component driver may attempt to workaround this by loading an invalid graph through the topology file. Be strict and fail topology loading when invalid graph is encountered. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240308090502.2136760-3-cezary.rojewski@intel.com Reviewed-by: Pierre-Louis Bossart Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index ba4890991f0d7..fad9432a10f16 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1083,8 +1083,15 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, break; } - /* add route, but keep going if some fail */ - snd_soc_dapm_add_routes(dapm, route, 1); + ret = snd_soc_dapm_add_routes(dapm, route, 1); + if (ret) { + if (!dapm->card->disable_route_checks) { + dev_err(tplg->dev, "ASoC: dapm_add_routes failed: %d\n", ret); + break; + } + dev_info(tplg->dev, + "ASoC: disable_route_checks set, ignoring dapm_add_routes errors\n"); + } } return ret; From e6719d48ba6329536c459dcee5a571e535687094 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 8 Mar 2024 10:05:00 +0100 Subject: [PATCH 007/603] ASoC: Intel: avs: ssm4567: Do not ignore route checks A copy-paste from intel/boards/skl_nau88l25_ssm4567.c made the avs's equivalent disable route checks as well. Such behavior is not desired. Fixes: 69ea14efe99b ("ASoC: Intel: avs: Add ssm4567 machine board") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240308090502.2136760-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/ssm4567.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index 4a0e136835ff5..b64be685dc23b 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -172,7 +172,6 @@ static int avs_ssm4567_probe(struct platform_device *pdev) card->dapm_routes = card_base_routes; card->num_dapm_routes = ARRAY_SIZE(card_base_routes); card->fully_routed = true; - card->disable_route_checks = true; ret = snd_soc_fixup_dai_links_platform_name(card, pname); if (ret) From 3a2be3f05110fa1a5c682ff72a26681ed4f54cef Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 8 Mar 2024 10:05:01 +0100 Subject: [PATCH 008/603] ASoC: Intel: avs: ssm4567: Board cleanup The card-name suffix and the DP-widgets are an unintended copy-paste from skl_nau88215_ssm4567.c. Both are redundant. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240308090502.2136760-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/ssm4567.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index b64be685dc23b..abb87bb88fffe 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -37,8 +37,6 @@ static const struct snd_kcontrol_new card_controls[] = { static const struct snd_soc_dapm_widget card_widgets[] = { SND_SOC_DAPM_SPK("Left Speaker", NULL), SND_SOC_DAPM_SPK("Right Speaker", NULL), - SND_SOC_DAPM_SPK("DP1", NULL), - SND_SOC_DAPM_SPK("DP2", NULL), }; static const struct snd_soc_dapm_route card_base_routes[] = { @@ -158,7 +156,7 @@ static int avs_ssm4567_probe(struct platform_device *pdev) if (!card) return -ENOMEM; - card->name = "avs_ssm4567-adi"; + card->name = "avs_ssm4567"; card->dev = dev; card->owner = THIS_MODULE; card->dai_link = dai_link; From 60c10c678b582d41532fefa12667d8adca75811b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 8 Mar 2024 10:05:02 +0100 Subject: [PATCH 009/603] ASoC: Intel: avs: i2s_test: Remove redundant dapm routes MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Remove unnecessary widgets and routes as they are created by snd_soc_dapm_connect_dai_link_widgets() automatically. Link: https://lore.kernel.org/all/20230612110958.592674-1-brent.lu@intel.com/ Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240308090502.2136760-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/i2s_test.c | 79 --------------------------- 1 file changed, 79 deletions(-) diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index 28f254eb0d03f..282256d18cc69 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -54,76 +54,13 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_create_dapm_routes(struct device *dev, int ssp_port, int tdm_slot, - struct snd_soc_dapm_route **routes, int *num_routes) -{ - struct snd_soc_dapm_route *dr; - const int num_dr = 2; - - dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); - if (!dr) - return -ENOMEM; - - dr[0].sink = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", "pb", ssp_port, tdm_slot)); - dr[0].source = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", " Tx", ssp_port, tdm_slot)); - if (!dr[0].sink || !dr[0].source) - return -ENOMEM; - - dr[1].sink = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", " Rx", ssp_port, tdm_slot)); - dr[1].source = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", "cp", ssp_port, tdm_slot)); - if (!dr[1].sink || !dr[1].source) - return -ENOMEM; - - *routes = dr; - *num_routes = num_dr; - - return 0; -} - -static int avs_create_dapm_widgets(struct device *dev, int ssp_port, int tdm_slot, - struct snd_soc_dapm_widget **widgets, int *num_widgets) -{ - struct snd_soc_dapm_widget *dw; - const int num_dw = 2; - - dw = devm_kcalloc(dev, num_dw, sizeof(*dw), GFP_KERNEL); - if (!dw) - return -ENOMEM; - - dw[0].id = snd_soc_dapm_hp; - dw[0].reg = SND_SOC_NOPM; - dw[0].name = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", "pb", ssp_port, tdm_slot)); - if (!dw[0].name) - return -ENOMEM; - - dw[1].id = snd_soc_dapm_mic; - dw[1].reg = SND_SOC_NOPM; - dw[1].name = devm_kasprintf(dev, GFP_KERNEL, - AVS_STRING_FMT("ssp", "cp", ssp_port, tdm_slot)); - if (!dw[1].name) - return -ENOMEM; - - *widgets = dw; - *num_widgets = num_dw; - - return 0; -} - static int avs_i2s_test_probe(struct platform_device *pdev) { - struct snd_soc_dapm_widget *widgets; - struct snd_soc_dapm_route *routes; struct snd_soc_dai_link *dai_link; struct snd_soc_acpi_mach *mach; struct snd_soc_card *card; struct device *dev = &pdev->dev; const char *pname; - int num_routes, num_widgets; int ssp_port, tdm_slot, ret; mach = dev_get_platdata(dev); @@ -156,26 +93,10 @@ static int avs_i2s_test_probe(struct platform_device *pdev) return ret; } - ret = avs_create_dapm_routes(dev, ssp_port, tdm_slot, &routes, &num_routes); - if (ret) { - dev_err(dev, "Failed to create dapm routes: %d\n", ret); - return ret; - } - - ret = avs_create_dapm_widgets(dev, ssp_port, tdm_slot, &widgets, &num_widgets); - if (ret) { - dev_err(dev, "Failed to create dapm widgets: %d\n", ret); - return ret; - } - card->dev = dev; card->owner = THIS_MODULE; card->dai_link = dai_link; card->num_links = 1; - card->dapm_routes = routes; - card->num_dapm_routes = num_routes; - card->dapm_widgets = widgets; - card->num_dapm_widgets = num_widgets; card->fully_routed = true; ret = snd_soc_fixup_dai_links_platform_name(card, pname); From 41f96cd53f2838ac4894491ac5eadb06f1e5b858 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 11 Mar 2024 20:13:45 +0900 Subject: [PATCH 010/603] ASoC: fsl: imx-pcm-rpmsg: Register component with rpmsg channel name Machine driver uses rpmsg channel name to link this platform component. However if the component is re-registerd card will not find this new created component in snd_soc_try_rebind_card(). Explicitly register this component with rpmsg channel name so that card can always find this component. Signed-off-by: Chancel Liu Link: https://msgid.link/r/20240311111349.723256-2-chancel.liu@nxp.com Acked-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-rpmsg.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index fb9244c1e9c5f..b84d1dfddba2f 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -732,9 +732,6 @@ static int imx_rpmsg_pcm_probe(struct platform_device *pdev) goto fail; } - /* platform component name is used by machine driver to link with */ - component->name = info->rpdev->id.name; - #ifdef CONFIG_DEBUG_FS component->debugfs_prefix = "rpmsg"; #endif @@ -822,9 +819,17 @@ static const struct dev_pm_ops imx_rpmsg_pcm_pm_ops = { imx_rpmsg_pcm_resume) }; +static const struct platform_device_id imx_rpmsg_pcm_id_table[] = { + { .name = "rpmsg-audio-channel" }, + { .name = "rpmsg-micfil-channel" }, + { }, +}; +MODULE_DEVICE_TABLE(platform, imx_rpmsg_pcm_id_table); + static struct platform_driver imx_pcm_rpmsg_driver = { .probe = imx_rpmsg_pcm_probe, .remove_new = imx_rpmsg_pcm_remove, + .id_table = imx_rpmsg_pcm_id_table, .driver = { .name = IMX_PCM_DRV_NAME, .pm = &imx_rpmsg_pcm_pm_ops, From dacc7459745df168152b5cceba34efade9b5e063 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 11 Mar 2024 20:13:46 +0900 Subject: [PATCH 011/603] ASoC: fsl: imx-audio-rpmsg: Register device with rpmsg channel name This rpmsg driver registers device for ASoC platform driver. To align with platform driver use rpmsg channel name to create device. Signed-off-by: Chancel Liu Link: https://msgid.link/r/20240311111349.723256-3-chancel.liu@nxp.com Acked-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audio-rpmsg.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-audio-rpmsg.c b/sound/soc/fsl/imx-audio-rpmsg.c index 289e47c03d402..40820d5ad92d8 100644 --- a/sound/soc/fsl/imx-audio-rpmsg.c +++ b/sound/soc/fsl/imx-audio-rpmsg.c @@ -87,8 +87,8 @@ static int imx_audio_rpmsg_probe(struct rpmsg_device *rpdev) /* Register platform driver for rpmsg routine */ data->rpmsg_pdev = platform_device_register_data(&rpdev->dev, - IMX_PCM_DRV_NAME, - PLATFORM_DEVID_AUTO, + rpdev->id.name, + PLATFORM_DEVID_NONE, NULL, 0); if (IS_ERR(data->rpmsg_pdev)) { dev_err(&rpdev->dev, "failed to register rpmsg platform.\n"); From c73524768e9e1a7ac9eb3a4d36a1ac0d34f22644 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 11 Mar 2024 20:13:47 +0900 Subject: [PATCH 012/603] ASoC: fsl: Let imx-audio-rpmsg register platform device for card Let imx-audio-rpmsg register platform device for card. So that card register and unregister can be controlled by rpmsg driver's register and unregister. Signed-off-by: Chancel Liu Link: https://msgid.link/r/20240311111349.723256-4-chancel.liu@nxp.com Acked-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_rpmsg.c | 11 ----------- sound/soc/fsl/imx-audio-rpmsg.c | 17 ++++++++++++++++- 2 files changed, 16 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index 00852f174a69c..53bd517e59d6b 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -240,17 +240,6 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; - rpmsg->card_pdev = platform_device_register_data(&pdev->dev, - "imx-audio-rpmsg", - PLATFORM_DEVID_AUTO, - NULL, - 0); - if (IS_ERR(rpmsg->card_pdev)) { - dev_err(&pdev->dev, "failed to register rpmsg card\n"); - ret = PTR_ERR(rpmsg->card_pdev); - goto err_pm_disable; - } - return 0; err_pm_disable: diff --git a/sound/soc/fsl/imx-audio-rpmsg.c b/sound/soc/fsl/imx-audio-rpmsg.c index 40820d5ad92d8..38aafb8954c76 100644 --- a/sound/soc/fsl/imx-audio-rpmsg.c +++ b/sound/soc/fsl/imx-audio-rpmsg.c @@ -12,6 +12,7 @@ */ struct imx_audio_rpmsg { struct platform_device *rpmsg_pdev; + struct platform_device *card_pdev; }; static int imx_audio_rpmsg_cb(struct rpmsg_device *rpdev, void *data, int len, @@ -95,6 +96,16 @@ static int imx_audio_rpmsg_probe(struct rpmsg_device *rpdev) ret = PTR_ERR(data->rpmsg_pdev); } + data->card_pdev = platform_device_register_data(&rpdev->dev, + "imx-audio-rpmsg", + PLATFORM_DEVID_AUTO, + rpdev->id.name, + strlen(rpdev->id.name) + 1); + if (IS_ERR(data->card_pdev)) { + dev_err(&rpdev->dev, "failed to register rpmsg card.\n"); + ret = PTR_ERR(data->card_pdev); + } + return ret; } @@ -105,6 +116,9 @@ static void imx_audio_rpmsg_remove(struct rpmsg_device *rpdev) if (data->rpmsg_pdev) platform_device_unregister(data->rpmsg_pdev); + if (data->card_pdev) + platform_device_unregister(data->card_pdev); + dev_info(&rpdev->dev, "audio rpmsg driver is removed\n"); } @@ -113,6 +127,7 @@ static struct rpmsg_device_id imx_audio_rpmsg_id_table[] = { { .name = "rpmsg-micfil-channel" }, { }, }; +MODULE_DEVICE_TABLE(rpmsg, imx_audio_rpmsg_id_table); static struct rpmsg_driver imx_audio_rpmsg_driver = { .drv.name = "imx_audio_rpmsg", @@ -126,5 +141,5 @@ module_rpmsg_driver(imx_audio_rpmsg_driver); MODULE_DESCRIPTION("Freescale SoC Audio RPMSG interface"); MODULE_AUTHOR("Shengjiu Wang "); -MODULE_ALIAS("platform:imx_audio_rpmsg"); +MODULE_ALIAS("rpmsg:imx_audio_rpmsg"); MODULE_LICENSE("GPL v2"); From 0aa7f5406afa828a93d84d75c9b9ac991cd75367 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 11 Mar 2024 20:13:48 +0900 Subject: [PATCH 013/603] ASoC: fsl: fsl_rpmsg: Register CPU DAI with name of rpmsg channel Each rpmsg sound card sits on one rpmsg channel. Register CPU DAI with name of rpmsg channel so that ASoC machine driver can easily link CPU DAI with rpmsg channel name. Signed-off-by: Chancel Liu Link: https://msgid.link/r/20240311111349.723256-5-chancel.liu@nxp.com Acked-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_rpmsg.c | 32 ++++++++++++++++++++++++++------ 1 file changed, 26 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index 53bd517e59d6b..bc41a06668566 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -135,7 +135,6 @@ static struct snd_soc_dai_driver fsl_rpmsg_dai = { static const struct snd_soc_component_driver fsl_component = { .name = "fsl-rpmsg", - .legacy_dai_naming = 1, }; static const struct fsl_rpmsg_soc_data imx7ulp_data = { @@ -190,19 +189,40 @@ MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids); static int fsl_rpmsg_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct snd_soc_dai_driver *dai_drv; + const char *dai_name; struct fsl_rpmsg *rpmsg; int ret; + dai_drv = devm_kzalloc(&pdev->dev, sizeof(struct snd_soc_dai_driver), GFP_KERNEL); + if (!dai_drv) + return -ENOMEM; + memcpy(dai_drv, &fsl_rpmsg_dai, sizeof(fsl_rpmsg_dai)); + rpmsg = devm_kzalloc(&pdev->dev, sizeof(struct fsl_rpmsg), GFP_KERNEL); if (!rpmsg) return -ENOMEM; rpmsg->soc_data = of_device_get_match_data(&pdev->dev); - fsl_rpmsg_dai.playback.rates = rpmsg->soc_data->rates; - fsl_rpmsg_dai.capture.rates = rpmsg->soc_data->rates; - fsl_rpmsg_dai.playback.formats = rpmsg->soc_data->formats; - fsl_rpmsg_dai.capture.formats = rpmsg->soc_data->formats; + if (rpmsg->soc_data) { + dai_drv->playback.rates = rpmsg->soc_data->rates; + dai_drv->capture.rates = rpmsg->soc_data->rates; + dai_drv->playback.formats = rpmsg->soc_data->formats; + dai_drv->capture.formats = rpmsg->soc_data->formats; + } + + /* Use rpmsg channel name as cpu dai name */ + ret = of_property_read_string(np, "fsl,rpmsg-channel-name", &dai_name); + if (ret) { + if (ret == -EINVAL) { + dai_name = "rpmsg-audio-channel"; + } else { + dev_err(&pdev->dev, "Failed to get rpmsg channel name: %d!\n", ret); + return ret; + } + } + dai_drv->name = dai_name; if (of_property_read_bool(np, "fsl,enable-lpa")) { rpmsg->enable_lpa = 1; @@ -236,7 +256,7 @@ static int fsl_rpmsg_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, - &fsl_rpmsg_dai, 1); + dai_drv, 1); if (ret) goto err_pm_disable; From c14445bdcb98feddf9afaeb05e6193cc1f8eec1a Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 11 Mar 2024 20:13:49 +0900 Subject: [PATCH 014/603] ASoC: fsl: imx-rpmsg: Update to correct DT node In order to support register and unregister rpmsg sound card through remoteproc platform device for card to probe is registered in imx-audio-rpmsg. ASoC machine driver no longer can get DT node of ASoC CPU DAI device through parent device. ASoC machine driver can get DT node of ASoC CPU DAI device with rpmsg channel name acquired from platform specific data. Signed-off-by: Chancel Liu Link: https://msgid.link/r/20240311111349.723256-6-chancel.liu@nxp.com Acked-by: Shengjiu Wang Signed-off-by: Mark Brown --- sound/soc/fsl/imx-rpmsg.c | 28 +++++++++++++++++++--------- 1 file changed, 19 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index e5bd63dab10cf..0f1ad7ad7d270 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -108,10 +108,8 @@ static int imx_rpmsg_late_probe(struct snd_soc_card *card) static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; - struct device *dev = pdev->dev.parent; - /* rpmsg_pdev is the platform device for the rpmsg node that probed us */ - struct platform_device *rpmsg_pdev = to_platform_device(dev); - struct device_node *np = rpmsg_pdev->dev.of_node; + struct snd_soc_dai *cpu_dai; + struct device_node *np = NULL; struct of_phandle_args args; const char *platform_name; struct imx_rpmsg *data; @@ -127,10 +125,6 @@ static int imx_rpmsg_probe(struct platform_device *pdev) goto fail; } - ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0); - if (ret) - dev_warn(&pdev->dev, "no reserved DMA memory\n"); - data->dai.cpus = &dlc[0]; data->dai.num_cpus = 1; data->dai.platforms = &dlc[1]; @@ -152,6 +146,23 @@ static int imx_rpmsg_probe(struct platform_device *pdev) */ data->dai.ignore_pmdown_time = 1; + data->dai.cpus->dai_name = pdev->dev.platform_data; + cpu_dai = snd_soc_find_dai(data->dai.cpus); + if (!cpu_dai) { + ret = -EPROBE_DEFER; + goto fail; + } + np = cpu_dai->dev->of_node; + if (!np) { + dev_err(&pdev->dev, "failed to parse CPU DAI device node\n"); + ret = -ENODEV; + goto fail; + } + + ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0); + if (ret) + dev_warn(&pdev->dev, "no reserved DMA memory\n"); + /* Optional codec node */ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args); if (ret) { @@ -170,7 +181,6 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->sysclk = clk_get_rate(clk); } - data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); if (!of_property_read_string(np, "fsl,rpmsg-channel-name", &platform_name)) data->dai.platforms->name = platform_name; else From 39938bb1bb23fd70f1c75ce9f52d92185403b89a Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Wed, 20 Mar 2024 16:30:11 +0800 Subject: [PATCH 015/603] ASoC: codecs: ES8326: Reducing power consumption For lower power consumption during hibernation, the configuration of es8326_suspend and es8326_remove will be adjusted. Adding es8326_i2c_shutdown and es8326_i2c_remove to cover different situations Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240320083012.4282-2-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 33 ++++++++++++++++++++++++++++++++- 1 file changed, 32 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 15289dadafea0..635320c86c004 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -1072,12 +1072,13 @@ static int es8326_suspend(struct snd_soc_component *component) es8326->calibrated = false; regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); regcache_cache_only(es8326->regmap, true); - regcache_mark_dirty(es8326->regmap); /* reset register value to default */ regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); usleep_range(1000, 3000); regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); + + regcache_mark_dirty(es8326->regmap); return 0; } @@ -1163,8 +1164,13 @@ static int es8326_set_jack(struct snd_soc_component *component, static void es8326_remove(struct snd_soc_component *component) { + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + es8326_disable_jack_detect(component); es8326_set_bias_level(component, SND_SOC_BIAS_OFF); + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); + usleep_range(1000, 3000); + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); } static const struct snd_soc_component_driver soc_component_dev_es8326 = { @@ -1236,6 +1242,29 @@ static int es8326_i2c_probe(struct i2c_client *i2c) &es8326_dai, 1); } + +static void es8326_i2c_shutdown(struct i2c_client *i2c) +{ + struct snd_soc_component *component; + struct es8326_priv *es8326; + + es8326 = i2c_get_clientdata(i2c); + component = es8326->component; + dev_dbg(component->dev, "Enter into %s\n", __func__); + cancel_delayed_work_sync(&es8326->jack_detect_work); + cancel_delayed_work_sync(&es8326->button_press_work); + + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); + usleep_range(1000, 3000); + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); + +} + +static void es8326_i2c_remove(struct i2c_client *i2c) +{ + es8326_i2c_shutdown(i2c); +} + static const struct i2c_device_id es8326_i2c_id[] = { {"es8326", 0 }, {} @@ -1265,6 +1294,8 @@ static struct i2c_driver es8326_i2c_driver = { .of_match_table = of_match_ptr(es8326_of_match), }, .probe = es8326_i2c_probe, + .shutdown = es8326_i2c_shutdown, + .remove = es8326_i2c_remove, .id_table = es8326_i2c_id, }; module_i2c_driver(es8326_i2c_driver); From e6913c6ef83c80aa7569c9e08204542222fbf542 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Wed, 20 Mar 2024 16:30:12 +0800 Subject: [PATCH 016/603] ASoC: codecs: ES8326: Delete unused REG_SUPPLY REG_SUPPLY mutes the DAC when switching between HDMI and speaker, so remove it to fix the mute issues Signed-off-by: Zhang Yi Link: https://msgid.link/r/20240320083012.4282-3-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 635320c86c004..e8526844337d0 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -292,11 +292,6 @@ static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, - 4, 7, 0, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, - 0, 7, 0, 0), - SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), }; @@ -316,9 +311,6 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = { {"LHPMIX", NULL, "Left DAC"}, {"RHPMIX", NULL, "Right DAC"}, - {"HPOR", NULL, "HPOR Supply"}, - {"HPOL", NULL, "HPOL Supply"}, - {"HPOL", NULL, "LHPMIX"}, {"HPOR", NULL, "RHPMIX"}, }; From cee28113db17f0de58df0eaea4e2756c404ee01f Mon Sep 17 00:00:00 2001 From: Tomi Valkeinen Date: Tue, 19 Mar 2024 10:22:36 +0200 Subject: [PATCH 017/603] ASoC: dmaengine_pcm: Allow passing component name via config At the moment we cannot instantiate two dmaengine_pcms with the same parent device, as the components will be named the same, leading to conflicts. Add 'name' field to the snd_dmaengine_pcm_config, and use that (if defined) as the component name instead of deriving the component name from the device. Signed-off-by: Tomi Valkeinen Link: https://msgid.link/r/20240319-xilinx-dp-audio-v2-1-92d6d3a7ca7e@ideasonboard.com Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 2 ++ sound/soc/soc-core.c | 8 +++++--- sound/soc/soc-generic-dmaengine-pcm.c | 3 +++ 3 files changed, 10 insertions(+), 3 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index d70c55f17df7c..c11aaf8079fbb 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -118,6 +118,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams( * which do not use devicetree. * @process: Callback used to apply processing on samples transferred from/to * user space. + * @name: Component name. If null, dev_name will be used. * @compat_filter_fn: Will be used as the filter function when requesting a * channel for platforms which do not use devicetree. The filter parameter * will be the DAI's DMA data. @@ -143,6 +144,7 @@ struct snd_dmaengine_pcm_config { int (*process)(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, unsigned long bytes); + const char *name; dma_filter_fn compat_filter_fn; struct device *dma_dev; const char *chan_names[SNDRV_PCM_STREAM_LAST + 1]; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2ec13d1634b63..3ab6626ad6802 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2796,10 +2796,12 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); if (!component->name) { - dev_err(dev, "ASoC: Failed to allocate name\n"); - return -ENOMEM; + component->name = fmt_single_name(dev, &component->id); + if (!component->name) { + dev_err(dev, "ASoC: Failed to allocate name\n"); + return -ENOMEM; + } } component->dev = dev; diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 092ca09f36319..83db1a83d8ba8 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -441,6 +441,9 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; + if (config->name) + pcm->component.name = config->name; + ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) goto err_free_dma; From 33a499668eb6781a4518e4facf2d85cc8cd1a650 Mon Sep 17 00:00:00 2001 From: Frank Li Date: Thu, 21 Mar 2024 11:04:00 -0400 Subject: [PATCH 018/603] ASoC: dt-bindings: fsl-audmix: Convert fsl,audmix.txt to yaml Convert fsl,audmix.txt to yaml to let dtb check tools check audmix part. Signed-off-by: Frank Li Reviewed-by: Rob Herring Link: https://msgid.link/r/20240321150401.2464783-1-Frank.Li@nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,audmix.txt | 50 ----------- .../devicetree/bindings/sound/fsl,audmix.yaml | 83 +++++++++++++++++++ 2 files changed, 83 insertions(+), 50 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.txt b/Documentation/devicetree/bindings/sound/fsl,audmix.txt deleted file mode 100644 index 840b7e0d6a631..0000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,audmix.txt +++ /dev/null @@ -1,50 +0,0 @@ -NXP Audio Mixer (AUDMIX). - -The Audio Mixer is a on-chip functional module that allows mixing of two -audio streams into a single audio stream. Audio Mixer has two input serial -audio interfaces. These are driven by two Synchronous Audio interface -modules (SAI). Each input serial interface carries 8 audio channels in its -frame in TDM manner. Mixer mixes audio samples of corresponding channels -from two interfaces into a single sample. Before mixing, audio samples of -two inputs can be attenuated based on configuration. The output of the -Audio Mixer is also a serial audio interface. Like input interfaces it has -the same TDM frame format. This output is used to drive the serial DAC TDM -interface of audio codec and also sent to the external pins along with the -receive path of normal audio SAI module for readback by the CPU. - -The output of Audio Mixer can be selected from any of the three streams - - serial audio input 1 - - serial audio input 2 - - mixed audio - -Mixing operation is independent of audio sample rate but the two audio -input streams must have same audio sample rate with same number of channels -in TDM frame to be eligible for mixing. - -Device driver required properties: -================================= - - compatible : Compatible list, contains "fsl,imx8qm-audmix" - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "ipg" for register access. - - - power-domains : Must contain the phandle to AUDMIX power domain node - - - dais : Must contain a list of phandles to AUDMIX connected - DAIs. The current implementation requires two phandles - to SAI interfaces to be provided, the first SAI in the - list being used to route the AUDMIX output. - -Device driver configuration example: -====================================== - audmix: audmix@59840000 { - compatible = "fsl,imx8qm-audmix"; - reg = <0x0 0x59840000 0x0 0x10000>; - clocks = <&clk IMX8QXP_AUD_AUDMIX_IPG>; - clock-names = "ipg"; - power-domains = <&pd_audmix>; - dais = <&sai4>, <&sai5>; - }; diff --git a/Documentation/devicetree/bindings/sound/fsl,audmix.yaml b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml new file mode 100644 index 0000000000000..9413b901cf778 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,audmix.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,audmix.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Mixer (AUDMIX). + +maintainers: + - Shengjiu Wang + - Frank Li + +description: | + The Audio Mixer is a on-chip functional module that allows mixing of two + audio streams into a single audio stream. Audio Mixer has two input serial + audio interfaces. These are driven by two Synchronous Audio interface + modules (SAI). Each input serial interface carries 8 audio channels in its + frame in TDM manner. Mixer mixes audio samples of corresponding channels + from two interfaces into a single sample. Before mixing, audio samples of + two inputs can be attenuated based on configuration. The output of the + Audio Mixer is also a serial audio interface. Like input interfaces it has + the same TDM frame format. This output is used to drive the serial DAC TDM + interface of audio codec and also sent to the external pins along with the + receive path of normal audio SAI module for readback by the CPU. + + The output of Audio Mixer can be selected from any of the three streams + - serial audio input 1 + - serial audio input 2 + - mixed audio + + Mixing operation is independent of audio sample rate but the two audio + input streams must have same audio sample rate with same number of channels + in TDM frame to be eligible for mixing. + +properties: + compatible: + const: fsl,imx8qm-audmix + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: ipg + + power-domains: + maxItems: 1 + + dais: + description: contain a list of phandles to AUDMIX connected DAIs. + $ref: /schemas/types.yaml#/definitions/phandle-array + minItems: 2 + items: + - description: the AUDMIX output + maxItems: 1 + - description: serial audio input 1 + maxItems: 1 + - description: serial audio input 2 + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + - power-domains + - dais + +unevaluatedProperties: false + +examples: + - | + audmix@59840000 { + compatible = "fsl,imx8qm-audmix"; + reg = <0x59840000 0x10000>; + clocks = <&amix_lpcg 0>; + clock-names = "ipg"; + power-domains = <&pd_audmix>; + dais = <&sai4>, <&sai5>; + }; From a806793f231e390510343837668d8f0eade75a72 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 22 Mar 2024 16:50:15 +0530 Subject: [PATCH 019/603] ASoC: amd: simplify soundwire dependencies for legacy stack The CONFIG_SOUNDWIRE_AMD is a user-visible option, it should be never selected by another driver. So replace the extra complexity with a normal Kconfig dependency in SND_SOC_AMD_SOUNDWIRE. Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240322112018.3063344-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index fa74635cee08e..3508f5a96b751 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -134,15 +134,14 @@ config SND_SOC_AMD_RPL_ACP6x config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE tristate - select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n select SND_AMD_SOUNDWIRE_ACPI if ACPI config SND_SOC_AMD_SOUNDWIRE tristate "Support for SoundWire based AMD platforms" default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE - depends on ACPI && SOUNDWIRE - depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y) + depends on ACPI + depends on SOUNDWIRE_AMD help This adds support for SoundWire for AMD platforms. Say Y if you want to enable SoundWire links with SOF. From 1fdc23aa57913d7a0a521f71dcdefdde9364d4f9 Mon Sep 17 00:00:00 2001 From: Kartik Agarwala Date: Mon, 18 Mar 2024 01:32:02 +0530 Subject: [PATCH 020/603] ASoC: dt-bindings: wm8776: Convert to dtschema Convert WM8776 audio CODEC bindings from text to dtschema. Signed-off-by: Kartik Agarwala Reviewed-by: Charles Keepax Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240317200201.119233-1-agarwala.kartik@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/wlf,wm8776.yaml | 41 +++++++++++++++++++ .../devicetree/bindings/sound/wm8776.txt | 18 -------- 2 files changed, 41 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8776.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8776.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml new file mode 100644 index 0000000000000..7bbc96ee81be3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8776.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8776.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8776 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8776 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8776"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8776.txt b/Documentation/devicetree/bindings/sound/wm8776.txt deleted file mode 100644 index 01173369c3ed3..0000000000000 --- a/Documentation/devicetree/bindings/sound/wm8776.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8776 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8776" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8776: codec@1a { - compatible = "wlf,wm8776"; - reg = <0x1a>; -}; From 3ca49e7f9475ac06614edf1bfece123eafbdf8f3 Mon Sep 17 00:00:00 2001 From: Alexander Stein Date: Thu, 14 Mar 2024 15:16:42 +0100 Subject: [PATCH 021/603] ASoC: fsl: fsl_ssi: Add dev_err_probe if PCM DMA init fails This happens especially if this driver is built-in, but SDMA driver is configured as module. Signed-off-by: Alexander Stein Link: https://msgid.link/r/20240314141642.2943605-1-alexander.stein@ew.tq-group.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ab6ec19748072..4ca3a16f7ac0d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1401,8 +1401,10 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, goto error_pcm; } else { ret = imx_pcm_dma_init(pdev); - if (ret) + if (ret) { + dev_err_probe(dev, ret, "Failed to init PCM DMA\n"); goto error_pcm; + } } return 0; From 9855f05e553637f05494cf47a3154cbf9a5cfc67 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 18 Mar 2024 22:07:56 +0200 Subject: [PATCH 022/603] ASoC: fsl: imx-es8328: Switch to using gpiod API This updates the driver to gpiod API, and removes yet another use of of_get_named_gpio(). Signed-off-by: Andy Shevchenko Link: https://msgid.link/r/20240318200804.181516-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-es8328.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 6f0d031c1d5fd..d67b8a149bfff 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -3,7 +3,7 @@ // Copyright 2012 Freescale Semiconductor, Inc. // Copyright 2012 Linaro Ltd. -#include +#include #include #include #include @@ -23,7 +23,7 @@ struct imx_es8328_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - int jack_gpio; + struct gpio_desc *jack_gpiod; }; static struct snd_soc_jack_gpio headset_jack_gpios[] = { @@ -54,8 +54,8 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) struct imx_es8328_data, card); int ret = 0; - /* Headphone jack detection */ - if (gpio_is_valid(data->jack_gpio)) { + if (data->jack_gpiod) { + /* Headphone jack detection */ ret = snd_soc_card_jack_new_pins(rtd->card, "Headphone", SND_JACK_HEADSET | SND_JACK_BTN_0, &headset_jack, @@ -64,7 +64,7 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - headset_jack_gpios[0].gpio = data->jack_gpio; + headset_jack_gpios[0].desc = data->jack_gpiod; ret = snd_soc_jack_add_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), headset_jack_gpios); @@ -174,7 +174,11 @@ static int imx_es8328_probe(struct platform_device *pdev) data->dev = dev; - data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + data->jack_gpiod = devm_gpiod_get_optional(dev, "jack", GPIOD_IN); + if (IS_ERR(data->jack_gpiod)) { + ret = PTR_ERR(data->jack_gpiod); + goto put_device; + } /* * CPU == Platform From 7aaaa22de56ce0dae15fd9f42a69a1d1a7a6e078 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Wed, 13 Mar 2024 12:01:26 +0100 Subject: [PATCH 023/603] ASoC: mediatek: mt8192-afe-pcm: Convert to devm_pm_runtime_enable() Switch from pm_runtime_enable() to devm_pm_runtime_enable(), allowing to remove all gotos from the probe function. Signed-off-by: AngeloGioacchino Del Regno Reviewed-by: Chen-Yu Tsai Link: https://msgid.link/r/20240313110147.1267793-2-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 46 ++++++++-------------- 1 file changed, 17 insertions(+), 29 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index bdd1e91824d9c..794419d16b01b 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -2217,21 +2217,20 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) return ret; } - pm_runtime_enable(&pdev->dev); - if (!pm_runtime_enabled(&pdev->dev)) - goto err_pm_disable; + ret = devm_pm_runtime_enable(&pdev->dev); + if (ret) + return ret; /* regmap init */ afe->regmap = syscon_node_to_regmap(dev->parent->of_node); if (IS_ERR(afe->regmap)) { dev_err(dev, "could not get regmap from parent\n"); - ret = PTR_ERR(afe->regmap); - goto err_pm_disable; + return PTR_ERR(afe->regmap); } ret = regmap_attach_dev(dev, afe->regmap, &mt8192_afe_regmap_config); if (ret) { dev_warn(dev, "regmap_attach_dev fail, ret %d\n", ret); - goto err_pm_disable; + return ret; } /* enable clock for regcache get default value from hw */ @@ -2241,7 +2240,7 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) ret = regmap_reinit_cache(afe->regmap, &mt8192_afe_regmap_config); if (ret) { dev_err(dev, "regmap_reinit_cache fail, ret %d\n", ret); - goto err_pm_disable; + return ret; } pm_runtime_put_sync(&pdev->dev); @@ -2254,10 +2253,8 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) afe->memif_size = MT8192_MEMIF_NUM; afe->memif = devm_kcalloc(dev, afe->memif_size, sizeof(*afe->memif), GFP_KERNEL); - if (!afe->memif) { - ret = -ENOMEM; - goto err_pm_disable; - } + if (!afe->memif) + return -ENOMEM; for (i = 0; i < afe->memif_size; i++) { afe->memif[i].data = &memif_data[i]; @@ -2271,26 +2268,22 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) afe->irqs_size = MT8192_IRQ_NUM; afe->irqs = devm_kcalloc(dev, afe->irqs_size, sizeof(*afe->irqs), GFP_KERNEL); - if (!afe->irqs) { - ret = -ENOMEM; - goto err_pm_disable; - } + if (!afe->irqs) + return -ENOMEM; for (i = 0; i < afe->irqs_size; i++) afe->irqs[i].irq_data = &irq_data[i]; /* request irq */ irq_id = platform_get_irq(pdev, 0); - if (irq_id < 0) { - ret = irq_id; - goto err_pm_disable; - } + if (irq_id < 0) + return irq_id; ret = devm_request_irq(dev, irq_id, mt8192_afe_irq_handler, IRQF_TRIGGER_NONE, "asys-isr", (void *)afe); if (ret) { dev_err(dev, "could not request_irq for Afe_ISR_Handle\n"); - goto err_pm_disable; + return ret; } /* init sub_dais */ @@ -2301,7 +2294,7 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) if (ret) { dev_warn(afe->dev, "dai register i %d fail, ret %d\n", i, ret); - goto err_pm_disable; + return ret; } } @@ -2310,7 +2303,7 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) if (ret) { dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", ret); - goto err_pm_disable; + return ret; } /* others */ @@ -2329,7 +2322,7 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) &mt8192_afe_component, NULL, 0); if (ret) { dev_warn(dev, "err_platform\n"); - goto err_pm_disable; + return ret; } ret = devm_snd_soc_register_component(&pdev->dev, @@ -2338,15 +2331,10 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) afe->num_dai_drivers); if (ret) { dev_warn(dev, "err_dai_component\n"); - goto err_pm_disable; + return ret; } return 0; - -err_pm_disable: - pm_runtime_disable(&pdev->dev); - - return ret; } static void mt8192_afe_pcm_dev_remove(struct platform_device *pdev) From 324c603a4efca7d1045e0bf3477ca54970eac72c Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Wed, 13 Mar 2024 12:01:27 +0100 Subject: [PATCH 024/603] ASoC: mediatek: mt8192-afe-pcm: Simplify with dev_err_probe() Simplify the probe function by switching error prints to return dev_err_probe(), lowering the lines count; while at it, also beautify some messages and change some others' level from warn to error. Signed-off-by: AngeloGioacchino Del Regno Reviewed-by: Chen-Yu Tsai Link: https://msgid.link/r/20240313110147.1267793-3-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 65 ++++++++-------------- 1 file changed, 22 insertions(+), 43 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index 794419d16b01b..aed22baef9fb3 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -2205,17 +2205,12 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) /* reset controller to reset audio regs before regmap cache */ rstc = devm_reset_control_get_exclusive(dev, "audiosys"); - if (IS_ERR(rstc)) { - ret = PTR_ERR(rstc); - dev_err(dev, "could not get audiosys reset:%d\n", ret); - return ret; - } + if (IS_ERR(rstc)) + return dev_err_probe(dev, PTR_ERR(rstc), "could not get audiosys reset\n"); ret = reset_control_reset(rstc); - if (ret) { - dev_err(dev, "failed to trigger audio reset:%d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, "failed to trigger audio reset\n"); ret = devm_pm_runtime_enable(&pdev->dev); if (ret) @@ -2223,25 +2218,21 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) /* regmap init */ afe->regmap = syscon_node_to_regmap(dev->parent->of_node); - if (IS_ERR(afe->regmap)) { - dev_err(dev, "could not get regmap from parent\n"); - return PTR_ERR(afe->regmap); - } + if (IS_ERR(afe->regmap)) + return dev_err_probe(dev, PTR_ERR(afe->regmap), + "could not get regmap from parent"); + ret = regmap_attach_dev(dev, afe->regmap, &mt8192_afe_regmap_config); - if (ret) { - dev_warn(dev, "regmap_attach_dev fail, ret %d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, "regmap_attach_dev fail\n"); /* enable clock for regcache get default value from hw */ afe_priv->pm_runtime_bypass_reg_ctl = true; pm_runtime_get_sync(&pdev->dev); ret = regmap_reinit_cache(afe->regmap, &mt8192_afe_regmap_config); - if (ret) { - dev_err(dev, "regmap_reinit_cache fail, ret %d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, "regmap_reinit_cache fail\n"); pm_runtime_put_sync(&pdev->dev); afe_priv->pm_runtime_bypass_reg_ctl = false; @@ -2281,30 +2272,22 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) ret = devm_request_irq(dev, irq_id, mt8192_afe_irq_handler, IRQF_TRIGGER_NONE, "asys-isr", (void *)afe); - if (ret) { - dev_err(dev, "could not request_irq for Afe_ISR_Handle\n"); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, "could not request_irq for Afe_ISR_Handle\n"); /* init sub_dais */ INIT_LIST_HEAD(&afe->sub_dais); for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { ret = dai_register_cbs[i](afe); - if (ret) { - dev_warn(afe->dev, "dai register i %d fail, ret %d\n", - i, ret); - return ret; - } + if (ret) + return dev_err_probe(afe->dev, ret, "dai %d register fail", i); } /* init dai_driver and component_driver */ ret = mtk_afe_combine_sub_dai(afe); - if (ret) { - dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(afe->dev, ret, "mtk_afe_combine_sub_dai fail\n"); /* others */ afe->mtk_afe_hardware = &mt8192_afe_hardware; @@ -2320,19 +2303,15 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) /* register platform */ ret = devm_snd_soc_register_component(&pdev->dev, &mt8192_afe_component, NULL, 0); - if (ret) { - dev_warn(dev, "err_platform\n"); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, "Couldn't register AFE component\n"); ret = devm_snd_soc_register_component(&pdev->dev, &mt8192_afe_pcm_component, afe->dai_drivers, afe->num_dai_drivers); - if (ret) { - dev_warn(dev, "err_dai_component\n"); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, "Couldn't register AFE-PCM component\n"); return 0; } From d6c01755805d346a1382d2d81c214b2ca557487a Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Wed, 13 Mar 2024 12:01:28 +0100 Subject: [PATCH 025/603] ASoC: mediatek: Commonize ADDA rate transform functions and enums Both the enumerations for UL/DL rates, delay data and the functions adda_{dl,ul}_rate_transform were duplicated for each MediaTek SoC dai-adda driver: move the common bits to a new mtk-dai-adda-common file and its header. While at it, also add the "mtk_" prefix to the exported functions. Signed-off-by: AngeloGioacchino Del Regno Link: https://msgid.link/r/20240313110147.1267793-4-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/common/Makefile | 2 + .../soc/mediatek/common/mtk-dai-adda-common.c | 70 ++++++++++++++ .../soc/mediatek/common/mtk-dai-adda-common.h | 45 +++++++++ sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 85 +---------------- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 90 +----------------- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 92 +------------------ sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 89 +----------------- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 90 +----------------- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 90 +----------------- 9 files changed, 135 insertions(+), 518 deletions(-) create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.c create mode 100644 sound/soc/mediatek/common/mtk-dai-adda-common.h diff --git a/sound/soc/mediatek/common/Makefile b/sound/soc/mediatek/common/Makefile index 42e636c10c1ee..363cc258a3d55 100644 --- a/sound/soc/mediatek/common/Makefile +++ b/sound/soc/mediatek/common/Makefile @@ -1,6 +1,8 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o mtk-soundcard-driver.o +snd-soc-mtk-common-objs += mtk-dai-adda-common.o + obj-$(CONFIG_SND_SOC_MEDIATEK) += snd-soc-mtk-common.o obj-$(CONFIG_SND_SOC_MTK_BTCVSD) += mtk-btcvsd.o diff --git a/sound/soc/mediatek/common/mtk-dai-adda-common.c b/sound/soc/mediatek/common/mtk-dai-adda-common.c new file mode 100644 index 0000000000000..4dc1412489d60 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-dai-adda-common.c @@ -0,0 +1,70 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek ALSA SoC Audio DAI ADDA Common + * + * Copyright (c) 2021 MediaTek Inc. + * Copyright (c) 2024 Collabora Ltd. + * AngeloGioacchino Del Regno + */ + +#include +#include + +#include "mtk-base-afe.h" +#include "mtk-dai-adda-common.h" + +unsigned int mtk_adda_dl_rate_transform(struct mtk_base_afe *afe, u32 rate) +{ + switch (rate) { + case 8000: + return MTK_AFE_ADDA_DL_RATE_8K; + case 11025: + return MTK_AFE_ADDA_DL_RATE_11K; + case 12000: + return MTK_AFE_ADDA_DL_RATE_12K; + case 16000: + return MTK_AFE_ADDA_DL_RATE_16K; + case 22050: + return MTK_AFE_ADDA_DL_RATE_22K; + case 24000: + return MTK_AFE_ADDA_DL_RATE_24K; + case 32000: + return MTK_AFE_ADDA_DL_RATE_32K; + case 44100: + return MTK_AFE_ADDA_DL_RATE_44K; + case 48000: + return MTK_AFE_ADDA_DL_RATE_48K; + case 96000: + return MTK_AFE_ADDA_DL_RATE_96K; + case 192000: + return MTK_AFE_ADDA_DL_RATE_192K; + default: + dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", + __func__, rate); + return MTK_AFE_ADDA_DL_RATE_48K; + } +} +EXPORT_SYMBOL_GPL(mtk_adda_dl_rate_transform); + +unsigned int mtk_adda_ul_rate_transform(struct mtk_base_afe *afe, u32 rate) +{ + switch (rate) { + case 8000: + return MTK_AFE_ADDA_UL_RATE_8K; + case 16000: + return MTK_AFE_ADDA_UL_RATE_16K; + case 32000: + return MTK_AFE_ADDA_UL_RATE_32K; + case 48000: + return MTK_AFE_ADDA_UL_RATE_48K; + case 96000: + return MTK_AFE_ADDA_UL_RATE_96K; + case 192000: + return MTK_AFE_ADDA_UL_RATE_192K; + default: + dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", + __func__, rate); + return MTK_AFE_ADDA_UL_RATE_48K; + } +} +EXPORT_SYMBOL_GPL(mtk_adda_ul_rate_transform); diff --git a/sound/soc/mediatek/common/mtk-dai-adda-common.h b/sound/soc/mediatek/common/mtk-dai-adda-common.h new file mode 100644 index 0000000000000..208b0dd89f577 --- /dev/null +++ b/sound/soc/mediatek/common/mtk-dai-adda-common.h @@ -0,0 +1,45 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2021 MediaTek Inc. + * Copyright (c) 2024 Collabora Ltd. + * AngeloGioacchino Del Regno + */ + +#ifndef _MTK_DAI_ADDA_COMMON_H_ +#define _MTK_DAI_ADDA_COMMON_H_ + +struct mtk_base_afe; + +enum adda_input_mode_rate { + MTK_AFE_ADDA_DL_RATE_8K = 0, + MTK_AFE_ADDA_DL_RATE_11K = 1, + MTK_AFE_ADDA_DL_RATE_12K = 2, + MTK_AFE_ADDA_DL_RATE_16K = 3, + MTK_AFE_ADDA_DL_RATE_22K = 4, + MTK_AFE_ADDA_DL_RATE_24K = 5, + MTK_AFE_ADDA_DL_RATE_32K = 6, + MTK_AFE_ADDA_DL_RATE_44K = 7, + MTK_AFE_ADDA_DL_RATE_48K = 8, + MTK_AFE_ADDA_DL_RATE_96K = 9, + MTK_AFE_ADDA_DL_RATE_192K = 10, +}; + +enum adda_voice_mode_rate { + MTK_AFE_ADDA_UL_RATE_8K = 0, + MTK_AFE_ADDA_UL_RATE_16K = 1, + MTK_AFE_ADDA_UL_RATE_32K = 2, + MTK_AFE_ADDA_UL_RATE_48K = 3, + MTK_AFE_ADDA_UL_RATE_96K = 4, + MTK_AFE_ADDA_UL_RATE_192K = 5, + MTK_AFE_ADDA_UL_RATE_48K_HD = 6, +}; + +enum adda_rxif_delay_data { + DELAY_DATA_MISO1 = 0, + DELAY_DATA_MISO0 = 1, + DELAY_DATA_MISO2 = 1, +}; + +unsigned int mtk_adda_dl_rate_transform(struct mtk_base_afe *afe, u32 rate); +unsigned int mtk_adda_ul_rate_transform(struct mtk_base_afe *afe, u32 rate); +#endif diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index 0ac6409c6d61f..78f3ad758c120 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -10,86 +10,7 @@ #include "mt6797-afe-common.h" #include "mt6797-interconnection.h" #include "mt6797-reg.h" - -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, - MTK_AFE_ADDA_UL_RATE_48K_HD = 6, -}; - -static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} +#include "../common/mtk-dai-adda-common.h" /* dai component */ static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { @@ -246,7 +167,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, regmap_write(afe->regmap, AFE_ADDA_PREDIS_CON1, 0); /* set input sampling rate */ - dl_src2_con0 = adda_dl_rate_transform(afe, rate) << 28; + dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << 28; /* set output mode */ switch (rate) { @@ -296,7 +217,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, 0x1 << 0, 0x0 << 0); - voice_mode = adda_ul_rate_transform(afe, rate); + voice_mode = mtk_adda_ul_rate_transform(afe, rate); ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c index 5b8a274419ed1..be69bcea2a786 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c @@ -10,6 +10,7 @@ #include "mt8183-afe-common.h" #include "mt8183-interconnection.h" #include "mt8183-reg.h" +#include "../common/mtk-dai-adda-common.h" enum { AUDIO_SDM_LEVEL_MUTE = 0, @@ -18,91 +19,6 @@ enum { /* you need to change formula of hp impedance and dc trim too */ }; -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO2, -}; - -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, - MTK_AFE_ADDA_UL_RATE_48K_HD = 6, -}; - -static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} - /* dai component */ static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1", AFE_CONN3, I_DL1_CH1, 1, 0), @@ -369,7 +285,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, regmap_write(afe->regmap, AFE_ADDA_PREDIS_CON1, 0); /* set sampling rate */ - dl_src2_con0 = adda_dl_rate_transform(afe, rate) << 28; + dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << 28; /* set output mode */ switch (rate) { @@ -420,7 +336,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, 0x1 << 0, 0x0 << 0); - voice_mode = adda_ul_rate_transform(afe, rate); + voice_mode = mtk_adda_ul_rate_transform(afe, rate); ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index ad6d4b5cf6979..dbd157d1a1ea2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -11,6 +11,7 @@ #include "mt8186-afe-common.h" #include "mt8186-afe-gpio.h" #include "mt8186-interconnection.h" +#include "../common/mtk-dai-adda-common.h" enum { UL_IIR_SW = 0, @@ -33,35 +34,6 @@ enum { AUDIO_SDM_3RD, }; -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO2, -}; - -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, - MTK_AFE_ADDA_UL_RATE_48K_HD = 6, -}; - #define SDM_AUTO_RESET_THRESHOLD 0x190000 struct mtk_afe_adda_priv { @@ -83,64 +55,6 @@ static struct mtk_afe_adda_priv *get_adda_priv_by_name(struct mtk_base_afe *afe, return afe_priv->dai_priv[dai_id]; } -static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_dbg(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - } - - return MTK_AFE_ADDA_DL_RATE_48K; -} - -static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_dbg(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - } - - return MTK_AFE_ADDA_UL_RATE_48K; -} - /* dai component */ static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1 Switch", AFE_CONN3, I_DL1_CH1, 1, 0), @@ -658,7 +572,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, adda_priv->dl_rate = rate; /* set sampling rate */ - dl_src2_con0 = adda_dl_rate_transform(afe, rate) << + dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << DL_2_INPUT_MODE_CTL_SFT; /* set output mode, UP_SAMPLING_RATE_X8 */ @@ -721,7 +635,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, } } else { unsigned int ul_src_con0 = 0; - unsigned int voice_mode = adda_ul_rate_transform(afe, rate); + unsigned int voice_mode = mtk_adda_ul_rate_transform(afe, rate); adda_priv->ul_rate = rate; ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-adda.c b/sound/soc/mediatek/mt8188/mt8188-dai-adda.c index 7dc029f2b4283..8a17d1935c48f 100644 --- a/sound/soc/mediatek/mt8188/mt8188-dai-adda.c +++ b/sound/soc/mediatek/mt8188/mt8188-dai-adda.c @@ -14,6 +14,7 @@ #include "mt8188-afe-clk.h" #include "mt8188-afe-common.h" #include "mt8188-reg.h" +#include "../common/mtk-dai-adda-common.h" #define ADDA_HIRES_THRES 48000 @@ -24,94 +25,10 @@ enum { SUPPLY_SEQ_ADDA_AFE_ON, }; -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, -}; - -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO0 = 1, -}; - struct mtk_dai_adda_priv { bool hires_required; }; -static unsigned int afe_adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_info(afe->dev, "%s(), rate %u invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int afe_adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_info(afe->dev, "%s(), rate %u invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} - static int mt8188_adda_mtkaif_init(struct mtk_base_afe *afe) { struct mt8188_afe_private *afe_priv = afe->platform_priv; @@ -440,7 +357,7 @@ static int mtk_dai_da_configure(struct mtk_base_afe *afe, /* set sampling rate */ mask |= DL_2_INPUT_MODE_CTL_MASK; val |= FIELD_PREP(DL_2_INPUT_MODE_CTL_MASK, - afe_adda_dl_rate_transform(afe, rate)); + mtk_adda_dl_rate_transform(afe, rate)); /* turn off saturation */ mask |= DL_2_CH1_SATURATION_EN_CTL; @@ -474,7 +391,7 @@ static int mtk_dai_ad_configure(struct mtk_base_afe *afe, mask = UL_VOICE_MODE_CTL_MASK; val = FIELD_PREP(UL_VOICE_MODE_CTL_MASK, - afe_adda_ul_rate_transform(afe, rate)); + mtk_adda_ul_rate_transform(afe, rate)); regmap_update_bits(afe->regmap, AFE_ADDA_UL_SRC_CON0, mask, val); diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-adda.c b/sound/soc/mediatek/mt8192/mt8192-dai-adda.c index 36d33032a37a6..99de85b876435 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-adda.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-adda.c @@ -13,6 +13,7 @@ #include "mt8192-afe-common.h" #include "mt8192-afe-gpio.h" #include "mt8192-interconnection.h" +#include "../common/mtk-dai-adda-common.h" enum { UL_IIR_SW = 0, @@ -35,93 +36,8 @@ enum { AUDIO_SDM_3RD, }; -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO2, -}; - -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, - MTK_AFE_ADDA_UL_RATE_48K_HD = 6, -}; - #define SDM_AUTO_RESET_THRESHOLD 0x190000 -static unsigned int adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} - /* dai component */ static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { SOC_DAPM_SINGLE_AUTODISABLE("DL1_CH1", AFE_CONN3, I_DL1_CH1, 1, 0), @@ -1156,7 +1072,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, unsigned int dl_src2_con1 = 0; /* set sampling rate */ - dl_src2_con0 = adda_dl_rate_transform(afe, rate) << + dl_src2_con0 = mtk_adda_dl_rate_transform(afe, rate) << DL_2_INPUT_MODE_CTL_SFT; /* set output mode, UP_SAMPLING_RATE_X8 */ @@ -1246,7 +1162,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, unsigned int voice_mode = 0; unsigned int ul_src_con0 = 0; /* default value */ - voice_mode = adda_ul_rate_transform(afe, rate); + voice_mode = mtk_adda_ul_rate_transform(afe, rate); ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c index 0dd35255066be..8da1587128ccf 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c @@ -12,6 +12,7 @@ #include "mt8195-afe-clk.h" #include "mt8195-afe-common.h" #include "mt8195-reg.h" +#include "../common/mtk-dai-adda-common.h" #define ADDA_DL_GAIN_LOOPBACK 0x1800 #define ADDA_HIRES_THRES 48000 @@ -25,35 +26,6 @@ enum { SUPPLY_SEQ_ADDA_AFE_ON, }; -enum { - MTK_AFE_ADDA_DL_RATE_8K = 0, - MTK_AFE_ADDA_DL_RATE_11K = 1, - MTK_AFE_ADDA_DL_RATE_12K = 2, - MTK_AFE_ADDA_DL_RATE_16K = 3, - MTK_AFE_ADDA_DL_RATE_22K = 4, - MTK_AFE_ADDA_DL_RATE_24K = 5, - MTK_AFE_ADDA_DL_RATE_32K = 6, - MTK_AFE_ADDA_DL_RATE_44K = 7, - MTK_AFE_ADDA_DL_RATE_48K = 8, - MTK_AFE_ADDA_DL_RATE_96K = 9, - MTK_AFE_ADDA_DL_RATE_192K = 10, -}; - -enum { - MTK_AFE_ADDA_UL_RATE_8K = 0, - MTK_AFE_ADDA_UL_RATE_16K = 1, - MTK_AFE_ADDA_UL_RATE_32K = 2, - MTK_AFE_ADDA_UL_RATE_48K = 3, - MTK_AFE_ADDA_UL_RATE_96K = 4, - MTK_AFE_ADDA_UL_RATE_192K = 5, -}; - -enum { - DELAY_DATA_MISO1 = 0, - DELAY_DATA_MISO0 = 1, - DELAY_DATA_MISO2 = 1, -}; - enum { MTK_AFE_ADDA, MTK_AFE_ADDA6, @@ -63,62 +35,6 @@ struct mtk_dai_adda_priv { bool hires_required; }; -static unsigned int afe_adda_dl_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_DL_RATE_8K; - case 11025: - return MTK_AFE_ADDA_DL_RATE_11K; - case 12000: - return MTK_AFE_ADDA_DL_RATE_12K; - case 16000: - return MTK_AFE_ADDA_DL_RATE_16K; - case 22050: - return MTK_AFE_ADDA_DL_RATE_22K; - case 24000: - return MTK_AFE_ADDA_DL_RATE_24K; - case 32000: - return MTK_AFE_ADDA_DL_RATE_32K; - case 44100: - return MTK_AFE_ADDA_DL_RATE_44K; - case 48000: - return MTK_AFE_ADDA_DL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_DL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_DL_RATE_192K; - default: - dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_DL_RATE_48K; - } -} - -static unsigned int afe_adda_ul_rate_transform(struct mtk_base_afe *afe, - unsigned int rate) -{ - switch (rate) { - case 8000: - return MTK_AFE_ADDA_UL_RATE_8K; - case 16000: - return MTK_AFE_ADDA_UL_RATE_16K; - case 32000: - return MTK_AFE_ADDA_UL_RATE_32K; - case 48000: - return MTK_AFE_ADDA_UL_RATE_48K; - case 96000: - return MTK_AFE_ADDA_UL_RATE_96K; - case 192000: - return MTK_AFE_ADDA_UL_RATE_192K; - default: - dev_info(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", - __func__, rate); - return MTK_AFE_ADDA_UL_RATE_48K; - } -} - static int mt8195_adda_mtkaif_init(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; @@ -644,7 +560,7 @@ static int mtk_dai_da_configure(struct mtk_base_afe *afe, /* set sampling rate */ mask |= DL_2_INPUT_MODE_CTL_MASK; - val |= DL_2_INPUT_MODE_CTL(afe_adda_dl_rate_transform(afe, rate)); + val |= DL_2_INPUT_MODE_CTL(mtk_adda_dl_rate_transform(afe, rate)); /* turn off saturation */ mask |= DL_2_CH1_SATURATION_EN_CTL; @@ -681,7 +597,7 @@ static int mtk_dai_ad_configure(struct mtk_base_afe *afe, unsigned int mask = 0; mask |= UL_VOICE_MODE_CTL_MASK; - val |= UL_VOICE_MODE_CTL(afe_adda_ul_rate_transform(afe, rate)); + val |= UL_VOICE_MODE_CTL(mtk_adda_ul_rate_transform(afe, rate)); switch (id) { case MT8195_AFE_IO_UL_SRC1: From 5f39231888c63f0a7708abc86b51b847476379d8 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Wed, 13 Mar 2024 12:01:29 +0100 Subject: [PATCH 026/603] ASoC: mediatek: Assign dummy when codec not specified for a DAI link MediaTek sound card drivers are checking whether a DAI link is present and used on a board to assign the correct parameters and this is done by checking the codec DAI names at probe time. If no real codec is present, assign the dummy codec to the DAI link to avoid NULL pointer during string comparison. Fixes: 4302187d955f ("ASoC: mediatek: common: add soundcard driver common code") Signed-off-by: AngeloGioacchino Del Regno Link: https://msgid.link/r/20240313110147.1267793-5-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-soundcard-driver.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.c b/sound/soc/mediatek/common/mtk-soundcard-driver.c index a58e1e3674dec..000a086a8cf44 100644 --- a/sound/soc/mediatek/common/mtk-soundcard-driver.c +++ b/sound/soc/mediatek/common/mtk-soundcard-driver.c @@ -22,7 +22,11 @@ static int set_card_codec_info(struct snd_soc_card *card, codec_node = of_get_child_by_name(sub_node, "codec"); if (!codec_node) { - dev_dbg(dev, "%s no specified codec\n", dai_link->name); + dev_dbg(dev, "%s no specified codec: setting dummy.\n", dai_link->name); + + dai_link->codecs = &snd_soc_dummy_dlc; + dai_link->num_codecs = 1; + dai_link->dynamic = 1; return 0; } From 885dd75f41f9fff5b277bc6ab28ad798f98a37b4 Mon Sep 17 00:00:00 2001 From: Frank Li Date: Fri, 22 Mar 2024 10:54:05 -0400 Subject: [PATCH 027/603] ASoC: dt-bindings: fsl-esai: Convert fsl,esai.txt to yaml Convert fsl,esai.txt to yaml. So DTB_CHECK tools can verify dts file about esai part. clock-names 'spba' is optional according to description. So minItems of clocks and clock-names is 3. Signed-off-by: Frank Li Reviewed-by: Rob Herring Link: https://msgid.link/r/20240322145406.2613256-1-Frank.Li@nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,esai.txt | 68 ---------- .../devicetree/bindings/sound/fsl,esai.yaml | 116 ++++++++++++++++++ 2 files changed, 116 insertions(+), 68 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,esai.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt deleted file mode 100644 index 90112ca1ff423..0000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,esai.txt +++ /dev/null @@ -1,68 +0,0 @@ -Freescale Enhanced Serial Audio Interface (ESAI) Controller - -The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port -for serial communication with a variety of serial devices, including industry -standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and -other DSPs. It has up to six transmitters and four receivers. - -Required properties: - - - compatible : Compatible list, should contain one of the following - compatibles: - "fsl,imx35-esai", - "fsl,vf610-esai", - "fsl,imx6ull-esai", - "fsl,imx8qm-esai", - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "core" The core clock used to access registers - "extal" The esai baud clock for esai controller used to - derive HCK, SCK and FS. - "fsys" The system clock derived from ahb clock used to - derive HCK, SCK and FS. - "spba" The spba clock is required when ESAI is placed as a - bus slave of the Shared Peripheral Bus and when two - or more bus masters (CPU, DMA or DSP) try to access - it. This property is optional depending on the SoC - design. - - - fsl,fifo-depth : The number of elements in the transmit and receive - FIFOs. This number is the maximum allowed value for - TFCR[TFWM] or RFCR[RFWM]. - - - fsl,esai-synchronous: This is a boolean property. If present, indicating - that ESAI would work in the synchronous mode, which - means all the settings for Receiving would be - duplicated from Transmission related registers. - -Optional properties: - - - big-endian : If this property is absent, the native endian mode - will be in use as default, or the big endian mode - will be in use for all the device registers. - -Example: - -esai: esai@2024000 { - compatible = "fsl,imx35-esai"; - reg = <0x02024000 0x4000>; - interrupts = <0 51 0x04>; - clocks = <&clks 208>, <&clks 118>, <&clks 208>; - clock-names = "core", "extal", "fsys"; - dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; - dma-names = "rx", "tx"; - fsl,fifo-depth = <128>; - fsl,esai-synchronous; - big-endian; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml new file mode 100644 index 0000000000000..f167f1634d7e3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,esai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Enhanced Serial Audio Interface (ESAI) Controller + +maintainers: + - Shengjiu Wang + - Frank Li + +description: + The Enhanced Serial Audio Interface (ESAI) provides a full-duplex serial port + for serial communication with a variety of serial devices, including industry + standard codecs, Sony/Phillips Digital Interface (S/PDIF) transceivers, and + other DSPs. It has up to six transmitters and four receivers. + +properties: + compatible: + enum: + - fsl,imx35-esai + - fsl,imx6ull-esai + - fsl,imx8qm-esai + - fsl,vf610-esai + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + minItems: 3 + items: + - description: + The core clock used to access registers. + - description: + The esai baud clock for esai controller used to + derive HCK, SCK and FS. + - description: + The system clock derived from ahb clock used to + derive HCK, SCK and FS. + - description: + The spba clock is required when ESAI is placed as a + bus slave of the Shared Peripheral Bus and when two + or more bus masters (CPU, DMA or DSP) try to access + it. This property is optional depending on the SoC + design. + + clock-names: + minItems: 3 + items: + - const: core + - const: extal + - const: fsys + - const: spba + + dmas: + minItems: 2 + maxItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + fsl,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The number of elements in the transmit and receive + FIFOs. This number is the maximum allowed value for + TFCR[TFWM] or RFCR[RFWM]. + + fsl,esai-synchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: + This is a boolean property. If present, indicating + that ESAI would work in the synchronous mode, which + means all the settings for Receiving would be + duplicated from Transmission related registers. + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: + If this property is absent, the native endian mode + will be in use as default, or the big endian mode + will be in use for all the device registers. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - fsl,fifo-depth + - fsl,esai-synchronous + +unevaluatedProperties: false + +examples: + - | + esai@2024000 { + compatible = "fsl,imx35-esai"; + reg = <0x02024000 0x4000>; + interrupts = <0 51 0x04>; + clocks = <&clks 208>, <&clks 118>, <&clks 208>; + clock-names = "core", "extal", "fsys"; + dmas = <&sdma 23 21 0>, <&sdma 24 21 0>; + dma-names = "rx", "tx"; + fsl,fifo-depth = <128>; + fsl,esai-synchronous; + big-endian; + }; From b340f56a74b62d8ce8617650c8ab4a26c87ba5c5 Mon Sep 17 00:00:00 2001 From: Kartik Agarwala Date: Mon, 25 Mar 2024 23:49:42 +0530 Subject: [PATCH 028/603] ASoC: dt-bindings: wm8974: Convert to dtschema Convert WM8974 audio CODEC bindings from text to dtschema. Signed-off-by: Kartik Agarwala Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240325181943.116733-1-agarwala.kartik@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/wlf,wm8974.txt | 15 ------- .../devicetree/bindings/sound/wlf,wm8974.yaml | 41 +++++++++++++++++++ 2 files changed, 41 insertions(+), 15 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.txt create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.yaml diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt b/Documentation/devicetree/bindings/sound/wlf,wm8974.txt deleted file mode 100644 index 01d3a7c834197..0000000000000 --- a/Documentation/devicetree/bindings/sound/wlf,wm8974.txt +++ /dev/null @@ -1,15 +0,0 @@ -WM8974 audio CODEC - -This device supports both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - compatible: "wlf,wm8974" - - reg: the I2C address or SPI chip select number of the device - -Examples: - -codec: wm8974@1a { - compatible = "wlf,wm8974"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml new file mode 100644 index 0000000000000..d27300207c67e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8974.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8974.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8974 audio CODEC + +maintainers: + - patches@opensource.cirrus.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: wlf,wm8974 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8974"; + reg = <0x1a>; + }; + }; From b020aff0ec6ec089bd3e07bd6b58623826752ff7 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:39 -0500 Subject: [PATCH 029/603] ASoC: Intel: board_helpers: support DAI link ID customization Add an new field link_id_overwrite to sof_card_private structure to support machine drivers which DAI link ID is fixed number or discontinue (i.e. no-codec boards). If this field is zero, DAI array index will be used as link ID. Otherwise the value extracted from link_id_overwrite will be used. The field link_id_overwrite is supposed to be initialized by SOF_LINK_IDS macro like following example. ctx->link_id_overwrite = SOF_LINK_IDS(HEADPHONE_BE_ID, \ DMIC01_BE_ID, \ DMIC16K_BE_ID, \ IDISP_HDMI_BE_ID, \ SPK_BE_ID, \ BT_OFFLOAD_BE_ID, \ HDMI_IN_BE_ID) An exception is that, if you use link_order_overwrite to overwrite DAI link order, then you need to use the same order to build link_id_overwrite variable as well. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 36 ++++++++++++++++------ sound/soc/intel/boards/sof_board_helpers.h | 18 +++++++++++ 2 files changed, 45 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 088894ff41657..e853dc851c422 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -427,6 +427,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, int ret; int ssp_hdmi_in = 0; unsigned long link_order, link; + unsigned long link_ids, be_id; num_links = calculate_num_links(ctx); @@ -440,12 +441,26 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, else link_order = DEFAULT_LINK_ORDER; - dev_dbg(dev, "create dai links, link_order 0x%lx\n", link_order); + if (ctx->link_id_overwrite) + link_ids = ctx->link_id_overwrite; + else + link_ids = 0; + + dev_dbg(dev, "create dai links, link_order 0x%lx, id_overwrite 0x%lx\n", + link_order, link_ids); while (link_order) { link = link_order & SOF_LINK_ORDER_MASK; link_order >>= SOF_LINK_ORDER_SHIFT; + if (ctx->link_id_overwrite) { + be_id = link_ids & SOF_LINK_IDS_MASK; + link_ids >>= SOF_LINK_IDS_SHIFT; + } else { + /* use array index as link id */ + be_id = idx; + } + switch (link) { case SOF_LINK_CODEC: /* headphone codec */ @@ -453,7 +468,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, continue; ret = sof_intel_board_set_codec_link(dev, &links[idx], - idx, + be_id, ctx->codec_type, ctx->ssp_codec); if (ret) { @@ -472,7 +487,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, /* at least we have dmic01 */ ret = sof_intel_board_set_dmic_link(dev, &links[idx], - idx, SOF_DMIC_01); + be_id, SOF_DMIC_01); if (ret) { dev_err(dev, "fail to set dmic01 link, ret %d\n", ret); @@ -488,7 +503,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, /* set up 2 BE links at most */ ret = sof_intel_board_set_dmic_link(dev, &links[idx], - idx, SOF_DMIC_16K); + be_id, SOF_DMIC_16K); if (ret) { dev_err(dev, "fail to set dmic16k link, ret %d\n", ret); @@ -502,7 +517,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, for (i = 1; i <= ctx->hdmi_num; i++) { ret = sof_intel_board_set_intel_hdmi_link(dev, &links[idx], - idx, i, + be_id, + i, ctx->hdmi.idisp_codec); if (ret) { dev_err(dev, "fail to set hdmi link, ret %d\n", @@ -511,6 +527,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, } idx++; + be_id++; } break; case SOF_LINK_AMP: @@ -519,7 +536,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, continue; ret = sof_intel_board_set_ssp_amp_link(dev, &links[idx], - idx, + be_id, ctx->amp_type, ctx->ssp_amp); if (ret) { @@ -536,8 +553,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, if (!ctx->bt_offload_present) continue; - ret = sof_intel_board_set_bt_link(dev, &links[idx], idx, - ctx->ssp_bt); + ret = sof_intel_board_set_bt_link(dev, &links[idx], + be_id, ctx->ssp_bt); if (ret) { dev_err(dev, "fail to set bt link, ret %d\n", ret); @@ -551,7 +568,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, for_each_set_bit(ssp_hdmi_in, &ctx->ssp_mask_hdmi_in, 32) { ret = sof_intel_board_set_hdmi_in_link(dev, &links[idx], - idx, + be_id, ssp_hdmi_in); if (ret) { dev_err(dev, "fail to set hdmi-in link, ret %d\n", @@ -560,6 +577,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, } idx++; + be_id++; } break; case SOF_LINK_NONE: diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index f42d5d6403219..162a5f4841da0 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -33,6 +33,18 @@ enum { (((k6) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 5)) | \ (((k7) & SOF_LINK_ORDER_MASK) << (SOF_LINK_ORDER_SHIFT * 6))) +#define SOF_LINK_IDS_MASK (0xF) +#define SOF_LINK_IDS_SHIFT (4) + +#define SOF_LINK_IDS(k1, k2, k3, k4, k5, k6, k7) \ + ((((k1) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 0)) | \ + (((k2) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 1)) | \ + (((k3) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 2)) | \ + (((k4) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 3)) | \ + (((k5) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 4)) | \ + (((k6) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 5)) | \ + (((k7) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 6))) + /* * sof_rt5682_private: private data for rt5682 machine driver * @@ -61,6 +73,7 @@ struct sof_rt5682_private { * @codec_link: pointer to headset codec dai link * @amp_link: pointer to speaker amplifier dai link * @link_order_overwrite: custom DAI link order + * @link_id_overwrite: custom DAI link ID * @rt5682: private data for rt5682 machine driver */ struct sof_card_private { @@ -84,6 +97,11 @@ struct sof_card_private { struct snd_soc_dai_link *amp_link; unsigned long link_order_overwrite; + /* + * A variable stores id for all BE DAI links, use SOF_LINK_IDS macro to + * build the value; use DAI link array index as id if zero. + */ + unsigned long link_id_overwrite; union { struct sof_rt5682_private rt5682; From 8906d8663d72642b05108963a47a069b7a04b23f Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:40 -0500 Subject: [PATCH 030/603] ASoC: Intel: sof_ssp_amp: use common module for DAI link generation Use intel_board module to generate DAI link array and update num_links field in snd_soc_card structure. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 180 ++++++++------------------- 1 file changed, 54 insertions(+), 126 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index ee2e813bf4c05..0b8adab76755e 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -75,126 +75,64 @@ static struct snd_soc_card sof_ssp_amp_card = { #define HDMI_IN_BE_ID 0 #define SPK_BE_ID 2 #define DMIC01_BE_ID 3 -#define DMIC16K_BE_ID 4 #define INTEL_HDMI_BE_ID 5 - -static struct snd_soc_dai_link * -sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, - int ssp_amp, int dmic_be_num, int hdmi_num, - bool idisp_codec) +/* extra BE links to support no-hdmi-in boards */ +#define DMIC16K_BE_ID 4 +#define BT_OFFLOAD_BE_ID 8 + +#define SSP_AMP_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_HDMI_IN, \ + SOF_LINK_AMP, \ + SOF_LINK_DMIC01, \ + SOF_LINK_DMIC16K, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_BT_OFFLOAD, \ + SOF_LINK_NONE) + +#define SSP_AMP_LINK_IDS SOF_LINK_ORDER(HDMI_IN_BE_ID, \ + SPK_BE_ID, \ + DMIC01_BE_ID, \ + DMIC16K_BE_ID, \ + INTEL_HDMI_BE_ID, \ + BT_OFFLOAD_BE_ID, \ + 0) + +static int +sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx) { - struct snd_soc_dai_link *links; - int i; - int id = 0; int ret; - bool fixed_be = false; - int be_id; - unsigned long ssp_mask_hdmi_in; - - links = devm_kcalloc(dev, sof_ssp_amp_card.num_links, - sizeof(struct snd_soc_dai_link), GFP_KERNEL); - if (!links) - return NULL; - - /* HDMI-In SSP */ - ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >> - SOF_HDMI_CAPTURE_SSP_MASK_SHIFT; - - if (ssp_mask_hdmi_in) { - int port = 0; - - /* the topology supports HDMI-IN uses fixed BE ID for DAI links */ - fixed_be = true; - - be_id = HDMI_IN_BE_ID; - for_each_set_bit(port, &ssp_mask_hdmi_in, 32) { - ret = sof_intel_board_set_hdmi_in_link(dev, &links[id], - be_id, port); - if (ret) - return NULL; - - id++; - be_id++; - } - } - - /* codec SSP */ - if (amp_type != CODEC_NONE) { - be_id = fixed_be ? SPK_BE_ID : id; - ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], be_id, - amp_type, ssp_amp); - if (ret) - return NULL; - - /* codec-specific fields */ - switch (amp_type) { - case CODEC_CS35L41: - cs35l41_set_dai_link(&links[id]); - break; - case CODEC_RT1308: - sof_rt1308_dai_link(&links[id]); - break; - default: - dev_err(dev, "invalid amp type %d\n", amp_type); - return NULL; - } - - id++; - } - - /* dmic */ - if (dmic_be_num > 0) { - /* at least we have dmic01 */ - be_id = fixed_be ? DMIC01_BE_ID : id; - ret = sof_intel_board_set_dmic_link(dev, &links[id], be_id, - SOF_DMIC_01); - if (ret) - return NULL; - id++; - } - - if (dmic_be_num > 1) { - /* set up 2 BE links at most */ - be_id = fixed_be ? DMIC16K_BE_ID : id; - ret = sof_intel_board_set_dmic_link(dev, &links[id], be_id, - SOF_DMIC_16K); - if (ret) - return NULL; - - id++; - } + ret = sof_intel_board_set_dai_link(dev, card, ctx); + if (ret) + return ret; - /* HDMI playback */ - for (i = 1; i <= hdmi_num; i++) { - be_id = fixed_be ? (INTEL_HDMI_BE_ID + i - 1) : id; - ret = sof_intel_board_set_intel_hdmi_link(dev, &links[id], be_id, - i, idisp_codec); - if (ret) - return NULL; + if (ctx->amp_type == CODEC_NONE) + return 0; - id++; + if (!ctx->amp_link) { + dev_err(dev, "amp link not available"); + return -EINVAL; } - /* BT audio offload */ - if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { - int port = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ret = sof_intel_board_set_bt_link(dev, &links[id], id, port); - if (ret) - return NULL; - - id++; + /* codec-specific fields for speaker amplifier */ + switch (ctx->amp_type) { + case CODEC_CS35L41: + cs35l41_set_dai_link(ctx->amp_link); + break; + case CODEC_RT1308: + sof_rt1308_dai_link(ctx->amp_link); + break; + default: + dev_err(dev, "invalid amp type %d\n", ctx->amp_type); + return -EINVAL; } - return links; + return 0; } static int sof_ssp_amp_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; - struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; int ret; @@ -221,15 +159,6 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) ctx->ssp_amp = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK; - /* set number of dai links */ - sof_ssp_amp_card.num_links = ctx->dmic_be_num; - - if (ctx->amp_type != CODEC_NONE) - sof_ssp_amp_card.num_links++; - - if (ctx->ssp_mask_hdmi_in) - sof_ssp_amp_card.num_links += hweight32(ctx->ssp_mask_hdmi_in); - if (sof_ssp_amp_quirk & SOF_HDMI_PLAYBACK_PRESENT) { ctx->hdmi_num = (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_PLAYBACK_MASK) >> SOF_NO_OF_HDMI_PLAYBACK_SHIFT; @@ -239,25 +168,24 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - - sof_ssp_amp_card.num_links += ctx->hdmi_num; } else { ctx->hdmi_num = 0; } - if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) ctx->bt_offload_present = true; - sof_ssp_amp_card.num_links++; - } - dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ctx->ssp_amp, ctx->dmic_be_num, - ctx->hdmi_num, - ctx->hdmi.idisp_codec); - if (!dai_links) - return -ENOMEM; + ctx->link_order_overwrite = SSP_AMP_LINK_ORDER; - sof_ssp_amp_card.dai_link = dai_links; + if (ctx->ssp_mask_hdmi_in) { + /* the topology supports HDMI-IN uses fixed BE ID for DAI links */ + ctx->link_id_overwrite = SSP_AMP_LINK_IDS; + } + + /* update dai_link */ + ret = sof_card_dai_links_create(&pdev->dev, &sof_ssp_amp_card, ctx); + if (ret) + return ret; /* update codec_conf */ switch (ctx->amp_type) { From 3b6378bb8baf519e641151dcdb9171c2bdecb7b7 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:41 -0500 Subject: [PATCH 031/603] ASoC: Intel: board_helpers: change dai link helpers to static function Since there is a helper function to generate entire DAI link array, we switch individual dai link helpers to static function. No functional change in this commit. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 85 +++++++++------------- sound/soc/intel/boards/sof_board_helpers.h | 24 ------ 2 files changed, 35 insertions(+), 74 deletions(-) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index e853dc851c422..feba1a5225279 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -74,6 +74,11 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) * DAI Link Helpers */ +enum sof_dmic_be_type { + SOF_DMIC_01, + SOF_DMIC_16K, +}; + /* DEFAULT_LINK_ORDER: the order used in sof_rt5682 */ #define DEFAULT_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_CODEC, \ SOF_LINK_DMIC01, \ @@ -97,13 +102,13 @@ static struct snd_soc_dai_link_component platform_component[] = { } }; -int sof_intel_board_set_codec_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_ssp_codec codec_type, int ssp_codec) +static int set_ssp_codec_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, enum sof_ssp_codec codec_type, + int ssp_codec) { struct snd_soc_dai_link_component *cpus; - dev_dbg(dev, "link %d: codec %s, ssp %d\n", be_id, + dev_dbg(dev, "link %d: ssp codec %s, ssp %d\n", be_id, sof_ssp_get_codec_name(codec_type), ssp_codec); /* link name */ @@ -144,11 +149,9 @@ int sof_intel_board_set_codec_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_codec_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_dmic_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_dmic_be_type be_type) +static int set_dmic_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, enum sof_dmic_be_type be_type) { struct snd_soc_dai_link_component *cpus; @@ -196,16 +199,14 @@ int sof_intel_board_set_dmic_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_dmic_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_intel_hdmi_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int hdmi_id, bool idisp_codec) +static int set_idisp_hdmi_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, int hdmi_id, bool idisp_codec) { struct snd_soc_dai_link_component *cpus, *codecs; - dev_dbg(dev, "link %d: intel hdmi, hdmi id %d, idisp codec %d\n", - be_id, hdmi_id, idisp_codec); + dev_dbg(dev, "link %d: idisp hdmi %d, idisp codec %d\n", be_id, hdmi_id, + idisp_codec); /* link name */ link->name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", hdmi_id); @@ -256,11 +257,9 @@ int sof_intel_board_set_intel_hdmi_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_intel_hdmi_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_ssp_amp_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_ssp_codec amp_type, int ssp_amp) +static int set_ssp_amp_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, enum sof_ssp_codec amp_type, int ssp_amp) { struct snd_soc_dai_link_component *cpus; @@ -298,11 +297,9 @@ int sof_intel_board_set_ssp_amp_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_ssp_amp_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_bt_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int ssp_bt) +static int set_bt_offload_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, int ssp_bt) { struct snd_soc_dai_link_component *cpus; @@ -341,11 +338,9 @@ int sof_intel_board_set_bt_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_bt_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); -int sof_intel_board_set_hdmi_in_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int ssp_hdmi) +static int set_hdmi_in_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, int ssp_hdmi) { struct snd_soc_dai_link_component *cpus; @@ -383,7 +378,6 @@ int sof_intel_board_set_hdmi_in_link(struct device *dev, return 0; } -EXPORT_SYMBOL_NS(sof_intel_board_set_hdmi_in_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); static int calculate_num_links(struct sof_card_private *ctx) { @@ -467,10 +461,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, if (ctx->codec_type == CODEC_NONE) continue; - ret = sof_intel_board_set_codec_link(dev, &links[idx], - be_id, - ctx->codec_type, - ctx->ssp_codec); + ret = set_ssp_codec_link(dev, &links[idx], be_id, + ctx->codec_type, ctx->ssp_codec); if (ret) { dev_err(dev, "fail to set codec link, ret %d\n", ret); @@ -486,8 +478,7 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, continue; /* at least we have dmic01 */ - ret = sof_intel_board_set_dmic_link(dev, &links[idx], - be_id, SOF_DMIC_01); + ret = set_dmic_link(dev, &links[idx], be_id, SOF_DMIC_01); if (ret) { dev_err(dev, "fail to set dmic01 link, ret %d\n", ret); @@ -502,8 +493,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, continue; /* set up 2 BE links at most */ - ret = sof_intel_board_set_dmic_link(dev, &links[idx], - be_id, SOF_DMIC_16K); + ret = set_dmic_link(dev, &links[idx], be_id, + SOF_DMIC_16K); if (ret) { dev_err(dev, "fail to set dmic16k link, ret %d\n", ret); @@ -515,11 +506,9 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, case SOF_LINK_IDISP_HDMI: /* idisp HDMI */ for (i = 1; i <= ctx->hdmi_num; i++) { - ret = sof_intel_board_set_intel_hdmi_link(dev, - &links[idx], - be_id, - i, - ctx->hdmi.idisp_codec); + ret = set_idisp_hdmi_link(dev, &links[idx], + be_id, i, + ctx->hdmi.idisp_codec); if (ret) { dev_err(dev, "fail to set hdmi link, ret %d\n", ret); @@ -535,10 +524,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, if (ctx->amp_type == CODEC_NONE) continue; - ret = sof_intel_board_set_ssp_amp_link(dev, &links[idx], - be_id, - ctx->amp_type, - ctx->ssp_amp); + ret = set_ssp_amp_link(dev, &links[idx], be_id, + ctx->amp_type, ctx->ssp_amp); if (ret) { dev_err(dev, "fail to set amp link, ret %d\n", ret); @@ -553,8 +540,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, if (!ctx->bt_offload_present) continue; - ret = sof_intel_board_set_bt_link(dev, &links[idx], - be_id, ctx->ssp_bt); + ret = set_bt_offload_link(dev, &links[idx], be_id, + ctx->ssp_bt); if (ret) { dev_err(dev, "fail to set bt link, ret %d\n", ret); @@ -566,10 +553,8 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, case SOF_LINK_HDMI_IN: /* HDMI-In */ for_each_set_bit(ssp_hdmi_in, &ctx->ssp_mask_hdmi_in, 32) { - ret = sof_intel_board_set_hdmi_in_link(dev, - &links[idx], - be_id, - ssp_hdmi_in); + ret = set_hdmi_in_link(dev, &links[idx], be_id, + ssp_hdmi_in); if (ret) { dev_err(dev, "fail to set hdmi-in link, ret %d\n", ret); diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 162a5f4841da0..38e459e6af9b9 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -108,34 +108,10 @@ struct sof_card_private { }; }; -enum sof_dmic_be_type { - SOF_DMIC_01, - SOF_DMIC_16K, -}; - int sof_intel_board_card_late_probe(struct snd_soc_card *card); int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, struct sof_card_private *ctx); -int sof_intel_board_set_codec_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_ssp_codec codec_type, int ssp_codec); -int sof_intel_board_set_dmic_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_dmic_be_type be_type); -int sof_intel_board_set_intel_hdmi_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int hdmi_id, bool idisp_codec); -int sof_intel_board_set_ssp_amp_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - enum sof_ssp_codec amp_type, int ssp_amp); -int sof_intel_board_set_bt_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int ssp_bt); -int sof_intel_board_set_hdmi_in_link(struct device *dev, - struct snd_soc_dai_link *link, int be_id, - int ssp_hdmi); - struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, const char * const dai_name[], int num_dais); From 54bac56954a7c36eb8d1c7047b63133635999eb8 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:42 -0500 Subject: [PATCH 032/603] ASoC: Intel: sof_da7219: add rpl_mx98360_da7219 board config This configuration supports RPL boards which implement DA7219 on SSP0 and MAX98360A on SSP1. DA7219 uses PLL bypass mode to avoid WCLK locking problem. To use this mode, the MCLK frequency must be 12.288 or 24.576MHz in the topology binary. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 4 ++++ sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 7 +++++++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 6eb5a6144e97c..290bf75bdaef6 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -545,6 +545,10 @@ static const struct platform_device_id board_ids[] = { .name = "adl_mx98360_da7219", /* no quirk needed for this board */ }, + { + .name = "rpl_mx98360_da7219", + /* no quirk needed for this board */ + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 00a21af210fa4..f797937cf9d89 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -455,6 +455,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .drv_name = "rpl_lt6911_hdmi_ssp", .sof_tplg_filename = "sof-rpl-nocodec-hdmi-ssp02.tplg" }, + { + .id = "DLGS7219", + .drv_name = "rpl_mx98360_da7219", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rpl_max98360a_amp, + .sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines); From 94e9dd5b38bc3d04f86c4a876f3b4b397add248b Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:43 -0500 Subject: [PATCH 033/603] ASoC: Intel: sof_rt5682: support ALC5650 on RPL boards This commit supports RPL boards which implement ALC5650 dual I2S interface codec. SSP port usage: HP: SSP0 -> AIF1 SPK: SSP1 -> AIF2 BT: SSP2 -> BT Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index f797937cf9d89..c43e537e60d2c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -462,6 +462,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .quirk_data = &rpl_max98360a_amp, .sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg", }, + { + .id = "10EC5650", + .drv_name = "rpl_rt5682_def", + .sof_tplg_filename = "sof-rpl-rt5650.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines); From 109896246a5311aa05692ecf38c0d71e1837fe23 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:44 -0500 Subject: [PATCH 034/603] ASoC: Intel: sof_cs42l42: rename BT offload quirk Rename the quirk in preparation for future changes: common quriks will be defined and handled in board helper module. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 323b86c42ef95..330d596b2eb6d 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -34,7 +34,7 @@ #define SOF_CS42L42_NUM_HDMIDEV_MASK (GENMASK(9, 7)) #define SOF_CS42L42_NUM_HDMIDEV(quirk) \ (((quirk) << SOF_CS42L42_NUM_HDMIDEV_SHIFT) & SOF_CS42L42_NUM_HDMIDEV_MASK) -#define SOF_BT_OFFLOAD_PRESENT BIT(25) +#define SOF_CS42L42_BT_OFFLOAD_PRESENT BIT(25) #define SOF_CS42L42_SSP_BT_SHIFT 26 #define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) #define SOF_CS42L42_SSP_BT(quirk) \ @@ -268,7 +268,7 @@ static int sof_audio_probe(struct platform_device *pdev) ctx->ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK; - if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + if (sof_cs42l42_quirk & SOF_CS42L42_BT_OFFLOAD_PRESENT) ctx->bt_offload_present = true; /* update dai_link */ @@ -306,7 +306,7 @@ static const struct platform_device_id board_ids[] = { .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | SOF_CS42L42_SSP_AMP(1) | SOF_CS42L42_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_PRESENT | + SOF_CS42L42_BT_OFFLOAD_PRESENT | SOF_CS42L42_SSP_BT(2)), }, { } From cf22d15ecf61c783e6903f3409e78d439f9c9b4a Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:45 -0500 Subject: [PATCH 035/603] ASoC: Intel: board_helpers: support sof_card_private initialization Add a helper function for machine drivers to initialize common part of sof_card_private structure. Also unify the macros of board quirks for the initialization. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 43 ++++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 40 ++++++++++++++++++++ 2 files changed, 83 insertions(+) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index feba1a5225279..a5135be94f32a 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -587,6 +587,49 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, } EXPORT_SYMBOL_NS(sof_intel_board_set_dai_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); +struct sof_card_private * +sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk) +{ + struct sof_card_private *ctx; + + dev_dbg(dev, "create ctx, board_quirk 0x%lx\n", board_quirk); + + ctx = devm_kzalloc(dev, sizeof(struct sof_card_private), GFP_KERNEL); + if (!ctx) + return NULL; + + ctx->codec_type = sof_ssp_detect_codec_type(dev); + ctx->amp_type = sof_ssp_detect_amp_type(dev); + + ctx->dmic_be_num = 2; + ctx->hdmi_num = (board_quirk & SOF_NUM_IDISP_HDMI_MASK) >> + SOF_NUM_IDISP_HDMI_SHIFT; + /* default number of HDMI DAI's */ + if (!ctx->hdmi_num) + ctx->hdmi_num = 3; + + /* port number/mask of peripherals attached to ssp interface */ + if (ctx->codec_type != CODEC_NONE) + ctx->ssp_codec = (board_quirk & SOF_SSP_PORT_CODEC_MASK) >> + SOF_SSP_PORT_CODEC_SHIFT; + + if (ctx->amp_type != CODEC_NONE) + ctx->ssp_amp = (board_quirk & SOF_SSP_PORT_AMP_MASK) >> + SOF_SSP_PORT_AMP_SHIFT; + + if (board_quirk & SOF_BT_OFFLOAD_PRESENT) { + ctx->bt_offload_present = true; + ctx->ssp_bt = (board_quirk & SOF_SSP_PORT_BT_OFFLOAD_MASK) >> + SOF_SSP_PORT_BT_OFFLOAD_SHIFT; + } + + ctx->ssp_mask_hdmi_in = (board_quirk & SOF_SSP_MASK_HDMI_CAPTURE_MASK) >> + SOF_SSP_MASK_HDMI_CAPTURE_SHIFT; + + return ctx; +} +EXPORT_SYMBOL_NS(sof_intel_board_get_ctx, SND_SOC_INTEL_SOF_BOARD_HELPERS); + struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, const char * const dai_name[], int num_dais) { diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 38e459e6af9b9..0d0a8d97843b0 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -10,6 +10,44 @@ #include "sof_hdmi_common.h" #include "sof_ssp_common.h" +/* + * Common board quirks: from bit 8 to 31, LSB 8 bits reserved for machine + * drivers + */ + +/* SSP port number for headphone codec: 3 bits */ +#define SOF_SSP_PORT_CODEC_SHIFT 8 +#define SOF_SSP_PORT_CODEC_MASK (GENMASK(10, 8)) +#define SOF_SSP_PORT_CODEC(quirk) \ + (((quirk) << SOF_SSP_PORT_CODEC_SHIFT) & SOF_SSP_PORT_CODEC_MASK) + +/* SSP port number for speaker amplifier: 3 bits */ +#define SOF_SSP_PORT_AMP_SHIFT 11 +#define SOF_SSP_PORT_AMP_MASK (GENMASK(13, 11)) +#define SOF_SSP_PORT_AMP(quirk) \ + (((quirk) << SOF_SSP_PORT_AMP_SHIFT) & SOF_SSP_PORT_AMP_MASK) + +/* SSP port number for BT audio offload: 3 bits */ +#define SOF_SSP_PORT_BT_OFFLOAD_SHIFT 14 +#define SOF_SSP_PORT_BT_OFFLOAD_MASK (GENMASK(16, 14)) +#define SOF_SSP_PORT_BT_OFFLOAD(quirk) \ + (((quirk) << SOF_SSP_PORT_BT_OFFLOAD_SHIFT) & SOF_SSP_PORT_BT_OFFLOAD_MASK) + +/* SSP port mask for HDMI capture: 6 bits */ +#define SOF_SSP_MASK_HDMI_CAPTURE_SHIFT 17 +#define SOF_SSP_MASK_HDMI_CAPTURE_MASK (GENMASK(22, 17)) +#define SOF_SSP_MASK_HDMI_CAPTURE(quirk) \ + (((quirk) << SOF_SSP_MASK_HDMI_CAPTURE_SHIFT) & SOF_SSP_MASK_HDMI_CAPTURE_MASK) + +/* Number of idisp HDMI BE link: 3 bits */ +#define SOF_NUM_IDISP_HDMI_SHIFT 23 +#define SOF_NUM_IDISP_HDMI_MASK (GENMASK(25, 23)) +#define SOF_NUM_IDISP_HDMI(quirk) \ + (((quirk) << SOF_NUM_IDISP_HDMI_SHIFT) & SOF_NUM_IDISP_HDMI_MASK) + +/* Board uses BT audio offload */ +#define SOF_BT_OFFLOAD_PRESENT BIT(26) + enum { SOF_LINK_NONE = 0, SOF_LINK_CODEC, @@ -111,6 +149,8 @@ struct sof_card_private { int sof_intel_board_card_late_probe(struct snd_soc_card *card); int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, struct sof_card_private *ctx); +struct sof_card_private * +sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk); struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, const char * const dai_name[], int num_dais); From 96258c3348e1a2cb6842175a0ac895ef3963f2e2 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:46 -0500 Subject: [PATCH 036/603] ASoC: Intel: sof_cs42l42: use common module for sof_card_private initialization Use intel_board module to initialize sof_card_private structure. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 72 +++++++--------------------- 1 file changed, 16 insertions(+), 56 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 330d596b2eb6d..40ecfeaa1d263 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -22,23 +22,6 @@ #include "../common/soc-intel-quirks.h" #include "sof_board_helpers.h" #include "sof_maxim_common.h" -#include "sof_ssp_common.h" - -#define SOF_CS42L42_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) -#define SOF_CS42L42_SSP_CODEC_MASK (GENMASK(2, 0)) -#define SOF_CS42L42_SSP_AMP_SHIFT 4 -#define SOF_CS42L42_SSP_AMP_MASK (GENMASK(6, 4)) -#define SOF_CS42L42_SSP_AMP(quirk) \ - (((quirk) << SOF_CS42L42_SSP_AMP_SHIFT) & SOF_CS42L42_SSP_AMP_MASK) -#define SOF_CS42L42_NUM_HDMIDEV_SHIFT 7 -#define SOF_CS42L42_NUM_HDMIDEV_MASK (GENMASK(9, 7)) -#define SOF_CS42L42_NUM_HDMIDEV(quirk) \ - (((quirk) << SOF_CS42L42_NUM_HDMIDEV_SHIFT) & SOF_CS42L42_NUM_HDMIDEV_MASK) -#define SOF_CS42L42_BT_OFFLOAD_PRESENT BIT(25) -#define SOF_CS42L42_SSP_BT_SHIFT 26 -#define SOF_CS42L42_SSP_BT_MASK (GENMASK(28, 26)) -#define SOF_CS42L42_SSP_BT(quirk) \ - (((quirk) << SOF_CS42L42_SSP_BT_SHIFT) & SOF_CS42L42_SSP_BT_MASK) static struct snd_soc_jack_pin jack_pins[] = { { @@ -52,7 +35,7 @@ static struct snd_soc_jack_pin jack_pins[] = { }; /* Default: SSP2 */ -static unsigned long sof_cs42l42_quirk = SOF_CS42L42_SSP_CODEC(2); +static unsigned long sof_cs42l42_quirk = SOF_SSP_PORT_CODEC(2); static int sof_cs42l42_init(struct snd_soc_pcm_runtime *rtd) { @@ -229,48 +212,26 @@ static int sof_audio_probe(struct platform_device *pdev) struct sof_card_private *ctx; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - if (pdev->id_entry && pdev->id_entry->driver_data) sof_cs42l42_quirk = (unsigned long)pdev->id_entry->driver_data; - ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev); - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); + dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); + + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, sof_cs42l42_quirk); + if (!ctx) + return -ENOMEM; if (soc_intel_is_glk()) { ctx->dmic_be_num = 1; - ctx->hdmi_num = 3; /* overwrite the DAI link order for GLK boards */ ctx->link_order_overwrite = GLK_LINK_ORDER; - } else { - ctx->dmic_be_num = 2; - ctx->hdmi_num = (sof_cs42l42_quirk & SOF_CS42L42_NUM_HDMIDEV_MASK) >> - SOF_CS42L42_NUM_HDMIDEV_SHIFT; - /* default number of HDMI DAI's */ - if (!ctx->hdmi_num) - ctx->hdmi_num = 3; } if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); - - /* port number of peripherals attached to ssp interface */ - ctx->ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> - SOF_CS42L42_SSP_BT_SHIFT; - - ctx->ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> - SOF_CS42L42_SSP_AMP_SHIFT; - - ctx->ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK; - - if (sof_cs42l42_quirk & SOF_CS42L42_BT_OFFLOAD_PRESENT) - ctx->bt_offload_present = true; - /* update dai_link */ ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_cs42l42, ctx); if (ret) @@ -293,21 +254,21 @@ static int sof_audio_probe(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { .name = "glk_cs4242_mx98357a", - .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(2) | - SOF_CS42L42_SSP_AMP(1)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(1)), }, { .name = "jsl_cs4242_mx98360a", - .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | - SOF_CS42L42_SSP_AMP(1)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "adl_mx98360a_cs4242", - .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | - SOF_CS42L42_SSP_AMP(1) | - SOF_CS42L42_NUM_HDMIDEV(4) | - SOF_CS42L42_BT_OFFLOAD_PRESENT | - SOF_CS42L42_SSP_BT(2)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_SSP_PORT_BT_OFFLOAD(2)), }, { } }; @@ -329,4 +290,3 @@ MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); From bfa8c0088f02440653bfdc7e72144302f2f5f22b Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:47 -0500 Subject: [PATCH 037/603] ASoC: Intel: sof_nau8825: use common module for sof_card_private initialization Use intel_board module to initialize sof_card_private structure. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 88 ++++++++-------------------- 1 file changed, 23 insertions(+), 65 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 719c2fbaf515a..b9db6e236a931 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -24,27 +24,8 @@ #include "sof_realtek_common.h" #include "sof_maxim_common.h" #include "sof_nuvoton_common.h" -#include "sof_ssp_common.h" - -#define SOF_NAU8825_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) -#define SOF_NAU8825_SSP_CODEC_MASK (GENMASK(2, 0)) -#define SOF_NAU8825_SSP_AMP_SHIFT 4 -#define SOF_NAU8825_SSP_AMP_MASK (GENMASK(6, 4)) -#define SOF_NAU8825_SSP_AMP(quirk) \ - (((quirk) << SOF_NAU8825_SSP_AMP_SHIFT) & SOF_NAU8825_SSP_AMP_MASK) -#define SOF_NAU8825_NUM_HDMIDEV_SHIFT 7 -#define SOF_NAU8825_NUM_HDMIDEV_MASK (GENMASK(9, 7)) -#define SOF_NAU8825_NUM_HDMIDEV(quirk) \ - (((quirk) << SOF_NAU8825_NUM_HDMIDEV_SHIFT) & SOF_NAU8825_NUM_HDMIDEV_MASK) - -/* BT audio offload: reserve 3 bits for future */ -#define SOF_BT_OFFLOAD_SSP_SHIFT 10 -#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(12, 10)) -#define SOF_BT_OFFLOAD_SSP(quirk) \ - (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) -#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(13) - -static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); + +static unsigned long sof_nau8825_quirk = SOF_SSP_PORT_CODEC(0); static struct snd_soc_jack_pin jack_pins[] = { { @@ -264,41 +245,19 @@ static int sof_audio_probe(struct platform_device *pdev) struct sof_card_private *ctx; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - if (pdev->id_entry && pdev->id_entry->driver_data) sof_nau8825_quirk = (unsigned long)pdev->id_entry->driver_data; - ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev); - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); - dev_dbg(&pdev->dev, "sof_nau8825_quirk = %lx\n", sof_nau8825_quirk); - /* default number of DMIC DAI's */ - ctx->dmic_be_num = 2; - ctx->hdmi_num = (sof_nau8825_quirk & SOF_NAU8825_NUM_HDMIDEV_MASK) >> - SOF_NAU8825_NUM_HDMIDEV_SHIFT; - /* default number of HDMI DAI's */ - if (!ctx->hdmi_num) - ctx->hdmi_num = 3; + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, sof_nau8825_quirk); + if (!ctx) + return -ENOMEM; if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - /* port number of peripherals attached to ssp interface */ - ctx->ssp_bt = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ctx->ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> - SOF_NAU8825_SSP_AMP_SHIFT; - - ctx->ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; - - if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) - ctx->bt_offload_present = true; - /* update dai_link */ ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_nau8825, ctx); if (ret) @@ -340,33 +299,33 @@ static int sof_audio_probe(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { .name = "sof_nau8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "adl_rt1019p_8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(2) | - SOF_NAU8825_NUM_HDMIDEV(4)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .name = "adl_nau8825_def", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "rpl_nau8825_def", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { } }; @@ -392,4 +351,3 @@ MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_NUVOTON_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); From 8efcd48646526dc41dc54a5044b434cd8a139d82 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:48 -0500 Subject: [PATCH 038/603] ASoC: Intel: sof_rt5682: use common module for sof_card_private initialization Use intel_board module to initialize sof_card_private structure Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 220 +++++++++++----------------- 1 file changed, 89 insertions(+), 131 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 640d17c6cd356..672636f07456d 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -27,37 +27,14 @@ #include "sof_board_helpers.h" #include "sof_maxim_common.h" #include "sof_realtek_common.h" -#include "sof_ssp_common.h" - -#define SOF_RT5682_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) -#define SOF_RT5682_SSP_CODEC_MASK (GENMASK(2, 0)) -#define SOF_RT5682_MCLK_EN BIT(3) -#define SOF_RT5682_SSP_AMP_SHIFT 6 -#define SOF_RT5682_SSP_AMP_MASK (GENMASK(8, 6)) -#define SOF_RT5682_SSP_AMP(quirk) \ - (((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK) -#define SOF_RT5682_MCLK_BYTCHT_EN BIT(9) -#define SOF_RT5682_NUM_HDMIDEV_SHIFT 10 -#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) -#define SOF_RT5682_NUM_HDMIDEV(quirk) \ - ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) - -/* BT audio offload: reserve 3 bits for future */ -#define SOF_BT_OFFLOAD_SSP_SHIFT 19 -#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(21, 19)) -#define SOF_BT_OFFLOAD_SSP(quirk) \ - (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) -#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22) - -/* HDMI capture*/ -#define SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT 27 -#define SOF_SSP_HDMI_CAPTURE_PRESENT_MASK (GENMASK(30, 27)) -#define SOF_HDMI_CAPTURE_SSP_MASK(quirk) \ - (((quirk) << SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT) & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) + +/* Driver-specific board quirks: from bit 0 to 7 */ +#define SOF_RT5682_MCLK_EN BIT(0) +#define SOF_RT5682_MCLK_BYTCHT_EN BIT(1) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0); + SOF_SSP_PORT_CODEC(0); static int sof_rt5682_quirk_cb(const struct dmi_system_id *id) { @@ -72,7 +49,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max"), }, - .driver_data = (void *)(SOF_RT5682_SSP_CODEC(2)), + .driver_data = (void *)(SOF_SSP_PORT_CODEC(2)), }, { .callback = sof_rt5682_quirk_cb, @@ -80,7 +57,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), DMI_MATCH(DMI_PRODUCT_NAME, "UP-CHT01"), }, - .driver_data = (void *)(SOF_RT5682_SSP_CODEC(2)), + .driver_data = (void *)(SOF_SSP_PORT_CODEC(2)), }, { .callback = sof_rt5682_quirk_cb, @@ -89,7 +66,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "WhiskeyLake Client"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(1)), + SOF_SSP_PORT_CODEC(1)), }, { .callback = sof_rt5682_quirk_cb, @@ -97,8 +74,8 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Hatch"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .callback = sof_rt5682_quirk_cb, @@ -107,7 +84,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0)), + SOF_SSP_PORT_CODEC(0)), }, { .callback = sof_rt5682_quirk_cb, @@ -116,9 +93,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98373_ALC5682I_I2S_UP4"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .callback = sof_rt5682_quirk_cb, @@ -128,9 +105,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_OEM_STRING, "AUDIO-ADL_MAX98373_ALC5682I_I2S"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .callback = sof_rt5682_quirk_cb, @@ -139,9 +116,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .callback = sof_rt5682_quirk_cb, @@ -150,9 +127,9 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S_AMP_SSP2"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .callback = sof_rt5682_quirk_cb, @@ -160,11 +137,11 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT + SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(0) | + SOF_NUM_IDISP_HDMI(3) | + SOF_SSP_PORT_BT_OFFLOAD(1) | + SOF_BT_OFFLOAD_PRESENT ), }, {} @@ -630,19 +607,29 @@ static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct sof_card_private *ctx; + bool is_legacy_cpu; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - if (pdev->id_entry && pdev->id_entry->driver_data) sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; dmi_check_system(sof_rt5682_quirk_table); - ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev); - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); + if (soc_intel_is_byt() || soc_intel_is_cht()) { + is_legacy_cpu = true; + + /* default quirk for legacy cpu */ + sof_rt5682_quirk = SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_BYTCHT_EN | + SOF_SSP_PORT_CODEC(2); + } + + dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); + + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, sof_rt5682_quirk); + if (!ctx) + return -ENOMEM; if (ctx->codec_type == CODEC_RT5650) { sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", @@ -653,23 +640,12 @@ static int sof_audio_probe(struct platform_device *pdev) ctx->amp_type = CODEC_RT5650; } - if (soc_intel_is_byt() || soc_intel_is_cht()) { + if (is_legacy_cpu) { ctx->rt5682.is_legacy_cpu = true; ctx->dmic_be_num = 0; /* HDMI is not supported by SOF on Baytrail/CherryTrail */ ctx->hdmi_num = 0; - /* default quirk for legacy cpu */ - sof_rt5682_quirk = SOF_RT5682_MCLK_EN | - SOF_RT5682_MCLK_BYTCHT_EN | - SOF_RT5682_SSP_CODEC(2); } else { - ctx->dmic_be_num = 2; - ctx->hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >> - SOF_RT5682_NUM_HDMIDEV_SHIFT; - /* default number of HDMI DAI's */ - if (!ctx->hdmi_num) - ctx->hdmi_num = 3; - if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; } @@ -694,23 +670,6 @@ static int sof_audio_probe(struct platform_device *pdev) } } - dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); - - /* port number/mask of peripherals attached to ssp interface */ - ctx->ssp_mask_hdmi_in = (sof_rt5682_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) >> - SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT; - - ctx->ssp_bt = (sof_rt5682_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ctx->ssp_amp = (sof_rt5682_quirk & SOF_RT5682_SSP_AMP_MASK) >> - SOF_RT5682_SSP_AMP_SHIFT; - - ctx->ssp_codec = sof_rt5682_quirk & SOF_RT5682_SSP_CODEC_MASK; - - if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) - ctx->bt_offload_present = true; - /* update dai_link */ ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_rt5682, ctx); if (ret) @@ -766,96 +725,96 @@ static const struct platform_device_id board_ids[] = { { .name = "cml_rt1015_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "jsl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "tgl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "adl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "adl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .name = "adl_rt5682_c1_h02", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(1) | - SOF_RT5682_NUM_HDMIDEV(3) | + SOF_SSP_PORT_CODEC(1) | + SOF_NUM_IDISP_HDMI(3) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_HDMI_CAPTURE_SSP_MASK(0x5)), + SOF_SSP_MASK_HDMI_CAPTURE(0x5)), }, { .name = "rpl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(2) | - SOF_RT5682_NUM_HDMIDEV(4)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(2) | + SOF_NUM_IDISP_HDMI(4)), }, { .name = "rpl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "rpl_rt5682_c1_h02", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(1) | - SOF_RT5682_NUM_HDMIDEV(3) | + SOF_SSP_PORT_CODEC(1) | + SOF_NUM_IDISP_HDMI(3) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_HDMI_CAPTURE_SSP_MASK(0x5)), + SOF_SSP_MASK_HDMI_CAPTURE(0x5)), }, { .name = "mtl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(3) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "mtl_mx98360_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(0) | - SOF_RT5682_SSP_AMP(1) | - SOF_RT5682_NUM_HDMIDEV(3)), + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(3)), }, { .name = "mtl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(0) | + SOF_NUM_IDISP_HDMI(3) | + SOF_SSP_PORT_BT_OFFLOAD(1) | + SOF_BT_OFFLOAD_PRESENT), }, { } }; @@ -881,4 +840,3 @@ MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); From 495d5b48d4b54efb4fa9f972f8bb599c780d209e Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:49 -0500 Subject: [PATCH 039/603] ASoC: Intel: sof_ssp_amp: use common module for sof_card_private initialization Use intel_board module to initialize sof_card_private structure. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 93 ++++++++-------------------- 1 file changed, 26 insertions(+), 67 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 0b8adab76755e..b99af8bc4ccc8 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -20,34 +20,12 @@ #include "sof_board_helpers.h" #include "sof_realtek_common.h" #include "sof_cirrus_common.h" -#include "sof_ssp_common.h" - -/* SSP port ID for speaker amplifier */ -#define SOF_AMPLIFIER_SSP(quirk) ((quirk) & GENMASK(3, 0)) -#define SOF_AMPLIFIER_SSP_MASK (GENMASK(3, 0)) - -/* HDMI capture*/ -#define SOF_HDMI_CAPTURE_SSP_MASK_SHIFT 4 -#define SOF_HDMI_CAPTURE_SSP_MASK_MASK (GENMASK(9, 4)) -#define SOF_HDMI_CAPTURE_SSP_MASK(quirk) \ - (((quirk) << SOF_HDMI_CAPTURE_SSP_MASK_SHIFT) & SOF_HDMI_CAPTURE_SSP_MASK_MASK) - -/* HDMI playback */ -#define SOF_HDMI_PLAYBACK_PRESENT BIT(13) -#define SOF_NO_OF_HDMI_PLAYBACK_SHIFT 14 -#define SOF_NO_OF_HDMI_PLAYBACK_MASK (GENMASK(16, 14)) -#define SOF_NO_OF_HDMI_PLAYBACK(quirk) \ - (((quirk) << SOF_NO_OF_HDMI_PLAYBACK_SHIFT) & SOF_NO_OF_HDMI_PLAYBACK_MASK) - -/* BT audio offload */ -#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(17) -#define SOF_BT_OFFLOAD_SSP_SHIFT 18 -#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(20, 18)) -#define SOF_BT_OFFLOAD_SSP(quirk) \ - (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) + +/* Driver-specific board quirks: from bit 0 to 7 */ +#define SOF_HDMI_PLAYBACK_PRESENT BIT(0) /* Default: SSP2 */ -static unsigned long sof_ssp_amp_quirk = SOF_AMPLIFIER_SSP(2); +static unsigned long sof_ssp_amp_quirk = SOF_SSP_PORT_AMP(2); static const struct dmi_system_id chromebook_platforms[] = { { @@ -136,45 +114,27 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) struct sof_card_private *ctx; int ret; - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - if (pdev->id_entry && pdev->id_entry->driver_data) sof_ssp_amp_quirk = (unsigned long)pdev->id_entry->driver_data; - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); - - if (dmi_check_system(chromebook_platforms) || mach->mach_params.dmic_num > 0) - ctx->dmic_be_num = 2; - else - ctx->dmic_be_num = 0; - - /* port number/mask of peripherals attached to ssp interface */ - ctx->ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >> - SOF_HDMI_CAPTURE_SSP_MASK_SHIFT; + dev_dbg(&pdev->dev, "sof_ssp_amp_quirk = %lx\n", sof_ssp_amp_quirk); - ctx->ssp_bt = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, sof_ssp_amp_quirk); + if (!ctx) + return -ENOMEM; - ctx->ssp_amp = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK; + if (!dmi_check_system(chromebook_platforms) && + (mach->mach_params.dmic_num == 0)) + ctx->dmic_be_num = 0; if (sof_ssp_amp_quirk & SOF_HDMI_PLAYBACK_PRESENT) { - ctx->hdmi_num = (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_PLAYBACK_MASK) >> - SOF_NO_OF_HDMI_PLAYBACK_SHIFT; - /* default number of HDMI DAI's */ - if (!ctx->hdmi_num) - ctx->hdmi_num = 3; - if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; } else { ctx->hdmi_num = 0; } - if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) - ctx->bt_offload_present = true; - ctx->link_order_overwrite = SSP_AMP_LINK_ORDER; if (ctx->ssp_mask_hdmi_in) { @@ -220,38 +180,38 @@ static const struct platform_device_id board_ids[] = { }, { .name = "tgl_rt1308_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_AMPLIFIER_SSP(2) | - SOF_HDMI_CAPTURE_SSP_MASK(0x22)), + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_AMP(2) | + SOF_SSP_MASK_HDMI_CAPTURE(0x22)), /* SSP 1 and SSP 5 are used for HDMI IN */ }, { .name = "adl_cs35l41", - .driver_data = (kernel_ulong_t)(SOF_AMPLIFIER_SSP(1) | - SOF_NO_OF_HDMI_PLAYBACK(4) | + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | SOF_HDMI_PLAYBACK_PRESENT | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "adl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | + .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NO_OF_HDMI_PLAYBACK(3) | + SOF_NUM_IDISP_HDMI(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { .name = "rpl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | + .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NO_OF_HDMI_PLAYBACK(3) | + SOF_NUM_IDISP_HDMI(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { .name = "mtl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | - /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NO_OF_HDMI_PLAYBACK(3) | - SOF_HDMI_PLAYBACK_PRESENT), + .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | + /* SSP 0 and SSP 2 are used for HDMI IN */ + SOF_NUM_IDISP_HDMI(3) | + SOF_HDMI_PLAYBACK_PRESENT), }, { } }; @@ -274,4 +234,3 @@ MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_CIRRUS_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); From a8f408dc8733d83c1cbd0ff238fc90d8b43db1c4 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:50 -0500 Subject: [PATCH 040/603] ASoC: Intel: sof_da7219: use common module for DAI link generation Use intel_board module to generate DAI link array and update num_links field in snd_soc_card structure. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_board_helpers.h | 13 + sound/soc/intel/boards/sof_da7219.c | 391 ++++++--------------- 3 files changed, 129 insertions(+), 276 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 18ac3ce0752ec..e5df64fec3199 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -624,6 +624,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH select SND_SOC_MAX98357A select SND_SOC_MAX98373_I2C select SND_SOC_DMIC + select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON select SND_SOC_INTEL_SOF_SSP_COMMON help diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 0d0a8d97843b0..1701481ab6ae0 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -83,6 +83,17 @@ enum { (((k6) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 5)) | \ (((k7) & SOF_LINK_IDS_MASK) << (SOF_LINK_IDS_SHIFT * 6))) +/* + * sof_da7219_private: private data for da7219 machine driver + * + * @is_jsl_board: true for JSL boards + * @pll_bypass: true for PLL bypass mode + */ +struct sof_da7219_private { + bool is_jsl_board; + bool pll_bypass; +}; + /* * sof_rt5682_private: private data for rt5682 machine driver * @@ -112,6 +123,7 @@ struct sof_rt5682_private { * @amp_link: pointer to speaker amplifier dai link * @link_order_overwrite: custom DAI link order * @link_id_overwrite: custom DAI link ID + * @da7219: private data for da7219 machine driver * @rt5682: private data for rt5682 machine driver */ struct sof_card_private { @@ -142,6 +154,7 @@ struct sof_card_private { unsigned long link_id_overwrite; union { + struct sof_da7219_private da7219; struct sof_rt5682_private rt5682; }; }; diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 290bf75bdaef6..f0cb1572c6c41 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -15,35 +15,24 @@ #include #include #include "../../codecs/da7219.h" -#include "hda_dsp_common.h" -#include "sof_hdmi_common.h" +#include "sof_board_helpers.h" #include "sof_maxim_common.h" -#include "sof_ssp_common.h" -/* Board Quirks */ -#define SOF_DA7219_JSL_BOARD BIT(2) +/* Driver-specific board quirks: from bit 0 to 7 */ +#define SOF_DA7219_JSL_BOARD BIT(0) #define DIALOG_CODEC_DAI "da7219-hifi" -struct card_private { - struct snd_soc_jack headset_jack; - struct sof_hdmi_private hdmi; - enum sof_ssp_codec codec_type; - enum sof_ssp_codec amp_type; - - unsigned int pll_bypass:1; -}; - static int platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; - struct card_private *ctx = snd_soc_card_get_drvdata(card); + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_dai *codec_dai; int ret = 0; - if (ctx->pll_bypass) + if (ctx->da7219.pll_bypass) return ret; /* PLL SRM mode */ @@ -74,8 +63,6 @@ static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Line Out"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), }; static const struct snd_soc_dapm_widget widgets[] = { @@ -83,14 +70,9 @@ static const struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), - - SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -102,9 +84,6 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "Platform Clock" }, { "Headset Mic", NULL, "Platform Clock" }, { "Line Out", NULL, "Platform Clock" }, - - /* digital mics */ - {"DMic", NULL, "SoC DMIC"}, }; static struct snd_soc_jack_pin jack_pins[] = { @@ -124,7 +103,7 @@ static struct snd_soc_jack_pin jack_pins[] = { static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { - struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); struct snd_soc_component *component = codec_dai->component; struct snd_soc_jack *jack = &ctx->headset_jack; @@ -157,7 +136,7 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ctx->pll_bypass = 1; + ctx->da7219.pll_bypass = true; } /* @@ -222,215 +201,11 @@ static const struct snd_soc_ops max98373_ops = { .hw_params = max98373_hw_params, }; -static int hdmi_init(struct snd_soc_pcm_runtime *rtd) -{ - struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - - ctx->hdmi.hdmi_comp = dai->component; - - return 0; -} - static int card_late_probe(struct snd_soc_card *card) { - struct card_private *ctx = snd_soc_card_get_drvdata(card); - - if (!ctx->hdmi.idisp_codec) - return 0; - - if (!ctx->hdmi.hdmi_comp) - return -EINVAL; - - return hda_dsp_hdmi_build_controls(card, ctx->hdmi.hdmi_comp); + return sof_intel_board_card_late_probe(card); } -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", DIALOG_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); - -SND_SOC_DAILINK_DEF(ssp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin"))); -SND_SOC_DAILINK_DEF(dummy_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("snd-soc-dummy", "snd-soc-dummy-dai"))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(dmic16k_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(idisp4_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp4 Pin"))); -SND_SOC_DAILINK_DEF(idisp4_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi4"))); - -SND_SOC_DAILINK_DEF(platform, /* subject to be overridden during probe */ - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -static struct snd_soc_dai_link jsl_dais[] = { - /* Back End DAI links */ - { - .name = "SSP1-Codec", - .id = 0, - .ignore_pmdown_time = 1, - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, /* IV feedback */ - SND_SOC_DAILINK_REG(ssp1_pin, max_98373_components, platform), - }, - { - .name = "SSP0-Codec", - .id = 1, - .no_pcm = 1, - .init = da7219_codec_init, - .ignore_pmdown_time = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - .name = "dmic01", - .id = 2, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, - { - .name = "dmic16k", - .id = 6, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform), - } -}; - -static struct snd_soc_dai_link adl_dais[] = { - /* Back End DAI links */ - { - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .init = da7219_codec_init, - .ignore_pmdown_time = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - .name = "dmic01", - .id = 1, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "dmic16k", - .id = 2, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, - { - .name = "iDisp4", - .id = 6, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp4_pin, idisp4_codec, platform), - }, - { - .name = "SSP1-Codec", - .id = 7, - .no_pcm = 1, - .dpcm_playback = 1, - /* feedback stream or firmware-generated echo reference */ - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, max_98373_components, platform), - }, - { - .name = "SSP2-BT", - .id = 8, - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp2_pin, dummy_codec, platform), - }, -}; - static struct snd_soc_card card_da7219 = { .name = "da7219", /* the sof- prefix is added by the core */ .owner = THIS_MODULE, @@ -444,28 +219,100 @@ static struct snd_soc_card card_da7219 = { .late_probe = card_late_probe, }; +static struct snd_soc_dai_link_component da7219_component[] = { + { + .name = "i2c-DLGS7219:00", + .dai_name = DIALOG_CODEC_DAI, + } +}; + +static int +sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx) +{ + int ret; + + ret = sof_intel_board_set_dai_link(dev, card, ctx); + if (ret) + return ret; + + if (!ctx->codec_link) { + dev_err(dev, "codec link not available"); + return -EINVAL; + } + + /* codec-specific fields for headphone codec */ + ctx->codec_link->codecs = da7219_component; + ctx->codec_link->num_codecs = ARRAY_SIZE(da7219_component); + ctx->codec_link->init = da7219_codec_init; + + if (ctx->amp_type == CODEC_NONE) + return 0; + + if (!ctx->amp_link) { + dev_err(dev, "amp link not available"); + return -EINVAL; + } + + /* codec-specific fields for speaker amplifier */ + switch (ctx->amp_type) { + case CODEC_MAX98360A: + max_98360a_dai_link(ctx->amp_link); + break; + case CODEC_MAX98373: + ctx->amp_link->codecs = max_98373_components; + ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components); + ctx->amp_link->init = max_98373_spk_codec_init; + if (ctx->da7219.is_jsl_board) { + ctx->amp_link->ops = &max98373_ops; /* use local ops */ + } else { + /* TBD: implement the amp for later platform */ + dev_err(dev, "max98373 not support yet\n"); + return -EINVAL; + } + break; + default: + dev_err(dev, "invalid amp type %d\n", ctx->amp_type); + return -EINVAL; + } + + return 0; +} + +#define JSL_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \ + SOF_LINK_CODEC, \ + SOF_LINK_DMIC01, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_DMIC16K, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE) + static int audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; - struct snd_soc_dai_link *dai_links; - struct card_private *ctx; + struct sof_card_private *ctx; unsigned long board_quirk = 0; - int ret, amp_idx; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; + int ret; if (pdev->id_entry && pdev->id_entry->driver_data) board_quirk = (unsigned long)pdev->id_entry->driver_data; - ctx->codec_type = sof_ssp_detect_codec_type(&pdev->dev); - ctx->amp_type = sof_ssp_detect_amp_type(&pdev->dev); + dev_dbg(&pdev->dev, "board_quirk = %lx\n", board_quirk); + + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, board_quirk); + if (!ctx) + return -ENOMEM; if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; if (board_quirk & SOF_DA7219_JSL_BOARD) { + ctx->da7219.is_jsl_board = true; + + /* overwrite the DAI link order for JSL boards */ + ctx->link_order_overwrite = JSL_LINK_ORDER; + /* backward-compatible with existing devices */ switch (ctx->amp_type) { case CODEC_MAX98360A: @@ -480,46 +327,27 @@ static int audio_probe(struct platform_device *pdev) default: break; } - - dai_links = jsl_dais; - amp_idx = 0; - - card_da7219.num_links = ARRAY_SIZE(jsl_dais); - } else { - dai_links = adl_dais; - amp_idx = 7; - - card_da7219.num_links = ARRAY_SIZE(adl_dais); } - dev_dbg(&pdev->dev, "board_quirk = %lx\n", board_quirk); + /* update dai_link */ + ret = sof_card_dai_links_create(&pdev->dev, &card_da7219, ctx); + if (ret) + return ret; - /* speaker amp */ + /* update codec_conf */ switch (ctx->amp_type) { - case CODEC_MAX98360A: - max_98360a_dai_link(&dai_links[amp_idx]); - break; case CODEC_MAX98373: - dai_links[amp_idx].codecs = max_98373_components; - dai_links[amp_idx].num_codecs = ARRAY_SIZE(max_98373_components); - dai_links[amp_idx].init = max_98373_spk_codec_init; - if (board_quirk & SOF_DA7219_JSL_BOARD) { - dai_links[amp_idx].ops = &max98373_ops; /* use local ops */ - } else { - /* TBD: implement the amp for later platform */ - dev_err(&pdev->dev, "max98373 not support yet\n"); - return -EINVAL; - } - max_98373_set_codec_conf(&card_da7219); break; + case CODEC_MAX98360A: + case CODEC_NONE: + /* no codec conf required */ + break; default: dev_err(&pdev->dev, "invalid amp type %d\n", ctx->amp_type); return -EINVAL; } - card_da7219.dai_link = dai_links; - card_da7219.dev = &pdev->dev; ret = snd_soc_fixup_dai_links_platform_name(&card_da7219, @@ -535,19 +363,31 @@ static int audio_probe(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { .name = "jsl_mx98373_da7219", - .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD), + .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD | + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "jsl_mx98360_da7219", - .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD), + .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD | + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), }, { .name = "adl_mx98360_da7219", - /* no quirk needed for this board */ + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { .name = "rpl_mx98360_da7219", - /* no quirk needed for this board */ + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_SSP_PORT_BT_OFFLOAD(2) | + SOF_BT_OFFLOAD_PRESENT), }, { } }; @@ -568,6 +408,5 @@ MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver for Dialog codec"); MODULE_AUTHOR("Yong Zhi "); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL v2"); -MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); From a7963f6ecc4be8a331d7740cb276651f2045e92c Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:51 -0500 Subject: [PATCH 041/603] ASoC: Intel: sof_da7219: add codec exit function Add exit function to headphone codec dai link. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index f0cb1572c6c41..3d9c48107f587 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -167,6 +167,13 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static void da7219_codec_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_set_jack(component, NULL, NULL); +} + static int max98373_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -245,6 +252,7 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, ctx->codec_link->codecs = da7219_component; ctx->codec_link->num_codecs = ARRAY_SIZE(da7219_component); ctx->codec_link->init = da7219_codec_init; + ctx->codec_link->exit = da7219_codec_exit; if (ctx->amp_type == CODEC_NONE) return 0; From ed61b2ef5aa0036ef1dc593c191e762386a92cc5 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:52 -0500 Subject: [PATCH 042/603] ASoC: Intel: sof_da7219: add SOF_DA7219_MCLK_EN quirk PLL bypass mode requires mclk to be present. However, mclk pin is not connected in JSL boards. Here we add the SOF_DA7219_MCLK_EN quirk to improve driver readability. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-15-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.h | 2 ++ sound/soc/intel/boards/sof_da7219.c | 13 ++++++++++--- 2 files changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 1701481ab6ae0..2f27ad8726f8b 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -87,10 +87,12 @@ enum { * sof_da7219_private: private data for da7219 machine driver * * @is_jsl_board: true for JSL boards + * @mclk_en: true for mclk pin is connected * @pll_bypass: true for PLL bypass mode */ struct sof_da7219_private { bool is_jsl_board; + bool mclk_en; bool pll_bypass; }; diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 3d9c48107f587..4417e2b7adfda 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -20,6 +20,7 @@ /* Driver-specific board quirks: from bit 0 to 7 */ #define SOF_DA7219_JSL_BOARD BIT(0) +#define SOF_DA7219_MCLK_EN BIT(1) #define DIALOG_CODEC_DAI "da7219-hifi" @@ -126,7 +127,8 @@ static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd) * Use PLL bypass mode if MCLK is available, be sure to set the * frequency of MCLK to 12.288 or 24.576MHz on topology side. */ - if (mclk_rate == 12288000 || mclk_rate == 24576000) { + if (ctx->da7219.mclk_en && + (mclk_rate == 12288000 || mclk_rate == 24576000)) { /* PLL bypass mode */ dev_dbg(rtd->dev, "pll bypass mode, mclk rate %d\n", mclk_rate); @@ -337,6 +339,9 @@ static int audio_probe(struct platform_device *pdev) } } + if (board_quirk & SOF_DA7219_MCLK_EN) + ctx->da7219.mclk_en = true; + /* update dai_link */ ret = sof_card_dai_links_create(&pdev->dev, &card_da7219, ctx); if (ret) @@ -383,7 +388,8 @@ static const struct platform_device_id board_ids[] = { }, { .name = "adl_mx98360_da7219", - .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN | + SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1) | SOF_NUM_IDISP_HDMI(4) | SOF_SSP_PORT_BT_OFFLOAD(2) | @@ -391,7 +397,8 @@ static const struct platform_device_id board_ids[] = { }, { .name = "rpl_mx98360_da7219", - .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN | + SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1) | SOF_NUM_IDISP_HDMI(4) | SOF_SSP_PORT_BT_OFFLOAD(2) | From 45728dc6df48118b4e49f5abac4de87935bfc255 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:53 -0500 Subject: [PATCH 043/603] ASoC: Intel: sof_da7219: board id cleanup for jsl boards Introduce "jsl_da7219_def" board to reduce the number of jsl board configs. This config could support all boards which implement headphone codec on SSP0 and speaker amplifiers on SSP1. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-16-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 8 +------- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 4 ++-- 2 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 4417e2b7adfda..3057798a0dcad 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -375,13 +375,7 @@ static int audio_probe(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { - .name = "jsl_mx98373_da7219", - .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD | - SOF_SSP_PORT_CODEC(0) | - SOF_SSP_PORT_AMP(1)), - }, - { - .name = "jsl_mx98360_da7219", + .name = "jsl_da7219_def", .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD | SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1)), diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index a6ac2525df175..d4b397c53bcc8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -52,14 +52,14 @@ static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { { .id = "DLGS7219", - .drv_name = "jsl_mx98373_da7219", + .drv_name = "jsl_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98373_spk, .sof_tplg_filename = "sof-jsl-da7219.tplg", }, { .id = "DLGS7219", - .drv_name = "jsl_mx98360_da7219", + .drv_name = "jsl_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-da7219-mx98360a.tplg", From ef771be889f0e5e036ff54e4f0178b6eba177d56 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:54 -0500 Subject: [PATCH 044/603] ASoC: Intel: sof_da7219: board id cleanup for adl boards Introduce "adl_da7219_def" board to reduce the number of jsl board configs. This config could support all boards which implement headphone codec on SSP0 and speaker amplifiers on SSP1. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-17-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 2 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 3057798a0dcad..0da7aa2e31c29 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -381,7 +381,7 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_PORT_AMP(1)), }, { - .name = "adl_mx98360_da7219", + .name = "adl_da7219_def", .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN | SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1) | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 0da79a3ba1f0b..6c710e9a26f19 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -616,7 +616,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "adl_mx98360_da7219", + .drv_name = "adl_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, .sof_tplg_filename = "sof-adl-max98360a-da7219.tplg", From 9ca54c3356f488e7800b6bcdbccb1f6de90e1621 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:55 -0500 Subject: [PATCH 045/603] ASoC: Intel: sof_da7219: board id cleanup for rpl boards Introduce "rpl_da7219_def" board to reduce the number of rpl board configs. This config could support all boards which implement headphone codec on SSP0 and speaker amplifiers on SSP1. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-18-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 2 +- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 0da7aa2e31c29..fd3a7be993c19 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -390,7 +390,7 @@ static const struct platform_device_id board_ids[] = { SOF_BT_OFFLOAD_PRESENT), }, { - .name = "rpl_mx98360_da7219", + .name = "rpl_da7219_def", .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN | SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1) | diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index c43e537e60d2c..77c917897c8d1 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -457,7 +457,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "rpl_mx98360_da7219", + .drv_name = "rpl_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rpl_max98360a_amp, .sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg", From b497654f1d3d95c37c8f2c6950047e022be04a2e Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:56 -0500 Subject: [PATCH 046/603] ASoC: Intel: sof_rt5682: remove unnecessary idisp HDMI quirk Remove SOF_NUM_IDISP_HDMI(3) from board quirks since the value is 3 if not defined. Reviewed-by: Pierre-Louis Bossart Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-19-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 672636f07456d..f80fdbd8fdac5 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -139,7 +139,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { .driver_data = (void *)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(2) | SOF_SSP_PORT_AMP(0) | - SOF_NUM_IDISP_HDMI(3) | SOF_SSP_PORT_BT_OFFLOAD(1) | SOF_BT_OFFLOAD_PRESENT ), @@ -692,11 +691,11 @@ static int sof_audio_probe(struct platform_device *pdev) case CODEC_RT1015P: sof_rt1015p_codec_conf(&sof_audio_card_rt5682); break; - case CODEC_NONE: case CODEC_MAX98357A: case CODEC_MAX98360A: case CODEC_RT1019P: case CODEC_RT5650: + case CODEC_NONE: /* no codec conf required */ break; default: @@ -763,7 +762,6 @@ static const struct platform_device_id board_ids[] = { .name = "adl_rt5682_c1_h02", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(1) | - SOF_NUM_IDISP_HDMI(3) | /* SSP 0 and SSP 2 are used for HDMI IN */ SOF_SSP_MASK_HDMI_CAPTURE(0x5)), }, @@ -787,7 +785,6 @@ static const struct platform_device_id board_ids[] = { .name = "rpl_rt5682_c1_h02", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(1) | - SOF_NUM_IDISP_HDMI(3) | /* SSP 0 and SSP 2 are used for HDMI IN */ SOF_SSP_MASK_HDMI_CAPTURE(0x5)), }, @@ -796,7 +793,6 @@ static const struct platform_device_id board_ids[] = { .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1) | - SOF_NUM_IDISP_HDMI(3) | SOF_SSP_PORT_BT_OFFLOAD(2) | SOF_BT_OFFLOAD_PRESENT), }, @@ -804,15 +800,13 @@ static const struct platform_device_id board_ids[] = { .name = "mtl_mx98360_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(0) | - SOF_SSP_PORT_AMP(1) | - SOF_NUM_IDISP_HDMI(3)), + SOF_SSP_PORT_AMP(1)), }, { .name = "mtl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(2) | SOF_SSP_PORT_AMP(0) | - SOF_NUM_IDISP_HDMI(3) | SOF_SSP_PORT_BT_OFFLOAD(1) | SOF_BT_OFFLOAD_PRESENT), }, From 16e5700644bda0d236288f17fd68b15a1e2e6f2c Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:57 -0500 Subject: [PATCH 047/603] ASoC: Intel: sof_ssp_amp: remove unnecessary idisp HDMI quirk Remove SOF_NUM_IDISP_HDMI(3) from board quirks since the value is 3 if not defined. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-20-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index b99af8bc4ccc8..206c9b7238059 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -152,8 +152,8 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) case CODEC_CS35L41: cs35l41_set_codec_conf(&sof_ssp_amp_card); break; - case CODEC_NONE: case CODEC_RT1308: + case CODEC_NONE: /* no codec conf required */ break; default: @@ -196,21 +196,18 @@ static const struct platform_device_id board_ids[] = { .name = "adl_lt6911_hdmi_ssp", .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NUM_IDISP_HDMI(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { .name = "rpl_lt6911_hdmi_ssp", .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NUM_IDISP_HDMI(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { .name = "mtl_lt6911_hdmi_ssp", .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | /* SSP 0 and SSP 2 are used for HDMI IN */ - SOF_NUM_IDISP_HDMI(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { } From 2b384bcd2cb727edb5647692134f4eec2f0a32ae Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:58 -0500 Subject: [PATCH 048/603] ASoC: Intel: sof_nau8825: remove sof_nau8825 board id Remove sof_nau8825 board id and use adl_nau8825_def instead since SSP port assignment is the same. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-21-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 10 +--------- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- 2 files changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index b9db6e236a931..23fe8b4015cce 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -271,10 +271,10 @@ static int sof_audio_probe(struct platform_device *pdev) case CODEC_RT1015P: sof_rt1015p_codec_conf(&sof_audio_card_nau8825); break; - case CODEC_NONE: case CODEC_MAX98360A: case CODEC_NAU8318: case CODEC_RT1019P: + case CODEC_NONE: /* no codec conf required */ break; default: @@ -297,14 +297,6 @@ static int sof_audio_probe(struct platform_device *pdev) } static const struct platform_device_id board_ids[] = { - { - .name = "sof_nau8825", - .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | - SOF_NUM_IDISP_HDMI(4) | - SOF_SSP_PORT_BT_OFFLOAD(2) | - SOF_BT_OFFLOAD_PRESENT), - - }, { .name = "adl_rt1019p_8825", .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 6c710e9a26f19..7ce8aade07d70 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -563,7 +563,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10508825", - .drv_name = "sof_nau8825", + .drv_name = "adl_nau8825_def", .sof_tplg_filename = "sof-adl-nau8825.tplg", }, { From 61cafaeab5bca2d3e6a68ee8fa92b5c10b8610ca Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 25 Mar 2024 17:10:59 -0500 Subject: [PATCH 049/603] ASoC: Intel: sof_rt5682: board id cleanup for cml boards Introduce "cml_rt5682_def" for cml boards which implement headphone codec on SSP0 and speaker amplifiers on SSP1. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240325221059.206042-22-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index f80fdbd8fdac5..aadd341a202cc 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -722,7 +722,7 @@ static const struct platform_device_id board_ids[] = { .name = "sof_rt5682", }, { - .name = "cml_rt1015_rt5682", + .name = "cml_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1)), diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 5eab17820532e..d47a548959eaf 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -49,21 +49,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }, { .id = "10EC5682", - .drv_name = "cml_rt1015_rt5682", + .drv_name = "cml_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rt1015_spk_codecs, .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", }, { .id = "10EC5682", - .drv_name = "sof_rt5682", + .drv_name = "cml_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &max98357a_spk_codecs, .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", }, { .id = "10EC5682", - .drv_name = "sof_rt5682", + .drv_name = "cml_rt5682_def", .sof_tplg_filename = "sof-cml-rt5682.tplg", }, { From 1d717123bb1a7555a432e51a41709badf8545dba Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Mon, 25 Mar 2024 19:59:34 -0600 Subject: [PATCH 050/603] ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning Use the `DEFINE_FLEX()` helper for an on-stack definition of a flexible structure where the size of the flexible-array member is known at compile-time, and refactor the rest of the code, accordingly. So, with these changes, fix the following warning: sound/firewire/amdtp-stream.c:1184:46: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] Link: https://github.com/KSPP/linux/issues/202 Signed-off-by: Gustavo A. R. Silva Reviewed-by: Takashi Sakamoto Message-ID: Signed-off-by: Takashi Iwai --- sound/firewire/amdtp-stream.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index c9f153f85ae6b..d35d0a420ee08 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -1180,13 +1180,11 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ (void)fw_card_read_cycle_time(fw_parent_device(s->unit)->card, &curr_cycle_time); for (i = 0; i < packets; ++i) { - struct { - struct fw_iso_packet params; - __be32 header[CIP_HEADER_QUADLETS]; - } template = { {0}, {0} }; + DEFINE_FLEX(struct fw_iso_packet, template, header, + header_length, CIP_HEADER_QUADLETS); bool sched_irq = false; - build_it_pkt_header(s, desc->cycle, &template.params, pkt_header_length, + build_it_pkt_header(s, desc->cycle, template, pkt_header_length, desc->data_blocks, desc->data_block_counter, desc->syt, i, curr_cycle_time); @@ -1198,7 +1196,7 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ } } - if (queue_out_packet(s, &template.params, sched_irq) < 0) { + if (queue_out_packet(s, template, sched_irq) < 0) { cancel_stream(s); return; } From 6a92834166b16babd70e99c3e0ce9262893ad6ae Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 25 Mar 2024 21:13:41 +0200 Subject: [PATCH 051/603] ASoC: fsl: imx-es8328: Remove leftover gpio initialisation The gpio field is not used anymore, remove the leftover. This also fixes the compilation error after the ... Fixes: 9855f05e5536 ("ASoC: fsl: imx-es8328: Switch to using gpiod API") Signed-off-by: Andy Shevchenko Link: https://msgid.link/r/20240325191341.3977321-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-es8328.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index d67b8a149bfff..5b9648f3b087f 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -28,7 +28,6 @@ struct imx_es8328_data { static struct snd_soc_jack_gpio headset_jack_gpios[] = { { - .gpio = -1, .name = "headset-gpio", .report = SND_JACK_HEADSET, .invert = 0, From ea5fee227ff3dae209062ac9544906debe1e9ac1 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 25 Mar 2024 17:19:25 -0500 Subject: [PATCH 052/603] ASoC: hdac_hda: improve error logs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We have a couple of duplicate logs and missing information, add __func__ consistently and make sure useful error codes are logged. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://msgid.link/r/20240325221925.206507-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdac_hda.c | 44 ++++++++++++++++++++----------------- 1 file changed, 24 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 6aa3223985be8..29c88de5508b8 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -230,7 +230,8 @@ static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, format_val = snd_hdac_stream_format(params_channels(params), bits, params_rate(params)); if (!format_val) { dev_err(dai->dev, - "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + "%s: invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + __func__, params_rate(params), params_channels(params), params_format(params), maxbps); @@ -266,14 +267,12 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct hdac_device *hdev; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; int ret = 0; hda_pvt = snd_soc_component_get_drvdata(component); - hdev = &hda_pvt->codec->core; pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); if (!pcm) return -EINVAL; @@ -286,7 +285,7 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, ret = snd_hda_codec_prepare(hda_pvt->codec, hda_stream, stream, format_val, substream); if (ret < 0) - dev_err(&hdev->dev, "codec prepare failed %d\n", ret); + dev_err(dai->dev, "%s: failed %d\n", __func__, ret); return ret; } @@ -298,6 +297,7 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, struct hdac_hda_priv *hda_pvt; struct hda_pcm_stream *hda_stream; struct hda_pcm *pcm; + int ret; hda_pvt = snd_soc_component_get_drvdata(component); pcm = snd_soc_find_pcm_from_dai(hda_pvt, dai); @@ -308,7 +308,11 @@ static int hdac_hda_dai_open(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - return hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); + ret = hda_stream->ops.open(hda_stream, hda_pvt->codec, substream); + if (ret < 0) + dev_err(dai->dev, "%s: failed %d\n", __func__, ret); + + return ret; } static void hdac_hda_dai_close(struct snd_pcm_substream *substream, @@ -367,7 +371,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, pcm_name = "HDMI 3"; break; default: - dev_err(&hcodec->core.dev, "invalid dai id %d\n", dai->id); + dev_err(dai->dev, "%s: invalid dai id %d\n", __func__, dai->id); return NULL; } @@ -381,7 +385,7 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, } } - dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name); + dev_err(dai->dev, "%s: didn't find PCM for DAI %s\n", __func__, dai->name); return NULL; } @@ -411,7 +415,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return -EIO; } @@ -429,7 +433,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_device_new(hcodec->bus, component->card->snd_card, hdev->addr, hcodec, true); if (ret < 0) { - dev_err(&hdev->dev, "failed to create hda codec %d\n", ret); + dev_err(&hdev->dev, "%s: failed to create hda codec %d\n", __func__, ret); goto error_no_pm; } @@ -446,7 +450,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (fw) { ret = snd_hda_load_patch(hcodec->bus, fw->size, fw->data); if (ret < 0) { - dev_err(&hdev->dev, "failed to load hda patch %d\n", ret); + dev_err(&hdev->dev, "%s: failed to load hda patch %d\n", __func__, ret); goto error_no_pm; } release_firmware(fw); @@ -470,13 +474,13 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name); if (ret < 0) { - dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name); + dev_err(&hdev->dev, "%s: name failed %s\n", __func__, hcodec->preset->name); goto error_pm; } ret = snd_hdac_regmap_init(&hcodec->core); if (ret < 0) { - dev_err(&hdev->dev, "regmap init failed\n"); + dev_err(&hdev->dev, "%s: regmap init failed\n", __func__); goto error_pm; } @@ -484,16 +488,16 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (patch) { ret = patch(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "patch failed %d\n", ret); + dev_err(&hdev->dev, "%s: patch failed %d\n", __func__, ret); goto error_regmap; } } else { - dev_dbg(&hdev->dev, "no patch file found\n"); + dev_dbg(&hdev->dev, "%s: no patch file found\n", __func__); } ret = snd_hda_codec_parse_pcms(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + dev_err(&hdev->dev, "%s: unable to map pcms to dai %d\n", __func__, ret); goto error_patch; } @@ -501,8 +505,8 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (!is_hdmi_codec(hcodec)) { ret = snd_hda_codec_build_controls(hcodec); if (ret < 0) { - dev_err(&hdev->dev, "unable to create controls %d\n", - ret); + dev_err(&hdev->dev, "%s: unable to create controls %d\n", + __func__, ret); goto error_patch; } } @@ -548,7 +552,7 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component) hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return; } @@ -624,7 +628,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) /* hold the ref while we probe */ hlink = snd_hdac_ext_bus_get_hlink_by_name(hdev->bus, dev_name(&hdev->dev)); if (!hlink) { - dev_err(&hdev->dev, "hdac link not found\n"); + dev_err(&hdev->dev, "%s: hdac link not found\n", __func__); return -EIO; } snd_hdac_ext_bus_link_get(hdev->bus, hlink); @@ -640,7 +644,7 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) ARRAY_SIZE(hdac_hda_dais)); if (ret < 0) { - dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); + dev_err(&hdev->dev, "%s: failed to register HDA codec %d\n", __func__, ret); return ret; } From 1e90a846493c716e3e6b4d901fc0844e9eea6430 Mon Sep 17 00:00:00 2001 From: John Watts Date: Tue, 26 Mar 2024 22:48:00 +1100 Subject: [PATCH 053/603] ASoC: soc-dai: Note valid values of sysclock direction Clock direction is marked as 'unsigned int' but only two values are currently used in practice. Note these down in the documentation. Signed-off-by: John Watts Link: https://msgid.link/r/20240326-dai_mclk_hint-v1-1-653cbd2d78d9@jookia.org Signed-off-by: Mark Brown --- sound/soc/soc-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 6f8773a8fc053..fefe394dce72d 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -45,7 +45,7 @@ static inline int _soc_dai_ret(struct snd_soc_dai *dai, * @dai: DAI * @clk_id: DAI specific clock ID * @freq: new clock frequency in Hz - * @dir: new clock direction - input/output. + * @dir: new clock direction (SND_SOC_CLOCK_IN or SND_SOC_CLOCK_OUT) * * Configures the DAI master (MCLK) or system (SYSCLK) clocking. */ From cce4cbb157493483f03c21213753b66425a31430 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Tue, 5 Mar 2024 15:36:29 +0100 Subject: [PATCH 054/603] ASoC: dt-bindings: Add Rockchip RK3308 internal audio codec Add device tree bindings document for the internal audio codec of the Rockchip RK3308 SoC. Reviewed-by: Rob Herring Signed-off-by: Luca Ceresoli Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-2-312acdbe628f@bootlin.com Signed-off-by: Mark Brown --- .../bindings/sound/rockchip,rk3308-codec.yaml | 98 +++++++++++++++++++ MAINTAINERS | 5 + 2 files changed, 103 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml new file mode 100644 index 0000000000000..ecf3d7d968c80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3308-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip RK3308 Internal Codec + +description: | + This is the audio codec embedded in the Rockchip RK3308 + SoC. It has 8 24-bit ADCs and 2 24-bit DACs. The maximum supported + sampling rate is 192 kHz. + + It is connected internally to one out of a selection of the internal I2S + controllers. + + The RK3308 audio codec has 8 independent capture channels, but some + features work on stereo pairs called groups: + * grp 0 -- MIC1 / MIC2 + * grp 1 -- MIC3 / MIC4 + * grp 2 -- MIC5 / MIC6 + * grp 3 -- MIC7 / MIC8 + +maintainers: + - Luca Ceresoli + +properties: + compatible: + const: rockchip,rk3308-codec + + reg: + maxItems: 1 + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + Phandle to the General Register Files (GRF) + + clocks: + items: + - description: clock for TX + - description: clock for RX + - description: AHB clock driving the interface + + clock-names: + items: + - const: mclk_tx + - const: mclk_rx + - const: hclk + + resets: + maxItems: 1 + + reset-names: + items: + - const: codec + + "#sound-dai-cells": + const: 0 + + rockchip,micbias-avdd-percent: + description: | + Voltage setting for the MICBIAS pins expressed as a percentage of + AVDD. + + E.g. if rockchip,micbias-avdd-percent = 85 and AVDD = 3v3, then the + MIC BIAS voltage will be 3.3 V * 85% = 2.805 V. + + enum: [ 50, 55, 60, 65, 70, 75, 80, 85 ] + +required: + - compatible + - reg + - rockchip,grf + - clocks + - resets + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include + + audio_codec: audio-codec@ff560000 { + compatible = "rockchip,rk3308-codec"; + reg = <0xff560000 0x10000>; + rockchip,grf = <&grf>; + clock-names = "mclk_tx", "mclk_rx", "hclk"; + clocks = <&cru SCLK_I2S2_8CH_TX_OUT>, + <&cru SCLK_I2S2_8CH_RX_OUT>, + <&cru PCLK_ACODEC>; + reset-names = "codec"; + resets = <&cru SRST_ACODEC_P>; + #sound-dai-cells = <0>; + }; + +... diff --git a/MAINTAINERS b/MAINTAINERS index aa3b947fb0801..89a59660bf0f8 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -19046,6 +19046,11 @@ S: Maintained F: Documentation/devicetree/bindings/media/rockchip-rga.yaml F: drivers/media/platform/rockchip/rga/ +ROCKCHIP RK3308 INTERNAL AUDIO CODEC +M: Luca Ceresoli +S: Maintained +F: Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml + ROCKCHIP VIDEO DECODER DRIVER M: Ezequiel Garcia L: linux-media@vger.kernel.org From d75a21611a6e723d81db3f827e131ad39b69186c Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Tue, 5 Mar 2024 15:36:30 +0100 Subject: [PATCH 055/603] ASoC: core: add SOC_DOUBLE_RANGE_TLV() helper macro No macro currently allows handling a stereo control that has left and right in the same register and whose minimum register value is not zero. Add one that does that. Note that even though the snd_soc_*_volsw_range() look more appropriate given the _range suffix, they are not suitable because they don't honor the two shift values. The snd_soc_*_volsw() look more generic and are suitable for the task. Signed-off-by: Luca Ceresoli Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-3-312acdbe628f@bootlin.com Signed-off-by: Mark Brown --- include/sound/soc.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 39613b406b1d0..0376f7e4c15d9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -149,6 +149,18 @@ {.reg = xreg, .rreg = xreg, \ .shift = shift_left, .rshift = shift_right, \ .max = xmax, .min = xmin} } +#define SOC_DOUBLE_RANGE_TLV(xname, xreg, xshift_left, xshift_right, xmin, xmax, \ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, \ + .shift = xshift_left, .rshift = xshift_right, \ + .min = xmin, .max = xmax, .invert = xinvert} } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ From 4ed0915f5bc4bcc81bca783a5b984f3d81e9764e Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Tue, 5 Mar 2024 15:36:31 +0100 Subject: [PATCH 056/603] ASoC: codecs: Add RK3308 internal audio codec driver Add driver for the internal audio codec of the Rockchip RK3308 SoC. Initially based on the vendor kernel driver [0], with lots of cleanups, fixes, improvements, conversion to DAPM and removal of some features. [0] https://github.com/rockchip-linux/kernel/blob/develop-4.19/sound/soc/codecs/rk3308_codec.c Signed-off-by: Luca Ceresoli Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-4-312acdbe628f@bootlin.com Signed-off-by: Mark Brown --- MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 11 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk3308_codec.c | 974 ++++++++++++++++++++++++++++++++ sound/soc/codecs/rk3308_codec.h | 579 +++++++++++++++++++ 5 files changed, 1568 insertions(+) create mode 100644 sound/soc/codecs/rk3308_codec.c create mode 100644 sound/soc/codecs/rk3308_codec.h diff --git a/MAINTAINERS b/MAINTAINERS index 89a59660bf0f8..74ab768c0f6f1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -19050,6 +19050,8 @@ ROCKCHIP RK3308 INTERNAL AUDIO CODEC M: Luca Ceresoli S: Maintained F: Documentation/devicetree/bindings/sound/rockchip,rk3308-codec.yaml +F: sound/soc/codecs/rk3308_codec.c +F: sound/soc/codecs/rk3308_codec.h ROCKCHIP VIDEO DECODER DRIVER M: Ezequiel Garcia diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f78ea2f86fa64..1752814fffdda 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,6 +180,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_PEB2466 + imply SND_SOC_RK3308 imply SND_SOC_RK3328 imply SND_SOC_RK817 imply SND_SOC_RT274 @@ -1433,6 +1434,16 @@ config SND_SOC_PEB2466 To compile this driver as a module, choose M here: the module will be called snd-soc-peb2466. +config SND_SOC_RK3308 + tristate "Rockchip RK3308 audio CODEC" + select REGMAP_MMIO + help + This is a device driver for the audio codec embedded in the + Rockchip RK3308 SoC. + + It has 8 24-bit ADCs and 2 24-bit DACs. The maximum supported + sampling rate is 192 kHz. + config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7c075539dc476..9ba24fb870b11 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -205,6 +205,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-peb2466-objs := peb2466.o +snd-soc-rk3308-objs := rk3308_codec.o snd-soc-rk3328-objs := rk3328_codec.o snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o @@ -595,6 +596,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_PEB2466) += snd-soc-peb2466.o +obj-$(CONFIG_SND_SOC_RK3308) += snd-soc-rk3308.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o diff --git a/sound/soc/codecs/rk3308_codec.c b/sound/soc/codecs/rk3308_codec.c new file mode 100644 index 0000000000000..9d3e4691a7b53 --- /dev/null +++ b/sound/soc/codecs/rk3308_codec.c @@ -0,0 +1,974 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Rockchip RK3308 internal audio codec driver + * + * Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + * Copyright (c) 2024, Vivax-Metrotech Ltd + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "rk3308_codec.h" + +#define ADC_LR_GROUP_MAX 4 + +#define GRF_CHIP_ID 0x800 + +enum { + ACODEC_VERSION_A = 'A', + ACODEC_VERSION_B, + ACODEC_VERSION_C, +}; + +struct rk3308_codec_priv { + const struct device *dev; + struct regmap *regmap; + struct regmap *grf; + struct reset_control *reset; + struct clk *hclk; + struct clk *mclk_rx; + struct clk *mclk_tx; + struct snd_soc_component *component; + unsigned char codec_ver; +}; + +static struct clk_bulk_data rk3308_codec_clocks[] = { + { .id = "hclk" }, + { .id = "mclk_rx" }, + { .id = "mclk_tx" }, +}; + +static const DECLARE_TLV_DB_SCALE(rk3308_codec_adc_alc_gain_tlv, -1800, 150, 0); +static const DECLARE_TLV_DB_SCALE(rk3308_codec_dac_hpout_gain_tlv, -3900, 150, 0); +static const DECLARE_TLV_DB_SCALE(rk3308_codec_dac_hpmix_gain_tlv, -600, 600, 0); + +static const DECLARE_TLV_DB_RANGE(rk3308_codec_dac_lineout_gain_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-300, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-150, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(0, 0, 0), +); + +static const char * const rk3308_codec_hpf_cutoff_text[] = { + "20 Hz", "245 Hz", "612 Hz" +}; + +static SOC_ENUM_SINGLE_DECL(rk3308_codec_hpf_cutoff_enum12, RK3308_ADC_DIG_CON04(0), 0, + rk3308_codec_hpf_cutoff_text); +static SOC_ENUM_SINGLE_DECL(rk3308_codec_hpf_cutoff_enum34, RK3308_ADC_DIG_CON04(1), 0, + rk3308_codec_hpf_cutoff_text); +static SOC_ENUM_SINGLE_DECL(rk3308_codec_hpf_cutoff_enum56, RK3308_ADC_DIG_CON04(2), 0, + rk3308_codec_hpf_cutoff_text); +static SOC_ENUM_SINGLE_DECL(rk3308_codec_hpf_cutoff_enum78, RK3308_ADC_DIG_CON04(3), 0, + rk3308_codec_hpf_cutoff_text); + +static const struct snd_kcontrol_new rk3308_codec_controls[] = { + /* Despite the register names, these set the gain when AGC is OFF */ + SOC_SINGLE_RANGE_TLV("MIC1 Capture Volume", + RK3308_ADC_ANA_CON03(0), + RK3308_ADC_CH1_ALC_GAIN_SFT, + RK3308_ADC_CH1_ALC_GAIN_MIN, + RK3308_ADC_CH1_ALC_GAIN_MAX, + 0, rk3308_codec_adc_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("MIC2 Capture Volume", + RK3308_ADC_ANA_CON04(0), + RK3308_ADC_CH2_ALC_GAIN_SFT, + RK3308_ADC_CH2_ALC_GAIN_MIN, + RK3308_ADC_CH2_ALC_GAIN_MAX, + 0, rk3308_codec_adc_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("MIC3 Capture Volume", + RK3308_ADC_ANA_CON03(1), + RK3308_ADC_CH1_ALC_GAIN_SFT, + RK3308_ADC_CH1_ALC_GAIN_MIN, + RK3308_ADC_CH1_ALC_GAIN_MAX, + 0, rk3308_codec_adc_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("MIC4 Capture Volume", + RK3308_ADC_ANA_CON04(1), + RK3308_ADC_CH2_ALC_GAIN_SFT, + RK3308_ADC_CH2_ALC_GAIN_MIN, + RK3308_ADC_CH2_ALC_GAIN_MAX, + 0, rk3308_codec_adc_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("MIC5 Capture Volume", + RK3308_ADC_ANA_CON03(2), + RK3308_ADC_CH1_ALC_GAIN_SFT, + RK3308_ADC_CH1_ALC_GAIN_MIN, + RK3308_ADC_CH1_ALC_GAIN_MAX, + 0, rk3308_codec_adc_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("MIC6 Capture Volume", + RK3308_ADC_ANA_CON04(2), + RK3308_ADC_CH2_ALC_GAIN_SFT, + RK3308_ADC_CH2_ALC_GAIN_MIN, + RK3308_ADC_CH2_ALC_GAIN_MAX, + 0, rk3308_codec_adc_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("MIC7 Capture Volume", + RK3308_ADC_ANA_CON03(3), + RK3308_ADC_CH1_ALC_GAIN_SFT, + RK3308_ADC_CH1_ALC_GAIN_MIN, + RK3308_ADC_CH1_ALC_GAIN_MAX, + 0, rk3308_codec_adc_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("MIC8 Capture Volume", + RK3308_ADC_ANA_CON04(3), + RK3308_ADC_CH2_ALC_GAIN_SFT, + RK3308_ADC_CH2_ALC_GAIN_MIN, + RK3308_ADC_CH2_ALC_GAIN_MAX, + 0, rk3308_codec_adc_alc_gain_tlv), + + SOC_SINGLE("MIC1 Capture Switch", RK3308_ADC_ANA_CON00(0), 3, 1, 0), + SOC_SINGLE("MIC2 Capture Switch", RK3308_ADC_ANA_CON00(0), 7, 1, 0), + SOC_SINGLE("MIC3 Capture Switch", RK3308_ADC_ANA_CON00(1), 3, 1, 0), + SOC_SINGLE("MIC4 Capture Switch", RK3308_ADC_ANA_CON00(1), 7, 1, 0), + SOC_SINGLE("MIC5 Capture Switch", RK3308_ADC_ANA_CON00(2), 3, 1, 0), + SOC_SINGLE("MIC6 Capture Switch", RK3308_ADC_ANA_CON00(2), 7, 1, 0), + SOC_SINGLE("MIC7 Capture Switch", RK3308_ADC_ANA_CON00(3), 3, 1, 0), + SOC_SINGLE("MIC8 Capture Switch", RK3308_ADC_ANA_CON00(3), 7, 1, 0), + + SOC_SINGLE("MIC12 HPF Capture Switch", RK3308_ADC_DIG_CON04(0), 2, 1, 1), + SOC_SINGLE("MIC34 HPF Capture Switch", RK3308_ADC_DIG_CON04(1), 2, 1, 1), + SOC_SINGLE("MIC56 HPF Capture Switch", RK3308_ADC_DIG_CON04(2), 2, 1, 1), + SOC_SINGLE("MIC78 HPF Capture Switch", RK3308_ADC_DIG_CON04(3), 2, 1, 1), + + SOC_ENUM("MIC12 HPF Cutoff", rk3308_codec_hpf_cutoff_enum12), + SOC_ENUM("MIC34 HPF Cutoff", rk3308_codec_hpf_cutoff_enum34), + SOC_ENUM("MIC56 HPF Cutoff", rk3308_codec_hpf_cutoff_enum56), + SOC_ENUM("MIC78 HPF Cutoff", rk3308_codec_hpf_cutoff_enum78), + + SOC_DOUBLE_TLV("Line Out Playback Volume", + RK3308_DAC_ANA_CON04, + RK3308_DAC_L_LINEOUT_GAIN_SFT, + RK3308_DAC_R_LINEOUT_GAIN_SFT, + RK3308_DAC_x_LINEOUT_GAIN_MAX, + 0, rk3308_codec_dac_lineout_gain_tlv), + SOC_DOUBLE("Line Out Playback Switch", + RK3308_DAC_ANA_CON04, + RK3308_DAC_L_LINEOUT_MUTE_SFT, + RK3308_DAC_R_LINEOUT_MUTE_SFT, 1, 0), + SOC_DOUBLE_R_TLV("Headphone Playback Volume", + RK3308_DAC_ANA_CON05, + RK3308_DAC_ANA_CON06, + RK3308_DAC_x_HPOUT_GAIN_SFT, + RK3308_DAC_x_HPOUT_GAIN_MAX, + 0, rk3308_codec_dac_hpout_gain_tlv), + SOC_DOUBLE("Headphone Playback Switch", + RK3308_DAC_ANA_CON03, + RK3308_DAC_L_HPOUT_MUTE_SFT, + RK3308_DAC_R_HPOUT_MUTE_SFT, 1, 0), + SOC_DOUBLE_RANGE_TLV("DAC HPMIX Playback Volume", + RK3308_DAC_ANA_CON12, + RK3308_DAC_L_HPMIX_GAIN_SFT, + RK3308_DAC_R_HPMIX_GAIN_SFT, + 1, 2, 0, rk3308_codec_dac_hpmix_gain_tlv), +}; + +static int rk3308_codec_pop_sound_set(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component); + unsigned int val = (event == SND_SOC_DAPM_POST_PMU) ? + RK3308_DAC_HPOUT_POP_SOUND_x_WORK : + RK3308_DAC_HPOUT_POP_SOUND_x_INIT; + unsigned int mask = RK3308_DAC_HPOUT_POP_SOUND_x_MSK; + + regmap_update_bits(rk3308->regmap, RK3308_DAC_ANA_CON01, + mask << w->shift, val << w->shift); + + return 0; +} + +static const struct snd_soc_dapm_widget rk3308_codec_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + SND_SOC_DAPM_INPUT("MIC5"), + SND_SOC_DAPM_INPUT("MIC6"), + SND_SOC_DAPM_INPUT("MIC7"), + SND_SOC_DAPM_INPUT("MIC8"), + + SND_SOC_DAPM_SUPPLY("ADC_CURRENT_EN12", RK3308_ADC_ANA_CON06(0), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_CURRENT_EN34", RK3308_ADC_ANA_CON06(1), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_CURRENT_EN56", RK3308_ADC_ANA_CON06(2), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_CURRENT_EN78", RK3308_ADC_ANA_CON06(3), 0, 0, NULL, 0), + + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC1_EN", RK3308_ADC_ANA_CON00(0), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC2_EN", RK3308_ADC_ANA_CON00(0), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC3_EN", RK3308_ADC_ANA_CON00(1), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC4_EN", RK3308_ADC_ANA_CON00(1), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC5_EN", RK3308_ADC_ANA_CON00(2), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC6_EN", RK3308_ADC_ANA_CON00(2), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC7_EN", RK3308_ADC_ANA_CON00(3), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC8_EN", RK3308_ADC_ANA_CON00(3), 5, 1, 1, 0), + + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC1_WORK", RK3308_ADC_ANA_CON00(0), 2, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC2_WORK", RK3308_ADC_ANA_CON00(0), 6, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC3_WORK", RK3308_ADC_ANA_CON00(1), 2, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC4_WORK", RK3308_ADC_ANA_CON00(1), 6, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC5_WORK", RK3308_ADC_ANA_CON00(2), 2, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC6_WORK", RK3308_ADC_ANA_CON00(2), 6, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC7_WORK", RK3308_ADC_ANA_CON00(3), 2, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mic, "MIC8_WORK", RK3308_ADC_ANA_CON00(3), 6, 1, 1, 0), + + /* + * In theory MIC1 and MIC2 can switch to LINE IN, but this is not + * supported so all we can do is enabling the MIC input. + */ + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "CH1_IN_SEL", RK3308_ADC_ANA_CON07(0), 4, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "CH2_IN_SEL", RK3308_ADC_ANA_CON07(0), 6, 1, 1, 0), + + SND_SOC_DAPM_SUPPLY("ADC1_BUF_REF_EN", RK3308_ADC_ANA_CON00(0), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC2_BUF_REF_EN", RK3308_ADC_ANA_CON00(0), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC3_BUF_REF_EN", RK3308_ADC_ANA_CON00(1), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC4_BUF_REF_EN", RK3308_ADC_ANA_CON00(1), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC5_BUF_REF_EN", RK3308_ADC_ANA_CON00(2), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC6_BUF_REF_EN", RK3308_ADC_ANA_CON00(2), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC7_BUF_REF_EN", RK3308_ADC_ANA_CON00(3), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC8_BUF_REF_EN", RK3308_ADC_ANA_CON00(3), 4, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC_MCLK_GATE", RK3308_GLB_CON, 5, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("ADC1_CLK_EN", RK3308_ADC_ANA_CON05(0), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC2_CLK_EN", RK3308_ADC_ANA_CON05(0), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC3_CLK_EN", RK3308_ADC_ANA_CON05(1), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC4_CLK_EN", RK3308_ADC_ANA_CON05(1), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC5_CLK_EN", RK3308_ADC_ANA_CON05(2), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC6_CLK_EN", RK3308_ADC_ANA_CON05(2), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC7_CLK_EN", RK3308_ADC_ANA_CON05(3), 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC8_CLK_EN", RK3308_ADC_ANA_CON05(3), 4, 0, NULL, 0), + + /* The "ALC" name from the TRM is misleading, these are needed even without ALC/AGC */ + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC1_EN", RK3308_ADC_ANA_CON02(0), 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC2_EN", RK3308_ADC_ANA_CON02(0), 4, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC3_EN", RK3308_ADC_ANA_CON02(1), 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC4_EN", RK3308_ADC_ANA_CON02(1), 4, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC5_EN", RK3308_ADC_ANA_CON02(2), 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC6_EN", RK3308_ADC_ANA_CON02(2), 4, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC7_EN", RK3308_ADC_ANA_CON02(3), 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC8_EN", RK3308_ADC_ANA_CON02(3), 4, 1, 1, 0), + + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC1_EN", RK3308_ADC_ANA_CON05(0), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC2_EN", RK3308_ADC_ANA_CON05(0), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC3_EN", RK3308_ADC_ANA_CON05(1), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC4_EN", RK3308_ADC_ANA_CON05(1), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC5_EN", RK3308_ADC_ANA_CON05(2), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC6_EN", RK3308_ADC_ANA_CON05(2), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC7_EN", RK3308_ADC_ANA_CON05(3), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ADC8_EN", RK3308_ADC_ANA_CON05(3), 5, 1, 1, 0), + + SND_SOC_DAPM_ADC("ADC1_WORK", "Capture", RK3308_ADC_ANA_CON05(0), 2, 0), + SND_SOC_DAPM_ADC("ADC2_WORK", "Capture", RK3308_ADC_ANA_CON05(0), 6, 0), + SND_SOC_DAPM_ADC("ADC3_WORK", "Capture", RK3308_ADC_ANA_CON05(1), 2, 0), + SND_SOC_DAPM_ADC("ADC4_WORK", "Capture", RK3308_ADC_ANA_CON05(1), 6, 0), + SND_SOC_DAPM_ADC("ADC5_WORK", "Capture", RK3308_ADC_ANA_CON05(2), 2, 0), + SND_SOC_DAPM_ADC("ADC6_WORK", "Capture", RK3308_ADC_ANA_CON05(2), 6, 0), + SND_SOC_DAPM_ADC("ADC7_WORK", "Capture", RK3308_ADC_ANA_CON05(3), 2, 0), + SND_SOC_DAPM_ADC("ADC8_WORK", "Capture", RK3308_ADC_ANA_CON05(3), 6, 0), + + /* The "ALC" name from the TRM is misleading, these are needed even without ALC/AGC */ + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC1_WORK", RK3308_ADC_ANA_CON02(0), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC2_WORK", RK3308_ADC_ANA_CON02(0), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC3_WORK", RK3308_ADC_ANA_CON02(1), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC4_WORK", RK3308_ADC_ANA_CON02(1), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC5_WORK", RK3308_ADC_ANA_CON02(2), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC6_WORK", RK3308_ADC_ANA_CON02(2), 5, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC7_WORK", RK3308_ADC_ANA_CON02(3), 1, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_adc, "ALC8_WORK", RK3308_ADC_ANA_CON02(3), 5, 1, 1, 0), + + SND_SOC_DAPM_SUPPLY("MICBIAS Current", RK3308_ADC_ANA_CON08(0), 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RK3308_ADC_ANA_CON07(1), 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RK3308_ADC_ANA_CON07(2), 3, 0, NULL, 0), + + SND_SOC_DAPM_OUT_DRV("DAC_L_HPMIX_EN", RK3308_DAC_ANA_CON13, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("DAC_R_HPMIX_EN", RK3308_DAC_ANA_CON13, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("DAC_L_HPMIX_WORK", RK3308_DAC_ANA_CON13, 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("DAC_R_HPMIX_WORK", RK3308_DAC_ANA_CON13, 5, 0, NULL, 0), + /* HPMIX is not actually acting as a mixer as the only supported input is I2S */ + SND_SOC_DAPM_OUT_DRV("DAC_L_HPMIX_SEL", RK3308_DAC_ANA_CON12, 2, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("DAC_R_HPMIX_SEL", RK3308_DAC_ANA_CON12, 6, 0, NULL, 0), + SND_SOC_DAPM_MIXER("DAC HPMIX Left", RK3308_DAC_ANA_CON13, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("DAC HPMIX Right", RK3308_DAC_ANA_CON13, 6, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC_MCLK_GATE", RK3308_GLB_CON, 4, 1, NULL, 0), + + SND_SOC_DAPM_SUPPLY("DAC_CURRENT_EN", RK3308_DAC_ANA_CON00, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_L_REF_EN", RK3308_DAC_ANA_CON02, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_R_REF_EN", RK3308_DAC_ANA_CON02, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_L_CLK_EN", RK3308_DAC_ANA_CON02, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_R_CLK_EN", RK3308_DAC_ANA_CON02, 5, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC_L_DAC_WORK", NULL, RK3308_DAC_ANA_CON02, 3, 0), + SND_SOC_DAPM_DAC("DAC_R_DAC_WORK", NULL, RK3308_DAC_ANA_CON02, 7, 0), + + SND_SOC_DAPM_SUPPLY("DAC_BUF_REF_L", RK3308_DAC_ANA_CON01, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_BUF_REF_R", RK3308_DAC_ANA_CON01, 6, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("HPOUT_POP_SOUND_L", SND_SOC_NOPM, 0, 0, NULL, 0, + rk3308_codec_pop_sound_set, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUT_DRV_E("HPOUT_POP_SOUND_R", SND_SOC_NOPM, 4, 0, NULL, 0, + rk3308_codec_pop_sound_set, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUT_DRV("L_HPOUT_EN", RK3308_DAC_ANA_CON03, 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("R_HPOUT_EN", RK3308_DAC_ANA_CON03, 5, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("L_HPOUT_WORK", RK3308_DAC_ANA_CON03, 2, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("R_HPOUT_WORK", RK3308_DAC_ANA_CON03, 6, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HPOUT_L"), + SND_SOC_DAPM_OUTPUT("HPOUT_R"), + + SND_SOC_DAPM_OUT_DRV("L_LINEOUT_EN", RK3308_DAC_ANA_CON04, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("R_LINEOUT_EN", RK3308_DAC_ANA_CON04, 4, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("LINEOUT_L"), + SND_SOC_DAPM_OUTPUT("LINEOUT_R"), +}; + +static const struct snd_soc_dapm_route rk3308_codec_dapm_routes[] = { + { "MICBIAS1", NULL, "MICBIAS Current" }, + { "MICBIAS2", NULL, "MICBIAS Current" }, + + { "MIC1_EN", NULL, "MIC1" }, + { "MIC2_EN", NULL, "MIC2" }, + { "MIC3_EN", NULL, "MIC3" }, + { "MIC4_EN", NULL, "MIC4" }, + { "MIC5_EN", NULL, "MIC5" }, + { "MIC6_EN", NULL, "MIC6" }, + { "MIC7_EN", NULL, "MIC7" }, + { "MIC8_EN", NULL, "MIC8" }, + + { "MIC1_WORK", NULL, "MIC1_EN" }, + { "MIC2_WORK", NULL, "MIC2_EN" }, + { "MIC3_WORK", NULL, "MIC3_EN" }, + { "MIC4_WORK", NULL, "MIC4_EN" }, + { "MIC5_WORK", NULL, "MIC5_EN" }, + { "MIC6_WORK", NULL, "MIC6_EN" }, + { "MIC7_WORK", NULL, "MIC7_EN" }, + { "MIC8_WORK", NULL, "MIC8_EN" }, + + { "CH1_IN_SEL", NULL, "MIC1_WORK" }, + { "CH2_IN_SEL", NULL, "MIC2_WORK" }, + + { "ALC1_EN", NULL, "CH1_IN_SEL" }, + { "ALC2_EN", NULL, "CH2_IN_SEL" }, + { "ALC3_EN", NULL, "MIC3_WORK" }, + { "ALC4_EN", NULL, "MIC4_WORK" }, + { "ALC5_EN", NULL, "MIC5_WORK" }, + { "ALC6_EN", NULL, "MIC6_WORK" }, + { "ALC7_EN", NULL, "MIC7_WORK" }, + { "ALC8_EN", NULL, "MIC8_WORK" }, + + { "ADC1_EN", NULL, "ALC1_EN" }, + { "ADC2_EN", NULL, "ALC2_EN" }, + { "ADC3_EN", NULL, "ALC3_EN" }, + { "ADC4_EN", NULL, "ALC4_EN" }, + { "ADC5_EN", NULL, "ALC5_EN" }, + { "ADC6_EN", NULL, "ALC6_EN" }, + { "ADC7_EN", NULL, "ALC7_EN" }, + { "ADC8_EN", NULL, "ALC8_EN" }, + + { "ADC1_WORK", NULL, "ADC1_EN" }, + { "ADC2_WORK", NULL, "ADC2_EN" }, + { "ADC3_WORK", NULL, "ADC3_EN" }, + { "ADC4_WORK", NULL, "ADC4_EN" }, + { "ADC5_WORK", NULL, "ADC5_EN" }, + { "ADC6_WORK", NULL, "ADC6_EN" }, + { "ADC7_WORK", NULL, "ADC7_EN" }, + { "ADC8_WORK", NULL, "ADC8_EN" }, + + { "ADC1_BUF_REF_EN", NULL, "ADC_CURRENT_EN12" }, + { "ADC2_BUF_REF_EN", NULL, "ADC_CURRENT_EN12" }, + { "ADC3_BUF_REF_EN", NULL, "ADC_CURRENT_EN34" }, + { "ADC4_BUF_REF_EN", NULL, "ADC_CURRENT_EN34" }, + { "ADC5_BUF_REF_EN", NULL, "ADC_CURRENT_EN56" }, + { "ADC6_BUF_REF_EN", NULL, "ADC_CURRENT_EN56" }, + { "ADC7_BUF_REF_EN", NULL, "ADC_CURRENT_EN78" }, + { "ADC8_BUF_REF_EN", NULL, "ADC_CURRENT_EN78" }, + + { "ADC1_WORK", NULL, "ADC1_BUF_REF_EN" }, + { "ADC2_WORK", NULL, "ADC2_BUF_REF_EN" }, + { "ADC3_WORK", NULL, "ADC3_BUF_REF_EN" }, + { "ADC4_WORK", NULL, "ADC4_BUF_REF_EN" }, + { "ADC5_WORK", NULL, "ADC5_BUF_REF_EN" }, + { "ADC6_WORK", NULL, "ADC6_BUF_REF_EN" }, + { "ADC7_WORK", NULL, "ADC7_BUF_REF_EN" }, + { "ADC8_WORK", NULL, "ADC8_BUF_REF_EN" }, + + { "ADC1_CLK_EN", NULL, "ADC_MCLK_GATE" }, + { "ADC2_CLK_EN", NULL, "ADC_MCLK_GATE" }, + { "ADC3_CLK_EN", NULL, "ADC_MCLK_GATE" }, + { "ADC4_CLK_EN", NULL, "ADC_MCLK_GATE" }, + { "ADC5_CLK_EN", NULL, "ADC_MCLK_GATE" }, + { "ADC6_CLK_EN", NULL, "ADC_MCLK_GATE" }, + { "ADC7_CLK_EN", NULL, "ADC_MCLK_GATE" }, + { "ADC8_CLK_EN", NULL, "ADC_MCLK_GATE" }, + + { "ADC1_WORK", NULL, "ADC1_CLK_EN" }, + { "ADC2_WORK", NULL, "ADC2_CLK_EN" }, + { "ADC3_WORK", NULL, "ADC3_CLK_EN" }, + { "ADC4_WORK", NULL, "ADC4_CLK_EN" }, + { "ADC5_WORK", NULL, "ADC5_CLK_EN" }, + { "ADC6_WORK", NULL, "ADC6_CLK_EN" }, + { "ADC7_WORK", NULL, "ADC7_CLK_EN" }, + { "ADC8_WORK", NULL, "ADC8_CLK_EN" }, + + { "ALC1_WORK", NULL, "ADC1_WORK" }, + { "ALC2_WORK", NULL, "ADC2_WORK" }, + { "ALC3_WORK", NULL, "ADC3_WORK" }, + { "ALC4_WORK", NULL, "ADC4_WORK" }, + { "ALC5_WORK", NULL, "ADC5_WORK" }, + { "ALC6_WORK", NULL, "ADC6_WORK" }, + { "ALC7_WORK", NULL, "ADC7_WORK" }, + { "ALC8_WORK", NULL, "ADC8_WORK" }, + + { "HiFi Capture", NULL, "ALC1_WORK" }, + { "HiFi Capture", NULL, "ALC2_WORK" }, + { "HiFi Capture", NULL, "ALC3_WORK" }, + { "HiFi Capture", NULL, "ALC4_WORK" }, + { "HiFi Capture", NULL, "ALC5_WORK" }, + { "HiFi Capture", NULL, "ALC6_WORK" }, + { "HiFi Capture", NULL, "ALC7_WORK" }, + { "HiFi Capture", NULL, "ALC8_WORK" }, + + { "DAC_L_HPMIX_EN", NULL, "HiFi Playback" }, + { "DAC_R_HPMIX_EN", NULL, "HiFi Playback" }, + { "DAC_L_HPMIX_WORK", NULL, "DAC_L_HPMIX_EN" }, + { "DAC_R_HPMIX_WORK", NULL, "DAC_R_HPMIX_EN" }, + { "DAC HPMIX Left", NULL, "DAC_L_HPMIX_WORK" }, + { "DAC HPMIX Right", NULL, "DAC_R_HPMIX_WORK" }, + + { "DAC_L_DAC_WORK", NULL, "DAC HPMIX Left" }, + { "DAC_R_DAC_WORK", NULL, "DAC HPMIX Right" }, + + { "DAC_L_REF_EN", NULL, "DAC_CURRENT_EN" }, + { "DAC_R_REF_EN", NULL, "DAC_CURRENT_EN" }, + { "DAC_L_CLK_EN", NULL, "DAC_L_REF_EN" }, + { "DAC_R_CLK_EN", NULL, "DAC_R_REF_EN" }, + { "DAC_L_CLK_EN", NULL, "DAC_MCLK_GATE" }, + { "DAC_R_CLK_EN", NULL, "DAC_MCLK_GATE" }, + { "DAC_L_DAC_WORK", NULL, "DAC_L_CLK_EN" }, + { "DAC_R_DAC_WORK", NULL, "DAC_R_CLK_EN" }, + { "DAC_L_HPMIX_SEL", NULL, "DAC_L_DAC_WORK" }, + { "DAC_R_HPMIX_SEL", NULL, "DAC_R_DAC_WORK" }, + + { "HPOUT_L", NULL, "DAC_BUF_REF_L" }, + { "HPOUT_R", NULL, "DAC_BUF_REF_R" }, + { "L_HPOUT_EN", NULL, "DAC_L_HPMIX_SEL" }, + { "R_HPOUT_EN", NULL, "DAC_R_HPMIX_SEL" }, + { "L_HPOUT_WORK", NULL, "L_HPOUT_EN" }, + { "R_HPOUT_WORK", NULL, "R_HPOUT_EN" }, + { "HPOUT_POP_SOUND_L", NULL, "L_HPOUT_WORK" }, + { "HPOUT_POP_SOUND_R", NULL, "R_HPOUT_WORK" }, + { "HPOUT_L", NULL, "HPOUT_POP_SOUND_L" }, + { "HPOUT_R", NULL, "HPOUT_POP_SOUND_R" }, + + { "L_LINEOUT_EN", NULL, "DAC_L_HPMIX_SEL" }, + { "R_LINEOUT_EN", NULL, "DAC_R_HPMIX_SEL" }, + { "LINEOUT_L", NULL, "L_LINEOUT_EN" }, + { "LINEOUT_R", NULL, "R_LINEOUT_EN" }, +}; + +static int rk3308_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component); + const unsigned int inv_bits = fmt & SND_SOC_DAIFMT_INV_MASK; + const bool inv_bitclk = + (inv_bits & SND_SOC_DAIFMT_IB_IF) || + (inv_bits & SND_SOC_DAIFMT_IB_NF); + const bool inv_frmclk = + (inv_bits & SND_SOC_DAIFMT_IB_IF) || + (inv_bits & SND_SOC_DAIFMT_NB_IF); + const unsigned int dac_master_bits = rk3308->codec_ver < ACODEC_VERSION_C ? + RK3308_DAC_IO_MODE_MASTER | RK3308_DAC_MODE_MASTER : + RK3308BS_DAC_IO_MODE_MASTER | RK3308BS_DAC_MODE_MASTER; + unsigned int adc_aif1 = 0, adc_aif2 = 0, dac_aif1 = 0, dac_aif2 = 0; + bool is_master = false; + int grp; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + break; + case SND_SOC_DAIFMT_CBP_CFP: + adc_aif2 |= RK3308_ADC_IO_MODE_MASTER; + adc_aif2 |= RK3308_ADC_MODE_MASTER; + dac_aif2 |= dac_master_bits; + is_master = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + adc_aif1 |= RK3308_ADC_I2S_MODE_PCM; + dac_aif1 |= RK3308_DAC_I2S_MODE_PCM; + break; + case SND_SOC_DAIFMT_I2S: + adc_aif1 |= RK3308_ADC_I2S_MODE_I2S; + dac_aif1 |= RK3308_DAC_I2S_MODE_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + adc_aif1 |= RK3308_ADC_I2S_MODE_RJ; + dac_aif1 |= RK3308_DAC_I2S_MODE_RJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + adc_aif1 |= RK3308_ADC_I2S_MODE_LJ; + dac_aif1 |= RK3308_DAC_I2S_MODE_LJ; + break; + default: + return -EINVAL; + } + + if (inv_bitclk) { + adc_aif2 |= RK3308_ADC_I2S_BIT_CLK_POL_REVERSAL; + dac_aif2 |= RK3308_DAC_I2S_BIT_CLK_POL_REVERSAL; + } + + if (inv_frmclk) { + adc_aif1 |= RK3308_ADC_I2S_LRC_POL_REVERSAL; + dac_aif1 |= RK3308_DAC_I2S_LRC_POL_REVERSAL; + } + + /* + * Hold ADC Digital registers start at master mode + * + * There are 8 ADCs which use the same internal SCLK and LRCK for + * master mode. We need to make sure that they are in effect at the + * same time, otherwise they will cause abnormal clocks. + */ + if (is_master) + regmap_clear_bits(rk3308->regmap, RK3308_GLB_CON, RK3308_ADC_DIG_WORK); + + for (grp = 0; grp < ADC_LR_GROUP_MAX; grp++) { + regmap_update_bits(rk3308->regmap, RK3308_ADC_DIG_CON01(grp), + RK3308_ADC_I2S_LRC_POL_REVERSAL | + RK3308_ADC_I2S_MODE_MSK, + adc_aif1); + regmap_update_bits(rk3308->regmap, RK3308_ADC_DIG_CON02(grp), + RK3308_ADC_IO_MODE_MASTER | + RK3308_ADC_MODE_MASTER | + RK3308_ADC_I2S_BIT_CLK_POL_REVERSAL, + adc_aif2); + } + + /* Hold ADC Digital registers end at master mode */ + if (is_master) + regmap_set_bits(rk3308->regmap, RK3308_GLB_CON, RK3308_ADC_DIG_WORK); + + regmap_update_bits(rk3308->regmap, RK3308_DAC_DIG_CON01, + RK3308_DAC_I2S_LRC_POL_REVERSAL | + RK3308_DAC_I2S_MODE_MSK, + dac_aif1); + regmap_update_bits(rk3308->regmap, RK3308_DAC_DIG_CON02, + dac_master_bits | RK3308_DAC_I2S_BIT_CLK_POL_REVERSAL, + dac_aif2); + + return 0; +} + +static int rk3308_codec_dac_dig_config(struct rk3308_codec_priv *rk3308, + struct snd_pcm_hw_params *params) +{ + unsigned int dac_aif1 = 0; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + dac_aif1 |= RK3308_DAC_I2S_VALID_LEN_16BITS; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + dac_aif1 |= RK3308_DAC_I2S_VALID_LEN_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dac_aif1 |= RK3308_DAC_I2S_VALID_LEN_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dac_aif1 |= RK3308_DAC_I2S_VALID_LEN_32BITS; + break; + default: + return -EINVAL; + } + + regmap_update_bits(rk3308->regmap, RK3308_DAC_DIG_CON01, + RK3308_DAC_I2S_VALID_LEN_MSK, dac_aif1); + regmap_set_bits(rk3308->regmap, RK3308_DAC_DIG_CON02, RK3308_DAC_I2S_WORK); + + return 0; +} + +static int rk3308_codec_adc_dig_config(struct rk3308_codec_priv *rk3308, + struct snd_pcm_hw_params *params) +{ + unsigned int adc_aif1 = 0; + /* + * grp 0 = ADC1 and ADC2 + * grp 1 = ADC3 and ADC4 + * grp 2 = ADC5 and ADC6 + * grp 3 = ADC7 and ADC8 + */ + u32 used_adc_grps; + int grp; + + switch (params_channels(params)) { + case 1: + adc_aif1 |= RK3308_ADC_I2S_MONO; + used_adc_grps = 1; + break; + case 2: + case 4: + case 6: + case 8: + used_adc_grps = params_channels(params) / 2; + break; + default: + dev_err(rk3308->dev, "Invalid channel number %d\n", params_channels(params)); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + adc_aif1 |= RK3308_ADC_I2S_VALID_LEN_16BITS; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + adc_aif1 |= RK3308_ADC_I2S_VALID_LEN_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + adc_aif1 |= RK3308_ADC_I2S_VALID_LEN_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + adc_aif1 |= RK3308_ADC_I2S_VALID_LEN_32BITS; + break; + default: + return -EINVAL; + } + + for (grp = 0; grp < used_adc_grps; grp++) { + regmap_update_bits(rk3308->regmap, + RK3308_ADC_DIG_CON03(grp), + RK3308_ADC_L_CH_BIST_MSK | RK3308_ADC_R_CH_BIST_MSK, + RK3308_ADC_L_CH_NORMAL_LEFT | RK3308_ADC_R_CH_NORMAL_RIGHT); + regmap_update_bits(rk3308->regmap, RK3308_ADC_DIG_CON01(grp), + RK3308_ADC_I2S_VALID_LEN_MSK | RK3308_ADC_I2S_MONO, adc_aif1); + regmap_set_bits(rk3308->regmap, RK3308_ADC_DIG_CON02(grp), RK3308_ADC_I2S_WORK); + } + + return 0; +} + +static int rk3308_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component); + + return (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + rk3308_codec_dac_dig_config(rk3308, params) : + rk3308_codec_adc_dig_config(rk3308, params); +} + +static const struct snd_soc_dai_ops rk3308_codec_dai_ops = { + .hw_params = rk3308_codec_hw_params, + .set_fmt = rk3308_codec_set_dai_fmt, +}; + +static struct snd_soc_dai_driver rk3308_codec_dai_driver = { + .name = "rk3308-hifi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE), + }, + .ops = &rk3308_codec_dai_ops, +}; + +static void rk3308_codec_reset(struct snd_soc_component *component) +{ + struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component); + + reset_control_assert(rk3308->reset); + usleep_range(10000, 11000); /* estimated value */ + reset_control_deassert(rk3308->reset); + + regmap_write(rk3308->regmap, RK3308_GLB_CON, 0x00); + usleep_range(10000, 11000); /* estimated value */ + regmap_write(rk3308->regmap, RK3308_GLB_CON, + RK3308_SYS_WORK | + RK3308_DAC_DIG_WORK | + RK3308_ADC_DIG_WORK); +} + +/* + * Initialize register whose default after HW reset is problematic or which + * are never modified. + */ +static int rk3308_codec_initialize(struct rk3308_codec_priv *rk3308) +{ + int grp; + + /* + * Init ADC digital vol to 0 dB (reset value is 0xff, undocumented). + * Range: -97dB ~ +32dB. + */ + if (rk3308->codec_ver == ACODEC_VERSION_C) { + for (grp = 0; grp < ADC_LR_GROUP_MAX; grp++) { + regmap_write(rk3308->regmap, RK3308_ADC_DIG_CON05(grp), + RK3308_ADC_DIG_VOL_CON_x_0DB); + regmap_write(rk3308->regmap, RK3308_ADC_DIG_CON06(grp), + RK3308_ADC_DIG_VOL_CON_x_0DB); + } + } + + /* set HPMIX default gains (reset value is 0, which is illegal) */ + regmap_update_bits(rk3308->regmap, RK3308_DAC_ANA_CON12, + RK3308_DAC_L_HPMIX_GAIN_MSK | + RK3308_DAC_R_HPMIX_GAIN_MSK, + RK3308_DAC_L_HPMIX_GAIN_NDB_6 | + RK3308_DAC_R_HPMIX_GAIN_NDB_6); + + /* recover DAC digital gain to 0 dB (reset value is 0xff, undocumented) */ + if (rk3308->codec_ver == ACODEC_VERSION_C) + regmap_write(rk3308->regmap, RK3308_DAC_DIG_CON04, + RK3308BS_DAC_DIG_GAIN_0DB); + + /* + * Unconditionally enable zero-cross detection (needed for AGC, + * harmless without AGC) + */ + for (grp = 0; grp < ADC_LR_GROUP_MAX; grp++) + regmap_set_bits(rk3308->regmap, RK3308_ADC_ANA_CON02(grp), + RK3308_ADC_CH1_ZEROCROSS_DET_EN | + RK3308_ADC_CH2_ZEROCROSS_DET_EN); + + return 0; +} + +static int rk3308_codec_probe(struct snd_soc_component *component) +{ + struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component); + + rk3308->component = component; + + rk3308_codec_reset(component); + rk3308_codec_initialize(rk3308); + + return 0; +} + +static int rk3308_codec_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) != SND_SOC_BIAS_OFF) + break; + + /* Sequence from TRM Section 8.6.3 "Power Up" */ + regmap_set_bits(rk3308->regmap, RK3308_DAC_ANA_CON02, + RK3308_DAC_L_DAC_EN | RK3308_DAC_R_DAC_EN); + regmap_update_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0), + RK3308_ADC_CURRENT_CHARGE_MSK, 1); + regmap_set_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0), + RK3308_ADC_REF_EN); + regmap_update_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0), + RK3308_ADC_CURRENT_CHARGE_MSK, 0x7f); + msleep(20); /* estimated value */ + break; + case SND_SOC_BIAS_OFF: + /* Sequence from TRM Section 8.6.4 "Power Down" */ + regmap_update_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0), + RK3308_ADC_CURRENT_CHARGE_MSK, 1); + regmap_clear_bits(rk3308->regmap, RK3308_ADC_ANA_CON10(0), + RK3308_ADC_REF_EN); + regmap_clear_bits(rk3308->regmap, RK3308_DAC_ANA_CON02, + RK3308_DAC_L_DAC_EN | RK3308_DAC_R_DAC_EN); + msleep(20); /* estimated value */ + break; + } + return 0; +} + +static const struct snd_soc_component_driver rk3308_codec_component_driver = { + .probe = rk3308_codec_probe, + .set_bias_level = rk3308_codec_set_bias_level, + .controls = rk3308_codec_controls, + .num_controls = ARRAY_SIZE(rk3308_codec_controls), + .dapm_widgets = rk3308_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk3308_codec_dapm_widgets), + .dapm_routes = rk3308_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rk3308_codec_dapm_routes), +}; + +static const struct regmap_config rk3308_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = RK3308_DAC_ANA_CON15, +}; + +static int rk3308_codec_get_version(struct rk3308_codec_priv *rk3308) +{ + unsigned int chip_id; + int err; + + err = regmap_read(rk3308->grf, GRF_CHIP_ID, &chip_id); + if (err) + return err; + + switch (chip_id) { + case 3306: + rk3308->codec_ver = ACODEC_VERSION_A; + break; + case 0x3308: + rk3308->codec_ver = ACODEC_VERSION_B; + return dev_err_probe(rk3308->dev, -EINVAL, "Chip version B not supported\n"); + case 0x3308c: + rk3308->codec_ver = ACODEC_VERSION_C; + break; + default: + return dev_err_probe(rk3308->dev, -EINVAL, "Unknown chip_id: 0x%x\n", chip_id); + } + + dev_info(rk3308->dev, "Found codec version %c\n", rk3308->codec_ver); + return 0; +} + +static int rk3308_codec_set_micbias_level(struct rk3308_codec_priv *rk3308) +{ + struct device_node *np = rk3308->dev->of_node; + u32 percent; + u32 mult; + int err; + + err = of_property_read_u32(np, "rockchip,micbias-avdd-percent", &percent); + if (err == -EINVAL) + return 0; + if (err) + return dev_err_probe(rk3308->dev, err, + "Error reading 'rockchip,micbias-avdd-percent'\n"); + + /* Convert percent to register value, linerarly (50% -> 0, 5% step = +1) */ + mult = (percent - 50) / 5; + + /* Check range and that the percent was an exact value allowed */ + if (mult > RK3308_ADC_LEVEL_RANGE_MICBIAS_MAX || mult * 5 + 50 != percent) + return dev_err_probe(rk3308->dev, -EINVAL, + "Invalid value %u for 'rockchip,micbias-avdd-percent'\n", + percent); + + regmap_update_bits(rk3308->regmap, RK3308_ADC_ANA_CON07(0), + RK3308_ADC_LEVEL_RANGE_MICBIAS_MSK, + mult << RK3308_ADC_LEVEL_RANGE_MICBIAS_SFT); + + return 0; +} + +static int rk3308_codec_platform_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device *dev = &pdev->dev; + struct rk3308_codec_priv *rk3308; + void __iomem *base; + int err; + + rk3308 = devm_kzalloc(&pdev->dev, sizeof(*rk3308), GFP_KERNEL); + if (!rk3308) + return -ENOMEM; + + rk3308->dev = dev; + + rk3308->grf = syscon_regmap_lookup_by_phandle(np, "rockchip,grf"); + if (IS_ERR(rk3308->grf)) + return dev_err_probe(dev, PTR_ERR(rk3308->grf), "Error getting GRF\n"); + + rk3308->reset = devm_reset_control_get_optional_exclusive(dev, "codec"); + if (IS_ERR(rk3308->reset)) + return dev_err_probe(dev, PTR_ERR(rk3308->reset), "Failed to get reset control\n"); + + err = devm_clk_bulk_get(dev, ARRAY_SIZE(rk3308_codec_clocks), rk3308_codec_clocks); + if (err) + return dev_err_probe(dev, err, "Failed to get clocks\n"); + + err = clk_bulk_prepare_enable(ARRAY_SIZE(rk3308_codec_clocks), rk3308_codec_clocks); + if (err) + return dev_err_probe(dev, err, "Failed to enable clocks\n"); + + err = rk3308_codec_get_version(rk3308); + if (err) + return err; + + base = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(base)) + return PTR_ERR(base); + + rk3308->regmap = devm_regmap_init_mmio(dev, base, &rk3308_codec_regmap_config); + if (IS_ERR(rk3308->regmap)) + return dev_err_probe(dev, PTR_ERR(rk3308->regmap), + "Failed to init regmap\n"); + + platform_set_drvdata(pdev, rk3308); + + err = rk3308_codec_set_micbias_level(rk3308); + if (err) + return err; + + err = devm_snd_soc_register_component(dev, &rk3308_codec_component_driver, + &rk3308_codec_dai_driver, 1); + if (err) + return dev_err_probe(dev, err, "Failed to register codec\n"); + + return 0; +} + +static const struct of_device_id rk3308_codec_of_match[] = { + { .compatible = "rockchip,rk3308-codec", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rk3308_codec_of_match); + +static struct platform_driver rk3308_codec_driver = { + .driver = { + .name = "rk3308-acodec", + .of_match_table = of_match_ptr(rk3308_codec_of_match), + }, + .probe = rk3308_codec_platform_probe, +}; +module_platform_driver(rk3308_codec_driver); + +MODULE_AUTHOR("Xing Zheng "); +MODULE_AUTHOR("Luca Ceresoli "); +MODULE_DESCRIPTION("ASoC RK3308 Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rk3308_codec.h b/sound/soc/codecs/rk3308_codec.h new file mode 100644 index 0000000000000..a4226b235ab70 --- /dev/null +++ b/sound/soc/codecs/rk3308_codec.h @@ -0,0 +1,579 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Rockchip RK3308 internal audio codec driver -- register definitions + * + * Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + * Copyright (c) 2022, Vivax-Metrotech Ltd + */ + +#ifndef __RK3308_CODEC_H__ +#define __RK3308_CODEC_H__ + +#define RK3308_GLB_CON 0x00 + +/* ADC DIGITAL REGISTERS */ + +/* + * The ADC group are 0 ~ 3, that control: + * + * CH0: left_0(ADC1) and right_0(ADC2) + * CH1: left_1(ADC3) and right_1(ADC4) + * CH2: left_2(ADC5) and right_2(ADC6) + * CH3: left_3(ADC7) and right_3(ADC8) + */ +#define RK3308_ADC_DIG_OFFSET(ch) (((ch) & 0x3) * 0xc0 + 0x0) + +#define RK3308_ADC_DIG_CON01(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x04) +#define RK3308_ADC_DIG_CON02(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x08) +#define RK3308_ADC_DIG_CON03(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x0c) +#define RK3308_ADC_DIG_CON04(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x10) +#define RK3308_ADC_DIG_CON05(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x14) // ver.C only +#define RK3308_ADC_DIG_CON06(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x18) // ver.C only +#define RK3308_ADC_DIG_CON07(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x1c) + +#define RK3308_ALC_L_DIG_CON00(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x40) +#define RK3308_ALC_L_DIG_CON01(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x44) +#define RK3308_ALC_L_DIG_CON02(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x48) +#define RK3308_ALC_L_DIG_CON03(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x4c) +#define RK3308_ALC_L_DIG_CON04(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x50) +#define RK3308_ALC_L_DIG_CON05(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x54) +#define RK3308_ALC_L_DIG_CON06(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x58) +#define RK3308_ALC_L_DIG_CON07(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x5c) +#define RK3308_ALC_L_DIG_CON08(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x60) +#define RK3308_ALC_L_DIG_CON09(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x64) +#define RK3308_ALC_L_DIG_CON12(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x70) + +#define RK3308_ALC_R_DIG_CON00(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x80) +#define RK3308_ALC_R_DIG_CON01(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x84) +#define RK3308_ALC_R_DIG_CON02(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x88) +#define RK3308_ALC_R_DIG_CON03(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x8c) +#define RK3308_ALC_R_DIG_CON04(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x90) +#define RK3308_ALC_R_DIG_CON05(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x94) +#define RK3308_ALC_R_DIG_CON06(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x98) +#define RK3308_ALC_R_DIG_CON07(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0x9c) +#define RK3308_ALC_R_DIG_CON08(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0xa0) +#define RK3308_ALC_R_DIG_CON09(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0xa4) +#define RK3308_ALC_R_DIG_CON12(ch) (RK3308_ADC_DIG_OFFSET((ch)) + 0xb0) + +/* DAC DIGITAL REGISTERS */ +#define RK3308_DAC_DIG_OFFSET 0x300 +#define RK3308_DAC_DIG_CON01 (RK3308_DAC_DIG_OFFSET + 0x04) +#define RK3308_DAC_DIG_CON02 (RK3308_DAC_DIG_OFFSET + 0x08) +#define RK3308_DAC_DIG_CON03 (RK3308_DAC_DIG_OFFSET + 0x0c) +#define RK3308_DAC_DIG_CON04 (RK3308_DAC_DIG_OFFSET + 0x10) +#define RK3308_DAC_DIG_CON05 (RK3308_DAC_DIG_OFFSET + 0x14) +#define RK3308_DAC_DIG_CON10 (RK3308_DAC_DIG_OFFSET + 0x28) +#define RK3308_DAC_DIG_CON11 (RK3308_DAC_DIG_OFFSET + 0x2c) +#define RK3308_DAC_DIG_CON13 (RK3308_DAC_DIG_OFFSET + 0x34) +#define RK3308_DAC_DIG_CON14 (RK3308_DAC_DIG_OFFSET + 0x38) + +/* ADC ANALOG REGISTERS */ +/* + * The ADC group are 0 ~ 3, that control: + * + * CH0: left_0(ADC1) and right_0(ADC2) + * CH1: left_1(ADC3) and right_1(ADC4) + * CH2: left_2(ADC5) and right_2(ADC6) + * CH3: left_3(ADC7) and right_3(ADC8) + */ +#define RK3308_ADC_ANA_OFFSET(ch) (((ch) & 0x3) * 0x40 + 0x340) +#define RK3308_ADC_ANA_CON00(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x00) +#define RK3308_ADC_ANA_CON01(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x04) +#define RK3308_ADC_ANA_CON02(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x08) +#define RK3308_ADC_ANA_CON03(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x0c) +#define RK3308_ADC_ANA_CON04(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x10) +#define RK3308_ADC_ANA_CON05(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x14) +#define RK3308_ADC_ANA_CON06(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x18) +#define RK3308_ADC_ANA_CON07(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x1c) +#define RK3308_ADC_ANA_CON08(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x20) +#define RK3308_ADC_ANA_CON10(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x28) +#define RK3308_ADC_ANA_CON11(ch) (RK3308_ADC_ANA_OFFSET((ch)) + 0x2c) + +/* DAC ANALOG REGISTERS */ +#define RK3308_DAC_ANA_OFFSET 0x440 +#define RK3308_DAC_ANA_CON00 (RK3308_DAC_ANA_OFFSET + 0x00) +#define RK3308_DAC_ANA_CON01 (RK3308_DAC_ANA_OFFSET + 0x04) +#define RK3308_DAC_ANA_CON02 (RK3308_DAC_ANA_OFFSET + 0x08) +#define RK3308_DAC_ANA_CON03 (RK3308_DAC_ANA_OFFSET + 0x0c) +#define RK3308_DAC_ANA_CON04 (RK3308_DAC_ANA_OFFSET + 0x10) +#define RK3308_DAC_ANA_CON05 (RK3308_DAC_ANA_OFFSET + 0x14) +#define RK3308_DAC_ANA_CON06 (RK3308_DAC_ANA_OFFSET + 0x18) +#define RK3308_DAC_ANA_CON07 (RK3308_DAC_ANA_OFFSET + 0x1c) +#define RK3308_DAC_ANA_CON08 (RK3308_DAC_ANA_OFFSET + 0x20) +#define RK3308_DAC_ANA_CON12 (RK3308_DAC_ANA_OFFSET + 0x30) +#define RK3308_DAC_ANA_CON13 (RK3308_DAC_ANA_OFFSET + 0x34) +#define RK3308_DAC_ANA_CON14 (RK3308_DAC_ANA_OFFSET + 0x38) +#define RK3308_DAC_ANA_CON15 (RK3308_DAC_ANA_OFFSET + 0x3c) + +/* + * These are the bits for registers + */ + +/* RK3308_GLB_CON - REG: 0x0000 */ +#define RK3308_ADC_BIST_WORK BIT(7) +#define RK3308_DAC_BIST_WORK BIT(6) +#define RK3308_ADC_MCLK_GATING BIT(5) +#define RK3308_DAC_MCLK_GATING BIT(4) +#define RK3308_ADC_DIG_WORK BIT(2) +#define RK3308_DAC_DIG_WORK BIT(1) +#define RK3308_SYS_WORK BIT(0) + +/* RK3308_ADC_DIG_CON01 - REG: 0x0004 */ +#define RK3308_ADC_I2S_LRC_POL_REVERSAL BIT(7) +#define RK3308_ADC_I2S_VALID_LEN_SFT 5 +#define RK3308_ADC_I2S_VALID_LEN_MSK (0x3 << RK3308_ADC_I2S_VALID_LEN_SFT) +#define RK3308_ADC_I2S_VALID_LEN_32BITS (0x3 << RK3308_ADC_I2S_VALID_LEN_SFT) +#define RK3308_ADC_I2S_VALID_LEN_24BITS (0x2 << RK3308_ADC_I2S_VALID_LEN_SFT) +#define RK3308_ADC_I2S_VALID_LEN_20BITS (0x1 << RK3308_ADC_I2S_VALID_LEN_SFT) +#define RK3308_ADC_I2S_VALID_LEN_16BITS (0x0 << RK3308_ADC_I2S_VALID_LEN_SFT) +#define RK3308_ADC_I2S_MODE_SFT 3 +#define RK3308_ADC_I2S_MODE_MSK (0x3 << RK3308_ADC_I2S_MODE_SFT) +#define RK3308_ADC_I2S_MODE_PCM (0x3 << RK3308_ADC_I2S_MODE_SFT) +#define RK3308_ADC_I2S_MODE_I2S (0x2 << RK3308_ADC_I2S_MODE_SFT) +#define RK3308_ADC_I2S_MODE_LJ (0x1 << RK3308_ADC_I2S_MODE_SFT) +#define RK3308_ADC_I2S_MODE_RJ (0x0 << RK3308_ADC_I2S_MODE_SFT) +#define RK3308_ADC_I2S_LR_SWAP BIT(1) +#define RK3308_ADC_I2S_MONO BIT(0) + +/* RK3308_ADC_DIG_CON02 - REG: 0x0008 */ +#define RK3308_ADC_IO_MODE_MASTER BIT(5) +#define RK3308_ADC_MODE_MASTER BIT(4) +#define RK3308_ADC_I2S_FRAME_LEN_SFT 2 +#define RK3308_ADC_I2S_FRAME_LEN_MSK (0x3 << RK3308_ADC_I2S_FRAME_LEN_SFT) +#define RK3308_ADC_I2S_FRAME_32BITS (0x3 << RK3308_ADC_I2S_FRAME_LEN_SFT) +#define RK3308_ADC_I2S_FRAME_24BITS (0x2 << RK3308_ADC_I2S_FRAME_LEN_SFT) +#define RK3308_ADC_I2S_FRAME_20BITS (0x1 << RK3308_ADC_I2S_FRAME_LEN_SFT) +#define RK3308_ADC_I2S_FRAME_16BITS (0x0 << RK3308_ADC_I2S_FRAME_LEN_SFT) +#define RK3308_ADC_I2S_WORK BIT(1) +#define RK3308_ADC_I2S_BIT_CLK_POL_REVERSAL BIT(0) + +/* RK3308_ADC_DIG_CON03 - REG: 0x000c */ +#define RK3308_ADC_L_CH_BIST_SFT 2 +#define RK3308_ADC_L_CH_BIST_MSK (0x3 << RK3308_ADC_L_CH_BIST_SFT) +#define RK3308_ADC_L_CH_NORMAL_RIGHT (0x3 << RK3308_ADC_L_CH_BIST_SFT) /* normal mode */ +#define RK3308_ADC_L_CH_BIST_CUBE (0x2 << RK3308_ADC_L_CH_BIST_SFT) +#define RK3308_ADC_L_CH_BIST_SINE (0x1 << RK3308_ADC_L_CH_BIST_SFT) +#define RK3308_ADC_L_CH_NORMAL_LEFT (0x0 << RK3308_ADC_L_CH_BIST_SFT) /* normal mode */ +#define RK3308_ADC_R_CH_BIST_SFT 0 +#define RK3308_ADC_R_CH_BIST_MSK (0x3 << RK3308_ADC_R_CH_BIST_SFT) +#define RK3308_ADC_R_CH_NORMAL_LEFT (0x3 << RK3308_ADC_R_CH_BIST_SFT) /* normal mode */ +#define RK3308_ADC_R_CH_BIST_CUBE (0x2 << RK3308_ADC_R_CH_BIST_SFT) +#define RK3308_ADC_R_CH_BIST_SINE (0x1 << RK3308_ADC_R_CH_BIST_SFT) +#define RK3308_ADC_R_CH_NORMAL_RIGHT (0x0 << RK3308_ADC_R_CH_BIST_SFT) /* normal mode */ + +/* RK3308_ADC_DIG_CON04 - REG: 0x0010 */ +#define RK3308_ADC_HPF_PATH_DIS BIT(2) +#define RK3308_ADC_HPF_CUTOFF_SFT 0 +#define RK3308_ADC_HPF_CUTOFF_MSK (0x3 << RK3308_ADC_HPF_CUTOFF_SFT) +#define RK3308_ADC_HPF_CUTOFF_612HZ (0x2 << RK3308_ADC_HPF_CUTOFF_SFT) +#define RK3308_ADC_HPF_CUTOFF_245HZ (0x1 << RK3308_ADC_HPF_CUTOFF_SFT) +#define RK3308_ADC_HPF_CUTOFF_20HZ (0x0 << RK3308_ADC_HPF_CUTOFF_SFT) + +/* RK3308_ADC_DIG_CON07 - REG: 0x001c */ +#define RK3308_ADCL_DATA_SFT 4 +#define RK3308_ADCR_DATA_SFT 2 +#define RK3308_ADCL_DATA_SEL_ADCL BIT(1) +#define RK3308_ADCR_DATA_SEL_ADCR BIT(0) + +/* + * RK3308_ALC_L_DIG_CON00 - REG: 0x0040 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON00 - REG: 0x0080 + ch * 0xc0 + */ +#define RK3308_GAIN_ATTACK_JACK BIT(6) +#define RK3308_CTRL_GEN_SFT 4 +#define RK3308_CTRL_GEN_MSK (0x3 << RK3308_ALC_CTRL_GEN_SFT) +#define RK3308_CTRL_GEN_JACK3 (0x3 << RK3308_ALC_CTRL_GEN_SFT) +#define RK3308_CTRL_GEN_JACK2 (0x2 << RK3308_ALC_CTRL_GEN_SFT) +#define RK3308_CTRL_GEN_JACK1 (0x1 << RK3308_ALC_CTRL_GEN_SFT) +#define RK3308_CTRL_GEN_NORMAL (0x0 << RK3308_ALC_CTRL_GEN_SFT) +#define RK3308_AGC_HOLD_TIME_SFT 0 +#define RK3308_AGC_HOLD_TIME_MSK (0xf << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_1S (0xa << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_512MS (0x9 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_256MS (0x8 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_128MS (0x7 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_64MS (0x6 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_32MS (0x5 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_16MS (0x4 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_8MS (0x3 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_4MS (0x2 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_2MS (0x1 << RK3308_AGC_HOLD_TIME_SFT) +#define RK3308_AGC_HOLD_TIME_0MS (0x0 << RK3308_AGC_HOLD_TIME_SFT) + +/* + * RK3308_ALC_L_DIG_CON01 - REG: 0x0044 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON01 - REG: 0x0084 + ch * 0xc0 + */ +#define RK3308_AGC_DECAY_TIME_SFT 4 +#define RK3308_AGC_ATTACK_TIME_SFT 0 + +/* + * RK3308_ALC_L_DIG_CON02 - REG: 0x0048 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON02 - REG: 0x0088 + ch * 0xc0 + */ +#define RK3308_AGC_MODE_LIMITER BIT(7) +#define RK3308_AGC_ZERO_CRO_EN BIT(6) +#define RK3308_AGC_AMP_RECOVER_GAIN BIT(5) +#define RK3308_AGC_FAST_DEC_EN BIT(4) +#define RK3308_AGC_NOISE_GATE_EN BIT(3) +#define RK3308_AGC_NOISE_GATE_THRESH_SFT 0 +#define RK3308_AGC_NOISE_GATE_THRESH_MSK (0x7 << RK3308_AGC_NOISE_GATE_THRESH_SFT) + +/* + * RK3308_ALC_L_DIG_CON03 - REG: 0x004c + ch * 0xc0 + * RK3308_ALC_R_DIG_CON03 - REG: 0x008c + ch * 0xc0 + */ +#define RK3308_AGC_PGA_ZERO_CRO_EN BIT(5) +#define RK3308_AGC_PGA_GAIN_MAX 0x1f +#define RK3308_AGC_PGA_GAIN_MIN 0 +#define RK3308_AGC_PGA_GAIN_SFT 0 + +/* + * RK3308_ALC_L_DIG_CON04 - REG: 0x0050 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON04 - REG: 0x0090 + ch * 0xc0 + */ +#define RK3308_AGC_SLOW_CLK_EN BIT(3) +#define RK3308_AGC_APPROX_RATE_SFT 0 +#define RK3308_AGC_APPROX_RATE_MSK (0x7 << RK3308_AGC_APPROX_RATE_SFT) + +/* + * RK3308_ALC_L_DIG_CON05 - REG: 0x0054 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON05 - REG: 0x0094 + ch * 0xc0 + */ +#define RK3308_AGC_LO_8BITS_AGC_MAX_MSK 0xff + +/* + * RK3308_ALC_L_DIG_CON06 - REG: 0x0058 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON06 - REG: 0x0098 + ch * 0xc0 + */ +#define RK3308_AGC_HI_8BITS_AGC_MAX_MSK 0xff + +/* + * RK3308_ALC_L_DIG_CON07 - REG: 0x005c + ch * 0xc0 + * RK3308_ALC_R_DIG_CON07 - REG: 0x009c + ch * 0xc0 + */ +#define RK3308_AGC_LO_8BITS_AGC_MIN_MSK 0xff + +/* + * RK3308_ALC_L_DIG_CON08 - REG: 0x0060 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON08 - REG: 0x00a0 + ch * 0xc0 + */ +#define RK3308_AGC_HI_8BITS_AGC_MIN_MSK 0xff + +/* + * RK3308_ALC_L_DIG_CON09 - REG: 0x0064 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON09 - REG: 0x00a4 + ch * 0xc0 + */ +#define RK3308_AGC_FUNC_SEL BIT(6) +#define RK3308_AGC_MAX_GAIN_PGA_MAX 0x7 +#define RK3308_AGC_MAX_GAIN_PGA_MIN 0 +#define RK3308_AGC_MAX_GAIN_PGA_SFT 3 +#define RK3308_AGC_MAX_GAIN_PGA_MSK (0x7 << RK3308_AGC_MAX_GAIN_PGA_SFT) +#define RK3308_AGC_MIN_GAIN_PGA_MAX 0x7 +#define RK3308_AGC_MIN_GAIN_PGA_MIN 0 +#define RK3308_AGC_MIN_GAIN_PGA_SFT 0 +#define RK3308_AGC_MIN_GAIN_PGA_MSK (0x7 << RK3308_AGC_MIN_GAIN_PGA_SFT) + +/* + * RK3308_ALC_L_DIG_CON12 - REG: 0x0068 + ch * 0xc0 + * RK3308_ALC_R_DIG_CON12 - REG: 0x00a8 + ch * 0xc0 + */ +#define RK3308_AGC_GAIN_MSK 0x1f + +/* RK3308_DAC_DIG_CON01 - REG: 0x0304 */ +#define RK3308_DAC_I2S_LRC_POL_REVERSAL BIT(7) +#define RK3308_DAC_I2S_VALID_LEN_SFT 5 +#define RK3308_DAC_I2S_VALID_LEN_MSK (0x3 << RK3308_DAC_I2S_VALID_LEN_SFT) +#define RK3308_DAC_I2S_VALID_LEN_32BITS (0x3 << RK3308_DAC_I2S_VALID_LEN_SFT) +#define RK3308_DAC_I2S_VALID_LEN_24BITS (0x2 << RK3308_DAC_I2S_VALID_LEN_SFT) +#define RK3308_DAC_I2S_VALID_LEN_20BITS (0x1 << RK3308_DAC_I2S_VALID_LEN_SFT) +#define RK3308_DAC_I2S_VALID_LEN_16BITS (0x0 << RK3308_DAC_I2S_VALID_LEN_SFT) +#define RK3308_DAC_I2S_MODE_SFT 3 +#define RK3308_DAC_I2S_MODE_MSK (0x3 << RK3308_DAC_I2S_MODE_SFT) +#define RK3308_DAC_I2S_MODE_PCM (0x3 << RK3308_DAC_I2S_MODE_SFT) +#define RK3308_DAC_I2S_MODE_I2S (0x2 << RK3308_DAC_I2S_MODE_SFT) +#define RK3308_DAC_I2S_MODE_LJ (0x1 << RK3308_DAC_I2S_MODE_SFT) +#define RK3308_DAC_I2S_MODE_RJ (0x0 << RK3308_DAC_I2S_MODE_SFT) +#define RK3308_DAC_I2S_LR_SWAP BIT(2) + +/* RK3308_DAC_DIG_CON02 - REG: 0x0308 */ +#define RK3308BS_DAC_IO_MODE_MASTER BIT(7) +#define RK3308BS_DAC_MODE_MASTER BIT(6) +#define RK3308_DAC_IO_MODE_MASTER BIT(5) +#define RK3308_DAC_MODE_MASTER BIT(4) +#define RK3308_DAC_I2S_FRAME_LEN_SFT 2 +#define RK3308_DAC_I2S_FRAME_LEN_MSK (0x3 << RK3308_DAC_I2S_FRAME_LEN_SFT) +#define RK3308_DAC_I2S_FRAME_32BITS (0x3 << RK3308_DAC_I2S_FRAME_LEN_SFT) +#define RK3308_DAC_I2S_FRAME_24BITS (0x2 << RK3308_DAC_I2S_FRAME_LEN_SFT) +#define RK3308_DAC_I2S_FRAME_20BITS (0x1 << RK3308_DAC_I2S_FRAME_LEN_SFT) +#define RK3308_DAC_I2S_FRAME_16BITS (0x0 << RK3308_DAC_I2S_FRAME_LEN_SFT) +#define RK3308_DAC_I2S_WORK BIT(1) +#define RK3308_DAC_I2S_BIT_CLK_POL_REVERSAL BIT(0) + +/* RK3308_DAC_DIG_CON03 - REG: 0x030C */ +#define RK3308_DAC_L_CH_BIST_SFT 2 +#define RK3308_DAC_L_CH_BIST_MSK (0x3 << RK3308_DAC_L_CH_BIST_SFT) +#define RK3308_DAC_L_CH_BIST_LEFT (0x3 << RK3308_DAC_L_CH_BIST_SFT) /* normal mode */ +#define RK3308_DAC_L_CH_BIST_CUBE (0x2 << RK3308_DAC_L_CH_BIST_SFT) +#define RK3308_DAC_L_CH_BIST_SINE (0x1 << RK3308_DAC_L_CH_BIST_SFT) +#define RK3308_DAC_L_CH_BIST_RIGHT (0x0 << RK3308_DAC_L_CH_BIST_SFT) /* normal mode */ +#define RK3308_DAC_R_CH_BIST_SFT 0 +#define RK3308_DAC_R_CH_BIST_MSK (0x3 << RK3308_DAC_R_CH_BIST_SFT) +#define RK3308_DAC_R_CH_BIST_LEFT (0x3 << RK3308_DAC_R_CH_BIST_SFT) /* normal mode */ +#define RK3308_DAC_R_CH_BIST_CUBE (0x2 << RK3308_DAC_R_CH_BIST_SFT) +#define RK3308_DAC_R_CH_BIST_SINE (0x1 << RK3308_DAC_R_CH_BIST_SFT) +#define RK3308_DAC_R_CH_BIST_RIGHT (0x0 << RK3308_DAC_R_CH_BIST_SFT) /* normal mode */ + +/* RK3308_DAC_DIG_CON04 - REG: 0x0310 */ +/* Versions up to B: */ +#define RK3308_DAC_MODULATOR_GAIN_SFT 4 +#define RK3308_DAC_MODULATOR_GAIN_MSK (0x7 << RK3308_DAC_MODULATOR_GAIN_SFT) +#define RK3308_DAC_CIC_IF_GAIN_SFT 0 +#define RK3308_DAC_CIC_IF_GAIN_MSK (0x7 << RK3308_DAC_CIC_IF_GAIN_SFT) +/* Version C: */ +#define RK3308BS_DAC_DIG_GAIN_SFT 0 +#define RK3308BS_DAC_DIG_GAIN_MSK (0xff << RK3308BS_DAC_DIG_GAIN_SFT) +#define RK3308BS_DAC_DIG_GAIN_0DB (0xed << RK3308BS_DAC_DIG_GAIN_SFT) + +/* RK3308BS_ADC_DIG_CON05..06 (Version C only) */ +#define RK3308_ADC_DIG_VOL_CON_x_SFT 0 +#define RK3308_ADC_DIG_VOL_CON_x_MSK (0xff << RK3308_ADC_DIG_VOL_CON_x_SFT) +#define RK3308_ADC_DIG_VOL_CON_x_0DB (0xc2 << RK3308_ADC_DIG_VOL_CON_x_SFT) + +/* RK3308_DAC_DIG_CON05 - REG: 0x0314 */ +#define RK3308_DAC_L_REG_CTL_INDATA BIT(2) +#define RK3308_DAC_R_REG_CTL_INDATA BIT(1) + +/* RK3308_DAC_DIG_CON10 - REG: 0x0328 */ +#define RK3308_DAC_DATA_HI4(x) ((x) & 0xf) + +/* RK3308_DAC_DIG_CON11 - REG: 0x032c */ +#define RK3308_DAC_DATA_LO8(x) ((x) & 0xff) + +/* RK3308_ADC_ANA_CON00 - REG: 0x0340 */ +#define RK3308_ADC_CH1_CH2_MIC_ALL_MSK (0xff << 0) +#define RK3308_ADC_CH1_CH2_MIC_ALL 0xff +#define RK3308_ADC_CH2_MIC_UNMUTE BIT(7) +#define RK3308_ADC_CH2_MIC_WORK BIT(6) +#define RK3308_ADC_CH2_MIC_EN BIT(5) +#define RK3308_ADC_CH2_BUF_REF_EN BIT(4) +#define RK3308_ADC_CH1_MIC_UNMUTE BIT(3) +#define RK3308_ADC_CH1_MIC_WORK BIT(2) +#define RK3308_ADC_CH1_MIC_EN BIT(1) +#define RK3308_ADC_CH1_BUF_REF_EN BIT(0) + +/* RK3308_ADC_ANA_CON01 - REG: 0x0344 + * + * The PGA of MIC-INs: + * - HW version A: + * 0x0 - MIC1~MIC8 0 dB (recommended when ADC used as loopback) + * 0x3 - MIC1~MIC8 20 dB (recommended when ADC used as MIC input) + * - HW version B: + * 0x0 - MIC1~MIC8 0 dB + * 0x1 - MIC1~MIC8 6.6 dB + * 0x2 - MIC1~MIC8 13 dB + * 0x3 - MIC1~MIC8 20 dB + */ +#define RK3308_ADC_CH2_MIC_GAIN_MAX 0x3 +#define RK3308_ADC_CH2_MIC_GAIN_MIN 0 +#define RK3308_ADC_CH2_MIC_GAIN_SFT 4 +#define RK3308_ADC_CH2_MIC_GAIN_MSK (0x3 << RK3308_ADC_CH2_MIC_GAIN_SFT) +#define RK3308_ADC_CH2_MIC_GAIN_20DB (0x3 << RK3308_ADC_CH2_MIC_GAIN_SFT) +#define RK3308_ADC_CH2_MIC_GAIN_13DB (0x2 << RK3308_ADC_CH2_MIC_GAIN_SFT) +#define RK3308_ADC_CH2_MIC_GAIN_6_6DB (0x1 << RK3308_ADC_CH2_MIC_GAIN_SFT) +#define RK3308_ADC_CH2_MIC_GAIN_0DB (0x0 << RK3308_ADC_CH2_MIC_GAIN_SFT) + +#define RK3308_ADC_CH1_MIC_GAIN_MAX 0x3 +#define RK3308_ADC_CH1_MIC_GAIN_MIN 0 +#define RK3308_ADC_CH1_MIC_GAIN_SFT 0 +#define RK3308_ADC_CH1_MIC_GAIN_MSK (0x3 << RK3308_ADC_CH1_MIC_GAIN_SFT) +#define RK3308_ADC_CH1_MIC_GAIN_20DB (0x3 << RK3308_ADC_CH1_MIC_GAIN_SFT) +#define RK3308_ADC_CH1_MIC_GAIN_13DB (0x2 << RK3308_ADC_CH1_MIC_GAIN_SFT) +#define RK3308_ADC_CH1_MIC_GAIN_6_6DB (0x1 << RK3308_ADC_CH1_MIC_GAIN_SFT) +#define RK3308_ADC_CH1_MIC_GAIN_0DB (0x0 << RK3308_ADC_CH1_MIC_GAIN_SFT) + +/* RK3308_ADC_ANA_CON02 - REG: 0x0348 */ +#define RK3308_ADC_CH2_ZEROCROSS_DET_EN BIT(6) +#define RK3308_ADC_CH2_ALC_WORK BIT(5) +#define RK3308_ADC_CH2_ALC_EN BIT(4) +#define RK3308_ADC_CH1_ZEROCROSS_DET_EN BIT(2) +#define RK3308_ADC_CH1_ALC_WORK BIT(1) +#define RK3308_ADC_CH1_ALC_EN BIT(0) + +/* RK3308_ADC_ANA_CON03 - REG: 0x034c */ +#define RK3308_ADC_CH1_ALC_GAIN_MAX 0x1f +#define RK3308_ADC_CH1_ALC_GAIN_MIN 0 +#define RK3308_ADC_CH1_ALC_GAIN_SFT 0 +#define RK3308_ADC_CH1_ALC_GAIN_MSK (0x1f << RK3308_ADC_CH1_ALC_GAIN_SFT) +#define RK3308_ADC_CH1_ALC_GAIN_0DB (0x0c << RK3308_ADC_CH1_ALC_GAIN_SFT) + +/* RK3308_ADC_ANA_CON04 - REG: 0x0350 */ +#define RK3308_ADC_CH2_ALC_GAIN_MAX 0x1f +#define RK3308_ADC_CH2_ALC_GAIN_MIN 0 +#define RK3308_ADC_CH2_ALC_GAIN_SFT 0 +#define RK3308_ADC_CH2_ALC_GAIN_MSK (0x1f << RK3308_ADC_CH2_ALC_GAIN_SFT) +#define RK3308_ADC_CH2_ALC_GAIN_0DB (0x0c << RK3308_ADC_CH2_ALC_GAIN_SFT) + +/* RK3308_ADC_ANA_CON05 - REG: 0x0354 */ +#define RK3308_ADC_CH2_ADC_WORK BIT(6) +#define RK3308_ADC_CH2_ADC_EN BIT(5) +#define RK3308_ADC_CH2_CLK_EN BIT(4) +#define RK3308_ADC_CH1_ADC_WORK BIT(2) +#define RK3308_ADC_CH1_ADC_EN BIT(1) +#define RK3308_ADC_CH1_CLK_EN BIT(0) + +/* RK3308_ADC_ANA_CON06 - REG: 0x0358 */ +#define RK3308_ADC_CURRENT_EN BIT(0) + +/* RK3308_ADC_ANA_CON07 - REG: 0x035c */ +/* Note: The register configuration is only valid for ADC2 */ +#define RK3308_ADC_CH2_IN_SEL_SFT 6 +#define RK3308_ADC_CH2_IN_SEL_MSK (0x3 << RK3308_ADC_CH2_IN_SEL_SFT) +#define RK3308_ADC_CH2_IN_LINEIN_MIC (0x3 << RK3308_ADC_CH2_IN_SEL_SFT) +#define RK3308_ADC_CH2_IN_LINEIN (0x2 << RK3308_ADC_CH2_IN_SEL_SFT) +#define RK3308_ADC_CH2_IN_MIC (0x1 << RK3308_ADC_CH2_IN_SEL_SFT) +#define RK3308_ADC_CH2_IN_NONE (0x0 << RK3308_ADC_CH2_IN_SEL_SFT) +/* Note: The register configuration is only valid for ADC1 */ +#define RK3308_ADC_CH1_IN_SEL_SFT 4 +#define RK3308_ADC_CH1_IN_SEL_MSK (0x3 << RK3308_ADC_CH1_IN_SEL_SFT) +#define RK3308_ADC_CH1_IN_LINEIN_MIC (0x3 << RK3308_ADC_CH1_IN_SEL_SFT) +#define RK3308_ADC_CH1_IN_LINEIN (0x2 << RK3308_ADC_CH1_IN_SEL_SFT) +#define RK3308_ADC_CH1_IN_MIC (0x1 << RK3308_ADC_CH1_IN_SEL_SFT) +#define RK3308_ADC_CH1_IN_NONE (0x0 << RK3308_ADC_CH1_IN_SEL_SFT) +#define RK3308_ADC_MIC_BIAS_BUF_EN BIT(3) +#define RK3308_ADC_LEVEL_RANGE_MICBIAS_MAX 7 +#define RK3308_ADC_LEVEL_RANGE_MICBIAS_SFT 0 +#define RK3308_ADC_LEVEL_RANGE_MICBIAS_MSK (0x7 << RK3308_ADC_LEVEL_RANGE_MICBIAS_SFT) + +/* RK3308_ADC_ANA_CON08 - REG: 0x0360 */ +#define RK3308_ADC_MICBIAS_CURRENT_EN BIT(4) + +/* RK3308_ADC_ANA_CON10 - REG: 0x0368 */ +#define RK3308_ADC_REF_EN BIT(7) +#define RK3308_ADC_CURRENT_CHARGE_SFT 0 +#define RK3308_ADC_CURRENT_CHARGE_MSK (0x7f << RK3308_ADC_CURRENT_CHARGE_SFT) + +/* RK3308_ADC_ANA_CON11 - REG: 0x036c */ +#define RK3308_ADC_ALCR_CON_GAIN_PGAR_EN BIT(1) +#define RK3308_ADC_ALCL_CON_GAIN_PGAL_EN BIT(0) + +/* RK3308_DAC_ANA_CON00 - REG: 0x0440 */ +#define RK3308_DAC_HEADPHONE_DET_EN BIT(1) +#define RK3308_DAC_CURRENT_EN BIT(0) + +/* RK3308_DAC_ANA_CON01 - REG: 0x0444 */ +#define RK3308_DAC_BUF_REF_R_EN BIT(6) +#define RK3308_DAC_BUF_REF_L_EN BIT(2) +#define RK3308_DAC_HPOUT_POP_SOUND_R_SFT 4 +#define RK3308_DAC_HPOUT_POP_SOUND_L_SFT 0 +// unshifted values for both L and R: +#define RK3308_DAC_HPOUT_POP_SOUND_x_MSK 0x3 +#define RK3308_DAC_HPOUT_POP_SOUND_x_WORK 0x2 +#define RK3308_DAC_HPOUT_POP_SOUND_x_INIT 0x1 + +/* RK3308_DAC_ANA_CON02 - REG: 0x0448 */ +#define RK3308_DAC_R_DAC_WORK BIT(7) +#define RK3308_DAC_R_DAC_EN BIT(6) +#define RK3308_DAC_R_CLK_EN BIT(5) +#define RK3308_DAC_R_REF_EN BIT(4) +#define RK3308_DAC_L_DAC_WORK BIT(3) +#define RK3308_DAC_L_DAC_EN BIT(2) +#define RK3308_DAC_L_CLK_EN BIT(1) +#define RK3308_DAC_L_REF_EN BIT(0) + +/* RK3308_DAC_ANA_CON03 - REG: 0x044c */ +#define RK3308_DAC_R_HPOUT_WORK BIT(6) +#define RK3308_DAC_R_HPOUT_EN BIT(5) +#define RK3308_DAC_R_HPOUT_MUTE_SFT 4 +#define RK3308_DAC_L_HPOUT_WORK BIT(2) +#define RK3308_DAC_L_HPOUT_EN BIT(1) +#define RK3308_DAC_L_HPOUT_MUTE_SFT 0 + +/* RK3308_DAC_ANA_CON04 - REG: 0x0450 */ +#define RK3308_DAC_x_LINEOUT_GAIN_MAX 0x3 +#define RK3308_DAC_R_LINEOUT_GAIN_SFT 6 +#define RK3308_DAC_R_LINEOUT_GAIN_MSK (0x3 << RK3308_DAC_R_LINEOUT_GAIN_SFT) +#define RK3308_DAC_R_LINEOUT_GAIN_0DB (0x3 << RK3308_DAC_R_LINEOUT_GAIN_SFT) +#define RK3308_DAC_R_LINEOUT_GAIN_NDB_1_5 (0x2 << RK3308_DAC_R_LINEOUT_GAIN_SFT) +#define RK3308_DAC_R_LINEOUT_GAIN_NDB_3 (0x1 << RK3308_DAC_R_LINEOUT_GAIN_SFT) +#define RK3308_DAC_R_LINEOUT_GAIN_NDB_6 (0x0 << RK3308_DAC_R_LINEOUT_GAIN_SFT) +#define RK3308_DAC_R_LINEOUT_MUTE_SFT 5 +#define RK3308_DAC_R_LINEOUT_EN BIT(4) +#define RK3308_DAC_L_LINEOUT_GAIN_SFT 2 +#define RK3308_DAC_L_LINEOUT_GAIN_MSK (0x3 << RK3308_DAC_L_LINEOUT_GAIN_SFT) +#define RK3308_DAC_L_LINEOUT_GAIN_0DB (0x3 << RK3308_DAC_L_LINEOUT_GAIN_SFT) +#define RK3308_DAC_L_LINEOUT_GAIN_NDB_1_5 (0x2 << RK3308_DAC_L_LINEOUT_GAIN_SFT) +#define RK3308_DAC_L_LINEOUT_GAIN_NDB_3 (0x1 << RK3308_DAC_L_LINEOUT_GAIN_SFT) +#define RK3308_DAC_L_LINEOUT_GAIN_NDB_6 (0x0 << RK3308_DAC_L_LINEOUT_GAIN_SFT) +#define RK3308_DAC_L_LINEOUT_MUTE_SFT 1 +#define RK3308_DAC_L_LINEOUT_EN BIT(0) + +/* RK3308_DAC_ANA_CON05 - REG: 0x0454, step is 1.5db */ +/* RK3308_DAC_ANA_CON06 - REG: 0x0458, step is 1.5db */ +#define RK3308_DAC_x_HPOUT_GAIN_MAX 0x1e +#define RK3308_DAC_x_HPOUT_GAIN_SFT 0 +#define RK3308_DAC_x_HPOUT_GAIN_MSK (0x1f << RK3308_DAC_x_HPOUT_GAIN_SFT) +#define RK3308_DAC_x_HPOUT_GAIN_MIN (0x00 << RK3308_DAC_x_HPOUT_GAIN_SFT) + +/* RK3308_DAC_ANA_CON07 - REG: 0x045c */ +#define RK3308_DAC_R_HPOUT_DRV_SFT 4 +#define RK3308_DAC_R_HPOUT_DRV_MSK (0xf << RK3308_DAC_R_HPOUT_DRV_SFT) +#define RK3308_DAC_L_HPOUT_DRV_SFT 0 +#define RK3308_DAC_L_HPOUT_DRV_MSK (0xf << RK3308_DAC_L_HPOUT_DRV_SFT) + +/* RK3308_DAC_ANA_CON08 - REG: 0x0460 */ +#define RK3308_DAC_R_LINEOUT_DRV_SFT 4 +#define RK3308_DAC_R_LINEOUT_DRV_MSK (0xf << RK3308_DAC_R_LINEOUT_DRV_SFT) +#define RK3308_DAC_L_LINEOUT_DRV_SFT 0 +#define RK3308_DAC_L_LINEOUT_DRV_MSK (0xf << RK3308_DAC_L_LINEOUT_DRV_SFT) + +/* RK3308_DAC_ANA_CON12 - REG: 0x0470 */ +#define RK3308_DAC_R_HPMIX_SEL_SFT 6 +#define RK3308_DAC_R_HPMIX_SEL_MSK (0x3 << RK3308_DAC_R_HPMIX_SEL_SFT) +#define RK3308_DAC_R_HPMIX_LINEIN_I2S (0x3 << RK3308_DAC_R_HPMIX_SEL_SFT) +#define RK3308_DAC_R_HPMIX_LINEIN (0x2 << RK3308_DAC_R_HPMIX_SEL_SFT) +#define RK3308_DAC_R_HPMIX_I2S (0x1 << RK3308_DAC_R_HPMIX_SEL_SFT) +#define RK3308_DAC_R_HPMIX_NONE (0x0 << RK3308_DAC_R_HPMIX_SEL_SFT) +#define RK3308_DAC_L_HPMIX_SEL_SFT 2 +#define RK3308_DAC_L_HPMIX_SEL_MSK (0x3 << RK3308_DAC_L_HPMIX_SEL_SFT) +#define RK3308_DAC_L_HPMIX_LINEIN_I2S (0x3 << RK3308_DAC_L_HPMIX_SEL_SFT) +#define RK3308_DAC_L_HPMIX_LINEIN (0x2 << RK3308_DAC_L_HPMIX_SEL_SFT) +#define RK3308_DAC_L_HPMIX_I2S (0x1 << RK3308_DAC_L_HPMIX_SEL_SFT) +#define RK3308_DAC_L_HPMIX_NONE (0x0 << RK3308_DAC_L_HPMIX_SEL_SFT) +#define RK3308_DAC_x_HPMIX_GAIN_MIN 0x1 /* 0x0 and 0x3 are reserved */ +#define RK3308_DAC_x_HPMIX_GAIN_MAX 0x2 +#define RK3308_DAC_R_HPMIX_GAIN_SFT 4 +#define RK3308_DAC_R_HPMIX_GAIN_MSK (0x3 << RK3308_DAC_R_HPMIX_GAIN_SFT) +#define RK3308_DAC_R_HPMIX_GAIN_0DB (0x2 << RK3308_DAC_R_HPMIX_GAIN_SFT) +#define RK3308_DAC_R_HPMIX_GAIN_NDB_6 (0x1 << RK3308_DAC_R_HPMIX_GAIN_SFT) +#define RK3308_DAC_L_HPMIX_GAIN_SFT 0 +#define RK3308_DAC_L_HPMIX_GAIN_MSK (0x3 << RK3308_DAC_L_HPMIX_GAIN_SFT) +#define RK3308_DAC_L_HPMIX_GAIN_0DB (0x2 << RK3308_DAC_L_HPMIX_GAIN_SFT) +#define RK3308_DAC_L_HPMIX_GAIN_NDB_6 (0x1 << RK3308_DAC_L_HPMIX_GAIN_SFT) + +/* RK3308_DAC_ANA_CON13 - REG: 0x0474 */ +#define RK3308_DAC_R_HPMIX_UNMUTE BIT(6) +#define RK3308_DAC_R_HPMIX_WORK BIT(5) +#define RK3308_DAC_R_HPMIX_EN BIT(4) +#define RK3308_DAC_L_HPMIX_UNMUTE BIT(2) +#define RK3308_DAC_L_HPMIX_WORK BIT(1) +#define RK3308_DAC_L_HPMIX_EN BIT(0) + +/* RK3308_DAC_ANA_CON14 - REG: 0x0478 */ +#define RK3308_DAC_VCM_LINEOUT_EN (0x1 << 4) +#define RK3308_DAC_CURRENT_CHARGE_SFT 0 +#define RK3308_DAC_CURRENT_CHARGE_MSK (0xf << RK3308_DAC_CURRENT_CHARGE_SFT) + +/* RK3308_DAC_ANA_CON15 - REG: 0x047C */ +#define RK3308_DAC_LINEOUT_POP_SOUND_R_SFT 4 +#define RK3308_DAC_LINEOUT_POP_SOUND_R_MSK (0x3 << RK3308_DAC_LINEOUT_POP_SOUND_R_SFT) +#define RK3308_DAC_R_SEL_DC_FROM_INTERNAL (0x2 << RK3308_DAC_LINEOUT_POP_SOUND_R_SFT) +#define RK3308_DAC_R_SEL_DC_FROM_VCM (0x1 << RK3308_DAC_LINEOUT_POP_SOUND_R_SFT) +#define RK3308_DAC_R_SEL_LINEOUT_FROM_INTERNAL (0x0 << RK3308_DAC_LINEOUT_POP_SOUND_R_SFT) +#define RK3308_DAC_LINEOUT_POP_SOUND_L_SFT 0 +#define RK3308_DAC_LINEOUT_POP_SOUND_L_MSK (0x3 << RK3308_DAC_LINEOUT_POP_SOUND_L_SFT) +#define RK3308_DAC_L_SEL_DC_FROM_INTERNAL (0x2 << RK3308_DAC_LINEOUT_POP_SOUND_L_SFT) +#define RK3308_DAC_L_SEL_DC_FROM_VCM (0x1 << RK3308_DAC_LINEOUT_POP_SOUND_L_SFT) +#define RK3308_DAC_L_SEL_LINEOUT_FROM_INTERNAL (0x0 << RK3308_DAC_LINEOUT_POP_SOUND_L_SFT) + +#endif /* __RK3308_CODEC_H__ */ From 087777347bea060f82fa97827e7d1f625c0f9376 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:03:56 -0500 Subject: [PATCH 057/603] ASoC: intel: sof_sdw: Make find_codec_info_part() return a pointer Rather than returning an index simply return a pointer to the located codec info, this simplifies all the callers which only want to access the codec info structure. Also remove the inline specifier the function is fairly large for an inline function, let the compiler decide. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 79 +++++++++++++++----------------- 1 file changed, 36 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 08f330ed5c2ea..08b19a32cf9cb 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1055,7 +1055,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { }, }; -static inline int find_codec_info_part(const u64 adr) +static struct sof_sdw_codec_info *find_codec_info_part(const u64 adr) { unsigned int part_id, sdw_version; int i; @@ -1070,9 +1070,9 @@ static inline int find_codec_info_part(const u64 adr) if (part_id == codec_info_list[i].part_id && (!codec_info_list[i].version_id || sdw_version == codec_info_list[i].version_id)) - return i; + return &codec_info_list[i]; - return -EINVAL; + return NULL; } @@ -1116,7 +1116,6 @@ static int get_dailink_info(struct device *dev, for (; adr_link->num_adr; adr_link++) { const struct snd_soc_acpi_endpoint *endpoint; struct sof_sdw_codec_info *codec_info; - int codec_index; int stream; u64 adr; @@ -1126,11 +1125,9 @@ static int get_dailink_info(struct device *dev, for (i = 0; i < adr_link->num_adr; i++) { adr = adr_link->adr_d[i].adr; - codec_index = find_codec_info_part(adr); - if (codec_index < 0) - return codec_index; - - codec_info = &codec_info_list[codec_index]; + codec_info = find_codec_info_part(adr); + if (!codec_info) + return -EINVAL; *codecs_num += codec_info->dai_num; @@ -1257,11 +1254,11 @@ static int fill_sdw_codec_dlc(struct device *dev, { unsigned int sdw_version, unique_id, mfg_id, link_id, part_id, class_id; u64 adr = adr_link->adr_d[adr_index].adr; - int codec_index; + struct sof_sdw_codec_info *codec_info; - codec_index = find_codec_info_part(adr); - if (codec_index < 0) - return codec_index; + codec_info = find_codec_info_part(adr); + if (!codec_info) + return -EINVAL; sdw_version = SDW_VERSION(adr); link_id = SDW_DISCO_LINK_ID(adr); @@ -1270,10 +1267,8 @@ static int fill_sdw_codec_dlc(struct device *dev, part_id = SDW_PART_ID(adr); class_id = SDW_CLASS_ID(adr); - if (codec_info_list[codec_index].codec_name) - codec->name = devm_kstrdup(dev, - codec_info_list[codec_index].codec_name, - GFP_KERNEL); + if (codec_info->codec_name) + codec->name = devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL); else if (is_unique_device(adr_link, sdw_version, mfg_id, part_id, class_id, adr_index)) codec->name = devm_kasprintf(dev, GFP_KERNEL, @@ -1287,7 +1282,7 @@ static int fill_sdw_codec_dlc(struct device *dev, if (!codec->name) return -ENOMEM; - codec->dai_name = codec_info_list[codec_index].dais[dai_index].dai_name; + codec->dai_name = codec_info->dais[dai_index].dai_name; return 0; } @@ -1309,22 +1304,22 @@ static int set_codec_init_func(struct snd_soc_card *card, * we should end immediately if it is not aggregated (group_id=0) */ for ( ; i < adr_link->num_adr; i++) { - int codec_index; + struct sof_sdw_codec_info *codec_info; - codec_index = find_codec_info_part(adr_link->adr_d[i].adr); - if (codec_index < 0) - return codec_index; + codec_info = find_codec_info_part(adr_link->adr_d[i].adr); + if (!codec_info) + return -EINVAL; /* The group_id is > 0 iff the codec is aggregated */ if (adr_link->adr_d[i].endpoints->group_id != group_id) continue; - if (codec_info_list[codec_index].dais[dai_index].init) - codec_info_list[codec_index].dais[dai_index].init(card, - adr_link, - dai_links, - &codec_info_list[codec_index], - playback); + if (codec_info->dais[dai_index].init) + codec_info->dais[dai_index].init(card, + adr_link, + dai_links, + codec_info, + playback); if (!group_id) return 0; } @@ -1481,7 +1476,6 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, int cpu_dai_num; unsigned int group_id; int codec_dlc_index = 0; - int codec_index; int codec_num; int stream; int i = 0; @@ -1541,10 +1535,9 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, } /* find codec info to create BE DAI */ - codec_index = find_codec_info_part(adr_link->adr_d[adr_index].adr); - if (codec_index < 0) - return codec_index; - codec_info = &codec_info_list[codec_index]; + codec_info = find_codec_info_part(adr_link->adr_d[adr_index].adr); + if (!codec_info) + return -EINVAL; if (codec_info->ignore_pch_dmic) *ignore_pch_dmic = true; @@ -1631,7 +1624,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++, playback, group_id, adr_index, dai_index); if (ret < 0) { - dev_err(dev, "failed to init codec %d\n", codec_index); + dev_err(dev, "failed to init codec 0x%x\n", codec_info->part_id); return ret; } } @@ -1649,6 +1642,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) const struct snd_soc_acpi_link_adr *adr_link = mach_params->links; bool aggregation = !(sof_sdw_quirk & SOF_SDW_NO_AGGREGATION); struct snd_soc_codec_conf *codec_conf; + struct sof_sdw_codec_info *codec_info; bool append_dai_type = false; bool ignore_pch_dmic = false; int codec_conf_num = 0; @@ -1660,7 +1654,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) char *name, *cpu_dai_name; char *codec_name, *codec_dai_name; int i, j, be_id = 0; - int codec_index; int hdmi_num; int ret; @@ -1729,10 +1722,10 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) */ for (i = 0; i < adr_link->num_adr; i++) { /* find codec info to get dai_num */ - codec_index = find_codec_info_part(adr_link->adr_d[i].adr); - if (codec_index < 0) - return codec_index; - if (codec_info_list[codec_index].dai_num > 1) { + codec_info = find_codec_info_part(adr_link->adr_d[i].adr); + if (!codec_info) + return -EINVAL; + if (codec_info->dai_num > 1) { append_dai_type = true; goto out; } @@ -1762,11 +1755,11 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) continue; /* find codec info to get dai_num */ - codec_index = find_codec_info_part(adr_link->adr_d[i].adr); - if (codec_index < 0) - return codec_index; + codec_info = find_codec_info_part(adr_link->adr_d[i].adr); + if (!codec_info) + return -EINVAL; - for (j = 0; j < codec_info_list[codec_index].dai_num ; j++) { + for (j = 0; j < codec_info->dai_num ; j++) { int current_be_id; ret = create_sdw_dailink(card, &link_index, dai_links, From c2c7a8b3848127f3355109d72c865b7741af9f0c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:03:57 -0500 Subject: [PATCH 058/603] ASoC: intel: sof_sdw: Make find_codec_info_acpi() return a pointer Rather than returning an index simply return a pointer to the located codec info, this simplifies all the callers which only want to access the codec info structure. Also remove the inline specifier the function is fairly large for an inline function, let the compiler decide. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++++---------------- 1 file changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 08b19a32cf9cb..517ff44de1f0f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1076,18 +1076,18 @@ static struct sof_sdw_codec_info *find_codec_info_part(const u64 adr) } -static inline int find_codec_info_acpi(const u8 *acpi_id) +static struct sof_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id) { int i; if (!acpi_id[0]) - return -EINVAL; + return NULL; for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) if (!memcmp(codec_info_list[i].acpi_id, acpi_id, ACPI_ID_LEN)) - return i; + return &codec_info_list[i]; - return -EINVAL; + return NULL; } /* @@ -1643,18 +1643,19 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) bool aggregation = !(sof_sdw_quirk & SOF_SDW_NO_AGGREGATION); struct snd_soc_codec_conf *codec_conf; struct sof_sdw_codec_info *codec_info; + struct sof_sdw_codec_info *ssp_info; bool append_dai_type = false; bool ignore_pch_dmic = false; int codec_conf_num = 0; int codec_conf_index = 0; bool group_generated[SDW_MAX_GROUPS] = { }; - int ssp_codec_index, ssp_mask; struct snd_soc_dai_link *dai_links; int num_links, link_index = 0; char *name, *cpu_dai_name; char *codec_name, *codec_dai_name; int i, j, be_id = 0; int hdmi_num; + int ssp_mask; int ret; ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); @@ -1669,8 +1670,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) * system only when I2S mode is supported, not sdw mode. * Here check ACPI ID to confirm I2S is supported. */ - ssp_codec_index = find_codec_info_acpi(mach->id); - if (ssp_codec_index >= 0) { + ssp_info = find_codec_info_acpi(mach->id); + if (ssp_info) { ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk); ssp_num = hweight_long(ssp_mask); } @@ -1788,30 +1789,28 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto DMIC; for (i = 0, j = 0; ssp_mask; i++, ssp_mask >>= 1) { - struct sof_sdw_codec_info *info; int playback, capture; if (!(ssp_mask & 0x1)) continue; - info = &codec_info_list[ssp_codec_index]; - name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", i); cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", - info->acpi_id, j++); + ssp_info->acpi_id, j++); - playback = info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; - capture = info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; + playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; + capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, name, playback, capture, cpu_dai_name, - codec_name, info->dais[0].dai_name, - NULL, info->ops); + codec_name, ssp_info->dais[0].dai_name, + NULL, ssp_info->ops); if (ret) return ret; - ret = info->dais[0].init(card, NULL, dai_links + link_index, info, 0); + ret = ssp_info->dais[0].init(card, NULL, dai_links + link_index, + ssp_info, 0); if (ret < 0) return ret; From 1329f5b0d9d0b26021b6bd469a41139b9ccef58a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:03:58 -0500 Subject: [PATCH 059/603] ASoC: intel: sof_sdw: Make find_codec_info_dai() return a pointer Rather than returning an index simply return a pointer to the located codec info, this simplifies all the callers which only want to access the codec info structure. Also remove the inline specifier the function is fairly large for an inline function, let the compiler decide. And move the function such that it is located with the other find_codec_info_*() functions. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 39 ++++++++++++++++---------------- 1 file changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 517ff44de1f0f..4bd9f62c48fcc 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1090,6 +1090,23 @@ static struct sof_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id) return NULL; } +static struct sof_sdw_codec_info *find_codec_info_dai(const char *dai_name, + int *dai_index) +{ + int i, j; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + for (j = 0; j < codec_info_list[i].dai_num; j++) { + if (!strcmp(codec_info_list[i].dais[j].dai_name, dai_name)) { + *dai_index = j; + return &codec_info_list[i]; + } + } + } + + return NULL; +} + /* * get BE dailink number and CPU DAI number based on sdw link adr. * Since some sdw slaves may be aggregated, the CPU DAI number @@ -1403,37 +1420,19 @@ static void set_dailink_map(struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps, } } -static inline int find_codec_info_dai(const char *dai_name, int *dai_index) -{ - int i, j; - - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { - for (j = 0; j < codec_info_list[i].dai_num; j++) { - if (!strcmp(codec_info_list[i].dais[j].dai_name, dai_name)) { - *dai_index = j; - return i; - } - } - } - - return -EINVAL; -} - static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct sof_sdw_codec_info *codec_info; struct snd_soc_dai *dai; - int codec_index; int dai_index; int ret; int i; for_each_rtd_codec_dais(rtd, i, dai) { - codec_index = find_codec_info_dai(dai->name, &dai_index); - if (codec_index < 0) + codec_info = find_codec_info_dai(dai->name, &dai_index); + if (!codec_info) return -EINVAL; - codec_info = &codec_info_list[codec_index]; /* * A codec dai can be connected to different dai links for capture and playback, * but we only need to call the rtd_init function once. From 961e694749fb8ddb8591512216e2fa6b4e3f42e2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:03:59 -0500 Subject: [PATCH 060/603] ASoC: intel: sof_sdw: Only pass codec_conf pointer around Rather than passing around a pointer to the codec_conf array, an index into it and a size, simply pass around a pointer to the current codec_conf. This reduces the amount of state moving around. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 28 ++++++++++++---------------- 1 file changed, 12 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 4bd9f62c48fcc..7b61d45df5da2 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1457,10 +1457,8 @@ static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic" static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, struct snd_soc_dai_link *dai_links, int sdw_be_num, const struct snd_soc_acpi_link_adr *adr_link, - struct snd_soc_codec_conf *codec_conf, - int codec_count, int *be_id, - int *codec_conf_index, - bool *ignore_pch_dmic, + struct snd_soc_codec_conf **codec_conf, + int *be_id, bool *ignore_pch_dmic, bool append_dai_type, int adr_index, int dai_index) @@ -1509,7 +1507,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, continue; /* sanity check */ - if (*codec_conf_index >= codec_count) { + if (*codec_conf >= card->codec_conf + card->num_configs) { dev_err(dev, "codec_conf array overflowed\n"); return -EINVAL; } @@ -1520,12 +1518,11 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, if (ret) return ret; - codec_conf[*codec_conf_index].dlc = codecs[codec_dlc_index]; - codec_conf[*codec_conf_index].name_prefix = - adr_link_next->adr_d[j].name_prefix; + (*codec_conf)->dlc = codecs[codec_dlc_index]; + (*codec_conf)->name_prefix = adr_link_next->adr_d[j].name_prefix; codec_dlc_index++; - (*codec_conf_index)++; + (*codec_conf)++; } j = 0; @@ -1646,7 +1643,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) bool append_dai_type = false; bool ignore_pch_dmic = false; int codec_conf_num = 0; - int codec_conf_index = 0; bool group_generated[SDW_MAX_GROUPS] = { }; struct snd_soc_dai_link *dai_links; int num_links, link_index = 0; @@ -1706,6 +1702,9 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (!codec_conf) return -ENOMEM; + card->codec_conf = codec_conf; + card->num_configs = codec_conf_num; + /* SDW */ if (!sdw_be_num) goto SSP; @@ -1764,9 +1763,9 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) ret = create_sdw_dailink(card, &link_index, dai_links, sdw_be_num, adr_link, - codec_conf, codec_conf_num, - ¤t_be_id, &codec_conf_index, - &ignore_pch_dmic, append_dai_type, i, j); + &codec_conf, ¤t_be_id, + &ignore_pch_dmic, + append_dai_type, i, j); if (ret < 0) { dev_err(dev, "failed to create dai link %d\n", link_index); return ret; @@ -1886,9 +1885,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) card->dai_link = dai_links; card->num_links = num_links; - card->codec_conf = codec_conf; - card->num_configs = codec_conf_num; - return 0; } From 634ffef9cbc41b9db2b45974969dda06219ffce1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:00 -0500 Subject: [PATCH 061/603] ASoC: intel: sof_sdw: Set channel map directly from endpoints The current code evenly distributes the CODEC DAIs across the CPU DAIs in the DAI link, but this is just an assumption about how the devices are connected to the host. All the information about which CODEC is connected to which CPU DAI is contained in the endpoints datastructures and those structures are already parsed to work out which CODECs to include in the DAI link. Simply fill in the mapping from CPUs -> CODECs as we parse through the endpoints structures, this will ensure that mapping matches the physical layout. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 31 ++++++++++--------------------- 1 file changed, 10 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 7b61d45df5da2..e2c1822b185fe 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1407,19 +1407,6 @@ static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, return 0; } -static void set_dailink_map(struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps, - int codec_num, int cpu_num) -{ - int step; - int i; - - step = codec_num / cpu_num; - for (i = 0; i < codec_num; i++) { - sdw_codec_ch_maps[i].cpu = i / step; - sdw_codec_ch_maps[i].codec = i; - } -} - static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct sof_sdw_codec_info *codec_info; @@ -1466,6 +1453,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *dev = card->dev; const struct snd_soc_acpi_link_adr *adr_link_next; + struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps; struct snd_soc_dai_link_component *codecs; struct snd_soc_dai_link_component *cpus; struct sof_sdw_codec_info *codec_info; @@ -1488,6 +1476,11 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, if (!codecs) return -ENOMEM; + sdw_codec_ch_maps = devm_kcalloc(dev, codec_num, + sizeof(*sdw_codec_ch_maps), GFP_KERNEL); + if (!sdw_codec_ch_maps) + return -ENOMEM; + /* generate codec name on different links in the same group */ j = adr_index; for (adr_link_next = adr_link; adr_link_next && adr_link_next->num_adr && @@ -1521,6 +1514,9 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, (*codec_conf)->dlc = codecs[codec_dlc_index]; (*codec_conf)->name_prefix = adr_link_next->adr_d[j].name_prefix; + sdw_codec_ch_maps[codec_dlc_index].cpu = i; + sdw_codec_ch_maps[codec_dlc_index].codec = codec_dlc_index; + codec_dlc_index++; (*codec_conf)++; } @@ -1539,7 +1535,6 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, *ignore_pch_dmic = true; for_each_pcm_streams(stream) { - struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps; char *name, *cpu_name; int playback, capture; static const char * const sdw_stream_name[] = { @@ -1558,11 +1553,6 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, return -EINVAL; } - sdw_codec_ch_maps = devm_kcalloc(dev, codec_num, - sizeof(*sdw_codec_ch_maps), GFP_KERNEL); - if (!sdw_codec_ch_maps) - return -ENOMEM; - /* create stream name according to first link id */ if (append_dai_type) { name = devm_kasprintf(dev, GFP_KERNEL, @@ -1614,9 +1604,8 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, * based on wait_for_completion(), tag them as 'nonatomic'. */ dai_links[*link_index].nonatomic = true; - - set_dailink_map(sdw_codec_ch_maps, codec_num, cpu_dai_num); dai_links[*link_index].ch_maps = sdw_codec_ch_maps; + ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++, playback, group_id, adr_index, dai_index); if (ret < 0) { From 0703329606a237c3604230603d58254a8bdf4b81 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:01 -0500 Subject: [PATCH 062/603] ASoC: Intel: sof_sdw: Move get_codec_dai_by_name() into sof_sdw itself Currently sof_sdw relies on sof_board_helpers to provide get_codec_dai_by_name(), but that is the only function from sof_board_helpers it uses and no other machine driver requires that function. There is no reason for sof_sdw to select SSP_COMMON but more and more functions in sof_board_helpers are gaining dependencies on functions that would require stubs for sof_sdw to build without it. Firstly it was sof_ssp_get_codec_name(), as was fixed in commit c1469c3a8a30 ("ASoC: Intel: ssp-common: Add stub for sof_ssp_get_codec_name"), now it is: ERROR: modpost: "sof_ssp_detect_amp_type" [sound/soc/intel/boards/snd-soc-intel-sof-board-helpers.ko] undefined! ERROR: modpost: "sof_ssp_detect_codec_type" [sound/soc/intel/boards/snd-soc-intel-sof-board-helpers.ko] undefined! Rather than adding more stubs, simply move the affected function to the sof_sdw machine driver itself and no longer select SND_SOC_INTEL_SOF_BOARD_HELPERS at all. This should allow work on SSP_COMMON to progress without affecting sof_sdw. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 - sound/soc/intel/boards/sof_board_helpers.c | 18 ------------------ sound/soc/intel/boards/sof_board_helpers.h | 3 --- sound/soc/intel/boards/sof_sdw.c | 18 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 4 ++++ sound/soc/intel/boards/sof_sdw_cs42l42.c | 1 - sound/soc/intel/boards/sof_sdw_rt5682.c | 1 - sound/soc/intel/boards/sof_sdw_rt700.c | 1 - sound/soc/intel/boards/sof_sdw_rt711.c | 1 - sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 1 - .../intel/boards/sof_sdw_rt_sdca_jack_common.c | 1 - sound/soc/intel/boards/sof_ssp_common.h | 7 ------- 12 files changed, 22 insertions(+), 35 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index e5df64fec3199..b7a6ac4b6c0ca 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -678,7 +678,6 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST depends on SOUNDWIRE - select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_MAX98363 select SND_SOC_MAX98373_I2C select SND_SOC_MAX98373_SDW diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index a5135be94f32a..586600680e84c 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -630,24 +630,6 @@ sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk) } EXPORT_SYMBOL_NS(sof_intel_board_get_ctx, SND_SOC_INTEL_SOF_BOARD_HELPERS); -struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, - const char * const dai_name[], int num_dais) -{ - struct snd_soc_dai *dai; - int index; - int i; - - for (index = 0; index < num_dais; index++) - for_each_rtd_codec_dais(rtd, i, dai) - if (strstr(dai->name, dai_name[index])) { - dev_dbg(rtd->card->dev, "get dai %s\n", dai->name); - return dai; - } - - return NULL; -} -EXPORT_SYMBOL_NS(get_codec_dai_by_name, SND_SOC_INTEL_SOF_BOARD_HELPERS); - MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 2f27ad8726f8b..b6a83ca14fa82 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -167,7 +167,4 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, struct sof_card_private * sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk); -struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, - const char * const dai_name[], int num_dais); - #endif /* __SOF_INTEL_BOARD_HELPERS_H */ diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e2c1822b185fe..45732884e3106 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -514,6 +514,24 @@ static struct snd_soc_dai_link_component platform_component[] = { } }; +struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, + const char * const dai_name[], + int num_dais) +{ + struct snd_soc_dai *dai; + int index; + int i; + + for (index = 0; index < num_dais; index++) + for_each_rtd_codec_dais(rtd, i, dai) + if (strstr(dai->name, dai_name[index])) { + dev_dbg(rtd->card->dev, "get dai %s\n", dai->name); + return dai; + } + + return NULL; +} + /* these wrappers are only needed to avoid typecast compilation errors */ int sdw_startup(struct snd_pcm_substream *substream) { diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index b1d57034361c4..a04cbab9bc70d 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -107,6 +107,10 @@ struct mc_private { extern unsigned long sof_sdw_quirk; +struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, + const char * const dai_name[], + int num_dais); + int sdw_startup(struct snd_pcm_substream *substream); int sdw_prepare(struct snd_pcm_substream *substream); int sdw_trigger(struct snd_pcm_substream *substream, int cmd); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index 0dc297f7de011..b999f4e7901a5 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -15,7 +15,6 @@ #include #include #include -#include "sof_board_helpers.h" #include "sof_sdw_common.h" static const struct snd_soc_dapm_widget cs42l42_widgets[] = { diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index 6b008a5a343bc..f812aea64322d 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -15,7 +15,6 @@ #include #include #include -#include "sof_board_helpers.h" #include "sof_sdw_common.h" static const struct snd_soc_dapm_widget rt5682_widgets[] = { diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c index 88e785a54b16f..a2648c900e747 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -13,7 +13,6 @@ #include #include #include -#include "sof_board_helpers.h" #include "sof_sdw_common.h" static const struct snd_soc_dapm_widget rt700_widgets[] = { diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index cdd1587b246c8..9762c48ba7a92 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -15,7 +15,6 @@ #include #include #include -#include "sof_board_helpers.h" #include "sof_sdw_common.h" /* diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index ebb4b58c198b9..9c898c7286af9 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -13,7 +13,6 @@ #include #include #include -#include "sof_board_helpers.h" #include "sof_sdw_common.h" static const struct snd_soc_dapm_widget rt712_spk_widgets[] = { diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 5253d8332780f..fe59a144efeff 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -15,7 +15,6 @@ #include #include #include -#include "sof_board_helpers.h" #include "sof_sdw_common.h" /* diff --git a/sound/soc/intel/boards/sof_ssp_common.h b/sound/soc/intel/boards/sof_ssp_common.h index d24888bc99fde..ff84154ce60a6 100644 --- a/sound/soc/intel/boards/sof_ssp_common.h +++ b/sound/soc/intel/boards/sof_ssp_common.h @@ -68,13 +68,6 @@ enum sof_ssp_codec { enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev); enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev); -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON) const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type); -#else -static inline const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type) -{ - return NULL; -} -#endif #endif /* __SOF_SSP_COMMON_H */ From d36bfa329ae6d94e435d11960936023c03df0d64 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:02 -0500 Subject: [PATCH 063/603] ASoC: Intel: sof_sdw: Move flags to private struct Move the flags ignore_pch_dmic and append_dai_type into the drivers private structure rather than passing them around between functions. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 21 +++++++-------------- sound/soc/intel/boards/sof_sdw_common.h | 2 ++ 2 files changed, 9 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 45732884e3106..5557db68a002b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1463,10 +1463,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, struct snd_soc_dai_link *dai_links, int sdw_be_num, const struct snd_soc_acpi_link_adr *adr_link, struct snd_soc_codec_conf **codec_conf, - int *be_id, bool *ignore_pch_dmic, - bool append_dai_type, - int adr_index, - int dai_index) + int *be_id, int adr_index, int dai_index) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *dev = card->dev; @@ -1549,8 +1546,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, if (!codec_info) return -EINVAL; - if (codec_info->ignore_pch_dmic) - *ignore_pch_dmic = true; + ctx->ignore_pch_dmic |= codec_info->ignore_pch_dmic; for_each_pcm_streams(stream) { char *name, *cpu_name; @@ -1572,7 +1568,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, } /* create stream name according to first link id */ - if (append_dai_type) { + if (ctx->append_dai_type) { name = devm_kasprintf(dev, GFP_KERNEL, sdw_stream_name[stream + 2], cpu_dai_id[0], type_strings[codec_info->dais[dai_index].dai_type]); @@ -1647,8 +1643,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) struct snd_soc_codec_conf *codec_conf; struct sof_sdw_codec_info *codec_info; struct sof_sdw_codec_info *ssp_info; - bool append_dai_type = false; - bool ignore_pch_dmic = false; int codec_conf_num = 0; bool group_generated[SDW_MAX_GROUPS] = { }; struct snd_soc_dai_link *dai_links; @@ -1732,7 +1726,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (!codec_info) return -EINVAL; if (codec_info->dai_num > 1) { - append_dai_type = true; + ctx->append_dai_type = true; goto out; } for (j = 0; j < i; j++) { @@ -1740,7 +1734,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) SDW_PART_ID(adr_link->adr_d[j].adr)) || (SDW_MFG_ID(adr_link->adr_d[i].adr) != SDW_MFG_ID(adr_link->adr_d[j].adr))) { - append_dai_type = true; + ctx->append_dai_type = true; goto out; } } @@ -1771,8 +1765,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) ret = create_sdw_dailink(card, &link_index, dai_links, sdw_be_num, adr_link, &codec_conf, ¤t_be_id, - &ignore_pch_dmic, - append_dai_type, i, j); + i, j); if (ret < 0) { dev_err(dev, "failed to create dai link %d\n", link_index); return ret; @@ -1825,7 +1818,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) DMIC: /* dmic */ if (dmic_num > 0) { - if (ignore_pch_dmic) { + if (ctx->ignore_pch_dmic) { dev_warn(dev, "Ignoring PCH DMIC\n"); goto HDMI; } diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index a04cbab9bc70d..1daf98a5849f7 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -103,6 +103,8 @@ struct mc_private { struct device *amp_dev1, *amp_dev2; /* To store SDW Pin index for each SoundWire link */ unsigned int sdw_pin_index[SDW_MAX_LINKS]; + bool append_dai_type; + bool ignore_pch_dmic; }; extern unsigned long sof_sdw_quirk; From c577b747b9a0ad32047dcfa01d0ea7e2441cf590 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:03 -0500 Subject: [PATCH 064/603] ASoC: Intel: sof_sdw: Only pass dai_link pointer around Rather than passing around a pointer to the dai_link array and an index into this array, simply pass a pointer to the current dai_link. Also move the DAI link pointer sanity check to the end of the DAI link creation, and change it to a warn on. This check should only be hit if there is a serious bug in the machine driver, so checking it on each iteration is excessive. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 62 +++++++++++++++----------------- 1 file changed, 29 insertions(+), 33 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5557db68a002b..0ffa6e36dbed3 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1459,8 +1459,8 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; -static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, - struct snd_soc_dai_link *dai_links, int sdw_be_num, +static int create_sdw_dailink(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, const struct snd_soc_acpi_link_adr *adr_link, struct snd_soc_codec_conf **codec_conf, int *be_id, int adr_index, int dai_index) @@ -1597,35 +1597,28 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, cpus[k].dai_name = cpu_name; } - /* - * We create sdw dai links at first stage, so link index should - * not be larger than sdw_be_num - */ - if (*link_index >= sdw_be_num) { - dev_err(dev, "invalid dai link index %d\n", *link_index); - return -EINVAL; - } - playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); capture = (stream == SNDRV_PCM_STREAM_CAPTURE); - init_dai_link(dev, dai_links + *link_index, be_id, name, - playback, capture, cpus, cpu_dai_num, codecs, codec_num, + init_dai_link(dev, *dai_links, be_id, name, playback, capture, + cpus, cpu_dai_num, codecs, codec_num, sof_sdw_rtd_init, &sdw_ops); /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations * based on wait_for_completion(), tag them as 'nonatomic'. */ - dai_links[*link_index].nonatomic = true; - dai_links[*link_index].ch_maps = sdw_codec_ch_maps; + (*dai_links)->nonatomic = true; + (*dai_links)->ch_maps = sdw_codec_ch_maps; - ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++, + ret = set_codec_init_func(card, adr_link, *dai_links, playback, group_id, adr_index, dai_index); if (ret < 0) { dev_err(dev, "failed to init codec 0x%x\n", codec_info->part_id); return ret; } + + (*dai_links)++; } return 0; @@ -1646,7 +1639,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) int codec_conf_num = 0; bool group_generated[SDW_MAX_GROUPS] = { }; struct snd_soc_dai_link *dai_links; - int num_links, link_index = 0; + int num_links; char *name, *cpu_dai_name; char *codec_name, *codec_dai_name; int i, j, be_id = 0; @@ -1703,6 +1696,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (!codec_conf) return -ENOMEM; + card->dai_link = dai_links; + card->num_links = num_links; card->codec_conf = codec_conf; card->num_configs = codec_conf_num; @@ -1762,12 +1757,13 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) for (j = 0; j < codec_info->dai_num ; j++) { int current_be_id; - ret = create_sdw_dailink(card, &link_index, dai_links, - sdw_be_num, adr_link, + ret = create_sdw_dailink(card, &dai_links, adr_link, &codec_conf, ¤t_be_id, i, j); if (ret < 0) { - dev_err(dev, "failed to create dai link %d\n", link_index); + dev_err(dev, + "failed to create dai link %d on 0x%x\n", + j, codec_info->part_id); return ret; } @@ -1800,19 +1796,18 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; - ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, name, + ret = init_simple_dai_link(dev, dai_links, &be_id, name, playback, capture, cpu_dai_name, codec_name, ssp_info->dais[0].dai_name, NULL, ssp_info->ops); if (ret) return ret; - ret = ssp_info->dais[0].init(card, NULL, dai_links + link_index, - ssp_info, 0); + ret = ssp_info->dais[0].init(card, NULL, dai_links, ssp_info, 0); if (ret < 0) return ret; - link_index++; + dai_links++; } DMIC: @@ -1823,16 +1818,16 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto HDMI; } - ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, "dmic01", + ret = init_simple_dai_link(dev, dai_links, &be_id, "dmic01", 0, 1, // DMIC only supports capture "DMIC01 Pin", "dmic-codec", "dmic-hifi", sof_sdw_dmic_init, NULL); if (ret) return ret; - link_index++; + dai_links++; - ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, "dmic16k", + ret = init_simple_dai_link(dev, dai_links, &be_id, "dmic16k", 0, 1, // DMIC only supports capture "DMIC16k Pin", "dmic-codec", "dmic-hifi", /* don't call sof_sdw_dmic_init() twice */ @@ -1840,7 +1835,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (ret) return ret; - link_index++; + dai_links++; } HDMI: @@ -1858,14 +1853,14 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) codec_dai_name = "snd-soc-dummy-dai"; } - ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, name, + ret = init_simple_dai_link(dev, dai_links, &be_id, name, 1, 0, // HDMI only supports playback cpu_dai_name, codec_name, codec_dai_name, i == 0 ? sof_sdw_hdmi_init : NULL, NULL); if (ret) return ret; - link_index++; + dai_links++; } if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { @@ -1875,15 +1870,16 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); - ret = init_simple_dai_link(dev, dai_links + link_index, &be_id, name, + ret = init_simple_dai_link(dev, dai_links, &be_id, name, 1, 1, cpu_dai_name, snd_soc_dummy_dlc.name, snd_soc_dummy_dlc.dai_name, NULL, NULL); if (ret) return ret; + + dai_links++; } - card->dai_link = dai_links; - card->num_links = num_links; + WARN_ON(dai_links != card->dai_link + card->num_links); return 0; } From 2132dbc1a99480bddb995170abaa3c3e1cf8681d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:04 -0500 Subject: [PATCH 065/603] ASoC: Intel: sof_sdw: Use for_each_set_bit Rather than open coding a loop to process each bit use for_each_set_bit. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 0ffa6e36dbed3..a12283d3e1cca 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -5,6 +5,7 @@ * sof_sdw - ASOC Machine driver for Intel SoundWire platforms */ +#include #include #include #include @@ -1644,7 +1645,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) char *codec_name, *codec_dai_name; int i, j, be_id = 0; int hdmi_num; - int ssp_mask; + unsigned long ssp_mask; int ret; ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); @@ -1782,12 +1783,10 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (!ssp_num) goto DMIC; - for (i = 0, j = 0; ssp_mask; i++, ssp_mask >>= 1) { + j = 0; + for_each_set_bit(i, &ssp_mask, BITS_PER_TYPE(ssp_mask)) { int playback, capture; - if (!(ssp_mask & 0x1)) - continue; - name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", i); cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", From 4d96a7f000f04e8041606f074dec5cb21bb4824d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:05 -0500 Subject: [PATCH 066/603] ASoC: Intel: sof_sdw: Factor out SSP DAI creation Factor out the creation of the SSP DAI links into a helper function. No functional change. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 63 +++++++++++++++++++------------- 1 file changed, 37 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a12283d3e1cca..9980652c901b2 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1625,6 +1625,40 @@ static int create_sdw_dailink(struct snd_soc_card *card, return 0; } +static int create_ssp_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id, + struct sof_sdw_codec_info *ssp_info, + unsigned long ssp_mask) +{ + struct device *dev = card->dev; + int i, j = 0; + int ret; + + for_each_set_bit(i, &ssp_mask, BITS_PER_TYPE(ssp_mask)) { + char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", i); + char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); + char *codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", + ssp_info->acpi_id, j++); + int playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; + int capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; + + ret = init_simple_dai_link(dev, *dai_links, be_id, name, + playback, capture, cpu_dai_name, + codec_name, ssp_info->dais[0].dai_name, + NULL, ssp_info->ops); + if (ret) + return ret; + + ret = ssp_info->dais[0].init(card, NULL, *dai_links, ssp_info, 0); + if (ret < 0) + return ret; + + (*dai_links)++; + } + + return 0; +} + static int sof_card_dai_links_create(struct snd_soc_card *card) { struct device *dev = card->dev; @@ -1780,36 +1814,13 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) SSP: /* SSP */ - if (!ssp_num) - goto DMIC; - - j = 0; - for_each_set_bit(i, &ssp_mask, BITS_PER_TYPE(ssp_mask)) { - int playback, capture; - - name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", i); - cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); - codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", - ssp_info->acpi_id, j++); - - playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; - capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; - - ret = init_simple_dai_link(dev, dai_links, &be_id, name, - playback, capture, cpu_dai_name, - codec_name, ssp_info->dais[0].dai_name, - NULL, ssp_info->ops); + if (ssp_num) { + ret = create_ssp_dailinks(card, &dai_links, &be_id, + ssp_info, ssp_mask); if (ret) return ret; - - ret = ssp_info->dais[0].init(card, NULL, dai_links, ssp_info, 0); - if (ret < 0) - return ret; - - dai_links++; } -DMIC: /* dmic */ if (dmic_num > 0) { if (ctx->ignore_pch_dmic) { From c2473a0e50f74b1ea9cc0070048d932d9b57c3ac Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:06 -0500 Subject: [PATCH 067/603] ASoC: Intel: sof_sdw: Factor out DMIC DAI creation. Factor out the creation of the DMIC DAI links into a helper function. No functional change. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 53 +++++++++++++++++++------------- 1 file changed, 32 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 9980652c901b2..5571c9e85dd7e 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1659,6 +1659,34 @@ static int create_ssp_dailinks(struct snd_soc_card *card, return 0; } +static int create_dmic_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id) +{ + struct device *dev = card->dev; + int ret; + + ret = init_simple_dai_link(dev, *dai_links, be_id, "dmic01", + 0, 1, // DMIC only supports capture + "DMIC01 Pin", "dmic-codec", "dmic-hifi", + sof_sdw_dmic_init, NULL); + if (ret) + return ret; + + (*dai_links)++; + + ret = init_simple_dai_link(dev, *dai_links, be_id, "dmic16k", + 0, 1, // DMIC only supports capture + "DMIC16k Pin", "dmic-codec", "dmic-hifi", + /* don't call sof_sdw_dmic_init() twice */ + NULL, NULL); + if (ret) + return ret; + + (*dai_links)++; + + return 0; +} + static int sof_card_dai_links_create(struct snd_soc_card *card) { struct device *dev = card->dev; @@ -1825,30 +1853,13 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (dmic_num > 0) { if (ctx->ignore_pch_dmic) { dev_warn(dev, "Ignoring PCH DMIC\n"); - goto HDMI; + } else { + ret = create_dmic_dailinks(card, &dai_links, &be_id); + if (ret) + return ret; } - - ret = init_simple_dai_link(dev, dai_links, &be_id, "dmic01", - 0, 1, // DMIC only supports capture - "DMIC01 Pin", "dmic-codec", "dmic-hifi", - sof_sdw_dmic_init, NULL); - if (ret) - return ret; - - dai_links++; - - ret = init_simple_dai_link(dev, dai_links, &be_id, "dmic16k", - 0, 1, // DMIC only supports capture - "DMIC16k Pin", "dmic-codec", "dmic-hifi", - /* don't call sof_sdw_dmic_init() twice */ - NULL, NULL); - if (ret) - return ret; - - dai_links++; } -HDMI: /* HDMI */ for (i = 0; i < hdmi_num; i++) { name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1); From 914c43ab50f49656f378e748f894f9532ed19a26 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:07 -0500 Subject: [PATCH 068/603] ASoC: Intel: sof_sdw: Factor out HDMI DAI creation Factor out the creation of the HDMI DAI links into a helper function. No functional change. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 61 ++++++++++++++++++++------------ 1 file changed, 38 insertions(+), 23 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5571c9e85dd7e..7dc3559bf9881 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1687,6 +1687,41 @@ static int create_dmic_dailinks(struct snd_soc_card *card, return 0; } +static int create_hdmi_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id, + int hdmi_num) +{ + struct device *dev = card->dev; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + int i, ret; + + for (i = 0; i < hdmi_num; i++) { + char *name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1); + char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1); + char *codec_name, *codec_dai_name; + + if (ctx->hdmi.idisp_codec) { + codec_name = "ehdaudio0D2"; + codec_dai_name = devm_kasprintf(dev, GFP_KERNEL, + "intel-hdmi-hifi%d", i + 1); + } else { + codec_name = "snd-soc-dummy"; + codec_dai_name = "snd-soc-dummy-dai"; + } + + ret = init_simple_dai_link(dev, *dai_links, be_id, name, + 1, 0, // HDMI only supports playback + cpu_dai_name, codec_name, codec_dai_name, + i == 0 ? sof_sdw_hdmi_init : NULL, NULL); + if (ret) + return ret; + + (*dai_links)++; + } + + return 0; +} + static int sof_card_dai_links_create(struct snd_soc_card *card) { struct device *dev = card->dev; @@ -1704,7 +1739,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) struct snd_soc_dai_link *dai_links; int num_links; char *name, *cpu_dai_name; - char *codec_name, *codec_dai_name; int i, j, be_id = 0; int hdmi_num; unsigned long ssp_mask; @@ -1861,28 +1895,9 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) } /* HDMI */ - for (i = 0; i < hdmi_num; i++) { - name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1); - cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1); - - if (ctx->hdmi.idisp_codec) { - codec_name = "ehdaudio0D2"; - codec_dai_name = devm_kasprintf(dev, GFP_KERNEL, - "intel-hdmi-hifi%d", i + 1); - } else { - codec_name = "snd-soc-dummy"; - codec_dai_name = "snd-soc-dummy-dai"; - } - - ret = init_simple_dai_link(dev, dai_links, &be_id, name, - 1, 0, // HDMI only supports playback - cpu_dai_name, codec_name, codec_dai_name, - i == 0 ? sof_sdw_hdmi_init : NULL, NULL); - if (ret) - return ret; - - dai_links++; - } + ret = create_hdmi_dailinks(card, &dai_links, &be_id, hdmi_num); + if (ret) + return ret; if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { int port = (sof_sdw_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> From 0e2c1dd08607de04912b963f5df470d6a6969496 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:08 -0500 Subject: [PATCH 069/603] ASoC: Intel: sof_sdw: Factor out BlueTooth DAI creation Factor out the creation of the BlueTooth DAI links into a helper function. No functional change. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 35 +++++++++++++++++++++----------- 1 file changed, 23 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 7dc3559bf9881..6e2e934e1246c 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1722,6 +1722,27 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, return 0; } +static int create_bt_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id) +{ + struct device *dev = card->dev; + int port = (sof_sdw_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); + char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); + int ret; + + ret = init_simple_dai_link(dev, *dai_links, be_id, name, + 1, 1, cpu_dai_name, snd_soc_dummy_dlc.name, + snd_soc_dummy_dlc.dai_name, NULL, NULL); + if (ret) + return ret; + + (*dai_links)++; + + return 0; +} + static int sof_card_dai_links_create(struct snd_soc_card *card) { struct device *dev = card->dev; @@ -1738,7 +1759,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) bool group_generated[SDW_MAX_GROUPS] = { }; struct snd_soc_dai_link *dai_links; int num_links; - char *name, *cpu_dai_name; int i, j, be_id = 0; int hdmi_num; unsigned long ssp_mask; @@ -1899,20 +1919,11 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (ret) return ret; + /* BT */ if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { - int port = (sof_sdw_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); - cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); - - ret = init_simple_dai_link(dev, dai_links, &be_id, name, - 1, 1, cpu_dai_name, snd_soc_dummy_dlc.name, - snd_soc_dummy_dlc.dai_name, NULL, NULL); + ret = create_bt_dailinks(card, &dai_links, &be_id); if (ret) return ret; - - dai_links++; } WARN_ON(dai_links != card->dai_link + card->num_links); From b48f238585a49983ae51f77d6494bcfcaad8f217 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:09 -0500 Subject: [PATCH 070/603] ASoC: Intel: sof_sdw: Factor out codec name generation In preparation for future refactoring pull out a helper specifically for generating the codec name. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-15-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 47 ++++++++++++++++++-------------- 1 file changed, 27 insertions(+), 20 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6e2e934e1246c..1253a2dc33d22 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1283,12 +1283,37 @@ static bool is_unique_device(const struct snd_soc_acpi_link_adr *adr_link, return true; } +static const char *get_codec_name(struct device *dev, + const struct sof_sdw_codec_info *codec_info, + const struct snd_soc_acpi_link_adr *adr_link, + int adr_index) +{ + u64 adr = adr_link->adr_d[adr_index].adr; + unsigned int sdw_version = SDW_VERSION(adr); + unsigned int link_id = SDW_DISCO_LINK_ID(adr); + unsigned int unique_id = SDW_UNIQUE_ID(adr); + unsigned int mfg_id = SDW_MFG_ID(adr); + unsigned int part_id = SDW_PART_ID(adr); + unsigned int class_id = SDW_CLASS_ID(adr); + + if (codec_info->codec_name) + return devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL); + else if (is_unique_device(adr_link, sdw_version, mfg_id, part_id, + class_id, adr_index)) + return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x", + link_id, mfg_id, part_id, class_id); + else + return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x:%01x", + link_id, mfg_id, part_id, class_id, unique_id); + + return NULL; +} + static int fill_sdw_codec_dlc(struct device *dev, const struct snd_soc_acpi_link_adr *adr_link, struct snd_soc_dai_link_component *codec, int adr_index, int dai_index) { - unsigned int sdw_version, unique_id, mfg_id, link_id, part_id, class_id; u64 adr = adr_link->adr_d[adr_index].adr; struct sof_sdw_codec_info *codec_info; @@ -1296,25 +1321,7 @@ static int fill_sdw_codec_dlc(struct device *dev, if (!codec_info) return -EINVAL; - sdw_version = SDW_VERSION(adr); - link_id = SDW_DISCO_LINK_ID(adr); - unique_id = SDW_UNIQUE_ID(adr); - mfg_id = SDW_MFG_ID(adr); - part_id = SDW_PART_ID(adr); - class_id = SDW_CLASS_ID(adr); - - if (codec_info->codec_name) - codec->name = devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL); - else if (is_unique_device(adr_link, sdw_version, mfg_id, part_id, - class_id, adr_index)) - codec->name = devm_kasprintf(dev, GFP_KERNEL, - "sdw:0:%01x:%04x:%04x:%02x", link_id, - mfg_id, part_id, class_id); - else - codec->name = devm_kasprintf(dev, GFP_KERNEL, - "sdw:0:%01x:%04x:%04x:%02x:%01x", link_id, - mfg_id, part_id, class_id, unique_id); - + codec->name = get_codec_name(dev, codec_info, adr_link, adr_index); if (!codec->name) return -ENOMEM; From aa238217d69b15edc709887248eec5c01370b453 Mon Sep 17 00:00:00 2001 From: Balamurugan C Date: Tue, 26 Mar 2024 11:04:10 -0500 Subject: [PATCH 071/603] ASoC: Intel: soc-acpi-intel-arl-match: Add rt711 sdca codec support Adding rt711 sdca codec support for arl boards. Reviewed-by: Bard Liao Reviewed-by: Chao Song Signed-off-by: Balamurugan C Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-16-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-arl-match.c | 24 +++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c index e52797aae6e65..79d26e0f2c28d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c @@ -24,6 +24,15 @@ static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { + { + .adr = 0x000030025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + static const struct snd_soc_acpi_link_adr arl_rvp[] = { { .mask = BIT(0), @@ -33,6 +42,15 @@ static const struct snd_soc_acpi_link_adr arl_rvp[] = { {} }; +static const struct snd_soc_acpi_link_adr arl_sdca_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + {} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_machines[] = { {}, }; @@ -46,6 +64,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-arl-rt711.tplg", }, + { + .link_mask = 0x1, /* link0 required */ + .links = arl_sdca_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-arl-rt711-l0.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_arl_sdw_machines); From 17750bc6519f7fb4905e63e3855e4e32b01f9419 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:11 -0500 Subject: [PATCH 072/603] ASoC: Intel: sof_sdw: Remove no longer supported quirk The SOF_SDW_NO_AGGREGATION quirk is mostly for debug and no longer works correctly with the current state of the machine driver. Remove it from the code and add an error message if someone uses it. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-17-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 12 ++++-------- sound/soc/intel/boards/sof_sdw_common.h | 1 + 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1253a2dc33d22..fdedc32291f01 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -36,7 +36,7 @@ static void log_quirks(struct device *dev) dev_dbg(dev, "SSP port %ld\n", SOF_SSP_GET_PORT(sof_sdw_quirk)); if (sof_sdw_quirk & SOF_SDW_NO_AGGREGATION) - dev_dbg(dev, "quirk SOF_SDW_NO_AGGREGATION enabled\n"); + dev_err(dev, "quirk SOF_SDW_NO_AGGREGATION enabled but no longer supported\n"); } static int sof_sdw_quirk_cb(const struct dmi_system_id *id) @@ -1136,11 +1136,9 @@ static int get_dailink_info(struct device *dev, int *sdw_be_num, int *codecs_num) { bool group_visited[SDW_MAX_GROUPS]; - bool no_aggregation; int i; int j; - no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION; *sdw_be_num = 0; if (!adr_link) @@ -1187,7 +1185,7 @@ static int get_dailink_info(struct device *dev, continue; /* count BE for each non-aggregated slave or group */ - if (!endpoint->aggregated || no_aggregation || + if (!endpoint->aggregated || !group_visited[endpoint->group_id]) (*sdw_be_num)++; } @@ -1393,10 +1391,9 @@ static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, int *codec_num, unsigned int *group_id, int adr_index) { - bool no_aggregation = sof_sdw_quirk & SOF_SDW_NO_AGGREGATION; int i; - if (!adr_link->adr_d[adr_index].endpoints->aggregated || no_aggregation) { + if (!adr_link->adr_d[adr_index].endpoints->aggregated) { cpu_dai_id[0] = ffs(adr_link->mask) - 1; *cpu_dai_num = 1; *codec_num = 1; @@ -1758,7 +1755,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; const struct snd_soc_acpi_link_adr *adr_link = mach_params->links; - bool aggregation = !(sof_sdw_quirk & SOF_SDW_NO_AGGREGATION); struct snd_soc_codec_conf *codec_conf; struct sof_sdw_codec_info *codec_info; struct sof_sdw_codec_info *ssp_info; @@ -1896,7 +1892,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) be_id = current_be_id; } - if (aggregation && endpoint->aggregated) + if (endpoint->aggregated) group_generated[endpoint->group_id] = true; } } diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 1daf98a5849f7..e21ef79126b29 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -50,6 +50,7 @@ enum { #define SOF_SDW_PCH_DMIC BIT(6) #define SOF_SSP_PORT(x) (((x) & GENMASK(5, 0)) << 7) #define SOF_SSP_GET_PORT(quirk) (((quirk) >> 7) & GENMASK(5, 0)) +/* Deprecated and no longer supported by the code */ #define SOF_SDW_NO_AGGREGATION BIT(14) /* BT audio offload: reserve 3 bits for future */ From 8166bdd2c560e59e9a6ec0c868b996294d8428d1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:12 -0500 Subject: [PATCH 073/603] ASoC: intel: soc-acpi: Add missing cs42l43 endpoints Currently the cs42l43 just specifies a single endpoint, as the current machine driver only looks at the first endpoint specified. Future refactoring will process all endpoints, as such proper specification should be added for all the cs42l43 endpoints. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-18-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 25 +++++++++++++++++-- .../intel/common/soc-acpi-intel-tgl-match.c | 25 +++++++++++++++++-- 2 files changed, 46 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index e9a5da0790890..e06efc5b1457a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -338,11 +338,32 @@ static const struct snd_soc_acpi_link_adr mtl_712_only[] = { {} }; +static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { + { /* Jack Playback Endpoint */ + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* DMIC Capture Endpoint */ + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Jack Capture Endpoint */ + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = { { .adr = 0x00003001FA424301ull, - .num_endpoints = 1, - .endpoints = &single_endpoint, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, .name_prefix = "cs42l43" } }; diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 0fba0a60d9c79..007f1ac2327ae 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -414,11 +414,32 @@ static const struct snd_soc_acpi_link_adr tgl_712_only[] = { {} }; +static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { + { /* Jack Playback Endpoint */ + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* DMIC Capture Endpoint */ + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Jack Capture Endpoint */ + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + static const struct snd_soc_acpi_adr_device cs42l43_3_adr[] = { { .adr = 0x00033001FA424301ull, - .num_endpoints = 1, - .endpoints = &single_endpoint, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, .name_prefix = "cs42l43" } }; From 27fd36aefa0013bea1cf6948e2e825e9b8cff97a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:13 -0500 Subject: [PATCH 074/603] ASoC: Intel: sof-sdw: Add new code for parsing the snd_soc_acpi structs The current machine driver code has a lot of loops parsing through the ACPI structs (snd_soc_acpi_link_adr, snd_soc_acpi_adr_device, snd_soc_acpi_endpoint), this makes it hard to understand exactly what information is being extracted and for what purposes. As well as being slightly inefficient, as the same information is looked up multiple times. There are also some issues with the handling of multiple endpoints on a single device, only the first of the snd_soc_acpi_endpoint structures is currently fully processed by the driver. This means doing things like aggregating the second endpoint on a device with another device are not currently possible. Add new parsing code that will count the devices and endpoints, parse them into an intermediate datastructure, and then use that to create the DAI links. This patch does not actually utilise the results of the parsing, items will be moved across in the following patches. This parsing is based around two new structures which are temporarily allocated whilst parsing. Firstly, sof_sdw_endpoint, which represents a specific endpoint for audio on a device and is more or less directly equivalent to snd_soc_acpi_endpoint. Secondly, sof_sdw_dailink which represents a DAI link and contains a linked list of one or more sof_sdw_endpoints. A single trip through the snd_soc_acpi data structures is used to populate these. One important point to note here is the use of the num field in snd_soc_acpi_endpoint to address sof_sdw_dai_info array in the sof_sdw_codec_info struct. This expects a one to one mapping between endpoints on a device and dai infos. It would be fine for a specific system to not specify an endpoint for all of the dai infos available, but two endpoints mapping to the same dai info would make not sense. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-19-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 220 +++++++++++++++++++++++++++++-- 1 file changed, 206 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index fdedc32291f01..15bf8fbb3be93 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1462,8 +1462,161 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) return 0; } +struct sof_sdw_endpoint { + struct list_head list; + + u32 link_mask; + const char *codec_name; + + const struct snd_soc_acpi_link_adr *adr_link; + struct sof_sdw_codec_info *codec_info; + const struct sof_sdw_dai_info *dai_info; +}; + +struct sof_sdw_dailink { + bool initialised; + + u8 group_id; + u32 link_mask[SNDRV_PCM_STREAM_LAST + 1]; + int num_devs[SNDRV_PCM_STREAM_LAST + 1]; + struct list_head endpoints; +}; + static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; +static int count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *num_ends) +{ + struct device *dev = card->dev; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); + struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; + const struct snd_soc_acpi_link_adr *adr_link; + int i; + + for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) { + *num_devs += adr_link->num_adr; + + for (i = 0; i < adr_link->num_adr; i++) + *num_ends += adr_link->adr_d[i].num_endpoints; + } + + dev_dbg(dev, "Found %d devices with %d endpoints\n", *num_devs, *num_ends); + + return 0; +} + +static struct sof_sdw_dailink *find_dailink(struct sof_sdw_dailink *dailinks, + const struct snd_soc_acpi_endpoint *new) +{ + while (dailinks->initialised) { + if (new->aggregated && dailinks->group_id == new->group_id) + return dailinks; + + dailinks++; + } + + INIT_LIST_HEAD(&dailinks->endpoints); + dailinks->group_id = new->group_id; + dailinks->initialised = true; + + return dailinks; +} + +static int parse_sdw_endpoints(struct snd_soc_card *card, + struct sof_sdw_dailink *sof_dais, + struct sof_sdw_endpoint *sof_ends) +{ + struct device *dev = card->dev; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); + struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; + const struct snd_soc_acpi_link_adr *adr_link; + struct sof_sdw_endpoint *sof_end = sof_ends; + int num_dais = 0; + int i, j; + + for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) { + if (!is_power_of_2(adr_link->mask)) { + dev_err(dev, "link with multiple mask bits: 0x%x\n", + adr_link->mask); + return -EINVAL; + } + + for (i = 0; i < adr_link->num_adr; i++) { + const struct snd_soc_acpi_adr_device *adr_dev = &adr_link->adr_d[i]; + struct sof_sdw_codec_info *codec_info; + const char *codec_name; + + if (!adr_dev->name_prefix) { + dev_err(dev, "codec 0x%llx does not have a name prefix\n", + adr_dev->adr); + return -EINVAL; + } + + codec_info = find_codec_info_part(adr_dev->adr); + if (!codec_info) + return -EINVAL; + + codec_name = get_codec_name(dev, codec_info, adr_link, i); + if (!codec_name) + return -ENOMEM; + + for (j = 0; j < adr_dev->num_endpoints; j++) { + const struct snd_soc_acpi_endpoint *adr_end; + const struct sof_sdw_dai_info *dai_info; + struct sof_sdw_dailink *sof_dai; + int stream; + + adr_end = &adr_dev->endpoints[j]; + dai_info = &codec_info->dais[adr_end->num]; + sof_dai = find_dailink(sof_dais, adr_end); + + dev_dbg(dev, + "Add dev: %d, 0x%llx end: %d, %s, %c/%c to %s: %d\n", + ffs(adr_link->mask) - 1, adr_dev->adr, + adr_end->num, type_strings[dai_info->dai_type], + dai_info->direction[SNDRV_PCM_STREAM_PLAYBACK] ? 'P' : '-', + dai_info->direction[SNDRV_PCM_STREAM_CAPTURE] ? 'C' : '-', + adr_end->aggregated ? "group" : "solo", + adr_end->group_id); + + if (adr_end->num >= codec_info->dai_num) { + dev_err(dev, + "%d is too many endpoints for codec: 0x%x\n", + adr_end->num, codec_info->part_id); + return -EINVAL; + } + + for_each_pcm_streams(stream) { + if (dai_info->direction[stream] && + dai_info->dailink[stream] < 0) { + dev_err(dev, + "Invalid dailink id %d for codec: 0x%x\n", + dai_info->dailink[stream], + codec_info->part_id); + return -EINVAL; + } + + if (dai_info->direction[stream]) { + num_dais += !sof_dai->num_devs[stream]; + sof_dai->num_devs[stream]++; + sof_dai->link_mask[stream] |= adr_link->mask; + } + } + + list_add_tail(&sof_end->list, &sof_dai->endpoints); + + sof_end->link_mask = adr_link->mask; + sof_end->codec_name = codec_name; + sof_end->adr_link = adr_link; + sof_end->codec_info = codec_info; + sof_end->dai_info = dai_info; + sof_end++; + } + } + } + + return num_dais; +} + static int create_sdw_dailink(struct snd_soc_card *card, struct snd_soc_dai_link **dai_links, const struct snd_soc_acpi_link_adr *adr_link, @@ -1758,7 +1911,11 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) struct snd_soc_codec_conf *codec_conf; struct sof_sdw_codec_info *codec_info; struct sof_sdw_codec_info *ssp_info; + struct sof_sdw_endpoint *sof_ends; + struct sof_sdw_dailink *sof_dais; int codec_conf_num = 0; + int num_devs = 0; + int num_ends = 0; bool group_generated[SDW_MAX_GROUPS] = { }; struct snd_soc_dai_link *dai_links; int num_links; @@ -1767,10 +1924,32 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) unsigned long ssp_mask; int ret; + ret = count_sdw_endpoints(card, &num_devs, &num_ends); + if (ret < 0) { + dev_err(dev, "failed to count devices/endpoints: %d\n", ret); + return ret; + } + + /* One per DAI link, worst case is a DAI link for every endpoint */ + sof_dais = kcalloc(num_ends, sizeof(*sof_dais), GFP_KERNEL); + if (!sof_dais) + return -ENOMEM; + + /* One per endpoint, ie. each DAI on each codec/amp */ + sof_ends = kcalloc(num_ends, sizeof(*sof_ends), GFP_KERNEL); + if (!sof_ends) { + ret = -ENOMEM; + goto err_dai; + } + + ret = parse_sdw_endpoints(card, sof_dais, sof_ends); + if (ret < 0) + goto err_end; + ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); if (ret < 0) { dev_err(dev, "failed to get sdw link info %d\n", ret); - return ret; + goto err_end; } /* @@ -1807,14 +1986,18 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) /* allocate BE dailinks */ num_links = sdw_be_num + ssp_num + dmic_num + hdmi_num + bt_num; dai_links = devm_kcalloc(dev, num_links, sizeof(*dai_links), GFP_KERNEL); - if (!dai_links) - return -ENOMEM; + if (!dai_links) { + ret = -ENOMEM; + goto err_end; + } /* allocate codec conf, will be populated when dailinks are created */ codec_conf = devm_kcalloc(dev, codec_conf_num, sizeof(*codec_conf), GFP_KERNEL); - if (!codec_conf) - return -ENOMEM; + if (!codec_conf) { + ret = -ENOMEM; + goto err_end; + } card->dai_link = dai_links; card->num_links = num_links; @@ -1838,8 +2021,10 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) for (i = 0; i < adr_link->num_adr; i++) { /* find codec info to get dai_num */ codec_info = find_codec_info_part(adr_link->adr_d[i].adr); - if (!codec_info) - return -EINVAL; + if (!codec_info) { + ret = -EINVAL; + goto err_end; + } if (codec_info->dai_num > 1) { ctx->append_dai_type = true; goto out; @@ -1871,8 +2056,10 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) /* find codec info to get dai_num */ codec_info = find_codec_info_part(adr_link->adr_d[i].adr); - if (!codec_info) - return -EINVAL; + if (!codec_info) { + ret = -EINVAL; + goto err_end; + } for (j = 0; j < codec_info->dai_num ; j++) { int current_be_id; @@ -1903,7 +2090,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) ret = create_ssp_dailinks(card, &dai_links, &be_id, ssp_info, ssp_mask); if (ret) - return ret; + goto err_end; } /* dmic */ @@ -1913,25 +2100,30 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) } else { ret = create_dmic_dailinks(card, &dai_links, &be_id); if (ret) - return ret; + goto err_end; } } /* HDMI */ ret = create_hdmi_dailinks(card, &dai_links, &be_id, hdmi_num); if (ret) - return ret; + goto err_end; /* BT */ if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { ret = create_bt_dailinks(card, &dai_links, &be_id); if (ret) - return ret; + goto err_end; } WARN_ON(dai_links != card->dai_link + card->num_links); - return 0; +err_end: + kfree(sof_ends); +err_dai: + kfree(sof_dais); + + return ret; } static int sof_sdw_card_late_probe(struct snd_soc_card *card) From 0d7b9880db92e1eb07bdd4dc097e574512b894a9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:14 -0500 Subject: [PATCH 075/603] ASoC: Intel: sof_sdw: Move counting and codec_conf to new parsing Use the output of the new snd_soc_acpi struct parsing for the purposes of counting the number of SoundWire DAI links and physical devices attached to the SoundWire. These counts are already returned by the parser so those can just be used directly. But the population of the codec_conf structures, is moved from the old parsing code over to the new. As the two parsers currently co-exist it is better to not have them both attempt to modify the same structures. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-20-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 124 ++++++------------------------- 1 file changed, 22 insertions(+), 102 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 15bf8fbb3be93..0566ab843c2ed 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1126,79 +1126,6 @@ static struct sof_sdw_codec_info *find_codec_info_dai(const char *dai_name, return NULL; } -/* - * get BE dailink number and CPU DAI number based on sdw link adr. - * Since some sdw slaves may be aggregated, the CPU DAI number - * may be larger than the number of BE dailinks. - */ -static int get_dailink_info(struct device *dev, - const struct snd_soc_acpi_link_adr *adr_link, - int *sdw_be_num, int *codecs_num) -{ - bool group_visited[SDW_MAX_GROUPS]; - int i; - int j; - - *sdw_be_num = 0; - - if (!adr_link) - return -EINVAL; - - for (i = 0; i < SDW_MAX_GROUPS; i++) - group_visited[i] = false; - - for (; adr_link->num_adr; adr_link++) { - const struct snd_soc_acpi_endpoint *endpoint; - struct sof_sdw_codec_info *codec_info; - int stream; - u64 adr; - - /* make sure the link mask has a single bit set */ - if (!is_power_of_2(adr_link->mask)) - return -EINVAL; - - for (i = 0; i < adr_link->num_adr; i++) { - adr = adr_link->adr_d[i].adr; - codec_info = find_codec_info_part(adr); - if (!codec_info) - return -EINVAL; - - *codecs_num += codec_info->dai_num; - - if (!adr_link->adr_d[i].name_prefix) { - dev_err(dev, "codec 0x%llx does not have a name prefix\n", - adr_link->adr_d[i].adr); - return -EINVAL; - } - - endpoint = adr_link->adr_d[i].endpoints; - if (endpoint->aggregated && !endpoint->group_id) { - dev_err(dev, "invalid group id on link %x\n", - adr_link->mask); - return -EINVAL; - } - - for (j = 0; j < codec_info->dai_num; j++) { - /* count DAI number for playback and capture */ - for_each_pcm_streams(stream) { - if (!codec_info->dais[j].direction[stream]) - continue; - - /* count BE for each non-aggregated slave or group */ - if (!endpoint->aggregated || - !group_visited[endpoint->group_id]) - (*sdw_be_num)++; - } - } - - if (endpoint->aggregated) - group_visited[endpoint->group_id] = true; - } - } - - return 0; -} - static void init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, int *be_id, char *name, int playback, int capture, struct snd_soc_dai_link_component *cpus, int cpus_num, @@ -1528,6 +1455,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, struct device *dev = card->dev; struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; + struct snd_soc_codec_conf *codec_conf = card->codec_conf; const struct snd_soc_acpi_link_adr *adr_link; struct sof_sdw_endpoint *sof_end = sof_ends; int num_dais = 0; @@ -1559,6 +1487,13 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, if (!codec_name) return -ENOMEM; + codec_conf->dlc.name = codec_name; + codec_conf->name_prefix = adr_dev->name_prefix; + codec_conf++; + + dev_dbg(dev, "Adding prefix %s for %s\n", + adr_dev->name_prefix, codec_name); + for (j = 0; j < adr_dev->num_endpoints; j++) { const struct snd_soc_acpi_endpoint *adr_end; const struct sof_sdw_dai_info *dai_info; @@ -1614,13 +1549,14 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, } } + WARN_ON(codec_conf != card->codec_conf + card->num_configs); + return num_dais; } static int create_sdw_dailink(struct snd_soc_card *card, struct snd_soc_dai_link **dai_links, const struct snd_soc_acpi_link_adr *adr_link, - struct snd_soc_codec_conf **codec_conf, int *be_id, int adr_index, int dai_index) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -1672,26 +1608,16 @@ static int create_sdw_dailink(struct snd_soc_card *card, endpoints->group_id != group_id)) continue; - /* sanity check */ - if (*codec_conf >= card->codec_conf + card->num_configs) { - dev_err(dev, "codec_conf array overflowed\n"); - return -EINVAL; - } - ret = fill_sdw_codec_dlc(dev, adr_link_next, &codecs[codec_dlc_index], j, dai_index); if (ret) return ret; - (*codec_conf)->dlc = codecs[codec_dlc_index]; - (*codec_conf)->name_prefix = adr_link_next->adr_d[j].name_prefix; - sdw_codec_ch_maps[codec_dlc_index].cpu = i; sdw_codec_ch_maps[codec_dlc_index].codec = codec_dlc_index; codec_dlc_index++; - (*codec_conf)++; } j = 0; @@ -1913,7 +1839,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) struct sof_sdw_codec_info *ssp_info; struct sof_sdw_endpoint *sof_ends; struct sof_sdw_dailink *sof_dais; - int codec_conf_num = 0; int num_devs = 0; int num_ends = 0; bool group_generated[SDW_MAX_GROUPS] = { }; @@ -1942,15 +1867,21 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto err_dai; } + /* will be populated when acpi endpoints are parsed */ + codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL); + if (!codec_conf) { + ret = -ENOMEM; + goto err_end; + } + + card->codec_conf = codec_conf; + card->num_configs = num_devs; + ret = parse_sdw_endpoints(card, sof_dais, sof_ends); if (ret < 0) goto err_end; - ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); - if (ret < 0) { - dev_err(dev, "failed to get sdw link info %d\n", ret); - goto err_end; - } + sdw_be_num = ret; /* * on generic tgl platform, I2S or sdw mode is supported @@ -1991,18 +1922,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto err_end; } - /* allocate codec conf, will be populated when dailinks are created */ - codec_conf = devm_kcalloc(dev, codec_conf_num, sizeof(*codec_conf), - GFP_KERNEL); - if (!codec_conf) { - ret = -ENOMEM; - goto err_end; - } - card->dai_link = dai_links; card->num_links = num_links; - card->codec_conf = codec_conf; - card->num_configs = codec_conf_num; /* SDW */ if (!sdw_be_num) @@ -2065,8 +1986,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) int current_be_id; ret = create_sdw_dailink(card, &dai_links, adr_link, - &codec_conf, ¤t_be_id, - i, j); + ¤t_be_id, i, j); if (ret < 0) { dev_err(dev, "failed to create dai link %d on 0x%x\n", From 22f2a5e71030c5da938c4d3c50f2159582ee2362 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:15 -0500 Subject: [PATCH 076/603] ASoC: Intel: sof_sdw: Move ignore_pch_dmic to new parsing Trivial move of the handling of ignore_pch_dmic over to the new parsing code. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-21-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 0566ab843c2ed..cda759ee63458 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1453,6 +1453,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, struct sof_sdw_endpoint *sof_ends) { struct device *dev = card->dev; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; struct snd_soc_codec_conf *codec_conf = card->codec_conf; @@ -1483,6 +1484,8 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, if (!codec_info) return -EINVAL; + ctx->ignore_pch_dmic |= codec_info->ignore_pch_dmic; + codec_name = get_codec_name(dev, codec_info, adr_link, i); if (!codec_name) return -ENOMEM; @@ -1630,8 +1633,6 @@ static int create_sdw_dailink(struct snd_soc_card *card, if (!codec_info) return -EINVAL; - ctx->ignore_pch_dmic |= codec_info->ignore_pch_dmic; - for_each_pcm_streams(stream) { char *name, *cpu_name; int playback, capture; From 13e698e8ee70cebfcaead8188e77d3e90f94498d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:16 -0500 Subject: [PATCH 077/603] ASoC: Intel: sof_sdw: Move append_dai_type to new parsing append_dai_type should be set to true in any situation that would result in multiple DAI links existing on a single SoundWire bus, because the legacy naming used only the bus number to make things unique. The current code handles this by looking for codecs with multiple dai_info structs and looking for buses that include multiple types of device on them. The first of these assumes that all DAIs on a given device would be in use. The second, with dissimilar aggregation now being supported, isn't really an accurate check either since those devices could be aggregated into a single DAI link. Move the handling for this flag over to the new parsing code and simplify things a little by looking directly for SoundWire links that will contain multiple DAI links to set the flag. These changes should not cause any change in behaviour for any currently supported systems. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-22-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 36 +++++--------------------------- 1 file changed, 5 insertions(+), 31 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index cda759ee63458..89ce779616348 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1463,6 +1463,8 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, int i, j; for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) { + int num_link_dailinks = 0; + if (!is_power_of_2(adr_link->mask)) { dev_err(dev, "link with multiple mask bits: 0x%x\n", adr_link->mask); @@ -1540,6 +1542,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, } } + num_link_dailinks += !!list_empty(&sof_dai->endpoints); list_add_tail(&sof_end->list, &sof_dai->endpoints); sof_end->link_mask = adr_link->mask; @@ -1550,6 +1553,8 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, sof_end++; } } + + ctx->append_dai_type |= (num_link_dailinks > 1); } WARN_ON(codec_conf != card->codec_conf + card->num_configs); @@ -1933,37 +1938,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) for (i = 0; i < SDW_MAX_LINKS; i++) ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE; - for (; adr_link->num_adr; adr_link++) { - /* - * If there are two or more different devices on the same sdw link, we have to - * append the codec type to the dai link name to prevent duplicated dai link name. - * The same type devices on the same sdw link will be in the same - * snd_soc_acpi_adr_device array. They won't be described in different adr_links. - */ - for (i = 0; i < adr_link->num_adr; i++) { - /* find codec info to get dai_num */ - codec_info = find_codec_info_part(adr_link->adr_d[i].adr); - if (!codec_info) { - ret = -EINVAL; - goto err_end; - } - if (codec_info->dai_num > 1) { - ctx->append_dai_type = true; - goto out; - } - for (j = 0; j < i; j++) { - if ((SDW_PART_ID(adr_link->adr_d[i].adr) != - SDW_PART_ID(adr_link->adr_d[j].adr)) || - (SDW_MFG_ID(adr_link->adr_d[i].adr) != - SDW_MFG_ID(adr_link->adr_d[j].adr))) { - ctx->append_dai_type = true; - goto out; - } - } - } - } -out: - /* generate DAI links by each sdw link */ for (adr_link = mach_params->links ; adr_link->num_adr; adr_link++) { for (i = 0; i < adr_link->num_adr; i++) { From 5f14d70b7f6e9deb97893d5c09dd0986e92b7021 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:17 -0500 Subject: [PATCH 078/603] ASoC: Intel: sof_sdw: Move generation of DAI links to new parsing The only part left using the old parsing code is now the generation of the actual DAI links. Move this generation over to being based on the new parsing, which allows the removal of the last of the old parsing code. The new DAI link generation is a simple matter of creating a new DAI link for each sof_sdw_dailink struct, and adding a cpu, a codec, and a mapping for each sof_sdw_endpoint contained in that sof_sdw_dailink. Note that the CPUs can be inferred as the endpoint list is iterated, because the endpoints were added into the list sorted by link. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-23-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 340 +++++++------------------------ 1 file changed, 78 insertions(+), 262 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 89ce779616348..9f1c49df9cd2b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1234,129 +1234,6 @@ static const char *get_codec_name(struct device *dev, return NULL; } -static int fill_sdw_codec_dlc(struct device *dev, - const struct snd_soc_acpi_link_adr *adr_link, - struct snd_soc_dai_link_component *codec, - int adr_index, int dai_index) -{ - u64 adr = adr_link->adr_d[adr_index].adr; - struct sof_sdw_codec_info *codec_info; - - codec_info = find_codec_info_part(adr); - if (!codec_info) - return -EINVAL; - - codec->name = get_codec_name(dev, codec_info, adr_link, adr_index); - if (!codec->name) - return -ENOMEM; - - codec->dai_name = codec_info->dais[dai_index].dai_name; - - return 0; -} - -static int set_codec_init_func(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *adr_link, - struct snd_soc_dai_link *dai_links, - bool playback, int group_id, int adr_index, int dai_index) -{ - int i = adr_index; - - do { - /* - * Initialize the codec. If codec is part of an aggregated - * group (group_id>0), initialize all codecs belonging to - * same group. - * The first link should start with adr_link->adr_d[adr_index] - * because that is the device that we want to initialize and - * we should end immediately if it is not aggregated (group_id=0) - */ - for ( ; i < adr_link->num_adr; i++) { - struct sof_sdw_codec_info *codec_info; - - codec_info = find_codec_info_part(adr_link->adr_d[i].adr); - if (!codec_info) - return -EINVAL; - - /* The group_id is > 0 iff the codec is aggregated */ - if (adr_link->adr_d[i].endpoints->group_id != group_id) - continue; - - if (codec_info->dais[dai_index].init) - codec_info->dais[dai_index].init(card, - adr_link, - dai_links, - codec_info, - playback); - if (!group_id) - return 0; - } - - i = 0; - adr_link++; - } while (adr_link->mask); - - return 0; -} - -/* - * check endpoint status in slaves and gather link ID for all slaves in - * the same group to generate different CPU DAI. Now only support - * one sdw link with all slaves set with only single group id. - * - * one slave on one sdw link with aggregated = 0 - * one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI - * - * two or more slaves on one sdw link with aggregated = 0 - * one sdw BE DAI <---> one-cpu DAI <---> multi-codec DAIs - * - * multiple links with multiple slaves with aggregated = 1 - * one sdw BE DAI <---> 1 .. N CPU DAIs <----> 1 .. N codec DAIs - */ -static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, - struct device *dev, int *cpu_dai_id, int *cpu_dai_num, - int *codec_num, unsigned int *group_id, - int adr_index) -{ - int i; - - if (!adr_link->adr_d[adr_index].endpoints->aggregated) { - cpu_dai_id[0] = ffs(adr_link->mask) - 1; - *cpu_dai_num = 1; - *codec_num = 1; - *group_id = 0; - return 0; - } - - *codec_num = 0; - *cpu_dai_num = 0; - *group_id = adr_link->adr_d[adr_index].endpoints->group_id; - - /* Count endpoints with the same group_id in the adr_link */ - for (; adr_link && adr_link->num_adr; adr_link++) { - unsigned int link_codecs = 0; - - for (i = 0; i < adr_link->num_adr; i++) { - if (adr_link->adr_d[i].endpoints->aggregated && - adr_link->adr_d[i].endpoints->group_id == *group_id) - link_codecs++; - } - - if (link_codecs) { - *codec_num += link_codecs; - - if (*cpu_dai_num >= SDW_MAX_CPU_DAIS) { - dev_err(dev, "cpu_dai_id array overflowed\n"); - return -EINVAL; - } - - cpu_dai_id[(*cpu_dai_num)++] = ffs(adr_link->mask) - 1; - } - } - - return 0; -} - static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct sof_sdw_codec_info *codec_info; @@ -1563,135 +1440,102 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, } static int create_sdw_dailink(struct snd_soc_card *card, + struct sof_sdw_dailink *sof_dai, struct snd_soc_dai_link **dai_links, - const struct snd_soc_acpi_link_adr *adr_link, - int *be_id, int adr_index, int dai_index) + int *be_id) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *dev = card->dev; - const struct snd_soc_acpi_link_adr *adr_link_next; - struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps; - struct snd_soc_dai_link_component *codecs; - struct snd_soc_dai_link_component *cpus; - struct sof_sdw_codec_info *codec_info; - int cpu_dai_id[SDW_MAX_CPU_DAIS]; - int cpu_dai_num; - unsigned int group_id; - int codec_dlc_index = 0; - int codec_num; + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct sof_sdw_endpoint *sof_end; int stream; - int i = 0; - int j, k; - int ret; - - ret = get_slave_info(adr_link, dev, cpu_dai_id, &cpu_dai_num, &codec_num, - &group_id, adr_index); - if (ret) - return ret; - - codecs = devm_kcalloc(dev, codec_num, sizeof(*codecs), GFP_KERNEL); - if (!codecs) - return -ENOMEM; - - sdw_codec_ch_maps = devm_kcalloc(dev, codec_num, - sizeof(*sdw_codec_ch_maps), GFP_KERNEL); - if (!sdw_codec_ch_maps) - return -ENOMEM; - - /* generate codec name on different links in the same group */ - j = adr_index; - for (adr_link_next = adr_link; adr_link_next && adr_link_next->num_adr && - i < cpu_dai_num; adr_link_next++) { - /* skip the link excluded by this processed group */ - if (cpu_dai_id[i] != ffs(adr_link_next->mask) - 1) - continue; - - /* j reset after loop, adr_index only applies to first link */ - for (; j < adr_link_next->num_adr && codec_dlc_index < codec_num; j++) { - const struct snd_soc_acpi_endpoint *endpoints; - - endpoints = adr_link_next->adr_d[j].endpoints; - - if (group_id && (!endpoints->aggregated || - endpoints->group_id != group_id)) - continue; - - ret = fill_sdw_codec_dlc(dev, adr_link_next, - &codecs[codec_dlc_index], - j, dai_index); - if (ret) - return ret; - - sdw_codec_ch_maps[codec_dlc_index].cpu = i; - sdw_codec_ch_maps[codec_dlc_index].codec = codec_dlc_index; - - codec_dlc_index++; - } - j = 0; - - /* check next link to create codec dai in the processed group */ - i++; - } - - /* find codec info to create BE DAI */ - codec_info = find_codec_info_part(adr_link->adr_d[adr_index].adr); - if (!codec_info) - return -EINVAL; for_each_pcm_streams(stream) { - char *name, *cpu_name; - int playback, capture; static const char * const sdw_stream_name[] = { "SDW%d-Playback", "SDW%d-Capture", "SDW%d-Playback-%s", "SDW%d-Capture-%s", }; + struct snd_soc_dai_link_ch_map *codec_maps; + struct snd_soc_dai_link_component *codecs; + struct snd_soc_dai_link_component *cpus; + int num_cpus = hweight32(sof_dai->link_mask[stream]); + int num_codecs = sof_dai->num_devs[stream]; + int playback, capture; + int cur_link = 0; + int i = 0, j = 0; + char *name; - if (!codec_info->dais[dai_index].direction[stream]) + if (!sof_dai->num_devs[stream]) continue; - *be_id = codec_info->dais[dai_index].dailink[stream]; + sof_end = list_first_entry(&sof_dai->endpoints, + struct sof_sdw_endpoint, list); + + *be_id = sof_end->dai_info->dailink[stream]; if (*be_id < 0) { dev_err(dev, "Invalid dailink id %d\n", *be_id); return -EINVAL; } /* create stream name according to first link id */ - if (ctx->append_dai_type) { + if (ctx->append_dai_type) name = devm_kasprintf(dev, GFP_KERNEL, - sdw_stream_name[stream + 2], cpu_dai_id[0], - type_strings[codec_info->dais[dai_index].dai_type]); - } else { + sdw_stream_name[stream + 2], + ffs(sof_end->link_mask) - 1, + type_strings[sof_end->dai_info->dai_type]); + else name = devm_kasprintf(dev, GFP_KERNEL, - sdw_stream_name[stream], cpu_dai_id[0]); - } + sdw_stream_name[stream], + ffs(sof_end->link_mask) - 1); if (!name) return -ENOMEM; - cpus = devm_kcalloc(dev, cpu_dai_num, sizeof(*cpus), GFP_KERNEL); + cpus = devm_kcalloc(dev, num_cpus, sizeof(*cpus), GFP_KERNEL); if (!cpus) return -ENOMEM; - /* - * generate CPU DAI name base on the sdw link ID and - * PIN ID with offset of 2 according to sdw dai driver. - */ - for (k = 0; k < cpu_dai_num; k++) { - cpu_name = devm_kasprintf(dev, GFP_KERNEL, - "SDW%d Pin%d", cpu_dai_id[k], - ctx->sdw_pin_index[cpu_dai_id[k]]++); - if (!cpu_name) - return -ENOMEM; + codecs = devm_kcalloc(dev, num_codecs, sizeof(*codecs), GFP_KERNEL); + if (!codecs) + return -ENOMEM; + + codec_maps = devm_kcalloc(dev, num_codecs, sizeof(*codec_maps), GFP_KERNEL); + if (!codec_maps) + return -ENOMEM; + + list_for_each_entry(sof_end, &sof_dai->endpoints, list) { + if (!sof_end->dai_info->direction[stream]) + continue; + + if (cur_link != sof_end->link_mask) { + int link_num = ffs(sof_end->link_mask) - 1; + int pin_num = ctx->sdw_pin_index[link_num]++; + + cur_link = sof_end->link_mask; - cpus[k].dai_name = cpu_name; + cpus[i].dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SDW%d Pin%d", + link_num, pin_num); + if (!cpus[i].dai_name) + return -ENOMEM; + i++; + } + + codec_maps[j].cpu = i - 1; + codec_maps[j].codec = j; + + codecs[j].name = sof_end->codec_name; + codecs[j].dai_name = sof_end->dai_info->dai_name; + j++; } + WARN_ON(i != num_cpus || j != num_codecs); + playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); capture = (stream == SNDRV_PCM_STREAM_CAPTURE); init_dai_link(dev, *dai_links, be_id, name, playback, capture, - cpus, cpu_dai_num, codecs, codec_num, + cpus, num_cpus, codecs, num_codecs, sof_sdw_rtd_init, &sdw_ops); /* @@ -1699,13 +1543,14 @@ static int create_sdw_dailink(struct snd_soc_card *card, * based on wait_for_completion(), tag them as 'nonatomic'. */ (*dai_links)->nonatomic = true; - (*dai_links)->ch_maps = sdw_codec_ch_maps; - - ret = set_codec_init_func(card, adr_link, *dai_links, - playback, group_id, adr_index, dai_index); - if (ret < 0) { - dev_err(dev, "failed to init codec 0x%x\n", codec_info->part_id); - return ret; + (*dai_links)->ch_maps = codec_maps; + + list_for_each_entry(sof_end, &sof_dai->endpoints, list) { + if (sof_end->dai_info->init) + sof_end->dai_info->init(card, sof_end->adr_link, + *dai_links, + sof_end->codec_info, + playback); } (*dai_links)++; @@ -1839,18 +1684,15 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, bt_num = 0; struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; - const struct snd_soc_acpi_link_adr *adr_link = mach_params->links; struct snd_soc_codec_conf *codec_conf; - struct sof_sdw_codec_info *codec_info; struct sof_sdw_codec_info *ssp_info; struct sof_sdw_endpoint *sof_ends; struct sof_sdw_dailink *sof_dais; int num_devs = 0; int num_ends = 0; - bool group_generated[SDW_MAX_GROUPS] = { }; struct snd_soc_dai_link *dai_links; int num_links; - int i, j, be_id = 0; + int i, be_id = 0; int hdmi_num; unsigned long ssp_mask; int ret; @@ -1939,44 +1781,18 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE; /* generate DAI links by each sdw link */ - for (adr_link = mach_params->links ; adr_link->num_adr; adr_link++) { - for (i = 0; i < adr_link->num_adr; i++) { - const struct snd_soc_acpi_endpoint *endpoint; - - endpoint = adr_link->adr_d[i].endpoints; - - /* this group has been generated */ - if (endpoint->aggregated && - group_generated[endpoint->group_id]) - continue; - - /* find codec info to get dai_num */ - codec_info = find_codec_info_part(adr_link->adr_d[i].adr); - if (!codec_info) { - ret = -EINVAL; - goto err_end; - } + while (sof_dais->initialised) { + int current_be_id; - for (j = 0; j < codec_info->dai_num ; j++) { - int current_be_id; - - ret = create_sdw_dailink(card, &dai_links, adr_link, - ¤t_be_id, i, j); - if (ret < 0) { - dev_err(dev, - "failed to create dai link %d on 0x%x\n", - j, codec_info->part_id); - return ret; - } + ret = create_sdw_dailink(card, sof_dais, &dai_links, ¤t_be_id); + if (ret) + goto err_end; - /* Update the be_id to match the highest ID used for SDW link */ - if (be_id < current_be_id) - be_id = current_be_id; - } + /* Update the be_id to match the highest ID used for SDW link */ + if (be_id < current_be_id) + be_id = current_be_id; - if (endpoint->aggregated) - group_generated[endpoint->group_id] = true; - } + sof_dais++; } SSP: From 59bf457d80551003a06d32f5c3d1da7f64a3d420 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:18 -0500 Subject: [PATCH 079/603] ASoC: intel: sof_sdw: Factor out SoundWire DAI creation Factor out the creation of the SoundWire DAI links into a helper function. No functional change. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-24-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 50 ++++++++++++++++++++------------ 1 file changed, 31 insertions(+), 19 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 9f1c49df9cd2b..3e6a758c98ab4 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1559,6 +1559,34 @@ static int create_sdw_dailink(struct snd_soc_card *card, return 0; } +static int create_sdw_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id, + struct sof_sdw_dailink *sof_dais) +{ + struct mc_private *ctx = snd_soc_card_get_drvdata(card); + int ret, i; + + for (i = 0; i < SDW_MAX_LINKS; i++) + ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE; + + /* generate DAI links by each sdw link */ + while (sof_dais->initialised) { + int current_be_id; + + ret = create_sdw_dailink(card, sof_dais, dai_links, ¤t_be_id); + if (ret) + return ret; + + /* Update the be_id to match the highest ID used for SDW link */ + if (*be_id < current_be_id) + *be_id = current_be_id; + + sof_dais++; + } + + return 0; +} + static int create_ssp_dailinks(struct snd_soc_card *card, struct snd_soc_dai_link **dai_links, int *be_id, struct sof_sdw_codec_info *ssp_info, @@ -1692,7 +1720,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) int num_ends = 0; struct snd_soc_dai_link *dai_links; int num_links; - int i, be_id = 0; + int be_id = 0; int hdmi_num; unsigned long ssp_mask; int ret; @@ -1774,28 +1802,12 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) card->num_links = num_links; /* SDW */ - if (!sdw_be_num) - goto SSP; - - for (i = 0; i < SDW_MAX_LINKS; i++) - ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE; - - /* generate DAI links by each sdw link */ - while (sof_dais->initialised) { - int current_be_id; - - ret = create_sdw_dailink(card, sof_dais, &dai_links, ¤t_be_id); + if (sdw_be_num) { + ret = create_sdw_dailinks(card, &dai_links, &be_id, sof_dais); if (ret) goto err_end; - - /* Update the be_id to match the highest ID used for SDW link */ - if (be_id < current_be_id) - be_id = current_be_id; - - sof_dais++; } -SSP: /* SSP */ if (ssp_num) { ret = create_ssp_dailinks(card, &dai_links, &be_id, From bee2fe44679f1e6a5332d7f78587ccca4109919f Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Mar 2024 11:04:19 -0500 Subject: [PATCH 080/603] ASoC: Intel: sof_sdw: use generic rtd_init function for Realtek SDW DMICs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The only thing that the rt_xxx_rtd_init() functions do is to set card->components. And we can set card->components with name_prefix as rt712_sdca_dmic_rtd_init() does. And sof_sdw_rtd_init() will always select the first dai with the given dai->name from codec_info_list[]. Unfortunately, we have different codecs with the same dai name. For example, dai name of rt715 and rt715-sdca are both "rt715-aif2". Using a generic rtd_init allow sof_sdw_rtd_init() run the rtd_init() callback from a similar codec dai. Fixes: 8266c73126b7 ("ASoC: Intel: sof_sdw: add common sdw dai link init") Reviewed-by: Chao Song Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-25-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/sof_sdw.c | 12 +++--- sound/soc/intel/boards/sof_sdw_common.h | 1 + sound/soc/intel/boards/sof_sdw_rt_dmic.c | 52 ++++++++++++++++++++++++ 4 files changed, 60 insertions(+), 6 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw_rt_dmic.c diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index bbf796a5f7ba8..08cfd4baecdd7 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -42,6 +42,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ sof_sdw_rt712_sdca.o sof_sdw_rt715.o \ sof_sdw_rt715_sdca.o sof_sdw_rt722_sdca.o \ + sof_sdw_rt_dmic.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ sof_sdw_dmic.o \ diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3e6a758c98ab4..0e376cb0ce0b2 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -749,7 +749,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-dmic-aif1", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt712_sdca_dmic_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -779,7 +779,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-dmic-aif1", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt712_sdca_dmic_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -841,7 +841,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt715_sdca_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -856,7 +856,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt715_sdca_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -871,7 +871,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt715_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -886,7 +886,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt715_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 1, diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index e21ef79126b29..decaed97fa74f 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -197,6 +197,7 @@ int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c new file mode 100644 index 0000000000000..9091f5b5c6484 --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c @@ -0,0 +1,52 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (c) 2024 Intel Corporation + +/* + * sof_sdw_rt_dmic - Helpers to handle Realtek SDW DMIC from generic machine driver + */ + +#include +#include +#include +#include +#include "sof_board_helpers.h" +#include "sof_sdw_common.h" + +static const char * const dmics[] = { + "rt715", + "rt712-sdca-dmic", +}; + +int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct snd_soc_component *component; + struct snd_soc_dai *codec_dai; + char *mic_name; + + codec_dai = get_codec_dai_by_name(rtd, dmics, ARRAY_SIZE(dmics)); + if (!codec_dai) + return -EINVAL; + + component = codec_dai->component; + + /* + * rt715-sdca (aka rt714) is a special case that uses different name in card->components + * and component->name_prefix. + */ + if (!strcmp(component->name_prefix, "rt714")) + mic_name = devm_kasprintf(card->dev, GFP_KERNEL, "rt715-sdca"); + else + mic_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s", component->name_prefix); + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s mic:%s", card->components, + mic_name); + if (!card->components) + return -ENOMEM; + + dev_dbg(card->dev, "card->components: %s\n", card->components); + + return 0; +} +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); From 45bbc14fb94698b43636ec18d0df2440934139e7 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Mar 2024 11:04:20 -0500 Subject: [PATCH 081/603] ASoC: Intel: sof_sdw: remove unused rt dmic rtd_init MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Some Realtek SDW DMIC codecs use the generic rt_dmic_rtd_init callback now. Remove unused rtd_init callbacks. Reviewed-by: Chao Song Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-26-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Makefile | 3 +-- sound/soc/intel/boards/sof_sdw_common.h | 3 --- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 24 ------------------- sound/soc/intel/boards/sof_sdw_rt715.c | 26 --------------------- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 26 --------------------- 5 files changed, 1 insertion(+), 81 deletions(-) delete mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c delete mode 100644 sound/soc/intel/boards/sof_sdw_rt715_sdca.c diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 08cfd4baecdd7..8f711be46c43e 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -40,8 +40,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ sof_sdw_maxim.o sof_sdw_rt_amp.o \ sof_sdw_rt5682.o sof_sdw_rt700.o \ sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ - sof_sdw_rt712_sdca.o sof_sdw_rt715.o \ - sof_sdw_rt715_sdca.o sof_sdw_rt722_sdca.o \ + sof_sdw_rt712_sdca.o sof_sdw_rt722_sdca.o \ sof_sdw_rt_dmic.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index decaed97fa74f..84715d9ca9424 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -193,10 +193,7 @@ int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index 9c898c7286af9..0c4cd4cdbd45b 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -66,27 +66,3 @@ int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static const char * const dmics[] = { - "rt712-sdca-dmic" -}; - -int rt712_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai; - struct snd_soc_component *component; - - codec_dai = get_codec_dai_by_name(rtd, dmics, ARRAY_SIZE(dmics)); - if (!codec_dai) - return -EINVAL; - - component = codec_dai->component; - card->components = devm_kasprintf(card->dev, GFP_KERNEL, - "%s mic:%s", - card->components, component->name_prefix); - if (!card->components) - return -ENOMEM; - - return 0; -} -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); diff --git a/sound/soc/intel/boards/sof_sdw_rt715.c b/sound/soc/intel/boards/sof_sdw_rt715.c deleted file mode 100644 index b5a886cd595dc..0000000000000 --- a/sound/soc/intel/boards/sof_sdw_rt715.c +++ /dev/null @@ -1,26 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright (c) 2020 Intel Corporation - -/* - * sof_sdw_rt715 - Helpers to handle RT715 from generic machine driver - */ - -#include -#include -#include -#include -#include "sof_sdw_common.h" - -int rt715_rtd_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - - card->components = devm_kasprintf(card->dev, GFP_KERNEL, - "%s mic:rt715", - card->components); - if (!card->components) - return -ENOMEM; - - return 0; -} - diff --git a/sound/soc/intel/boards/sof_sdw_rt715_sdca.c b/sound/soc/intel/boards/sof_sdw_rt715_sdca.c deleted file mode 100644 index 4b37a8a6dd2e2..0000000000000 --- a/sound/soc/intel/boards/sof_sdw_rt715_sdca.c +++ /dev/null @@ -1,26 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright (c) 2020 Intel Corporation - -/* - * sof_sdw_rt715_sdca - Helpers to handle RT715-SDCA from generic machine driver - */ - -#include -#include -#include -#include -#include "sof_sdw_common.h" - -int rt715_sdca_rtd_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - - card->components = devm_kasprintf(card->dev, GFP_KERNEL, - "%s mic:rt715-sdca", - card->components); - if (!card->components) - return -ENOMEM; - - return 0; -} - From 266c9b27cb0a2c11de5956ee4bd7e1266d0baa36 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Mar 2024 11:04:21 -0500 Subject: [PATCH 082/603] ASoC: Intel: sof_sdw_rt722_sdca: set rtd_init in codec_info_list[] MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add rt722 into rt_sdca_jack_rtd_init() supported list, and set rtd_init callback directly in codec_info_list[]. No need to use a init callback to set the rtd_init callback. Besides, sof_sdw_rt_sdca_jack_init() doesn't set .rtd_init anymore. Reviewed-by: Chao Song Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-27-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 5 ++-- sound/soc/intel/boards/sof_sdw_common.h | 14 ++-------- sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 26 ++----------------- .../boards/sof_sdw_rt_sdca_jack_common.c | 2 +- 4 files changed, 8 insertions(+), 39 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 0e376cb0ce0b2..bb8429a834670 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -902,6 +902,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, + .rtd_init = rt_sdca_jack_rtd_init, }, { .direction = {true, false}, @@ -909,14 +910,14 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_AMP, /* No feedback capability is provided by rt722-sdca codec driver*/ .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, - .init = sof_sdw_rt722_spk_init, + .rtd_init = rt722_spk_rtd_init, }, { .direction = {false, true}, .dai_name = "rt722-sdca-aif3", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = sof_sdw_rt722_sdca_dmic_init, + .rtd_init = rt722_sdca_dmic_rtd_init, }, }, .dai_num = 3, diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 84715d9ca9424..315b2a298195b 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -157,18 +157,6 @@ int sof_sdw_rt_amp_init(struct snd_soc_card *card, bool playback); int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); -/* RT722-SDCA support */ -int sof_sdw_rt722_spk_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); -int sof_sdw_rt722_sdca_dmic_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); - /* MAXIM codec support */ int sof_sdw_maxim_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, @@ -194,6 +182,8 @@ int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); +int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c index fe3a2bff95bc7..4947d16c70dbe 100644 --- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c @@ -27,7 +27,7 @@ static const struct snd_kcontrol_new rt722_spk_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), }; -static int rt722_spk_init(struct snd_soc_pcm_runtime *rtd) +int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int ret; @@ -59,18 +59,7 @@ static int rt722_spk_init(struct snd_soc_pcm_runtime *rtd) return ret; } -int sof_sdw_rt722_spk_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - dai_links->init = rt722_spk_init; - - return 0; -} - -static int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); @@ -84,14 +73,3 @@ static int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) return 0; } - -int sof_sdw_rt722_sdca_dmic_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback) -{ - dai_links->init = rt722_sdca_dmic_rtd_init; - - return 0; -} diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index fe59a144efeff..0067714854861 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -85,7 +85,7 @@ static struct snd_soc_jack_pin rt_sdca_jack_pins[] = { }; static const char * const jack_codecs[] = { - "rt711", "rt712", "rt713" + "rt711", "rt712", "rt713", "rt722" }; int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) From df19c6cd0fd0418b779f9c627b159d7ab77bff71 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Mar 2024 11:04:22 -0500 Subject: [PATCH 083/603] ASoC: Intel: sof_sdw_rt722_sdca: use rt_dmic_rtd_init MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit rt_dmic_rtd_init() can be used for rt722 dmic, too. Reviewed-by: Chao Song Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-28-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_sdw_common.h | 1 - sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 14 -------------- sound/soc/intel/boards/sof_sdw_rt_dmic.c | 1 + 4 files changed, 2 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index bb8429a834670..3874da8d7fe4a 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -917,7 +917,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt722-sdca-aif3", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .rtd_init = rt722_sdca_dmic_rtd_init, + .rtd_init = rt_dmic_rtd_init, }, }, .dai_num = 3, diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 315b2a298195b..2132d3fc42f81 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -183,7 +183,6 @@ int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c index 4947d16c70dbe..e5c2a36e400b8 100644 --- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c @@ -59,17 +59,3 @@ int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } -int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_component *component = codec_dai->component; - - card->components = devm_kasprintf(card->dev, GFP_KERNEL, - "%s mic:%s", - card->components, component->name_prefix); - if (!card->components) - return -ENOMEM; - - return 0; -} diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c index 9091f5b5c6484..5aec8422656cd 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_dmic.c +++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c @@ -15,6 +15,7 @@ static const char * const dmics[] = { "rt715", "rt712-sdca-dmic", + "rt722-sdca", }; int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) From 13112a34d83e0b3c925ff9818e0819ad2fe97e42 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Mar 2024 11:04:23 -0500 Subject: [PATCH 084/603] ASoC: rt715-sdca: rename dai name with rt715-sdca prefix MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The dai name are the same as rt715 codec. Rename them with rt715-sdca prefix allow machine driver to distinguish rt715 and rt715-sdca from dai name. Reviewed-by: Chao Song Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-29-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt715-sdca.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 4533eedd7e189..cee46db79d733 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -933,7 +933,7 @@ static const struct snd_soc_dai_ops rt715_sdca_ops = { static struct snd_soc_dai_driver rt715_sdca_dai[] = { { - .name = "rt715-aif1", + .name = "rt715-sdca-aif1", .id = RT715_AIF1, .capture = { .stream_name = "DP6 Capture", @@ -945,7 +945,7 @@ static struct snd_soc_dai_driver rt715_sdca_dai[] = { .ops = &rt715_sdca_ops, }, { - .name = "rt715-aif2", + .name = "rt715-sdca-aif2", .id = RT715_AIF2, .capture = { .stream_name = "DP4 Capture", From a2e620e4ac87c80e0987bd74c0c345b0da02b33e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Mar 2024 11:04:24 -0500 Subject: [PATCH 085/603] ASoC: Intel: sof_sdw: change rt715-sdca dai name MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit As rt715-sdca dai name has been changed in codec driver, we should change it in machine driver, too. Changing the dai name in codec_info_list[] also help sof_sdw_rtd_init() to run .rtd_init() from the same codec. Reviewed-by: Chao Song Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-30-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 4 ++-- sound/soc/intel/boards/sof_sdw_rt_dmic.c | 1 + 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3874da8d7fe4a..1816ec5dd5806 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -838,7 +838,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dais = { { .direction = {false, true}, - .dai_name = "rt715-aif2", + .dai_name = "rt715-sdca-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, @@ -853,7 +853,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dais = { { .direction = {false, true}, - .dai_name = "rt715-aif2", + .dai_name = "rt715-sdca-aif2", .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c index 5aec8422656cd..2f7ed9b31e793 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_dmic.c +++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c @@ -14,6 +14,7 @@ static const char * const dmics[] = { "rt715", + "rt715-sdca", "rt712-sdca-dmic", "rt722-sdca", }; From aac976aa3c6a37175eec9d3eb912cd92aa8c3a0b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 26 Mar 2024 11:04:25 -0500 Subject: [PATCH 086/603] ASoC: Intel: change cs35l56 name_prefix To use sequential amp numbers. I,e, AMP1, AMP2, AMP3, AMP4 Reviewed-by: Chao Song Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-31-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 4 ++-- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index e06efc5b1457a..7e53266d831d8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -373,13 +373,13 @@ static const struct snd_soc_acpi_adr_device cs35l56_1_adr[] = { .adr = 0x00013701FA355601ull, .num_endpoints = 1, .endpoints = &spk_r_endpoint, - .name_prefix = "AMP8" + .name_prefix = "AMP3" }, { .adr = 0x00013601FA355601ull, .num_endpoints = 1, .endpoints = &spk_3_endpoint, - .name_prefix = "AMP7" + .name_prefix = "AMP4" } }; diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 007f1ac2327ae..0cef3d788a5b0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -464,13 +464,13 @@ static const struct snd_soc_acpi_adr_device cs35l56_1_adr[] = { .adr = 0x00013701FA355601ull, .num_endpoints = 1, .endpoints = &spk_l_endpoint, - .name_prefix = "AMP8" + .name_prefix = "AMP3" }, { .adr = 0x00013601FA355601ull, .num_endpoints = 1, .endpoints = &spk_2_endpoint, - .name_prefix = "AMP7" + .name_prefix = "AMP4" } }; From 84aa440e02f3b44e927e274b9946b4c79608de43 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:26 -0500 Subject: [PATCH 087/603] ASoC: Intel: sof_sdw: Don't pass acpi_link_adr to init functions There is no need to pass the snd_soc_acpi_link_adr structure to the init functions, it conveys no specific information about the codec. Remove this parameter from the callbacks Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-32-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 7 ++----- sound/soc/intel/boards/sof_sdw_common.h | 6 ------ sound/soc/intel/boards/sof_sdw_cs_amp.c | 1 - sound/soc/intel/boards/sof_sdw_maxim.c | 1 - sound/soc/intel/boards/sof_sdw_rt711.c | 1 - sound/soc/intel/boards/sof_sdw_rt_amp.c | 1 - sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 1 - 7 files changed, 2 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1816ec5dd5806..f3167e3498e0c 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1273,7 +1273,6 @@ struct sof_sdw_endpoint { u32 link_mask; const char *codec_name; - const struct snd_soc_acpi_link_adr *adr_link; struct sof_sdw_codec_info *codec_info; const struct sof_sdw_dai_info *dai_info; }; @@ -1425,7 +1424,6 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, sof_end->link_mask = adr_link->mask; sof_end->codec_name = codec_name; - sof_end->adr_link = adr_link; sof_end->codec_info = codec_info; sof_end->dai_info = dai_info; sof_end++; @@ -1548,8 +1546,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, list_for_each_entry(sof_end, &sof_dai->endpoints, list) { if (sof_end->dai_info->init) - sof_end->dai_info->init(card, sof_end->adr_link, - *dai_links, + sof_end->dai_info->init(card, *dai_links, sof_end->codec_info, playback); } @@ -1612,7 +1609,7 @@ static int create_ssp_dailinks(struct snd_soc_card *card, if (ret) return ret; - ret = ssp_info->dais[0].init(card, NULL, *dai_links, ssp_info, 0); + ret = ssp_info->dais[0].init(card, *dai_links, ssp_info, 0); if (ret < 0) return ret; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 2132d3fc42f81..8d8c2ca68a7ff 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -74,7 +74,6 @@ struct sof_sdw_dai_info { const int dai_type; const int dailink[2]; /* dailink id for each direction */ int (*init)(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); @@ -132,7 +131,6 @@ int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); /* RT711 support */ int sof_sdw_rt711_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); @@ -140,7 +138,6 @@ int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_l /* RT711-SDCA support */ int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); @@ -151,7 +148,6 @@ extern struct snd_soc_ops sof_sdw_rt1308_i2s_ops; /* generic amp support */ int sof_sdw_rt_amp_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); @@ -159,14 +155,12 @@ int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_ /* MAXIM codec support */ int sof_sdw_maxim_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); /* CS AMP support */ int sof_sdw_cs_amp_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index 56cf75bc6cc4c..e29a586ce7c0e 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -57,7 +57,6 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) } int sof_sdw_cs_amp_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index 0347304326713..432e5112415ab 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -139,7 +139,6 @@ static int mx8373_sdw_late_probe(struct snd_soc_card *card) } int sof_sdw_maxim_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 9762c48ba7a92..7e54fc5cbe097 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -158,7 +158,6 @@ int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_l } int sof_sdw_rt711_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index 202edab950001..1b415708500e3 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -281,7 +281,6 @@ int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_ } int sof_sdw_rt_amp_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 0067714854861..85c09513bc35c 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -191,7 +191,6 @@ int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link } int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) From 36f307d296ad15e3d679d6567112b9ec4c30babc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:27 -0500 Subject: [PATCH 088/603] ASoC: Intel: sof_sdw: Remove redundant initialisations Unspecified fields are by default initialised to zero, remove redundant init pointer initialisations. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-33-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f3167e3498e0c..5a7014d82d6cb 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1026,7 +1026,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "sdw-mockup-aif1", .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, - .init = NULL, }, }, .dai_num = 1, @@ -1040,7 +1039,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "sdw-mockup-aif1", .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, - .init = NULL, }, }, .dai_num = 1, @@ -1054,7 +1052,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "sdw-mockup-aif1", .dai_type = SOF_SDW_DAI_TYPE_AMP, .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, - .init = NULL, }, }, .dai_num = 1, @@ -1068,7 +1065,6 @@ static struct sof_sdw_codec_info codec_info_list[] = { .direction = {false, true}, .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, - .init = NULL, }, }, .dai_num = 1, From 9c09bef69fe9376953348bb367c869f3d16c758c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:28 -0500 Subject: [PATCH 089/603] ASoC: Intel: sof_sdw: Add quirk for optional codec speakers Some codecs have speaker drivers that may or may not be used in a given system. Add a quirk (SOF_CODEC_SPKR) that defines if a system uses the codec speaker, this is kept generic such that any codec with such a speaker DAI can reuse the quirk. For implementation of this, add a quirk field into the sof_sdw_dai_info structure. This field is not used to specify if a system uses a particular quirk, that is still done through the sof_sdw_quirk variable, this field is used to flag a DAI as conditional on a certain quirk. This means the DAI will only be included in the card if sof_sdw_quirk and the DAI quirk field have a matching set bit. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-34-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 5 +++++ sound/soc/intel/boards/sof_sdw_common.h | 3 +++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5a7014d82d6cb..4fc15f597e286 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -37,6 +37,8 @@ static void log_quirks(struct device *dev) SOF_SSP_GET_PORT(sof_sdw_quirk)); if (sof_sdw_quirk & SOF_SDW_NO_AGGREGATION) dev_err(dev, "quirk SOF_SDW_NO_AGGREGATION enabled but no longer supported\n"); + if (sof_sdw_quirk & SOF_CODEC_SPKR) + dev_dbg(dev, "quirk SOF_CODEC_SPKR enabled\n"); } static int sof_sdw_quirk_cb(const struct dmi_system_id *id) @@ -1382,6 +1384,9 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, dai_info = &codec_info->dais[adr_end->num]; sof_dai = find_dailink(sof_dais, adr_end); + if (dai_info->quirk && !(dai_info->quirk & sof_sdw_quirk)) + continue; + dev_dbg(dev, "Add dev: %d, 0x%llx end: %d, %s, %c/%c to %s: %d\n", ffs(adr_link->mask) - 1, adr_dev->adr, diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 8d8c2ca68a7ff..81181627b4060 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -52,6 +52,8 @@ enum { #define SOF_SSP_GET_PORT(quirk) (((quirk) >> 7) & GENMASK(5, 0)) /* Deprecated and no longer supported by the code */ #define SOF_SDW_NO_AGGREGATION BIT(14) +/* If a CODEC has an optional speaker output, this quirk will enable it */ +#define SOF_CODEC_SPKR BIT(15) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 15 @@ -80,6 +82,7 @@ struct sof_sdw_dai_info { int (*exit)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); int (*rtd_init)(struct snd_soc_pcm_runtime *rtd); bool rtd_init_done; /* Indicate that the rtd_init callback is done */ + unsigned long quirk; }; struct sof_sdw_codec_info { From 59ffeb15b2f7b44cf934fd778dc0d98a35aa6a84 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 26 Mar 2024 11:04:29 -0500 Subject: [PATCH 090/603] ASoC: Intel: sof_sdw: Add support for cs42l43 optional speaker output Add support for the optional speaker output on the cs42l43, this uses the new SOF_CODEC_SPKR quirk to conditional include the speaker DAI link. It is worth noting that currently no systems are included that utilise this feature, but the feature is added as several systems are on the horizon. As part of this SOF_SDW_MAX_DAI_NUM must be increased, it is currently 3 but cs42l43 will now have 4 DAI links. This value is increased to 8 to give some head room for future devices. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240326160429.13560-35-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 11 +++- sound/soc/intel/boards/sof_sdw_common.h | 9 +++- sound/soc/intel/boards/sof_sdw_cs42l43.c | 50 +++++++++++++++++++ .../intel/common/soc-acpi-intel-mtl-match.c | 6 +++ .../intel/common/soc-acpi-intel-tgl-match.c | 6 +++ 5 files changed, 80 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 4fc15f597e286..b94835448b1b4 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1016,8 +1016,17 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_UNUSED_DAI_ID, SDW_JACK_IN_DAI_ID}, }, + { + .direction = {true, false}, + .dai_name = "cs42l43-dp6", + .dai_type = SOF_SDW_DAI_TYPE_AMP, + .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .init = sof_sdw_cs42l43_spk_init, + .rtd_init = cs42l43_spk_rtd_init, + .quirk = SOF_CODEC_SPKR, + }, }, - .dai_num = 3, + .dai_num = 4, }, { .part_id = 0xaaaa, /* generic codec mockup */ diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 81181627b4060..8468487a6bd6d 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -66,7 +66,7 @@ enum { #define SOF_SDW_DAI_TYPE_AMP 1 #define SOF_SDW_DAI_TYPE_MIC 2 -#define SOF_SDW_MAX_DAI_NUM 3 +#define SOF_SDW_MAX_DAI_NUM 8 struct sof_sdw_codec_info; @@ -162,6 +162,12 @@ int sof_sdw_maxim_init(struct snd_soc_card *card, struct sof_sdw_codec_info *info, bool playback); +/* CS42L43 support */ +int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + /* CS AMP support */ int sof_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, @@ -172,6 +178,7 @@ int sof_sdw_cs_amp_init(struct snd_soc_card *card, int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd); int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd); +int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index a9b6edac2ecd5..5361249f0f536 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -30,6 +30,17 @@ static const struct snd_soc_dapm_route cs42l43_hs_map[] = { { "cs42l43 ADC1_IN1_N", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_widget cs42l43_spk_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route cs42l43_spk_map[] = { + { "Speaker", NULL, "cs42l43 AMP1_OUT_P", }, + { "Speaker", NULL, "cs42l43 AMP1_OUT_N", }, + { "Speaker", NULL, "cs42l43 AMP2_OUT_P", }, + { "Speaker", NULL, "cs42l43 AMP2_OUT_N", }, +}; + static const struct snd_soc_dapm_widget cs42l43_dmic_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; @@ -108,6 +119,45 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } +int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s spk:cs42l43-spk", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_spk_widgets, + ARRAY_SIZE(cs42l43_spk_widgets)); + if (ret) { + dev_err(card->dev, "cs42l43 speaker widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_spk_map, + ARRAY_SIZE(cs42l43_spk_map)); + if (ret) + dev_err(card->dev, "cs42l43 speaker map addition failed: %d\n", ret); + + return ret; +} + +int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* Do init on playback link only. */ + if (!playback) + return 0; + + info->amp_num++; + + return 0; +} + int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 7e53266d831d8..27d1313e66865 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -357,6 +357,12 @@ static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { .group_position = 0, .group_id = 0, }, + { /* Speaker Playback Endpoint */ + .num = 3, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, }; static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = { diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 0cef3d788a5b0..77226d1eb1cf5 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -433,6 +433,12 @@ static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { .group_position = 0, .group_id = 0, }, + { /* Speaker Playback Endpoint */ + .num = 3, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, }; static const struct snd_soc_acpi_adr_device cs42l43_3_adr[] = { From 7b95ee0db7e0a7f99077f1b926323c7bf0d2e8f8 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 26 Mar 2024 19:09:48 +0200 Subject: [PATCH 091/603] ASoC: soc-jack: Get rid of legacy GPIO support No more users. Signed-off-by: Andy Shevchenko Link: https://msgid.link/r/20240326171134.1414462-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-jack.h | 2 -- sound/soc/generic/simple-card-utils.c | 2 -- sound/soc/soc-jack.c | 23 ++++------------------- 3 files changed, 4 insertions(+), 23 deletions(-) diff --git a/include/sound/soc-jack.h b/include/sound/soc-jack.h index a0abb1ee51107..3a81d4b8ca8a2 100644 --- a/include/sound/soc-jack.h +++ b/include/sound/soc-jack.h @@ -44,7 +44,6 @@ struct snd_soc_jack_zone { /** * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection * - * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device * @gpiod_dev: GPIO consumer device @@ -59,7 +58,6 @@ struct snd_soc_jack_zone { * ADC). */ struct snd_soc_jack_gpio { - unsigned int gpio; unsigned int idx; struct device *gpiod_dev; const char *name; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 81077d16d22f5..b4876b4f259dd 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -752,8 +752,6 @@ int simple_util_init_jack(struct snd_soc_card *card, if (!prefix) prefix = ""; - sjack->gpio.gpio = -ENOENT; - if (is_hp) { snprintf(prop, sizeof(prop), "%shp-det", prefix); pin_name = pin ? pin : "Headphones"; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index b2cc13b9c77b7..63971396b708b 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -8,7 +8,6 @@ #include #include -#include #include #include #include @@ -345,21 +344,9 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, goto undo; } } else { - /* legacy GPIO number */ - if (!gpio_is_valid(gpios[i].gpio)) { - dev_err(jack->card->dev, - "ASoC: Invalid gpio %d\n", - gpios[i].gpio); - ret = -EINVAL; - goto undo; - } - - ret = gpio_request_one(gpios[i].gpio, GPIOF_IN, - gpios[i].name); - if (ret) - goto undo; - - gpios[i].desc = gpio_to_desc(gpios[i].gpio); + dev_err(jack->card->dev, "ASoC: Invalid gpio at index %d\n", i); + ret = -EINVAL; + goto undo; } got_gpio: INIT_DELAYED_WORK(&gpios[i].work, gpio_work); @@ -373,7 +360,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpios[i].name, &gpios[i]); if (ret < 0) - goto err; + goto undo; if (gpios[i].wake) { ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1); @@ -401,8 +388,6 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, devres_add(jack->card->dev, tbl); return 0; -err: - gpio_free(gpios[i].gpio); undo: jack_free_gpios(jack, i, gpios); devres_free(tbl); From 559aebe45a054a479fdbd2a3dfba999ffd73cc9d Mon Sep 17 00:00:00 2001 From: Ondrej Jirman Date: Wed, 27 Mar 2024 09:05:34 +0100 Subject: [PATCH 092/603] ASoC: sun8i-codec: Fix build with CONFIG_SND_JACK_INPUT_DEV disabled When CONFIG_SND_JACK_INPUT_DEV is disabled, struct snd_jack doesn't have 'type' field. We can't rely on this field being always present, so store the jack_type in the codec driver itself. Signed-off-by: Ondrej Jirman Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202403261243.kcXCPAYs-lkp@intel.com/ Link: https://msgid.link/r/20240327080542.3649840-1-megi@xff.cz Signed-off-by: Mark Brown --- sound/soc/sunxi/sun8i-codec.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index 43795362fed0a..b5dafb749c3f2 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -232,6 +232,7 @@ struct sun8i_codec { struct delayed_work jack_work; int jack_irq; int jack_status; + int jack_type; int jack_last_sample; ktime_t jack_hbias_ready; struct mutex jack_mutex; @@ -1352,7 +1353,6 @@ static void sun8i_codec_jack_work(struct work_struct *work) struct sun8i_codec *scodec = container_of(work, struct sun8i_codec, jack_work.work); unsigned int mdata; - int type_mask = scodec->jack->jack->type; int type; guard(mutex)(&scodec->jack_mutex); @@ -1363,7 +1363,7 @@ static void sun8i_codec_jack_work(struct work_struct *work) scodec->jack_last_sample = -1; - if (type_mask & SND_JACK_MICROPHONE) { + if (scodec->jack_type & SND_JACK_MICROPHONE) { /* * If we were in disconnected state, we enable HBIAS and * wait 600ms before reading initial HDATA value. @@ -1376,7 +1376,7 @@ static void sun8i_codec_jack_work(struct work_struct *work) scodec->jack_status = SUN8I_JACK_STATUS_WAITING_HBIAS; } else { snd_soc_jack_report(scodec->jack, SND_JACK_HEADPHONE, - type_mask); + scodec->jack_type); scodec->jack_status = SUN8I_JACK_STATUS_CONNECTED; } } else if (scodec->jack_status == SUN8I_JACK_STATUS_WAITING_HBIAS) { @@ -1417,17 +1417,17 @@ static void sun8i_codec_jack_work(struct work_struct *work) if (type == SND_JACK_HEADPHONE) sun8i_codec_set_hmic_bias(scodec, false); - snd_soc_jack_report(scodec->jack, type, type_mask); + snd_soc_jack_report(scodec->jack, type, scodec->jack_type); scodec->jack_status = SUN8I_JACK_STATUS_CONNECTED; } else if (scodec->jack_status == SUN8I_JACK_STATUS_CONNECTED) { if (scodec->last_hmic_irq != SUN8I_HMIC_STS_JACK_OUT_IRQ_ST) return; scodec->jack_status = SUN8I_JACK_STATUS_DISCONNECTED; - if (type_mask & SND_JACK_MICROPHONE) + if (scodec->jack_type & SND_JACK_MICROPHONE) sun8i_codec_set_hmic_bias(scodec, false); - snd_soc_jack_report(scodec->jack, 0, type_mask); + snd_soc_jack_report(scodec->jack, 0, scodec->jack_type); } } @@ -1491,7 +1491,7 @@ static irqreturn_t sun8i_codec_jack_irq(int irq, void *dev_id) if (scodec->jack_last_sample >= 0 && scodec->jack_last_sample == value) snd_soc_jack_report(scodec->jack, type, - scodec->jack->jack->type); + scodec->jack_type); scodec->jack_last_sample = value; } From c335412ced27459b6ce71e812344a0bbaf43ab6a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:36 +0100 Subject: [PATCH 093/603] ASoC: loongson: i2s/pci: drop driver owner assignment PCI core in pci_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-1-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_i2s_pci.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c index fa90361865c6c..ec18b122cd792 100644 --- a/sound/soc/loongson/loongson_i2s_pci.c +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -160,7 +160,6 @@ static struct pci_driver loongson_i2s_driver = { .id_table = loongson_i2s_ids, .probe = loongson_i2s_pci_probe, .driver = { - .owner = THIS_MODULE, .pm = pm_sleep_ptr(&loongson_i2s_pm), }, }; From 8a4836231c98bde3fa49ad657c09342d2d7e27a7 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:37 +0100 Subject: [PATCH 094/603] ASoC: tlv320aic32x4-spi: drop driver owner assignment Core in spi_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-2-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c index d5976c91766e0..92246243ff94d 100644 --- a/sound/soc/codecs/tlv320aic32x4-spi.c +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -56,7 +56,6 @@ MODULE_DEVICE_TABLE(of, aic32x4_of_id); static struct spi_driver aic32x4_spi_driver = { .driver = { .name = "tlv320aic32x4", - .owner = THIS_MODULE, .of_match_table = aic32x4_of_id, }, .probe = aic32x4_spi_probe, From 54b8a522e67a8dfe135867c00f9fa297f0841c92 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:38 +0100 Subject: [PATCH 095/603] ASoC: tlv320aic3x-spi: drop driver owner assignment Core in spi_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-3-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x-spi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x-spi.c b/sound/soc/codecs/tlv320aic3x-spi.c index deed6ec7e0816..f8c1c16eaa0e7 100644 --- a/sound/soc/codecs/tlv320aic3x-spi.c +++ b/sound/soc/codecs/tlv320aic3x-spi.c @@ -63,7 +63,6 @@ MODULE_DEVICE_TABLE(of, aic3x_of_id); static struct spi_driver aic3x_spi_driver = { .driver = { .name = "tlv320aic3x", - .owner = THIS_MODULE, .of_match_table = aic3x_of_id, }, .probe = aic3x_spi_probe, From fe2065d645445dc2de61d9b336c41113759eef8a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:39 +0100 Subject: [PATCH 096/603] ASoC: max98373-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-4-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98373-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index 383e551f3bc7b..26860882fd91a 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -872,7 +872,6 @@ MODULE_DEVICE_TABLE(sdw, max98373_id); static struct sdw_driver max98373_sdw_driver = { .driver = { .name = "max98373", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(max98373_of_match), .acpi_match_table = ACPI_PTR(max98373_acpi_match), .pm = &max98373_pm, From d95c19aa2ca60010bb1976399b5a822ff165a8ef Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:40 +0100 Subject: [PATCH 097/603] ASoC: rt1017-sdca-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-5-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt1017-sdca-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c index 4dbbd8bdaaac2..7c8103a0d562a 100644 --- a/sound/soc/codecs/rt1017-sdca-sdw.c +++ b/sound/soc/codecs/rt1017-sdca-sdw.c @@ -809,7 +809,6 @@ static const struct dev_pm_ops rt1017_sdca_pm = { static struct sdw_driver rt1017_sdca_sdw_driver = { .driver = { .name = "rt1017-sdca", - .owner = THIS_MODULE, .pm = &rt1017_sdca_pm, }, .probe = rt1017_sdca_sdw_probe, From b982047530d7a29eb68d42acaa19f249ac6ae5a7 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:41 +0100 Subject: [PATCH 098/603] ASoC: rt1308-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-6-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt1308-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 63d4abf964d45..563df483a466c 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -804,7 +804,6 @@ static const struct dev_pm_ops rt1308_pm = { static struct sdw_driver rt1308_sdw_driver = { .driver = { .name = "rt1308", - .owner = THIS_MODULE, .pm = &rt1308_pm, }, .probe = rt1308_sdw_probe, From 7d5a2656626a6fbf573bde981c1550428a261cf2 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:42 +0100 Subject: [PATCH 099/603] ASoC: rt1316-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-7-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt1316-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 47511f70119ae..aa7c0ca66877e 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -781,7 +781,6 @@ static const struct dev_pm_ops rt1316_pm = { static struct sdw_driver rt1316_sdw_driver = { .driver = { .name = "rt1316-sdca", - .owner = THIS_MODULE, .pm = &rt1316_pm, }, .probe = rt1316_sdw_probe, From 3b7859dec834e96e8e833d2b2f2b4434e0e3b286 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:43 +0100 Subject: [PATCH 100/603] ASoC: rt1318-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-8-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt1318-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c index ff364bde4a084..3f6c7c25967fc 100644 --- a/sound/soc/codecs/rt1318-sdw.c +++ b/sound/soc/codecs/rt1318-sdw.c @@ -855,7 +855,6 @@ static const struct dev_pm_ops rt1318_pm = { static struct sdw_driver rt1318_sdw_driver = { .driver = { .name = "rt1318-sdca", - .owner = THIS_MODULE, .pm = &rt1318_pm, }, .probe = rt1318_sdw_probe, From e140dfeb1028ef49c34ae238863398c2c7e792f0 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:44 +0100 Subject: [PATCH 101/603] ASoC: rt5682-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-9-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index e67c2e19cb1a7..43737d5cedd0f 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -798,7 +798,6 @@ static const struct dev_pm_ops rt5682_pm = { static struct sdw_driver rt5682_sdw_driver = { .driver = { .name = "rt5682", - .owner = THIS_MODULE, .pm = &rt5682_pm, }, .probe = rt5682_sdw_probe, From 554bec0c52dc6b6bbd605aa09462d15e9c41575b Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:45 +0100 Subject: [PATCH 102/603] ASoC: rt700-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-10-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt700-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index 52c33d56b143a..24cb895b759f3 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -558,7 +558,6 @@ static const struct dev_pm_ops rt700_pm = { static struct sdw_driver rt700_sdw_driver = { .driver = { .name = "rt700", - .owner = THIS_MODULE, .pm = &rt700_pm, }, .probe = rt700_sdw_probe, From 852298a5e1ce7e72cb3c491bce3bca7262f6ef16 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:46 +0100 Subject: [PATCH 103/603] ASoC: rt711-sdca-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-11-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdca-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 935e597022d32..a7aa69495a7be 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -474,7 +474,6 @@ static const struct dev_pm_ops rt711_sdca_pm = { static struct sdw_driver rt711_sdca_sdw_driver = { .driver = { .name = "rt711-sdca", - .owner = THIS_MODULE, .pm = &rt711_sdca_pm, }, .probe = rt711_sdca_sdw_probe, From 37dc3531c56b86143b69576c4b56434d8bfcbf59 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:47 +0100 Subject: [PATCH 104/603] ASoC: rt711-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-12-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 3f5773310ae8c..0d587f48439ae 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -569,7 +569,6 @@ static const struct dev_pm_ops rt711_pm = { static struct sdw_driver rt711_sdw_driver = { .driver = { .name = "rt711", - .owner = THIS_MODULE, .pm = &rt711_pm, }, .probe = rt711_sdw_probe, From e50d2fbb6371f4d8482f5e4e112155278db6fc34 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:48 +0100 Subject: [PATCH 105/603] ASoC: rt712-sdca-dmic: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-13-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt712-sdca-dmic.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c index 0926b26619bd4..0e4d7cbdc9515 100644 --- a/sound/soc/codecs/rt712-sdca-dmic.c +++ b/sound/soc/codecs/rt712-sdca-dmic.c @@ -976,7 +976,6 @@ static int rt712_sdca_dmic_sdw_remove(struct sdw_slave *slave) static struct sdw_driver rt712_sdca_dmic_sdw_driver = { .driver = { .name = "rt712-sdca-dmic", - .owner = THIS_MODULE, .pm = &rt712_sdca_dmic_pm, }, .probe = rt712_sdca_dmic_sdw_probe, From 7c00f1706e4b9ac02fdfb2a6b93b4487cbcefd3d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:49 +0100 Subject: [PATCH 106/603] ASoC: rt712-sdca-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-14-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt712-sdca-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index 01ac555cd79b8..f04e11980591e 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -474,7 +474,6 @@ static const struct dev_pm_ops rt712_sdca_pm = { static struct sdw_driver rt712_sdca_sdw_driver = { .driver = { .name = "rt712-sdca", - .owner = THIS_MODULE, .pm = &rt712_sdca_pm, }, .probe = rt712_sdca_sdw_probe, From d4f501e2926e32be76c0c564a634f06edef768bf Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:50 +0100 Subject: [PATCH 107/603] ASoC: rt715-sdca-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-15-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt715-sdca-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index ab54a67a27ebb..08db815d3c8ec 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -270,7 +270,6 @@ static const struct dev_pm_ops rt715_pm = { static struct sdw_driver rt715_sdw_driver = { .driver = { .name = "rt715-sdca", - .owner = THIS_MODULE, .pm = &rt715_pm, }, .probe = rt715_sdca_sdw_probe, From eb9e0757976fdf34b65d70c7d25a7f0d3d1ecea9 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:51 +0100 Subject: [PATCH 108/603] ASoC: rt715-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-16-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt715-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index 21f37babd148a..823b9cacc3e28 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -577,7 +577,6 @@ static const struct dev_pm_ops rt715_pm = { static struct sdw_driver rt715_sdw_driver = { .driver = { .name = "rt715", - .owner = THIS_MODULE, .pm = &rt715_pm, }, .probe = rt715_sdw_probe, From 8c70ce6108cda5c36b9c4e66b1fd7dc8ded2a7d6 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:52 +0100 Subject: [PATCH 109/603] ASoC: rt722-sdca-sdw: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-17-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca-sdw.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index eb76f4c675b67..27af5f883c08f 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -503,7 +503,6 @@ static const struct dev_pm_ops rt722_sdca_pm = { static struct sdw_driver rt722_sdca_sdw_driver = { .driver = { .name = "rt722-sdca", - .owner = THIS_MODULE, .pm = &rt722_sdca_pm, }, .probe = rt722_sdca_sdw_probe, From 29580cd7b9c6f975e88597ca66a001b16b97bae9 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 27 Mar 2024 18:44:53 +0100 Subject: [PATCH 110/603] ASoC: sdw-mockup: drop driver owner assignment Core in sdw_register_driver() already sets the .owner, so driver does not need to. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327-module-owner-var-v1-18-86d5002ba6dc@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/sdw-mockup.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/sdw-mockup.c b/sound/soc/codecs/sdw-mockup.c index 5498ff027c58a..574c08b14f0c2 100644 --- a/sound/soc/codecs/sdw-mockup.c +++ b/sound/soc/codecs/sdw-mockup.c @@ -262,7 +262,6 @@ MODULE_DEVICE_TABLE(sdw, sdw_mockup_id); static struct sdw_driver sdw_mockup_sdw_driver = { .driver = { .name = "sdw-mockup", - .owner = THIS_MODULE, }, .probe = sdw_mockup_sdw_probe, .remove = sdw_mockup_sdw_remove, From 94a944a8c4f9e0de87cc9c5bdf8861ae2f64d874 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:51 -0500 Subject: [PATCH 111/603] ASoC: Intel: ssp-common: relocate source file Moving ssp-common source and header files from sound/soc/intel/boards directory to sound/soc/intel/common directory and rename the source file. Keep file content unchanged for tracking purpose. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Makefile | 3 --- sound/soc/intel/boards/sof_board_helpers.h | 2 +- sound/soc/intel/boards/sof_cirrus_common.h | 2 +- sound/soc/intel/boards/sof_maxim_common.h | 2 +- sound/soc/intel/boards/sof_nuvoton_common.h | 2 +- sound/soc/intel/boards/sof_realtek_common.h | 2 +- sound/soc/intel/common/Makefile | 3 +++ .../sof_ssp_common.c => common/soc-acpi-intel-ssp-common.c} | 0 sound/soc/intel/{boards => common}/sof_ssp_common.h | 0 9 files changed, 8 insertions(+), 8 deletions(-) rename sound/soc/intel/{boards/sof_ssp_common.c => common/soc-acpi-intel-ssp-common.c} (100%) rename sound/soc/intel/{boards => common}/sof_ssp_common.h (100%) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 8f711be46c43e..a93b658ed6728 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -101,8 +101,5 @@ obj-$(CONFIG_SND_SOC_INTEL_SOF_CIRRUS_COMMON) += snd-soc-intel-sof-cirrus-common snd-soc-intel-sof-nuvoton-common-objs += sof_nuvoton_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_NUVOTON_COMMON) += snd-soc-intel-sof-nuvoton-common.o -snd-soc-intel-sof-ssp-common-objs += sof_ssp_common.o -obj-$(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON) += snd-soc-intel-sof-ssp-common.o - snd-soc-intel-sof-board-helpers-objs += sof_board_helpers.o obj-$(CONFIG_SND_SOC_INTEL_SOF_BOARD_HELPERS) += snd-soc-intel-sof-board-helpers.o diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index b6a83ca14fa82..bd5c48cd0649b 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -8,7 +8,7 @@ #include #include "sof_hdmi_common.h" -#include "sof_ssp_common.h" +#include "../common/sof_ssp_common.h" /* * Common board quirks: from bit 8 to 31, LSB 8 bits reserved for machine diff --git a/sound/soc/intel/boards/sof_cirrus_common.h b/sound/soc/intel/boards/sof_cirrus_common.h index d4ecf8d023d19..445621466f05c 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.h +++ b/sound/soc/intel/boards/sof_cirrus_common.h @@ -9,7 +9,7 @@ #define __SOF_CIRRUS_COMMON_H #include -#include "sof_ssp_common.h" +#include "../common/sof_ssp_common.h" /* * Cirrus Logic CS35L41/CS35L53 diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index fe0212fbad8ea..63054bcd287bd 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -11,7 +11,7 @@ #define __SOF_MAXIM_COMMON_H #include -#include "sof_ssp_common.h" +#include "../common/sof_ssp_common.h" /* * Maxim MAX98373 diff --git a/sound/soc/intel/boards/sof_nuvoton_common.h b/sound/soc/intel/boards/sof_nuvoton_common.h index 53a84f9a67c0e..af2e58b1d393b 100644 --- a/sound/soc/intel/boards/sof_nuvoton_common.h +++ b/sound/soc/intel/boards/sof_nuvoton_common.h @@ -9,7 +9,7 @@ #define __SOF_NUVOTON_COMMON_H #include -#include "sof_ssp_common.h" +#include "../common/sof_ssp_common.h" /* * Nuvoton NAU8318 diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index e3fa2924c1c1b..99b23d96521a2 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -11,7 +11,7 @@ #define __SOF_REALTEK_COMMON_H #include -#include "sof_ssp_common.h" +#include "../common/sof_ssp_common.h" /* * Realtek ALC1011 diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index f7370e5b4e9e4..15c8a64368c57 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -17,3 +17,6 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o + +snd-soc-intel-sof-ssp-common-objs += soc-acpi-intel-ssp-common.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON) += snd-soc-intel-sof-ssp-common.o diff --git a/sound/soc/intel/boards/sof_ssp_common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c similarity index 100% rename from sound/soc/intel/boards/sof_ssp_common.c rename to sound/soc/intel/common/soc-acpi-intel-ssp-common.c diff --git a/sound/soc/intel/boards/sof_ssp_common.h b/sound/soc/intel/common/sof_ssp_common.h similarity index 100% rename from sound/soc/intel/boards/sof_ssp_common.h rename to sound/soc/intel/common/sof_ssp_common.h From bd1222ad1746ab4325b982c720c7099c78c7b731 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:52 -0500 Subject: [PATCH 112/603] ASoC: Intel: ssp-common: relocate header file Moving ssp-common header file from sound/soc/intel/common directory to include/sound directory and rename the file. Keep file content unchanged for tracking purpose. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../sound/soc-acpi-intel-ssp-common.h | 0 sound/soc/intel/boards/sof_board_helpers.h | 2 +- sound/soc/intel/boards/sof_cirrus_common.h | 2 +- sound/soc/intel/boards/sof_maxim_common.h | 2 +- sound/soc/intel/boards/sof_nuvoton_common.h | 2 +- sound/soc/intel/boards/sof_realtek_common.h | 2 +- sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 2 +- 7 files changed, 6 insertions(+), 6 deletions(-) rename sound/soc/intel/common/sof_ssp_common.h => include/sound/soc-acpi-intel-ssp-common.h (100%) diff --git a/sound/soc/intel/common/sof_ssp_common.h b/include/sound/soc-acpi-intel-ssp-common.h similarity index 100% rename from sound/soc/intel/common/sof_ssp_common.h rename to include/sound/soc-acpi-intel-ssp-common.h diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index bd5c48cd0649b..34ea974c8493d 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -7,8 +7,8 @@ #define __SOF_INTEL_BOARD_HELPERS_H #include +#include #include "sof_hdmi_common.h" -#include "../common/sof_ssp_common.h" /* * Common board quirks: from bit 8 to 31, LSB 8 bits reserved for machine diff --git a/sound/soc/intel/boards/sof_cirrus_common.h b/sound/soc/intel/boards/sof_cirrus_common.h index 445621466f05c..1c87637b9ef7b 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.h +++ b/sound/soc/intel/boards/sof_cirrus_common.h @@ -9,7 +9,7 @@ #define __SOF_CIRRUS_COMMON_H #include -#include "../common/sof_ssp_common.h" +#include /* * Cirrus Logic CS35L41/CS35L53 diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 63054bcd287bd..8886f985de68b 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -11,7 +11,7 @@ #define __SOF_MAXIM_COMMON_H #include -#include "../common/sof_ssp_common.h" +#include /* * Maxim MAX98373 diff --git a/sound/soc/intel/boards/sof_nuvoton_common.h b/sound/soc/intel/boards/sof_nuvoton_common.h index af2e58b1d393b..8a0f283260e74 100644 --- a/sound/soc/intel/boards/sof_nuvoton_common.h +++ b/sound/soc/intel/boards/sof_nuvoton_common.h @@ -9,7 +9,7 @@ #define __SOF_NUVOTON_COMMON_H #include -#include "../common/sof_ssp_common.h" +#include /* * Nuvoton NAU8318 diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index 99b23d96521a2..289e85b616602 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -11,7 +11,7 @@ #define __SOF_REALTEK_COMMON_H #include -#include "../common/sof_ssp_common.h" +#include /* * Realtek ALC1011 diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c index 96072790e9c0f..37758f4e3f32d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c +++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c @@ -4,7 +4,7 @@ #include #include -#include "sof_ssp_common.h" +#include /* * Codec probe function From a17fea3880aea23fd8821ff0660268a680e0326b Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:53 -0500 Subject: [PATCH 113/603] ASoC: Intel: ssp-common: naming convention change As we moved ssp-common files to new locations with new names, changing the naming convention from sof_ssp_ to snd_soc_acpi_intel_. Reviewed-by: Pierre-Louis Bossart Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-ssp-common.h | 17 ++++++++++------- sound/soc/intel/boards/sof_board_helpers.c | 13 +++++++------ sound/soc/intel/boards/sof_board_helpers.h | 4 ++-- .../intel/common/soc-acpi-intel-ssp-common.c | 17 ++++++++++------- 4 files changed, 29 insertions(+), 22 deletions(-) diff --git a/include/sound/soc-acpi-intel-ssp-common.h b/include/sound/soc-acpi-intel-ssp-common.h index ff84154ce60a6..b568cda110519 100644 --- a/include/sound/soc-acpi-intel-ssp-common.h +++ b/include/sound/soc-acpi-intel-ssp-common.h @@ -3,8 +3,8 @@ * Copyright(c) 2023 Intel Corporation. */ -#ifndef __SOF_SSP_COMMON_H -#define __SOF_SSP_COMMON_H +#ifndef __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H +#define __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H /* Cirrus Logic */ #define CS35L41_ACPI_HID "CSC3541" @@ -37,7 +37,7 @@ #define RT5682_ACPI_HID "10EC5682" #define RT5682S_ACPI_HID "RTL5682" -enum sof_ssp_codec { +enum snd_soc_acpi_intel_codec { CODEC_NONE, /* headphone codec */ @@ -65,9 +65,12 @@ enum sof_ssp_codec { CODEC_RT1308, }; -enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev); -enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev); +enum snd_soc_acpi_intel_codec +snd_soc_acpi_intel_detect_codec_type(struct device *dev); +enum snd_soc_acpi_intel_codec +snd_soc_acpi_intel_detect_amp_type(struct device *dev); -const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type); +const char * +snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type); -#endif /* __SOF_SSP_COMMON_H */ +#endif /* __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H */ diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 586600680e84c..8819ab5694bf6 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -103,13 +103,13 @@ static struct snd_soc_dai_link_component platform_component[] = { }; static int set_ssp_codec_link(struct device *dev, struct snd_soc_dai_link *link, - int be_id, enum sof_ssp_codec codec_type, + int be_id, enum snd_soc_acpi_intel_codec codec_type, int ssp_codec) { struct snd_soc_dai_link_component *cpus; dev_dbg(dev, "link %d: ssp codec %s, ssp %d\n", be_id, - sof_ssp_get_codec_name(codec_type), ssp_codec); + snd_soc_acpi_intel_get_codec_name(codec_type), ssp_codec); /* link name */ link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_codec); @@ -259,12 +259,13 @@ static int set_idisp_hdmi_link(struct device *dev, struct snd_soc_dai_link *link } static int set_ssp_amp_link(struct device *dev, struct snd_soc_dai_link *link, - int be_id, enum sof_ssp_codec amp_type, int ssp_amp) + int be_id, enum snd_soc_acpi_intel_codec amp_type, + int ssp_amp) { struct snd_soc_dai_link_component *cpus; dev_dbg(dev, "link %d: ssp amp %s, ssp %d\n", be_id, - sof_ssp_get_codec_name(amp_type), ssp_amp); + snd_soc_acpi_intel_get_codec_name(amp_type), ssp_amp); /* link name */ link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_amp); @@ -598,8 +599,8 @@ sof_intel_board_get_ctx(struct device *dev, unsigned long board_quirk) if (!ctx) return NULL; - ctx->codec_type = sof_ssp_detect_codec_type(dev); - ctx->amp_type = sof_ssp_detect_amp_type(dev); + ctx->codec_type = snd_soc_acpi_intel_detect_codec_type(dev); + ctx->amp_type = snd_soc_acpi_intel_detect_amp_type(dev); ctx->dmic_be_num = 2; ctx->hdmi_num = (board_quirk & SOF_NUM_IDISP_HDMI_MASK) >> diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 34ea974c8493d..644f6b4a1b7bc 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -132,8 +132,8 @@ struct sof_card_private { struct snd_soc_jack headset_jack; struct sof_hdmi_private hdmi; - enum sof_ssp_codec codec_type; - enum sof_ssp_codec amp_type; + enum snd_soc_acpi_intel_codec codec_type; + enum snd_soc_acpi_intel_codec amp_type; int dmic_be_num; int hdmi_num; diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c index 37758f4e3f32d..716243a4fc9ad 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c +++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c @@ -19,7 +19,7 @@ struct codec_map { const char *name; const char *acpi_hid; - enum sof_ssp_codec codec_type; + enum snd_soc_acpi_intel_codec codec_type; }; static const struct codec_map codecs[] = { @@ -64,7 +64,8 @@ static const struct codec_map amps[] = { CODEC_MAP_ENTRY("RT1308", RT1308_ACPI_HID, CODEC_RT1308), }; -enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev) +enum snd_soc_acpi_intel_codec +snd_soc_acpi_intel_detect_codec_type(struct device *dev) { int i; @@ -78,9 +79,10 @@ enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev) return CODEC_NONE; } -EXPORT_SYMBOL_NS(sof_ssp_detect_codec_type, SND_SOC_INTEL_SOF_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_INTEL_SOF_SSP_COMMON); -enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev) +enum snd_soc_acpi_intel_codec +snd_soc_acpi_intel_detect_amp_type(struct device *dev) { int i; @@ -94,9 +96,10 @@ enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev) return CODEC_NONE; } -EXPORT_SYMBOL_NS(sof_ssp_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON); -const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type) +const char * +snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type) { int i; @@ -115,7 +118,7 @@ const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type) return NULL; } -EXPORT_SYMBOL_NS(sof_ssp_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON); MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers"); MODULE_AUTHOR("Brent Lu "); From e1ff45518fbe73c09c3d0dbe8bd5284bb3d25dc3 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:54 -0500 Subject: [PATCH 114/603] ASoC: Intel: ssp-common: module name change As we applied new naming convention to functions and structures, changing module name as well to reflect the new naming. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 6 ++++++ sound/soc/intel/boards/Kconfig | 13 +++++-------- sound/soc/intel/boards/sof_board_helpers.c | 2 +- sound/soc/intel/common/Makefile | 4 ++-- sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 6 +++--- 5 files changed, 17 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4b9e498e33037..e17d1a517567f 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -197,6 +197,12 @@ config SND_SOC_ACPI_INTEL_MATCH # this option controls the compilation of ACPI matching tables and # helpers and is not meant to be selected by the user. +config SND_SOC_ACPI_INTEL_SSP_COMMON + tristate + select SND_SOC_ACPI if ACPI + # this option provides helpers to detect codec type and is not meant + # to be selected by the user. + endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL config SND_SOC_INTEL_KEEMBAY diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index b7a6ac4b6c0ca..4d6ef68da1303 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -41,9 +41,6 @@ config SND_SOC_INTEL_SOF_CIRRUS_COMMON config SND_SOC_INTEL_SOF_NUVOTON_COMMON tristate -config SND_SOC_INTEL_SOF_SSP_COMMON - tristate - config SND_SOC_INTEL_SOF_BOARD_HELPERS tristate @@ -503,7 +500,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON select SND_SOC_INTEL_SOF_REALTEK_COMMON - select SND_SOC_INTEL_SOF_SSP_COMMON + select SND_SOC_ACPI_INTEL_SSP_COMMON help This adds support for ASoC machine driver for SOF platforms with rt5650 or rt5682 codec. @@ -521,7 +518,7 @@ config SND_SOC_INTEL_SOF_CS42L42_MACH select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON - select SND_SOC_INTEL_SOF_SSP_COMMON + select SND_SOC_ACPI_INTEL_SSP_COMMON help This adds support for ASoC machine driver for SOF platforms with cs42l42 codec. @@ -574,7 +571,7 @@ config SND_SOC_INTEL_SOF_NAU8825_MACH select SND_SOC_INTEL_SOF_MAXIM_COMMON select SND_SOC_INTEL_SOF_NUVOTON_COMMON select SND_SOC_INTEL_SOF_REALTEK_COMMON - select SND_SOC_INTEL_SOF_SSP_COMMON + select SND_SOC_ACPI_INTEL_SSP_COMMON help This adds support for ASoC machine driver for SOF platforms with nau8825 codec. @@ -626,7 +623,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH select SND_SOC_DMIC select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON - select SND_SOC_INTEL_SOF_SSP_COMMON + select SND_SOC_ACPI_INTEL_SSP_COMMON help This adds support for ASoC machine driver for SOF platforms with Dialog DA7219 I2S audio codec. @@ -646,7 +643,7 @@ config SND_SOC_INTEL_SOF_SSP_AMP_MACH select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_REALTEK_COMMON select SND_SOC_INTEL_SOF_CIRRUS_COMMON - select SND_SOC_INTEL_SOF_SSP_COMMON + select SND_SOC_ACPI_INTEL_SSP_COMMON help This adds support for ASoC machine driver for SOF platforms with RT1308/CS35L41 I2S audio codec. diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 8819ab5694bf6..ba5be5423626b 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -635,4 +635,4 @@ MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_SSP_COMMON); diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 15c8a64368c57..dcd552d815a20 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -18,5 +18,5 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o -snd-soc-intel-sof-ssp-common-objs += soc-acpi-intel-ssp-common.o -obj-$(CONFIG_SND_SOC_INTEL_SOF_SSP_COMMON) += snd-soc-intel-sof-ssp-common.o +snd-soc-acpi-intel-ssp-common-objs += soc-acpi-intel-ssp-common.o +obj-$(CONFIG_SND_SOC_ACPI_INTEL_SSP_COMMON) += snd-soc-acpi-intel-ssp-common.o diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c index 716243a4fc9ad..40b68c20a12a0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c +++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c @@ -79,7 +79,7 @@ snd_soc_acpi_intel_detect_codec_type(struct device *dev) return CODEC_NONE; } -EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_INTEL_SOF_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_ACPI_INTEL_SSP_COMMON); enum snd_soc_acpi_intel_codec snd_soc_acpi_intel_detect_amp_type(struct device *dev) @@ -96,7 +96,7 @@ snd_soc_acpi_intel_detect_amp_type(struct device *dev) return CODEC_NONE; } -EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_ACPI_INTEL_SSP_COMMON); const char * snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type) @@ -118,7 +118,7 @@ snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type) return NULL; } -EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_ACPI_INTEL_SSP_COMMON); MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers"); MODULE_AUTHOR("Brent Lu "); From fe2365c298b51dd08def916a3339a84ec2a1fd42 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:55 -0500 Subject: [PATCH 115/603] ASoC: Intel: ssp-common: delete module Moving all code to snd-soc-acpi-intel-match module and remove the snd-soc-acpi-intel-ssp-common module. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 6 ------ sound/soc/intel/boards/Kconfig | 10 +++++----- sound/soc/intel/boards/sof_board_helpers.c | 2 +- sound/soc/intel/common/Makefile | 5 ++--- sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 6 +++--- 5 files changed, 11 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e17d1a517567f..4b9e498e33037 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -197,12 +197,6 @@ config SND_SOC_ACPI_INTEL_MATCH # this option controls the compilation of ACPI matching tables and # helpers and is not meant to be selected by the user. -config SND_SOC_ACPI_INTEL_SSP_COMMON - tristate - select SND_SOC_ACPI if ACPI - # this option provides helpers to detect codec type and is not meant - # to be selected by the user. - endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL config SND_SOC_INTEL_KEEMBAY diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 4d6ef68da1303..9f4a85513702a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -500,7 +500,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON select SND_SOC_INTEL_SOF_REALTEK_COMMON - select SND_SOC_ACPI_INTEL_SSP_COMMON + select SND_SOC_ACPI_INTEL_MATCH help This adds support for ASoC machine driver for SOF platforms with rt5650 or rt5682 codec. @@ -518,7 +518,7 @@ config SND_SOC_INTEL_SOF_CS42L42_MACH select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON - select SND_SOC_ACPI_INTEL_SSP_COMMON + select SND_SOC_ACPI_INTEL_MATCH help This adds support for ASoC machine driver for SOF platforms with cs42l42 codec. @@ -571,7 +571,7 @@ config SND_SOC_INTEL_SOF_NAU8825_MACH select SND_SOC_INTEL_SOF_MAXIM_COMMON select SND_SOC_INTEL_SOF_NUVOTON_COMMON select SND_SOC_INTEL_SOF_REALTEK_COMMON - select SND_SOC_ACPI_INTEL_SSP_COMMON + select SND_SOC_ACPI_INTEL_MATCH help This adds support for ASoC machine driver for SOF platforms with nau8825 codec. @@ -623,7 +623,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH select SND_SOC_DMIC select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON - select SND_SOC_ACPI_INTEL_SSP_COMMON + select SND_SOC_ACPI_INTEL_MATCH help This adds support for ASoC machine driver for SOF platforms with Dialog DA7219 I2S audio codec. @@ -643,7 +643,7 @@ config SND_SOC_INTEL_SOF_SSP_AMP_MACH select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_REALTEK_COMMON select SND_SOC_INTEL_SOF_CIRRUS_COMMON - select SND_SOC_ACPI_INTEL_SSP_COMMON + select SND_SOC_ACPI_INTEL_MATCH help This adds support for ASoC machine driver for SOF platforms with RT1308/CS35L41 I2S audio codec. diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index ba5be5423626b..eb140e13153f8 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -635,4 +635,4 @@ MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); -MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_SSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_MATCH); diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index dcd552d815a20..a86457674726b 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -15,8 +15,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-hda-match.o \ soc-acpi-intel-sdw-mockup-match.o +snd-soc-acpi-intel-match-objs += soc-acpi-intel-ssp-common.o + obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o - -snd-soc-acpi-intel-ssp-common-objs += soc-acpi-intel-ssp-common.o -obj-$(CONFIG_SND_SOC_ACPI_INTEL_SSP_COMMON) += snd-soc-acpi-intel-ssp-common.o diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c index 40b68c20a12a0..68da7260fb1dd 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c +++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c @@ -79,7 +79,7 @@ snd_soc_acpi_intel_detect_codec_type(struct device *dev) return CODEC_NONE; } -EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_ACPI_INTEL_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_codec_type, SND_SOC_ACPI_INTEL_MATCH); enum snd_soc_acpi_intel_codec snd_soc_acpi_intel_detect_amp_type(struct device *dev) @@ -96,7 +96,7 @@ snd_soc_acpi_intel_detect_amp_type(struct device *dev) return CODEC_NONE; } -EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_ACPI_INTEL_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_detect_amp_type, SND_SOC_ACPI_INTEL_MATCH); const char * snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type) @@ -118,7 +118,7 @@ snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type) return NULL; } -EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_ACPI_INTEL_SSP_COMMON); +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_ACPI_INTEL_MATCH); MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers"); MODULE_AUTHOR("Brent Lu "); From 2e723a79ec609871116d216309c1b89d2f61b713 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:56 -0500 Subject: [PATCH 116/603] ASoC: Intel: ssp-common: get codec tplg suffix function Add helper functions to get tplg suffix string for specific headphone codec or speaker amplifier. The string could be used to compose the default topology file name for specific headphone codec and speaker amplifier combination. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-ssp-common.h | 5 ++ .../intel/common/soc-acpi-intel-ssp-common.c | 76 ++++++++++++++----- 2 files changed, 60 insertions(+), 21 deletions(-) diff --git a/include/sound/soc-acpi-intel-ssp-common.h b/include/sound/soc-acpi-intel-ssp-common.h index b568cda110519..b4597c8dac782 100644 --- a/include/sound/soc-acpi-intel-ssp-common.h +++ b/include/sound/soc-acpi-intel-ssp-common.h @@ -73,4 +73,9 @@ snd_soc_acpi_intel_detect_amp_type(struct device *dev); const char * snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type); +const char * +snd_soc_acpi_intel_get_codec_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type); +const char * +snd_soc_acpi_intel_get_amp_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type); + #endif /* __LINUX_SND_SOC_ACPI_INTEL_SSP_COMMON_H */ diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c index 68da7260fb1dd..a887f019afe25 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c +++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c @@ -9,59 +9,61 @@ /* * Codec probe function */ -#define CODEC_MAP_ENTRY(n, h, t) \ +#define CODEC_MAP_ENTRY(n, s, h, t) \ { \ .name = n, \ + .tplg_suffix = s, \ .acpi_hid = h, \ .codec_type = t, \ } struct codec_map { const char *name; + const char *tplg_suffix; const char *acpi_hid; enum snd_soc_acpi_intel_codec codec_type; }; static const struct codec_map codecs[] = { /* Cirrus Logic */ - CODEC_MAP_ENTRY("CS42L42", CS42L42_ACPI_HID, CODEC_CS42L42), + CODEC_MAP_ENTRY("CS42L42", "cs42l42", CS42L42_ACPI_HID, CODEC_CS42L42), /* Dialog */ - CODEC_MAP_ENTRY("DA7219", DA7219_ACPI_HID, CODEC_DA7219), + CODEC_MAP_ENTRY("DA7219", "da7219", DA7219_ACPI_HID, CODEC_DA7219), /* Everest */ - CODEC_MAP_ENTRY("ES8316", ES8316_ACPI_HID, CODEC_ES8316), - CODEC_MAP_ENTRY("ES8326", ES8326_ACPI_HID, CODEC_ES8326), - CODEC_MAP_ENTRY("ES8336", ES8336_ACPI_HID, CODEC_ES8336), + CODEC_MAP_ENTRY("ES8316", "es8336", ES8316_ACPI_HID, CODEC_ES8316), + CODEC_MAP_ENTRY("ES8326", "es8336", ES8326_ACPI_HID, CODEC_ES8326), + CODEC_MAP_ENTRY("ES8336", "es8336", ES8336_ACPI_HID, CODEC_ES8336), /* Nuvoton */ - CODEC_MAP_ENTRY("NAU8825", NAU8825_ACPI_HID, CODEC_NAU8825), + CODEC_MAP_ENTRY("NAU8825", "nau8825", NAU8825_ACPI_HID, CODEC_NAU8825), /* Realtek */ - CODEC_MAP_ENTRY("RT5650", RT5650_ACPI_HID, CODEC_RT5650), - CODEC_MAP_ENTRY("RT5682", RT5682_ACPI_HID, CODEC_RT5682), - CODEC_MAP_ENTRY("RT5682S", RT5682S_ACPI_HID, CODEC_RT5682S), + CODEC_MAP_ENTRY("RT5650", "rt5650", RT5650_ACPI_HID, CODEC_RT5650), + CODEC_MAP_ENTRY("RT5682", "rt5682", RT5682_ACPI_HID, CODEC_RT5682), + CODEC_MAP_ENTRY("RT5682S", "rt5682", RT5682S_ACPI_HID, CODEC_RT5682S), }; static const struct codec_map amps[] = { /* Cirrus Logic */ - CODEC_MAP_ENTRY("CS35L41", CS35L41_ACPI_HID, CODEC_CS35L41), + CODEC_MAP_ENTRY("CS35L41", "cs35l41", CS35L41_ACPI_HID, CODEC_CS35L41), /* Maxim */ - CODEC_MAP_ENTRY("MAX98357A", MAX_98357A_ACPI_HID, CODEC_MAX98357A), - CODEC_MAP_ENTRY("MAX98360A", MAX_98360A_ACPI_HID, CODEC_MAX98360A), - CODEC_MAP_ENTRY("MAX98373", MAX_98373_ACPI_HID, CODEC_MAX98373), - CODEC_MAP_ENTRY("MAX98390", MAX_98390_ACPI_HID, CODEC_MAX98390), + CODEC_MAP_ENTRY("MAX98357A", "max98357a", MAX_98357A_ACPI_HID, CODEC_MAX98357A), + CODEC_MAP_ENTRY("MAX98360A", "max98360a", MAX_98360A_ACPI_HID, CODEC_MAX98360A), + CODEC_MAP_ENTRY("MAX98373", "max98373", MAX_98373_ACPI_HID, CODEC_MAX98373), + CODEC_MAP_ENTRY("MAX98390", "max98390", MAX_98390_ACPI_HID, CODEC_MAX98390), /* Nuvoton */ - CODEC_MAP_ENTRY("NAU8318", NAU8318_ACPI_HID, CODEC_NAU8318), + CODEC_MAP_ENTRY("NAU8318", "nau8318", NAU8318_ACPI_HID, CODEC_NAU8318), /* Realtek */ - CODEC_MAP_ENTRY("RT1011", RT1011_ACPI_HID, CODEC_RT1011), - CODEC_MAP_ENTRY("RT1015", RT1015_ACPI_HID, CODEC_RT1015), - CODEC_MAP_ENTRY("RT1015P", RT1015P_ACPI_HID, CODEC_RT1015P), - CODEC_MAP_ENTRY("RT1019P", RT1019P_ACPI_HID, CODEC_RT1019P), - CODEC_MAP_ENTRY("RT1308", RT1308_ACPI_HID, CODEC_RT1308), + CODEC_MAP_ENTRY("RT1011", "rt1011", RT1011_ACPI_HID, CODEC_RT1011), + CODEC_MAP_ENTRY("RT1015", "rt1015", RT1015_ACPI_HID, CODEC_RT1015), + CODEC_MAP_ENTRY("RT1015P", "rt1015", RT1015P_ACPI_HID, CODEC_RT1015P), + CODEC_MAP_ENTRY("RT1019P", "rt1019", RT1019P_ACPI_HID, CODEC_RT1019P), + CODEC_MAP_ENTRY("RT1308", "rt1308", RT1308_ACPI_HID, CODEC_RT1308), }; enum snd_soc_acpi_intel_codec @@ -120,6 +122,38 @@ snd_soc_acpi_intel_get_codec_name(enum snd_soc_acpi_intel_codec codec_type) } EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_name, SND_SOC_ACPI_INTEL_MATCH); +const char * +snd_soc_acpi_intel_get_codec_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(codecs); i++) { + if (codecs[i].codec_type != codec_type) + continue; + + return codecs[i].tplg_suffix; + } + + return NULL; +} +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_codec_tplg_suffix, SND_SOC_ACPI_INTEL_MATCH); + +const char * +snd_soc_acpi_intel_get_amp_tplg_suffix(enum snd_soc_acpi_intel_codec codec_type) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(amps); i++) { + if (amps[i].codec_type != codec_type) + continue; + + return amps[i].tplg_suffix; + } + + return NULL; +} +EXPORT_SYMBOL_NS(snd_soc_acpi_intel_get_amp_tplg_suffix, SND_SOC_ACPI_INTEL_MATCH); + MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); From 1504a768f6045157437693fbfb50ae63ca86ec61 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:57 -0500 Subject: [PATCH 117/603] ASoC: SOF: Intel: support tplg suffix detection Add new flags to tplg_quirk_mask to detect and append codec/amplifier tplg suffix to topology file name at runtime. With this feature we could implement an enumeration entry for all boards which implement same headphone codec regardless the speaker amplifier type. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 12 ++++++ sound/soc/sof/intel/hda.c | 82 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 94 insertions(+) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 23d6d6bfb0736..1d8f35ca1d6f3 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -151,6 +151,18 @@ struct snd_soc_acpi_link_adr { */ #define SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER BIT(2) +/* + * when set the speaker amplifier name suffix (i.e. "-max98360a") will be + * appended to topology file name + */ +#define SND_SOC_ACPI_TPLG_INTEL_AMP_NAME BIT(3) + +/* + * when set the headphone codec name suffix (i.e. "-rt5682") will be appended to + * topology file name + */ +#define SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME BIT(4) + /** * snd_soc_acpi_mach: ACPI-based machine descriptor. Most of the fields are * related to the hardware, except for the firmware and topology file names. diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 7fe72b0654514..e26b8fd682e5e 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include @@ -1676,13 +1677,36 @@ void hda_set_mach_params(struct snd_soc_acpi_mach *mach, mach_params->dai_drivers = desc->ops->drv; } +static int check_tplg_quirk_mask(struct snd_soc_acpi_mach *mach) +{ + u32 dmic_ssp_quirk; + u32 codec_amp_name_quirk; + + /* + * In current implementation dmic and ssp quirks are designed for es8336 + * machine driver and could not be mixed with codec name and amp name + * quirks. + */ + dmic_ssp_quirk = mach->tplg_quirk_mask & + (SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER | SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER); + codec_amp_name_quirk = mach->tplg_quirk_mask & + (SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME); + + if (dmic_ssp_quirk && codec_amp_name_quirk) + return -EINVAL; + + return 0; +} + struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) { u32 interface_mask = hda_get_interface_mask(sdev); struct snd_sof_pdata *sof_pdata = sdev->pdata; const struct sof_dev_desc *desc = sof_pdata->desc; struct snd_soc_acpi_mach *mach = NULL; + enum snd_soc_acpi_intel_codec codec_type; const char *tplg_filename; + const char *tplg_suffix; /* Try I2S or DMIC if it is supported */ if (interface_mask & (BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC))) @@ -1701,6 +1725,17 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) tplg_fixup = true; } + /* + * Checking quirk mask integrity; some quirk flags could not be + * set concurrently. + */ + if (tplg_fixup && + check_tplg_quirk_mask(mach)) { + dev_err(sdev->dev, "Invalid tplg quirk mask 0x%x\n", + mach->tplg_quirk_mask); + return NULL; + } + /* report to machine driver if any DMICs are found */ mach->mach_params.dmic_num = check_dmic_num(sdev); @@ -1775,6 +1810,52 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) } } + codec_type = snd_soc_acpi_intel_detect_amp_type(sdev->dev); + + if (tplg_fixup && + mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_AMP_NAME && + codec_type != CODEC_NONE) { + tplg_suffix = snd_soc_acpi_intel_get_amp_tplg_suffix(codec_type); + if (!tplg_suffix) { + dev_err(sdev->dev, "no tplg suffix found, amp %d\n", + codec_type); + return NULL; + } + + tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, + "%s-%s", + sof_pdata->tplg_filename, + tplg_suffix); + if (!tplg_filename) + return NULL; + + sof_pdata->tplg_filename = tplg_filename; + add_extension = true; + } + + codec_type = snd_soc_acpi_intel_detect_codec_type(sdev->dev); + + if (tplg_fixup && + mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME && + codec_type != CODEC_NONE) { + tplg_suffix = snd_soc_acpi_intel_get_codec_tplg_suffix(codec_type); + if (!tplg_suffix) { + dev_err(sdev->dev, "no tplg suffix found, codec %d\n", + codec_type); + return NULL; + } + + tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, + "%s-%s", + sof_pdata->tplg_filename, + tplg_suffix); + if (!tplg_filename) + return NULL; + + sof_pdata->tplg_filename = tplg_filename; + add_extension = true; + } + if (tplg_fixup && add_extension) { tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, "%s%s", @@ -1842,3 +1923,4 @@ MODULE_IMPORT_NS(SND_INTEL_SOUNDWIRE_ACPI); MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT); MODULE_IMPORT_NS(SOUNDWIRE_INTEL); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK); +MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_MATCH); From 1934906b26bf8d1bd798fa11bf7058bf380101a0 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:58 -0500 Subject: [PATCH 118/603] ASoC: Intel: sof_nau8825: mach cleanup for adl boards Add a common entry in enumeration table for all nau8825 boards with/without speaker amplifier. All other adl_nau8825_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-adl-match.c | 54 ++++--------------- 1 file changed, 11 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 7ce8aade07d70..f8b8864b5ec65 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -7,6 +7,7 @@ #include #include +#include static const struct snd_soc_acpi_codecs essx_83x6 = { .num_codecs = 3, @@ -467,11 +468,6 @@ static const struct snd_soc_acpi_codecs adl_rt5682_rt5682s_hp = { .codecs = {"10EC5682", "RTL5682"}, }; -static const struct snd_soc_acpi_codecs adl_rt1015p_amp = { - .num_codecs = 1, - .codecs = {"RTL1015"} -}; - static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { .num_codecs = 1, .codecs = {"RTL1019"} @@ -487,11 +483,6 @@ static const struct snd_soc_acpi_codecs adl_lt6911_hdmi = { .codecs = {"INTC10B0"} }; -static const struct snd_soc_acpi_codecs adl_nau8318_amp = { - .num_codecs = 1, - .codecs = {"NVTN2012"} -}; - static struct snd_soc_acpi_codecs adl_rt5650_amp = { .num_codecs = 1, .codecs = {"10EC5650"} @@ -526,20 +517,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_rt1019p_amp, .sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg", }, - { - .id = "10508825", - .drv_name = "adl_nau8825_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_max98373_amp, - .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg", - }, - { - .id = "10508825", - .drv_name = "adl_nau8825_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_max98360a_amp, - .sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg", - }, { .comp_ids = &adl_rt5682_rt5682s_hp, .drv_name = "adl_rt5682_def", @@ -547,25 +524,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_rt1019p_amp, .sof_tplg_filename = "sof-adl-rt1019-rt5682.tplg", }, - { - .id = "10508825", - .drv_name = "adl_nau8825_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_rt1015p_amp, - .sof_tplg_filename = "sof-adl-rt1015-nau8825.tplg", - }, - { - .id = "10508825", - .drv_name = "adl_nau8825_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_nau8318_amp, - .sof_tplg_filename = "sof-adl-nau8318-nau8825.tplg", - }, - { - .id = "10508825", - .drv_name = "adl_nau8825_def", - .sof_tplg_filename = "sof-adl-nau8825.tplg", - }, { .comp_ids = &adl_rt5682_rt5682s_hp, .drv_name = "adl_rt5682_def", @@ -621,6 +579,16 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_max98360a_amp, .sof_tplg_filename = "sof-adl-max98360a-da7219.tplg", }, + /* place boards for each headphone codec: sof driver will complete the + * tplg name and machine driver will detect the amp type + */ + { + .id = NAU8825_ACPI_HID, + .drv_name = "adl_nau8825_def", + .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, /* place amp-only boards in the end of table */ { .id = "CSC3541", From 3822d41469fc20fdff0d83b20324b382ee7bd0e7 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:23:59 -0500 Subject: [PATCH 119/603] ASoC: Intel: sof_nau8825: mach cleanup for rpl boards Add a common entry in enumeration table for all nau8825 boards with/without speaker amplifier. All other rpl_nau8825_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-rpl-match.c | 43 ++++++------------- 1 file changed, 12 insertions(+), 31 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 77c917897c8d1..44be9077b4e08 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -7,6 +7,7 @@ #include #include +#include static const struct snd_soc_acpi_endpoint single_endpoint = { .num = 0, @@ -365,21 +366,11 @@ static const struct snd_soc_acpi_codecs rpl_max98360a_amp = { .codecs = {"MX98360A"}, }; -static const struct snd_soc_acpi_codecs rpl_max98373_amp = { - .num_codecs = 1, - .codecs = {"MX98373"} -}; - static const struct snd_soc_acpi_codecs rpl_lt6911_hdmi = { .num_codecs = 1, .codecs = {"INTC10B0"} }; -static const struct snd_soc_acpi_codecs rpl_nau8318_amp = { - .num_codecs = 1, - .codecs = {"NVTN2012"} -}; - static const struct snd_soc_acpi_codecs rpl_rt1019p_amp = { .num_codecs = 1, .codecs = {"RTL1019"} @@ -400,27 +391,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .quirk_data = &rpl_max98360a_amp, .sof_tplg_filename = "sof-rpl-max98360a-rt5682.tplg", }, - { - .id = "10508825", - .drv_name = "rpl_nau8825_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &rpl_max98373_amp, - .sof_tplg_filename = "sof-rpl-max98373-nau8825.tplg", - }, - { - .id = "10508825", - .drv_name = "rpl_nau8825_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &rpl_max98360a_amp, - .sof_tplg_filename = "sof-rpl-max98360a-nau8825.tplg", - }, - { - .id = "10508825", - .drv_name = "rpl_nau8825_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &rpl_nau8318_amp, - .sof_tplg_filename = "sof-rpl-nau8318-nau8825.tplg", - }, { .comp_ids = &rpl_rt5682_hp, .drv_name = "rpl_rt5682_def", @@ -450,6 +420,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + /* place boards for each headphone codec: sof driver will complete the + * tplg name and machine driver will detect the amp type + */ + { + .id = NAU8825_ACPI_HID, + .drv_name = "rpl_nau8825_def", + .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, + /* place amp-only boards in the end of table */ { .id = "INTC10B0", .drv_name = "rpl_lt6911_hdmi_ssp", From d5dd7f4fc0b5633fc9fb665803f5e95d564c7331 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:00 -0500 Subject: [PATCH 120/603] ASoC: Intel: sof_nau8825: add mtl_nau8825_def for mtl boards Add the board config mtl_nau8825_def to nau8825 machine driver for all mtl boards using default SSP port allocation (headphone codec on SSP2, speaker amplifiers on SSP0, and BT offload on SSP1). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 7 +++++++ sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 11 +++++++++++ 2 files changed, 18 insertions(+) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 23fe8b4015cce..fe5f9e8dd6520 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -319,6 +319,13 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_PORT_BT_OFFLOAD(2) | SOF_BT_OFFLOAD_PRESENT), }, + { + .name = "mtl_nau8825_def", + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(0) | + SOF_SSP_PORT_BT_OFFLOAD(1) | + SOF_BT_OFFLOAD_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 27d1313e66865..0125e81af9e12 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -8,6 +8,7 @@ #include #include +#include #include "soc-acpi-intel-sdw-mockup-match.h" static const struct snd_soc_acpi_codecs mtl_max98357a_amp = { @@ -89,6 +90,16 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_rt5650_amp, .sof_tplg_filename = "sof-mtl-rt5650.tplg", }, + /* place boards for each headphone codec: sof driver will complete the + * tplg name and machine driver will detect the amp type + */ + { + .id = NAU8825_ACPI_HID, + .drv_name = "mtl_nau8825_def", + .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, /* place amp-only boards in the end of table */ { .id = "INTC10B0", From 4b95706b4a25c239aa7a7d7feb455c364c181db2 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:01 -0500 Subject: [PATCH 121/603] ASoC: Intel: sof_rt5682: mach cleanup for tgl boards Add a common entry in enumeration table for all rt5682 boards with/without speaker amplifier. All other tgl_rt5682_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-tgl-match.c | 50 +++++-------------- 1 file changed, 13 insertions(+), 37 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 77226d1eb1cf5..161ba532d270a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -8,6 +8,7 @@ #include #include +#include #include "soc-acpi-intel-sdw-mockup-match.h" static const struct snd_soc_acpi_codecs essx_83x6 = { @@ -15,11 +16,6 @@ static const struct snd_soc_acpi_codecs essx_83x6 = { .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, }; -static const struct snd_soc_acpi_codecs tgl_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - static const struct snd_soc_acpi_endpoint single_endpoint = { .num = 0, .aggregated = 0, @@ -499,19 +495,9 @@ static const struct snd_soc_acpi_link_adr tgl_cs42l43_cs35l56[] = { {} }; -static const struct snd_soc_acpi_codecs tgl_max98373_amp = { - .num_codecs = 1, - .codecs = {"MX98373"} -}; - -static const struct snd_soc_acpi_codecs tgl_rt1011_amp = { - .num_codecs = 1, - .codecs = {"10EC1011"} -}; - static const struct snd_soc_acpi_codecs tgl_rt5682_rt5682s_hp = { .num_codecs = 2, - .codecs = {"10EC5682", "RTL5682"}, + .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID}, }; static const struct snd_soc_acpi_codecs tgl_lt6911_hdmi = { @@ -520,27 +506,6 @@ static const struct snd_soc_acpi_codecs tgl_lt6911_hdmi = { }; struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { - { - .comp_ids = &tgl_rt5682_rt5682s_hp, - .drv_name = "tgl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &tgl_codecs, - .sof_tplg_filename = "sof-tgl-max98357a-rt5682.tplg", - }, - { - .comp_ids = &tgl_rt5682_rt5682s_hp, - .drv_name = "tgl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &tgl_max98373_amp, - .sof_tplg_filename = "sof-tgl-max98373-rt5682.tplg", - }, - { - .comp_ids = &tgl_rt5682_rt5682s_hp, - .drv_name = "tgl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &tgl_rt1011_amp, - .sof_tplg_filename = "sof-tgl-rt1011-rt5682.tplg", - }, { .comp_ids = &essx_83x6, .drv_name = "sof-essx8336", @@ -549,6 +514,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + /* place boards for each headphone codec: sof driver will complete the + * tplg name and machine driver will detect the amp type + */ + { + .comp_ids = &tgl_rt5682_rt5682s_hp, + .drv_name = "tgl_rt5682_def", + .sof_tplg_filename = "sof-tgl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, + /* place amp-only boards in the end of table */ { .id = "10EC1308", .drv_name = "tgl_rt1308_hdmi_ssp", From 34f256537685437122c4d326266739dd3b3678c6 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:02 -0500 Subject: [PATCH 122/603] ASoC: Intel: sof_rt5682: mach cleanup for adl boards Add two common entries in enumeration table for all rt5682/rt5650 boards with/without speaker amplifier. All other adl_rt5682_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-adl-match.c | 71 ++++--------------- 1 file changed, 15 insertions(+), 56 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index f8b8864b5ec65..9ac9310dd2be6 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -448,11 +448,6 @@ static const struct snd_soc_acpi_link_adr adl_chromebook_base[] = { {} }; -static const struct snd_soc_acpi_codecs adl_max98373_amp = { - .num_codecs = 1, - .codecs = {"MX98373"} -}; - static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .num_codecs = 1, .codecs = {"MX98357A"} @@ -465,7 +460,7 @@ static const struct snd_soc_acpi_codecs adl_max98360a_amp = { static const struct snd_soc_acpi_codecs adl_rt5682_rt5682s_hp = { .num_codecs = 2, - .codecs = {"10EC5682", "RTL5682"}, + .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID}, }; static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { @@ -473,29 +468,12 @@ static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { .codecs = {"RTL1019"} }; -static const struct snd_soc_acpi_codecs adl_max98390_amp = { - .num_codecs = 1, - .codecs = {"MX98390"} -}; - static const struct snd_soc_acpi_codecs adl_lt6911_hdmi = { .num_codecs = 1, .codecs = {"INTC10B0"} }; -static struct snd_soc_acpi_codecs adl_rt5650_amp = { - .num_codecs = 1, - .codecs = {"10EC5650"} -}; - struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { - { - .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_max98373_amp, - .sof_tplg_filename = "sof-adl-max98373-rt5682.tplg", - }, { .comp_ids = &adl_rt5682_rt5682s_hp, .drv_name = "adl_mx98357_rt5682", @@ -503,13 +481,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_max98357a_amp, .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, - { - .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_max98360a_amp, - .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", - }, { .id = "10508825", .drv_name = "adl_rt1019p_8825", @@ -517,20 +488,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_rt1019p_amp, .sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg", }, - { - .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_rt1019p_amp, - .sof_tplg_filename = "sof-adl-rt1019-rt5682.tplg", - }, - { - .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_max98390_amp, - .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", - }, { .comp_ids = &adl_rt5682_rt5682s_hp, .drv_name = "adl_rt5682_c1_h02", @@ -538,11 +495,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_lt6911_hdmi, .sof_tplg_filename = "sof-adl-rt5682-ssp1-hdmi-ssp02.tplg", }, - { - .comp_ids = &adl_rt5682_rt5682s_hp, - .drv_name = "adl_rt5682_def", - .sof_tplg_filename = "sof-adl-rt5682.tplg", - }, { .id = "10134242", .drv_name = "adl_mx98360a_cs4242", @@ -565,13 +517,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, - { - .id = "10EC5650", - .drv_name = "adl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_rt5650_amp, - .sof_tplg_filename = "sof-adl-rt5650.tplg", - }, { .id = "DLGS7219", .drv_name = "adl_da7219_def", @@ -589,6 +534,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, }, + { + .id = RT5650_ACPI_HID, + .drv_name = "adl_rt5682_def", + .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, + { + .comp_ids = &adl_rt5682_rt5682s_hp, + .drv_name = "adl_rt5682_def", + .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, /* place amp-only boards in the end of table */ { .id = "CSC3541", From 7b24d86c845ce3b1c7bdd9175cd7c51aefe9ba46 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:03 -0500 Subject: [PATCH 123/603] ASoC: Intel: sof_rt5682: mach cleanup for rpl boards Add two common entries in enumeration table for all rt5682/rt5650 boards with/without speaker amplifier. All other rpl_rt5682_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-rpl-match.c | 40 +++++++------------ 1 file changed, 15 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 44be9077b4e08..e28a21667780a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -348,7 +348,7 @@ static const struct snd_soc_acpi_link_adr rplp_crb[] = { static const struct snd_soc_acpi_codecs rpl_rt5682_hp = { .num_codecs = 2, - .codecs = {"10EC5682", "RTL5682"}, + .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID}, }; static const struct snd_soc_acpi_codecs rpl_essx_83x6 = { @@ -371,11 +371,6 @@ static const struct snd_soc_acpi_codecs rpl_lt6911_hdmi = { .codecs = {"INTC10B0"} }; -static const struct snd_soc_acpi_codecs rpl_rt1019p_amp = { - .num_codecs = 1, - .codecs = {"RTL1019"} -}; - struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { { .comp_ids = &rpl_rt5682_hp, @@ -384,20 +379,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .quirk_data = &rpl_max98357a_amp, .sof_tplg_filename = "sof-rpl-max98357a-rt5682.tplg", }, - { - .comp_ids = &rpl_rt5682_hp, - .drv_name = "rpl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &rpl_max98360a_amp, - .sof_tplg_filename = "sof-rpl-max98360a-rt5682.tplg", - }, - { - .comp_ids = &rpl_rt5682_hp, - .drv_name = "rpl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &rpl_rt1019p_amp, - .sof_tplg_filename = "sof-rpl-rt1019-rt5682.tplg", - }, { .comp_ids = &rpl_rt5682_hp, .drv_name = "rpl_rt5682_c1_h02", @@ -430,6 +411,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, }, + { + .id = RT5650_ACPI_HID, + .drv_name = "rpl_rt5682_def", + .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, + { + .comp_ids = &rpl_rt5682_hp, + .drv_name = "rpl_rt5682_def", + .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, /* place amp-only boards in the end of table */ { .id = "INTC10B0", @@ -443,11 +438,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .quirk_data = &rpl_max98360a_amp, .sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg", }, - { - .id = "10EC5650", - .drv_name = "rpl_rt5682_def", - .sof_tplg_filename = "sof-rpl-rt5650.tplg", - }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines); From a2e678fe951c3820db705a2cb20ec0a32d752033 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:04 -0500 Subject: [PATCH 124/603] ASoC: Intel: sof_rt5682: mach cleanup for mtl boards Add two common entries in enumeration table for all rt5682/rt5650 boards with/without speaker amplifier. All other mtl_rt5682_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-15-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 40 +++++++------------ 1 file changed, 15 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 0125e81af9e12..7f422848a3ba6 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -21,14 +21,9 @@ static const struct snd_soc_acpi_codecs mtl_max98360a_amp = { .codecs = {"MX98360A"} }; -static const struct snd_soc_acpi_codecs mtl_rt1019p_amp = { - .num_codecs = 1, - .codecs = {"RTL1019"} -}; - static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { .num_codecs = 2, - .codecs = {"10EC5682", "RTL5682"}, + .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID}, }; static const struct snd_soc_acpi_codecs mtl_essx_83x6 = { @@ -41,11 +36,6 @@ static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = { .codecs = {"INTC10B0"} }; -static const struct snd_soc_acpi_codecs mtl_rt5650_amp = { - .num_codecs = 1, - .codecs = {"10EC5650"} -}; - struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { { .comp_ids = &mtl_rt5682_rt5682s_hp, @@ -61,13 +51,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .quirk_data = &mtl_max98360a_amp, .sof_tplg_filename = "sof-mtl-max98360a-rt5682.tplg", }, - { - .comp_ids = &mtl_rt5682_rt5682s_hp, - .drv_name = "mtl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &mtl_rt1019p_amp, - .sof_tplg_filename = "sof-mtl-rt1019-rt5682.tplg", - }, { .comp_ids = &mtl_essx_83x6, .drv_name = "mtl_es83x6_c1_h02", @@ -83,13 +66,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, - { - .id = "10EC5650", - .drv_name = "mtl_rt5682_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &mtl_rt5650_amp, - .sof_tplg_filename = "sof-mtl-rt5650.tplg", - }, /* place boards for each headphone codec: sof driver will complete the * tplg name and machine driver will detect the amp type */ @@ -100,6 +76,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, }, + { + .id = RT5650_ACPI_HID, + .drv_name = "mtl_rt5682_def", + .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, + { + .comp_ids = &mtl_rt5682_rt5682s_hp, + .drv_name = "mtl_rt5682_def", + .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, /* place amp-only boards in the end of table */ { .id = "INTC10B0", From 2872f3b5df54ac14adc5ea62241312d66bd001ba Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:05 -0500 Subject: [PATCH 125/603] ASoC: Intel: sof_cs42l42: board id cleanup for adl boards Introduce "adl_cs42l42_def" for adl boards which implement headphone codec on SSP0 and speaker amplifiers on SSP1. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-16-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 2 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 40ecfeaa1d263..8a2833a0e3c52 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -263,7 +263,7 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_PORT_AMP(1)), }, { - .name = "adl_mx98360a_cs4242", + .name = "adl_cs42l42_def", .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1) | SOF_NUM_IDISP_HDMI(4) | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 9ac9310dd2be6..7977a6fb9a5c8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -497,7 +497,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10134242", - .drv_name = "adl_mx98360a_cs4242", + .drv_name = "adl_cs42l42_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, .sof_tplg_filename = "sof-adl-max98360a-cs42l42.tplg", From ae33c9134d6e1a88c2e49422aee1474c03a53233 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:06 -0500 Subject: [PATCH 126/603] ASoC: Intel: sof_cs42l42: mach cleanup for adl boards Add a common entry in enumeration table for all cs42l42 boards with/without speaker amplifier. All other adl_cs42l42_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-17-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 7977a6fb9a5c8..1ea2d9c582ee1 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -495,13 +495,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_lt6911_hdmi, .sof_tplg_filename = "sof-adl-rt5682-ssp1-hdmi-ssp02.tplg", }, - { - .id = "10134242", - .drv_name = "adl_cs42l42_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_max98360a_amp, - .sof_tplg_filename = "sof-adl-max98360a-cs42l42.tplg", - }, { .comp_ids = &essx_83x6, .drv_name = "adl_es83x6_c1_h02", @@ -527,6 +520,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { /* place boards for each headphone codec: sof driver will complete the * tplg name and machine driver will detect the amp type */ + { + .id = CS42L42_ACPI_HID, + .drv_name = "adl_cs42l42_def", + .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, { .id = NAU8825_ACPI_HID, .drv_name = "adl_nau8825_def", From 6d90e02ae8aa6bedf77110075e7eef3f24f6a8a3 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:07 -0500 Subject: [PATCH 127/603] ASoC: Intel: sof_cs42l42: add rpl_cs42l42_def for rpl boards Add the board config rpl_cs42l42_def to cs42l42 machine driver for all rpl boards using default SSP port allocation (headphone codec on SSP0, speaker amplifiers on SSP1, and BT offload on SSP2). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-18-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 8 ++++++++ sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 7 +++++++ 2 files changed, 15 insertions(+) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 8a2833a0e3c52..3fc76184bdb3d 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -270,6 +270,14 @@ static const struct platform_device_id board_ids[] = { SOF_BT_OFFLOAD_PRESENT | SOF_SSP_PORT_BT_OFFLOAD(2)), }, + { + .name = "rpl_cs42l42_def", + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1) | + SOF_NUM_IDISP_HDMI(4) | + SOF_BT_OFFLOAD_PRESENT | + SOF_SSP_PORT_BT_OFFLOAD(2)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index e28a21667780a..34588db6138af 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -404,6 +404,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { /* place boards for each headphone codec: sof driver will complete the * tplg name and machine driver will detect the amp type */ + { + .id = CS42L42_ACPI_HID, + .drv_name = "rpl_cs42l42_def", + .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, { .id = NAU8825_ACPI_HID, .drv_name = "rpl_nau8825_def", From 57ad033ce09d4d0c866ac558fc3c4cf53cfb2599 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Wed, 27 Mar 2024 11:24:08 -0500 Subject: [PATCH 128/603] ASoC: Intel: sof_cs42l42: add mtl_cs42l42_def for mtl boards Add the board config mtl_cs42l42_def to cs42l42 machine driver for all mtl boards using default SSP port allocation (headphone codec on SSP2, speaker amplifiers on SSP0, and BT offload on SSP1). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240327162408.63953-19-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 7 +++++++ sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 7 +++++++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 3fc76184bdb3d..f4fee2ee0d63b 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -278,6 +278,13 @@ static const struct platform_device_id board_ids[] = { SOF_BT_OFFLOAD_PRESENT | SOF_SSP_PORT_BT_OFFLOAD(2)), }, + { + .name = "mtl_cs42l42_def", + .driver_data = (kernel_ulong_t)(SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(0) | + SOF_BT_OFFLOAD_PRESENT | + SOF_SSP_PORT_BT_OFFLOAD(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 7f422848a3ba6..75935b454e5df 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -69,6 +69,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { /* place boards for each headphone codec: sof driver will complete the * tplg name and machine driver will detect the amp type */ + { + .id = CS42L42_ACPI_HID, + .drv_name = "mtl_cs42l42_def", + .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, { .id = NAU8825_ACPI_HID, .drv_name = "mtl_nau8825_def", From 2e4c003b3672085267a358ebf82129b4e08dd64f Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Wed, 27 Mar 2024 15:57:54 +0800 Subject: [PATCH 129/603] ASoC: dt-bindings: Added schema for "nuvoton,nau8325" Add a DT schema for describing nau8325 audio amplifiers. The key features are as follows: - Low SPK_VDD Quiescent Current - Gain Setting with 2-wire interface - Powerful Stereo Class-D Amplifier - Low Output Noise - Low Current Shutdown Mode - Click-and Pop Suppression More resources : https://www.nuvoton.com/products/smart-home-audio/audio-amplifiers/class-d-series/nau8325yg/ Signed-off-by: Seven Lee Reviewed-by: Rob Herring Link: https://msgid.link/r/20240327075755.3410381-2-wtli@nuvoton.com Signed-off-by: Mark Brown --- .../bindings/sound/nuvoton,nau8325.yaml | 80 +++++++++++++++++++ 1 file changed, 80 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml new file mode 100644 index 0000000000000..979be0d336da9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8325.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8325 audio Amplifier + +maintainers: + - Seven Lee + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8325 + + reg: + maxItems: 1 + + nuvoton,vref-impedance-ohms: + description: + The vref impedance to be used in ohms. Middle of voltage enables + Tie-Off selection options. Due to the high impedance of the VREF + pin, it is important to use a low-leakage capacitor. + + enum: [0, 25000, 125000, 2500] + + nuvoton,dac-vref-microvolt: + description: + The DAC vref to be used in voltage. DAC reference voltage setting. Can + be used for minor tuning of the output level. Since the VDDA is range + between 1.62 to 1.98 voltage, the typical value for design is 1.8V. After + the minor tuning, the final microvolt are as the below. + + enum: [1800000, 2700000, 2880000, 3060000] + + nuvoton,alc-enable: + description: + Enable digital automatic level control (ALC) function. + type: boolean + + nuvoton,clock-detection-disable: + description: + When clock detection is enabled, it will detect whether MCLK + and FS are within the range. MCLK range is from 2.048MHz to 24.576MHz. + FS range is from 8kHz to 96kHz. And also needs to detect the ratio + MCLK_SRC/LRCK of 256, 400 or 500, and needs to detect the BCLK + to make sure data is present. There needs to be at least 8 BCLK + cycles per Frame Sync. + type: boolean + + nuvoton,clock-det-data: + description: + Request clock detection to require 2048 non-zero samples before enabling + the audio paths. If set then non-zero samples is required, otherwise it + doesn't matter. + type: boolean + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@21 { + compatible = "nuvoton,nau8325"; + reg = <0x21>; + nuvoton,vref-impedance-ohms = <125000>; + nuvoton,dac-vref-microvolt = <2880000>; + nuvoton,alc-enable; + nuvoton,clock-det-data; + }; + }; From c0a3873b9938bfaa77bd337cad33266a50a6583f Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Wed, 27 Mar 2024 15:57:55 +0800 Subject: [PATCH 130/603] ASoC: nau8325: new driver The driver is for amplifiers NAU8325 of Nuvoton Technology Corporation. The NAU8325 is a stereo high efficiency filter-free Class-D audio amplifier, which is capable of driving a 4ohm load with up to 3W output power. Signed-off-by: Seven Lee Link: https://msgid.link/r/20240327075755.3410381-3-wtli@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8325.c | 893 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++++++++++ 2 files changed, 1284 insertions(+) create mode 100644 sound/soc/codecs/nau8325.c create mode 100644 sound/soc/codecs/nau8325.h diff --git a/sound/soc/codecs/nau8325.c b/sound/soc/codecs/nau8325.c new file mode 100644 index 0000000000000..a00a30364c897 --- /dev/null +++ b/sound/soc/codecs/nau8325.c @@ -0,0 +1,893 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// nau8325.c -- Nuvoton NAU8325 audio codec driver +// +// Copyright 2023 Nuvoton Technology Crop. +// Author: Seven Lee +// David Lin +// + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "nau8325.h" + +/* Range of Master Clock MCLK (Hz) */ +#define MASTER_CLK_MAX 49152000 +#define MASTER_CLK_MIN 2048000 + +/* scaling for MCLK source */ +#define CLK_PROC_BYPASS (-1) + +/* the maximum CLK_DAC */ +#define CLK_DA_AD_MAX 6144000 + +/* from MCLK input */ +#define MCLK_SRC 4 + +static const struct nau8325_src_attr mclk_n1_div[] = { + { 1, 0x0 }, + { 2, 0x1 }, + { 3, 0x2 }, +}; + +/* over sampling rate */ +static const struct nau8325_osr_attr osr_dac_sel[] = { + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 256, 0 }, /* OSR 256, SRC 1 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 0, 0 }, + { 32, 3 }, /* OSR 32, SRC 1/8 */ +}; + +static const struct nau8325_src_attr mclk_n2_div[] = { + { 0, 0x0 }, + { 1, 0x1 }, + { 2, 0x2 }, + { 3, 0x3 }, + { 4, 0x4 }, +}; + +static const struct nau8325_src_attr mclk_n3_mult[] = { + { 0, 0x1 }, + { 1, 0x2 }, + { 2, 0x3 }, + { 3, 0x4 }, +}; + +/* Sample Rate and MCLK_SRC selections */ +static const struct nau8325_srate_attr target_srate_table[] = { + /* { FS, range, max, { MCLK source }} */ + { 48000, 2, true, { 12288000, 19200000, 24000000 } }, + { 16000, 1, false, { 4096000, 6400000, 8000000 } }, + { 8000, 0, false, { 2048000, 3200000, 4000000 }}, + { 44100, 2, true, { 11289600, 17640000, 22050000 }}, + { 64000, 3, false, { 16384000, 25600000, 32000000 } }, + { 96000, 3, true, { 24576000, 38400000, 48000000 } }, + { 12000, 0, true, { 3072000, 4800000, 6000000 } }, + { 24000, 1, true, { 6144000, 9600000, 12000000 } }, + { 32000, 2, false, { 8192000, 12800000, 16000000 } }, +}; + +static const struct reg_default nau8325_reg_defaults[] = { + { NAU8325_R00_HARDWARE_RST, 0x0000 }, + { NAU8325_R01_SOFTWARE_RST, 0x0000 }, + { NAU8325_R03_CLK_CTRL, 0x0000 }, + { NAU8325_R04_ENA_CTRL, 0x0000 }, + { NAU8325_R05_INTERRUPT_CTRL, 0x007f }, + { NAU8325_R09_IRQOUT, 0x0000 }, + { NAU8325_R0A_IO_CTRL, 0x0000 }, + { NAU8325_R0B_PDM_CTRL, 0x0000 }, + { NAU8325_R0C_TDM_CTRL, 0x0000 }, + { NAU8325_R0D_I2S_PCM_CTRL1, 0x000a }, + { NAU8325_R0E_I2S_PCM_CTRL2, 0x0000 }, + { NAU8325_R0F_L_TIME_SLOT, 0x0000 }, + { NAU8325_R10_R_TIME_SLOT, 0x0000 }, + { NAU8325_R11_HPF_CTRL, 0x0000 }, + { NAU8325_R12_MUTE_CTRL, 0x0000 }, + { NAU8325_R13_DAC_VOLUME, 0xf3f3 }, + { NAU8325_R29_DAC_CTRL1, 0x0081 }, + { NAU8325_R2A_DAC_CTRL2, 0x0000 }, + { NAU8325_R2C_ALC_CTRL1, 0x000e }, + { NAU8325_R2D_ALC_CTRL2, 0x8400 }, + { NAU8325_R2E_ALC_CTRL3, 0x0000 }, + { NAU8325_R2F_ALC_CTRL4, 0x003f }, + { NAU8325_R40_CLK_DET_CTRL, 0xa801 }, + { NAU8325_R50_MIXER_CTRL, 0x0000 }, + { NAU8325_R55_MISC_CTRL, 0x0000 }, + { NAU8325_R60_BIAS_ADJ, 0x0000 }, + { NAU8325_R61_ANALOG_CONTROL_1, 0x0000 }, + { NAU8325_R62_ANALOG_CONTROL_2, 0x0000 }, + { NAU8325_R63_ANALOG_CONTROL_3, 0x0000 }, + { NAU8325_R64_ANALOG_CONTROL_4, 0x0000 }, + { NAU8325_R65_ANALOG_CONTROL_5, 0x0000 }, + { NAU8325_R66_ANALOG_CONTROL_6, 0x0000 }, + { NAU8325_R69_CLIP_CTRL, 0x0000 }, + { NAU8325_R73_RDAC, 0x0008 }, +}; + +static bool nau8325_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8325_R02_DEVICE_ID ... NAU8325_R06_INT_CLR_STATUS: + case NAU8325_R09_IRQOUT ... NAU8325_R13_DAC_VOLUME: + case NAU8325_R1D_DEBUG_READ1: + case NAU8325_R1F_DEBUG_READ2: + case NAU8325_R22_DEBUG_READ3: + case NAU8325_R29_DAC_CTRL1 ... NAU8325_R2A_DAC_CTRL2: + case NAU8325_R2C_ALC_CTRL1 ... NAU8325_R2F_ALC_CTRL4: + case NAU8325_R40_CLK_DET_CTRL: + case NAU8325_R49_TEST_STATUS ... NAU8325_R4A_ANALOG_READ: + case NAU8325_R50_MIXER_CTRL: + case NAU8325_R55_MISC_CTRL: + case NAU8325_R60_BIAS_ADJ ... NAU8325_R66_ANALOG_CONTROL_6: + case NAU8325_R69_CLIP_CTRL: + case NAU8325_R73_RDAC: + return true; + default: + return false; + } +} + +static bool nau8325_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8325_R00_HARDWARE_RST: + case NAU8325_R03_CLK_CTRL ... NAU8325_R06_INT_CLR_STATUS: + case NAU8325_R09_IRQOUT ... NAU8325_R13_DAC_VOLUME: + case NAU8325_R29_DAC_CTRL1 ... NAU8325_R2A_DAC_CTRL2: + case NAU8325_R2C_ALC_CTRL1 ... NAU8325_R2F_ALC_CTRL4: + case NAU8325_R40_CLK_DET_CTRL: + case NAU8325_R50_MIXER_CTRL: + case NAU8325_R55_MISC_CTRL: + case NAU8325_R60_BIAS_ADJ ... NAU8325_R66_ANALOG_CONTROL_6: + case NAU8325_R69_CLIP_CTRL: + case NAU8325_R73_RDAC: + return true; + default: + return false; + } +} + +static bool nau8325_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8325_R00_HARDWARE_RST ... NAU8325_R02_DEVICE_ID: + case NAU8325_R06_INT_CLR_STATUS: + case NAU8325_R1D_DEBUG_READ1: + case NAU8325_R1F_DEBUG_READ2: + case NAU8325_R22_DEBUG_READ3: + case NAU8325_R4A_ANALOG_READ: + return true; + default: + return false; + } +} + +static const char * const nau8325_dac_oversampl[] = { + "64", "256", "128", "", "32" }; + +static const struct soc_enum nau8325_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8325_R29_DAC_CTRL1, NAU8325_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8325_dac_oversampl), + nau8325_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(dac_vol_tlv, -8000, 600); + +static const struct snd_kcontrol_new nau8325_snd_controls[] = { + SOC_ENUM("DAC Oversampling Rate", nau8325_dac_oversampl_enum), + SOC_DOUBLE_TLV("Speaker Volume", NAU8325_R13_DAC_VOLUME, + NAU8325_DAC_VOLUME_L_SFT, NAU8325_DAC_VOLUME_R_SFT, + NAU8325_DAC_VOLUME_R_EN, 0, dac_vol_tlv), + SOC_SINGLE("ALC Max Gain", NAU8325_R2C_ALC_CTRL1, + NAU8325_ALC_MAXGAIN_SFT, NAU8325_ALC_MAXGAIN_MAX, 0), + SOC_SINGLE("ALC Min Gain", NAU8325_R2C_ALC_CTRL1, + NAU8325_ALC_MINGAIN_SFT, NAU8325_ALC_MINGAIN_MAX, 0), + SOC_SINGLE("ALC Decay Timer", NAU8325_R2D_ALC_CTRL2, + NAU8325_ALC_DCY_SFT, NAU8325_ALC_DCY_MAX, 0), + SOC_SINGLE("ALC Attack Timer", NAU8325_R2D_ALC_CTRL2, + NAU8325_ALC_ATK_SFT, NAU8325_ALC_ATK_MAX, 0), + SOC_SINGLE("ALC Hold Time", NAU8325_R2D_ALC_CTRL2, + NAU8325_ALC_HLD_SFT, NAU8325_ALC_HLD_MAX, 0), + SOC_SINGLE("ALC Target Level", NAU8325_R2D_ALC_CTRL2, + NAU8325_ALC_LVL_SFT, NAU8325_ALC_LVL_MAX, 0), + SOC_SINGLE("ALC Enable Switch", NAU8325_R2E_ALC_CTRL3, + NAU8325_ALC_EN_SFT, 1, 0), +}; + +static int nau8325_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8325->regmap, NAU8325_R12_MUTE_CTRL, + NAU8325_SOFT_MUTE, 0); + msleep(30); + break; + case SND_SOC_DAPM_PRE_PMD: + /* Soft mute the output to prevent the pop noise. */ + regmap_update_bits(nau8325->regmap, NAU8325_R12_MUTE_CTRL, + NAU8325_SOFT_MUTE, NAU8325_SOFT_MUTE); + msleep(30); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8325_powerup_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component); + + if (nau8325->clock_detection) + return 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8325->regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_PWRUP_DFT, NAU8325_PWRUP_DFT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8325->regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_PWRUP_DFT, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget nau8325_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("Power Up", SND_SOC_NOPM, 0, 0, + nau8325_powerup_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("DACL", NULL, NAU8325_R04_ENA_CTRL, + NAU8325_DAC_LEFT_CH_EN_SFT, 0, nau8325_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DACR", NULL, NAU8325_R04_ENA_CTRL, + NAU8325_DAC_RIGHT_CH_EN_SFT, 0, nau8325_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), +}; + +static const struct snd_soc_dapm_route nau8325_dapm_routes[] = { + { "DACL", NULL, "Power Up" }, + { "DACR", NULL, "Power Up" }, + + { "DACL", NULL, "AIFRX" }, + { "DACR", NULL, "AIFRX" }, + { "SPKL", NULL, "DACL" }, + { "SPKR", NULL, "DACR" }, +}; + +static int nau8325_srate_clk_apply(struct nau8325 *nau8325, + const struct nau8325_srate_attr *srate_table, + int n1_sel, int mclk_mult_sel, int n2_sel) +{ + if (!srate_table || n2_sel < 0 || n2_sel >= ARRAY_SIZE(mclk_n2_div) || + n1_sel < 0 || n1_sel >= ARRAY_SIZE(mclk_n1_div)) { + dev_dbg(nau8325->dev, "The CLK isn't supported."); + return -EINVAL; + } + + regmap_update_bits(nau8325->regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_REG_SRATE_MASK | NAU8325_REG_DIV_MAX, + (srate_table->range << NAU8325_REG_SRATE_SFT) | + (srate_table->max ? NAU8325_REG_DIV_MAX : 0)); + regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL, + NAU8325_MCLK_SRC_MASK, mclk_n2_div[n2_sel].val); + regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL, + NAU8325_CLK_MUL_SRC_MASK, + mclk_n1_div[n1_sel].val << NAU8325_CLK_MUL_SRC_SFT); + + if (mclk_mult_sel != CLK_PROC_BYPASS) { + regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL, + NAU8325_MCLK_SEL_MASK, + mclk_n3_mult[mclk_mult_sel].val << + NAU8325_MCLK_SEL_SFT); + } else { + regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL, + NAU8325_MCLK_SEL_MASK, 0); + } + + switch (mclk_mult_sel) { + case 2: + regmap_update_bits(nau8325->regmap, NAU8325_R65_ANALOG_CONTROL_5, + NAU8325_MCLK4XEN_EN, NAU8325_MCLK4XEN_EN); + break; + case 3: + regmap_update_bits(nau8325->regmap, NAU8325_R65_ANALOG_CONTROL_5, + NAU8325_MCLK4XEN_EN | NAU8325_MCLK8XEN_EN, + NAU8325_MCLK4XEN_EN | NAU8325_MCLK8XEN_EN); + break; + default: + regmap_update_bits(nau8325->regmap, NAU8325_R65_ANALOG_CONTROL_5, + NAU8325_MCLK4XEN_EN | NAU8325_MCLK8XEN_EN, 0); + break; + } + + return 0; +} + +static int nau8325_clksrc_n2(struct nau8325 *nau8325, + const struct nau8325_srate_attr *srate_table, + int mclk, int *n2_sel) +{ + int i, mclk_src, ratio; + + ratio = NAU8325_MCLK_FS_RATIO_NUM; + for (i = 0; i < ARRAY_SIZE(mclk_n2_div); i++) { + mclk_src = mclk >> mclk_n2_div[i].param; + if (srate_table->mclk_src[NAU8325_MCLK_FS_RATIO_256] == mclk_src) { + ratio = NAU8325_MCLK_FS_RATIO_256; + break; + } else if (srate_table->mclk_src[NAU8325_MCLK_FS_RATIO_400] == mclk_src) { + ratio = NAU8325_MCLK_FS_RATIO_400; + break; + } else if (srate_table->mclk_src[NAU8325_MCLK_FS_RATIO_500] == mclk_src) { + ratio = NAU8325_MCLK_FS_RATIO_500; + break; + } + } + if (ratio != NAU8325_MCLK_FS_RATIO_NUM) + *n2_sel = i; + + return ratio; +} + +static const struct nau8325_srate_attr *target_srate_attribute(int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(target_srate_table); i++) + if (target_srate_table[i].fs == srate) + break; + + if (i == ARRAY_SIZE(target_srate_table)) + goto proc_err; + + return &target_srate_table[i]; + +proc_err: + return NULL; +} + +static int nau8325_clksrc_choose(struct nau8325 *nau8325, + const struct nau8325_srate_attr **srate_table, + int *n1_sel, int *mult_sel, int *n2_sel) +{ + int i, j, mclk, mclk_max, ratio, ratio_sel, n2_max; + + if (!nau8325->mclk || !nau8325->fs) + goto proc_err; + + /* select sampling rate and MCLK_SRC */ + *srate_table = target_srate_attribute(nau8325->fs); + if (!*srate_table) + goto proc_err; + + /* First check clock from MCLK directly, decide N2 for MCLK_SRC. + * If not good, consider 1/N1 and Multiplier. + */ + ratio = nau8325_clksrc_n2(nau8325, *srate_table, nau8325->mclk, n2_sel); + if (ratio != NAU8325_MCLK_FS_RATIO_NUM) { + *n1_sel = 0; + *mult_sel = CLK_PROC_BYPASS; + *n2_sel = MCLK_SRC; + goto proc_done; + } + + /* Get MCLK_SRC through 1/N, Multiplier, and then 1/N2. */ + mclk_max = 0; + for (i = 0; i < ARRAY_SIZE(mclk_n1_div); i++) { + for (j = 0; j < ARRAY_SIZE(mclk_n3_mult); j++) { + mclk = nau8325->mclk << mclk_n3_mult[j].param; + mclk = mclk / mclk_n1_div[i].param; + ratio = nau8325_clksrc_n2(nau8325, + *srate_table, mclk, n2_sel); + if (ratio != NAU8325_MCLK_FS_RATIO_NUM && + (mclk_max < mclk || i > *n1_sel)) { + mclk_max = mclk; + n2_max = *n2_sel; + *n1_sel = i; + *mult_sel = j; + ratio_sel = ratio; + goto proc_done; + } + } + } + if (mclk_max) { + *n2_sel = n2_max; + ratio = ratio_sel; + goto proc_done; + } + +proc_err: + dev_dbg(nau8325->dev, "The MCLK %d is invalid. It can't get MCLK_SRC of 256/400/500 FS (%d)", + nau8325->mclk, nau8325->fs); + return -EINVAL; +proc_done: + dev_dbg(nau8325->dev, "nau8325->fs=%d,range=0x%x, %s, (n1,mu,n2,dmu):(%d,%d,%d), MCLK_SRC=%uHz (%d)", + nau8325->fs, (*srate_table)->range, + (*srate_table)->max ? "MAX" : "MIN", + *n1_sel == CLK_PROC_BYPASS ? + CLK_PROC_BYPASS : mclk_n1_div[*n1_sel].param, + *mult_sel == CLK_PROC_BYPASS ? + CLK_PROC_BYPASS : 1 << mclk_n3_mult[*mult_sel].param, + 1 << mclk_n2_div[*n2_sel].param, + (*srate_table)->mclk_src[ratio], + (*srate_table)->mclk_src[ratio] / nau8325->fs); + + return 0; +} + +static int nau8325_clock_config(struct nau8325 *nau8325) +{ + const struct nau8325_srate_attr *srate_table; + int ret, n1_sel, mult_sel, n2_sel; + + ret = nau8325_clksrc_choose(nau8325, &srate_table, + &n1_sel, &mult_sel, &n2_sel); + if (ret) + goto err; + + ret = nau8325_srate_clk_apply(nau8325, srate_table, + n1_sel, mult_sel, n2_sel); + if (ret) + goto err; + + return 0; +err: + return ret; +} + +static const struct nau8325_osr_attr *nau8325_get_osr(struct nau8325 *nau8325) +{ + unsigned int osr; + + regmap_read(nau8325->regmap, NAU8325_R29_DAC_CTRL1, &osr); + osr &= NAU8325_DAC_OVERSAMPLE_MASK; + if (osr >= ARRAY_SIZE(osr_dac_sel)) + return NULL; + + return &osr_dac_sel[osr]; +} + +static int nau8325_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component); + const struct nau8325_osr_attr *osr; + + osr = nau8325_get_osr(nau8325); + if (!osr || !osr->osr) + return -EINVAL; + + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 0, CLK_DA_AD_MAX / osr->osr); +} + +static int nau8325_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component); + unsigned int val_len = 0; + const struct nau8325_osr_attr *osr; + int ret; + + nau8325->fs = params_rate(params); + osr = nau8325_get_osr(nau8325); + if (!osr || !osr->osr || nau8325->fs * osr->osr > CLK_DA_AD_MAX) { + ret = -EINVAL; + goto err; + } + regmap_update_bits(nau8325->regmap, NAU8325_R03_CLK_CTRL, + NAU8325_CLK_DAC_SRC_MASK, + osr->clk_src << NAU8325_CLK_DAC_SRC_SFT); + + ret = nau8325_clock_config(nau8325); + if (ret) + goto err; + + switch (params_width(params)) { + case 16: + val_len |= NAU8325_I2S_DL_16; + break; + case 20: + val_len |= NAU8325_I2S_DL_20; + break; + case 24: + val_len |= NAU8325_I2S_DL_24; + break; + case 32: + val_len |= NAU8325_I2S_DL_32; + break; + default: + ret = -EINVAL; + goto err; + } + + regmap_update_bits(nau8325->regmap, NAU8325_R0D_I2S_PCM_CTRL1, + NAU8325_I2S_DL_MASK, val_len); + + return 0; + +err: + return ret; +} + +static int nau8325_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component); + unsigned int ctrl1_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8325_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8325_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8325_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8325_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8325_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8325_I2S_DF_PCM_AB; + ctrl1_val |= NAU8325_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8325->regmap, NAU8325_R0D_I2S_PCM_CTRL1, + NAU8325_I2S_DF_MASK | NAU8325_I2S_BP_MASK | + NAU8325_I2S_PCMB_EN, ctrl1_val); + + return 0; +} + +static int nau8325_set_sysclk(struct snd_soc_component *component, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8325 *nau8325 = snd_soc_component_get_drvdata(component); + + if (freq < MASTER_CLK_MIN || freq > MASTER_CLK_MAX) { + dev_dbg(nau8325->dev, "MCLK exceeds the range, MCLK:%d", freq); + return -EINVAL; + } + + nau8325->mclk = freq; + dev_dbg(nau8325->dev, "MCLK %dHz", nau8325->mclk); + + return 0; +} + +static const struct snd_soc_component_driver nau8325_component_driver = { + .set_sysclk = nau8325_set_sysclk, + .suspend_bias_off = true, + .controls = nau8325_snd_controls, + .num_controls = ARRAY_SIZE(nau8325_snd_controls), + .dapm_widgets = nau8325_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8325_dapm_widgets), + .dapm_routes = nau8325_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8325_dapm_routes), +}; + +static const struct snd_soc_dai_ops nau8325_dai_ops = { + .startup = nau8325_dai_startup, + .hw_params = nau8325_hw_params, + .set_fmt = nau8325_set_fmt, +}; + +#define NAU8325_RATES SNDRV_PCM_RATE_8000_96000 +#define NAU8325_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE) + +static struct snd_soc_dai_driver nau8325_dai = { + .name = NAU8325_CODEC_DAI, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8325_RATES, + .formats = NAU8325_FORMATS, + }, + .ops = &nau8325_dai_ops, +}; + +static const struct regmap_config nau8325_regmap_config = { + .reg_bits = NAU8325_REG_ADDR_LEN, + .val_bits = NAU8325_REG_DATA_LEN, + + .max_register = NAU8325_REG_MAX, + .readable_reg = nau8325_readable_reg, + .writeable_reg = nau8325_writeable_reg, + .volatile_reg = nau8325_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8325_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8325_reg_defaults), +}; + +static void nau8325_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8325_R00_HARDWARE_RST, 0x0001); + regmap_write(regmap, NAU8325_R00_HARDWARE_RST, 0x0000); +} + +static void nau8325_init_regs(struct nau8325 *nau8325) +{ + struct regmap *regmap = nau8325->regmap; + struct device *dev = nau8325->dev; + + /* set ALC parameters */ + regmap_update_bits(regmap, NAU8325_R2C_ALC_CTRL1, + NAU8325_ALC_MAXGAIN_MASK, + 0x7 << NAU8325_ALC_MAXGAIN_SFT); + regmap_update_bits(regmap, NAU8325_R2D_ALC_CTRL2, + NAU8325_ALC_DCY_MASK | NAU8325_ALC_ATK_MASK | + NAU8325_ALC_HLD_MASK, (0x5 << NAU8325_ALC_DCY_SFT) | + (0x3 << NAU8325_ALC_ATK_SFT) | + (0x5 << NAU8325_ALC_HLD_SFT)); + /* Enable ALC to avoid signal distortion when battery low. */ + if (nau8325->alc_enable) + regmap_update_bits(regmap, NAU8325_R2E_ALC_CTRL3, + NAU8325_ALC_EN, NAU8325_ALC_EN); + if (nau8325->clock_detection) + regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_CLKPWRUP_DIS | + NAU8325_PWRUP_DFT, 0); + else + regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_CLKPWRUP_DIS | NAU8325_PWRUP_DFT, + NAU8325_CLKPWRUP_DIS); + if (nau8325->clock_det_data) + regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_APWRUP_EN, NAU8325_APWRUP_EN); + else + regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_APWRUP_EN, 0); + + /* DAC Reference Voltage Setting */ + switch (nau8325->dac_vref_microvolt) { + case 1800000: + regmap_update_bits(regmap, NAU8325_R73_RDAC, + NAU8325_DACVREFSEL_MASK, 0 << NAU8325_DACVREFSEL_SFT); + break; + case 2700000: + regmap_update_bits(regmap, NAU8325_R73_RDAC, + NAU8325_DACVREFSEL_MASK, 1 << NAU8325_DACVREFSEL_SFT); + break; + case 2880000: + regmap_update_bits(regmap, NAU8325_R73_RDAC, + NAU8325_DACVREFSEL_MASK, 2 << NAU8325_DACVREFSEL_SFT); + break; + case 3060000: + regmap_update_bits(regmap, NAU8325_R73_RDAC, + NAU8325_DACVREFSEL_MASK, 3 << NAU8325_DACVREFSEL_SFT); + break; + default: + dev_dbg(dev, "Invalid dac-vref-microvolt %d", nau8325->dac_vref_microvolt); + + } + + /* DAC Reference Voltage Decoupling Capacitors. */ + regmap_update_bits(regmap, NAU8325_R63_ANALOG_CONTROL_3, + NAU8325_CLASSD_COARSE_GAIN_MASK, 0x4); + /* Auto-Att Min Gain 0dB, Class-D N Driver Slew Rate -25%. */ + regmap_update_bits(regmap, NAU8325_R64_ANALOG_CONTROL_4, + NAU8325_CLASSD_SLEWN_MASK, 0x7); + + /* VMID Tieoff (VMID Resistor Selection) */ + switch (nau8325->vref_impedance_ohms) { + case 0: + regmap_update_bits(regmap, NAU8325_R60_BIAS_ADJ, + NAU8325_BIAS_VMID_SEL_MASK, 0 << NAU8325_BIAS_VMID_SEL_SFT); + break; + case 25000: + regmap_update_bits(regmap, NAU8325_R60_BIAS_ADJ, + NAU8325_BIAS_VMID_SEL_MASK, 1 << NAU8325_BIAS_VMID_SEL_SFT); + break; + case 125000: + regmap_update_bits(regmap, NAU8325_R60_BIAS_ADJ, + NAU8325_BIAS_VMID_SEL_MASK, 2 << NAU8325_BIAS_VMID_SEL_SFT); + break; + case 2500: + regmap_update_bits(regmap, NAU8325_R60_BIAS_ADJ, + NAU8325_BIAS_VMID_SEL_MASK, 3 << NAU8325_BIAS_VMID_SEL_SFT); + break; + default: + dev_dbg(dev, "Invalid vref-impedance-ohms %d", nau8325->vref_impedance_ohms); + } + + + /* enable VMID, BIAS, DAC, DCA CLOCK, Voltage/Current Amps + */ + regmap_update_bits(regmap, NAU8325_R61_ANALOG_CONTROL_1, + NAU8325_DACEN_MASK | NAU8325_DACCLKEN_MASK | + NAU8325_DACEN_R_MASK | NAU8325_DACCLKEN_R_MASK | + NAU8325_CLASSDEN_MASK | NAU8325_VMDFSTENB_MASK | + NAU8325_BIASEN_MASK | NAU8325_VMIDEN_MASK, + (0x1 << NAU8325_DACEN_SFT) | + (0x1 << NAU8325_DACCLKEN_SFT) | + (0x1 << NAU8325_DACEN_R_SFT) | + (0x1 << NAU8325_DACCLKEN_R_SFT) | + (0x1 << NAU8325_CLASSDEN_SFT) | + (0x1 << NAU8325_VMDFSTENB_SFT) | + (0x1 << NAU8325_BIASEN_SFT) | 0x3); + + /* Enable ALC to avoid signal distortion when battery low. */ + if (nau8325->alc_enable) + regmap_update_bits(regmap, NAU8325_R2E_ALC_CTRL3, + NAU8325_ALC_EN, NAU8325_ALC_EN); + if (nau8325->clock_det_data) + regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_APWRUP_EN, NAU8325_APWRUP_EN); + else + regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_APWRUP_EN, 0); + if (nau8325->clock_detection) + regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_CLKPWRUP_DIS | + NAU8325_PWRUP_DFT, 0); + else + regmap_update_bits(regmap, NAU8325_R40_CLK_DET_CTRL, + NAU8325_CLKPWRUP_DIS | NAU8325_PWRUP_DFT, + NAU8325_CLKPWRUP_DIS); + regmap_update_bits(regmap, NAU8325_R29_DAC_CTRL1, + NAU8325_DAC_OVERSAMPLE_MASK, + NAU8325_DAC_OVERSAMPLE_128); +} + +static void nau8325_print_device_properties(struct nau8325 *nau8325) +{ + struct device *dev = nau8325->dev; + + dev_dbg(dev, "vref-impedance-ohms: %d", nau8325->vref_impedance_ohms); + dev_dbg(dev, "dac-vref-microvolt: %d", nau8325->dac_vref_microvolt); + dev_dbg(dev, "alc-enable: %d", nau8325->alc_enable); + dev_dbg(dev, "clock-det-data: %d", nau8325->clock_det_data); + dev_dbg(dev, "clock-detection-disable: %d", nau8325->clock_detection); +} + +static int nau8325_read_device_properties(struct device *dev, + struct nau8325 *nau8325) +{ + int ret; + + nau8325->alc_enable = + device_property_read_bool(dev, "nuvoton,alc-enable"); + nau8325->clock_det_data = + device_property_read_bool(dev, "nuvoton,clock-det-data"); + nau8325->clock_detection = + !device_property_read_bool(dev, "nuvoton,clock-detection-disable"); + + ret = device_property_read_u32(dev, "nuvoton,vref-impedance-ohms", + &nau8325->vref_impedance_ohms); + if (ret) + nau8325->vref_impedance_ohms = 125000; + ret = device_property_read_u32(dev, "nuvoton,dac-vref-microvolt", + &nau8325->dac_vref_microvolt); + if (ret) + nau8325->dac_vref_microvolt = 2880000; + + return 0; +} + +static int nau8325_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8325 *nau8325 = dev_get_platdata(dev); + int ret, value; + + if (!nau8325) { + nau8325 = devm_kzalloc(dev, sizeof(*nau8325), GFP_KERNEL); + if (!nau8325) { + ret = -ENOMEM; + goto err; + } + ret = nau8325_read_device_properties(dev, nau8325); + if (ret) + goto err; + } + i2c_set_clientdata(i2c, nau8325); + + nau8325->regmap = devm_regmap_init_i2c(i2c, &nau8325_regmap_config); + if (IS_ERR(nau8325->regmap)) { + ret = PTR_ERR(nau8325->regmap); + goto err; + } + nau8325->dev = dev; + nau8325_print_device_properties(nau8325); + + nau8325_reset_chip(nau8325->regmap); + ret = regmap_read(nau8325->regmap, NAU8325_R02_DEVICE_ID, &value); + if (ret) { + dev_dbg(dev, "Failed to read device id (%d)", ret); + goto err; + } + nau8325_init_regs(nau8325); + + ret = devm_snd_soc_register_component(dev, &nau8325_component_driver, + &nau8325_dai, 1); +err: + return ret; +} + +static const struct i2c_device_id nau8325_i2c_ids[] = { + { "nau8325", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8325_i2c_ids); + +#ifdef CONFIG_OF +static const struct of_device_id nau8325_of_ids[] = { + { .compatible = "nuvoton,nau8325", }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8325_of_ids); +#endif + +static struct i2c_driver nau8325_i2c_driver = { + .driver = { + .name = "nau8325", + .of_match_table = of_match_ptr(nau8325_of_ids), + }, + .probe = nau8325_i2c_probe, + .id_table = nau8325_i2c_ids, +}; +module_i2c_driver(nau8325_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8325 driver"); +MODULE_AUTHOR("Seven Lee "); +MODULE_AUTHOR("David Lin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8325.h b/sound/soc/codecs/nau8325.h new file mode 100644 index 0000000000000..0d173b66a4d4e --- /dev/null +++ b/sound/soc/codecs/nau8325.h @@ -0,0 +1,391 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * nau8325.h -- Nuvoton NAU8325 audio codec driver + * + * Copyright 2023 Nuvoton Technology Crop. + * Author: Seven Lee + * David Lin + */ + +#ifndef __NAU8325_H__ +#define __NAU8325_H__ + +#define NAU8325_R00_HARDWARE_RST 0x00 +#define NAU8325_R01_SOFTWARE_RST 0x01 +#define NAU8325_R02_DEVICE_ID 0x02 +#define NAU8325_R03_CLK_CTRL 0x03 +#define NAU8325_R04_ENA_CTRL 0x04 +#define NAU8325_R05_INTERRUPT_CTRL 0x05 +#define NAU8325_R06_INT_CLR_STATUS 0x06 +#define NAU8325_R09_IRQOUT 0x09 +#define NAU8325_R0A_IO_CTRL 0x0a +#define NAU8325_R0B_PDM_CTRL 0x0b +#define NAU8325_R0C_TDM_CTRL 0x0c +#define NAU8325_R0D_I2S_PCM_CTRL1 0x0d +#define NAU8325_R0E_I2S_PCM_CTRL2 0x0e +#define NAU8325_R0F_L_TIME_SLOT 0x0f +#define NAU8325_R10_R_TIME_SLOT 0x10 +#define NAU8325_R11_HPF_CTRL 0x11 +#define NAU8325_R12_MUTE_CTRL 0x12 +#define NAU8325_R13_DAC_VOLUME 0x13 +#define NAU8325_R1D_DEBUG_READ1 0x1d +#define NAU8325_R1F_DEBUG_READ2 0x1f +#define NAU8325_R22_DEBUG_READ3 0x22 +#define NAU8325_R29_DAC_CTRL1 0x29 +#define NAU8325_R2A_DAC_CTRL2 0x2a +#define NAU8325_R2C_ALC_CTRL1 0x2c +#define NAU8325_R2D_ALC_CTRL2 0x2d +#define NAU8325_R2E_ALC_CTRL3 0x2e +#define NAU8325_R2F_ALC_CTRL4 0x2f +#define NAU8325_R40_CLK_DET_CTRL 0x40 +#define NAU8325_R49_TEST_STATUS 0x49 +#define NAU8325_R4A_ANALOG_READ 0x4a +#define NAU8325_R50_MIXER_CTRL 0x50 +#define NAU8325_R55_MISC_CTRL 0x55 +#define NAU8325_R60_BIAS_ADJ 0x60 +#define NAU8325_R61_ANALOG_CONTROL_1 0x61 +#define NAU8325_R62_ANALOG_CONTROL_2 0x62 +#define NAU8325_R63_ANALOG_CONTROL_3 0x63 +#define NAU8325_R64_ANALOG_CONTROL_4 0x64 +#define NAU8325_R65_ANALOG_CONTROL_5 0x65 +#define NAU8325_R66_ANALOG_CONTROL_6 0x66 +#define NAU8325_R69_CLIP_CTRL 0x69 +#define NAU8325_R73_RDAC 0x73 +#define NAU8325_REG_MAX NAU8325_R73_RDAC + +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8325_REG_ADDR_LEN 16 +#define NAU8325_REG_DATA_LEN 16 + +/* CLK_CTRL (0x03) */ +#define NAU8325_CLK_DAC_SRC_SFT 12 +#define NAU8325_CLK_DAC_SRC_MASK (0x3 << NAU8325_CLK_DAC_SRC_SFT) +#define NAU8325_CLK_MUL_SRC_SFT 6 +#define NAU8325_CLK_MUL_SRC_MASK (0x3 << NAU8325_CLK_MUL_SRC_SFT) +#define NAU8325_MCLK_SEL_SFT 3 +#define NAU8325_MCLK_SEL_MASK (0x7 << NAU8325_MCLK_SEL_SFT) +#define NAU8325_MCLK_SRC_MASK 0x7 + +/* ENA_CTRL (0x04) */ +#define NAU8325_DAC_LEFT_CH_EN_SFT 3 +#define NAU8325_DAC_LEFT_CH_EN (0x1 << NAU8325_DAC_LEFT_CH_EN_SFT) +#define NAU8325_DAC_RIGHT_CH_EN_SFT 2 +#define NAU8325_DAC_RIGHT_CH_EN (0x1 << NAU8325_DAC_RIGHT_CH_EN_SFT) + +/* INTERRUPT_CTRL (0x05) */ +#define NAU8325_ARP_DWN_INT_SFT 12 +#define NAU8325_ARP_DWN_INT_MASK (0x1 << NAU8325_ARP_DWN_INT_SFT) +#define NAU8325_CLIP_INT_SFT 11 +#define NAU8325_CLIP_INT_MASK (0x1 << NAU8325_CLIP_INT_SFT) +#define NAU8325_LVD_INT_SFT 10 +#define NAU8325_LVD_INT_MASK (0x1 << NAU8325_LVD_INT_SFT) +#define NAU8325_PWR_INT_DIS_SFT 8 +#define NAU8325_PWR_INT_DIS (0x1 << NAU8325_PWR_INT_DIS_SFT) +#define NAU8325_OCP_OTP_SHTDWN_INT_SFT 4 +#define NAU8325_OCP_OTP_SHTDWN_INT_MASK (0x1 << NAU8325_OCP_OTP_SHTDWN_INT_SFT) +#define NAU8325_CLIP_INT_DIS_SFT 3 +#define NAU8325_CLIP_INT_DIS (0x1 << NAU8325_CLIP_INT_DIS_SFT) +#define NAU8325_LVD_INT_DIS_SFT 2 +#define NAU8325_LVD_INT_DIS (0x1 << NAU8325_LVD_INT_DIS_SFT) +#define NAU8325_PWR_INT_MASK 0x1 + +/* INT_CLR_STATUS (0x06) */ +#define NAU8325_INT_STATUS_MASK 0x7f + +/* IRQOUT (0x9) */ +#define NAU8325_IRQOUT_SEL_SEF 12 +#define NAU8325_IRQOUT_SEL_MASK (0xf << NAU8325_IRQOUT_SEL_SEF) +#define NAU8325_DEM_DITH_SFT 7 +#define NAU8325_DEM_DITH_EN (0x1 << NAU8325_DEM_DITH_SFT) +#define NAU8325_GAINZI3_SFT 5 +#define NAU8325_GAINZI3_MASK (0x1 << NAU8325_GAINZI3_SFT) +#define NAU8325_GAINZI2_MASK 0x1f + +/* IO_CTRL (0x0a) */ +#define NAU8325_IRQ_PL_SFT 15 +#define NAU8325_IRQ_PL_ACT_HIGH (0x1 << NAU8325_IRQ_PL_SFT) +#define NAU8325_IRQ_PS_SFT 14 +#define NAU8325_IRQ_PS_UP (0x1 << NAU8325_IRQ_PS_SFT) +#define NAU8325_IRQ_PE_SFT 13 +#define NAU8325_IRQ_PE_EN (0x1 << NAU8325_IRQ_PE_SFT) +#define NAU8325_IRQ_DS_SFT 12 +#define NAU8325_IRQ_DS_HIGH (0x1 << NAU8325_IRQ_DS_SFT) +#define NAU8325_IRQ_OUTPUT_SFT 11 +#define NAU8325_IRQ_OUTPUT_EN (0x1 << NAU8325_IRQ_OUTPUT_SFT) +#define NAU8325_IRQ_PIN_DEBUG_SFT 10 +#define NAU8325_IRQ_PIN_DEBUG_EN (0x1 << NAU8325_IRQ_PIN_DEBUG_SFT) + +/* PDM_CTRL (0x0b) */ +#define NAU8325_PDM_LCH_EDGE_SFT 1 +#define NAU8325_PDM_LCH_EDGE__MASK (0x1 << NAU8325_PDM_LCH_EDGE_SFT) +#define NAU8325_PDM_MODE_EN 0x1 + +/* TDM_CTRL (0x0c) */ +#define NAU8325_TDM_SFT 15 +#define NAU8325_TDM_EN (0x1 << NAU8325_TDM_SFT) +#define NAU8325_PCM_OFFSET_CTRL_SFT 14 +#define NAU8325_PCM_OFFSET_CTRL_EN (0x1 << NAU8325_PCM_OFFSET_CTRL_SFT) +#define NAU8325_DAC_LEFT_SFT 6 +#define NAU8325_NAU8325_DAC_LEFT_MASK (0x7 << NAU8325_DAC_LEFT_SFT) +#define NAU8325_DAC_RIGHT_SFT 3 +#define NAU8325_DAC_RIGHT_MASK (0x7 << NAU8325_DAC_RIGHT_SFT) + +/* I2S_PCM_CTRL1 (0x0d) */ +#define NAU8325_DACCM_CTL_SFT 14 +#define NAU8325_DACCM_CTL_MASK (0x3 << NAU8325_DACCM_CTL_SFT) +#define NAU8325_CMB8_0_SFT 10 +#define NAU8325_CMB8_0_MASK (0x1 << NAU8325_CMB8_0_SFT) +#define NAU8325_UA_OFFSET_SFT 9 +#define NAU8325_UA_OFFSET_MASK (0x1 << NAU8325_UA_OFFSET_SFT) +#define NAU8325_I2S_BP_SFT 7 +#define NAU8325_I2S_BP_MASK (0x1 << NAU8325_I2S_BP_SFT) +#define NAU8325_I2S_BP_INV (0x1 << NAU8325_I2S_BP_SFT) +#define NAU8325_I2S_PCMB_SFT 6 +#define NAU8325_I2S_PCMB_EN (0x1 << NAU8325_I2S_PCMB_SFT) +#define NAU8325_I2S_DACPSHS0_SFT 5 +#define NAU8325_I2S_DACPSHS0_MASK (0x1 << NAU8325_I2S_DACPSHS0_SFT) +#define NAU8325_I2S_DL_SFT 2 +#define NAU8325_I2S_DL_MASK (0x3 << NAU8325_I2S_DL_SFT) +#define NAU8325_I2S_DL_32 (0x3 << NAU8325_I2S_DL_SFT) +#define NAU8325_I2S_DL_24 (0x2 << NAU8325_I2S_DL_SFT) +#define NAU8325_I2S_DL_20 (0x1 << NAU8325_I2S_DL_SFT) +#define NAU8325_I2S_DL_16 (0x0 << NAU8325_I2S_DL_SFT) +#define NAU8325_I2S_DF_MASK 0x3 +#define NAU8325_I2S_DF_RIGTH 0x0 +#define NAU8325_I2S_DF_LEFT 0x1 +#define NAU8325_I2S_DF_I2S 0x2 +#define NAU8325_I2S_DF_PCM_AB 0x3 + +/* I2S_PCM_CTRL2 (0x0e) */ +#define NAU8325_PCM_TS_SFT 10 +#define NAU8325_PCM_TS_EN (0x1 << NAU8325_PCM_TS_SFT) +#define NAU8325_PCM8BIT0_SFT 8 +#define NAU8325_PCM8BIT0_MASK (0x1 << NAU8325_PCM8BIT0_SFT) + +/* L_TIME_SLOT (0x0f)*/ +#define NAU8325_SHORT_FS_DET_SFT 13 +#define NAU8325_SHORT_FS_DET_DIS (0x1 << NAU8325_SHORT_FS_DET_SFT) +#define NAU8325_TSLOT_L0_MASK 0x3ff + +/* R_TIME_SLOT (0x10)*/ +#define NAU8325_TSLOT_R0_MASK 0x3ff + +/* HPF_CTRL (0x11)*/ +#define NAU8325_DAC_HPF_SFT 15 +#define NAU8325_DAC_HPF_EN (0x1 << NAU8325_DAC_HPF_SFT) +#define NAU8325_DAC_HPF_APP_SFT 14 +#define NAU8325_DAC_HPF_APP_MASK (0x1 << NAU8325_DAC_HPF_APP_SFT) +#define NAU8325_DAC_HPF_FCUT_SFT 11 +#define NAU8325_DAC_HPF_FCUT_MASK (0x7 << NAU8325_DAC_HPF_FCUT_SFT) + +/* MUTE_CTRL (0x12)*/ +#define NAU8325_SOFT_MUTE_SFT 15 +#define NAU8325_SOFT_MUTE (0x1 << NAU8325_SOFT_MUTE_SFT) +#define NAU8325_DAC_ZC_SFT 8 +#define NAU8325_DAC_ZC_EN (0x1 << NAU8325_DAC_ZC_SFT) +#define NAU8325_UNMUTE_CTL_SFT 6 +#define NAU8325_UNMUTE_CTL_MASK (0x3 << NAU8325_UNMUTE_CTL_SFT) +#define NAU8325_ANA_MUTE_SFT 4 +#define NAU8325_ANA_MUTE_MASK (0x3 << NAU8325_ANA_MUTE_SFT) +#define NAU8325_AUTO_MUTE_SFT 3 +#define NAU8325_AUTO_MUTE_DIS (0x1 << NAU8325_AUTO_MUTE_SFT) + +/* DAC_VOLUME (0x13) */ +#define NAU8325_DAC_VOLUME_L_SFT 8 +#define NAU8325_DAC_VOLUME_L_EN (0xff << NAU8325_DAC_VOLUME_L_SFT) +#define NAU8325_DAC_VOLUME_R_SFT 0 +#define NAU8325_DAC_VOLUME_R_EN (0xff << NAU8325_DAC_VOLUME_R_SFT) +#define NAU8325_DAC_VOL_MAX 0xff + +/* DEBUG_READ1 (0x1d)*/ +#define NAU8325_OSR100_MASK (0x1 << 6) +#define NAU8325_MIPS500_MASK (0x1 << 5) +#define NAU8325_SHUTDWNDRVR_R_MASK (0x1 << 4) +#define NAU8325_SHUTDWNDRVR_L_MASK (0x1 << 3) +#define NAU8325_MUTEB_MASK (0x1 << 2) +#define NAU8325_PDOSCB_MASK (0x1 << 1) +#define NAU8325_POWERDOWN1B_D_MASK 0x1 + +/* DEBUG_READ2 (0x1f)*/ +#define NAU8325_R_CHANNEL_Vol_SFT 8 +#define NAU8325_R_CHANNEL_Vol_MASK (0xff << NAU8325_R_CHANNEL_Vol_SFT) +#define NAU8325_L_CHANNEL_Vol_MASK 0xff + +/* DEBUG_READ3(0x22)*/ +#define NAU8325_PGAL_GAIN_MASK (0x3f << 7) +#define NAU8325_CLIP_MASK (0x1 << 6) +#define NAU8325_SCAN_MODE_MASK (0x1 << 5) +#define NAU8325_SDB_MASK (0x1 << 4) +#define NAU8325_TALARM_MASK (0x1 << 3) +#define NAU8325_SHORTR_MASK (0x1 << 2) +#define NAU8325_SHORTL_MASK (0x1 << 1) +#define NAU8325_TMDET_MASK 0x1 + +/* DAC_CTRL1 (0x29) */ +#define NAU8325_DAC_OVERSAMPLE_SFT 0 +#define NAU8325_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8325_DAC_OVERSAMPLE_256 1 +#define NAU8325_DAC_OVERSAMPLE_128 2 +#define NAU8325_DAC_OVERSAMPLE_64 0 +#define NAU8325_DAC_OVERSAMPLE_32 4 + +/* ALC_CTRL1 (0x2c) */ +#define NAU8325_ALC_MAXGAIN_SFT 5 +#define NAU8325_ALC_MAXGAIN_MAX 0x7 +#define NAU8325_ALC_MAXGAIN_MASK (0x7 << NAU8325_ALC_MAXGAIN_SFT) +#define NAU8325_ALC_MINGAIN_MAX 4 +#define NAU8325_ALC_MINGAIN_SFT 1 +#define NAU8325_ALC_MINGAIN_MASK (0x7 << NAU8325_ALC_MINGAIN_SFT) + +/* ALC_CTRL2 (0x2d) */ +#define NAU8325_ALC_DCY_SFT 12 +#define NAU8325_ALC_DCY_MAX 0xb +#define NAU8325_ALC_DCY_MASK (0xf << NAU8325_ALC_DCY_SFT) +#define NAU8325_ALC_ATK_SFT 8 +#define NAU8325_ALC_ATK_MAX 0xb +#define NAU8325_ALC_ATK_MASK (0xf << NAU8325_ALC_ATK_SFT) +#define NAU8325_ALC_HLD_SFT 4 +#define NAU8325_ALC_HLD_MAX 0xa +#define NAU8325_ALC_HLD_MASK (0xf << NAU8325_ALC_HLD_SFT) +#define NAU8325_ALC_LVL_SFT 0 +#define NAU8325_ALC_LVL_MAX 0xf +#define NAU8325_ALC_LVL_MASK 0xf + +/* ALC_CTRL3 (0x2e) */ +#define NAU8325_ALC_EN_SFT 15 +#define NAU8325_ALC_EN (0x1 << NAU8325_ALC_EN_SFT) + +/* TEMP_COMP_CTRL (0x30) */ +#define NAU8325_TEMP_COMP_ACT2_MASK 0xff + +/* LPF_CTRL (0x33) */ +#define NAU8325_LPF_IN1_EN_SFT 15 +#define NAU8325_LPF_IN1_EN (0x1 << NAU8325_LPF_IN1_EN_SFT) +#define NAU8325_LPF_IN1_TC_SFT 11 +#define NAU8325_LPF_IN1_TC_MASK (0xf << NAU8325_LPF_IN1_TC_SFT) +#define NAU8325_LPF_IN2_EN_SFT 10 +#define NAU8325_LPF_IN2_EN (0x1 << NAU8325_LPF_IN2_EN_SFT) +#define NAU8325_LPF_IN2_TC_SFT 6 +#define NAU8325_LPF_IN2_TC_MASK (0xf << NAU8325_LPF_IN2_TC_SFT) + +/* CLK_DET_CTRL (0x40) */ +#define NAU8325_APWRUP_SFT 15 +#define NAU8325_APWRUP_EN (0x1 << NAU8325_APWRUP_SFT) +#define NAU8325_CLKPWRUP_SFT 14 +#define NAU8325_CLKPWRUP_DIS (0x1 << NAU8325_CLKPWRUP_SFT) +#define NAU8325_PWRUP_DFT_SFT 13 +#define NAU8325_PWRUP_DFT (0x1 << NAU8325_PWRUP_DFT_SFT) +#define NAU8325_REG_SRATE_SFT 10 +#define NAU8325_REG_SRATE_MASK (0x7 << NAU8325_REG_SRATE_SFT) +#define NAU8325_REG_ALT_SRATE_SFT 9 +#define NAU8325_REG_ALT_SRATE_EN (0x1 << NAU8325_REG_ALT_SRATE_SFT) +#define NAU8325_REG_DIV_MAX 0x1 + +/* BIAS_ADJ (0x60) */ +#define NAU8325_BIAS_VMID_SEL_SFT 4 +#define NAU8325_BIAS_VMID_SEL_MASK (0x3 << NAU8325_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_1 (0x61) */ +#define NAU8325_VMDFSTENB_SFT 14 +#define NAU8325_VMDFSTENB_MASK (0x3 << NAU8325_VMDFSTENB_SFT) +#define NAU8325_CLASSDEN_SFT 12 +#define NAU8325_CLASSDEN_MASK (0x3 << NAU8325_CLASSDEN_SFT) +#define NAU8325_DACCLKEN_R_SFT 10 +#define NAU8325_DACCLKEN_R_MASK (0x3 << NAU8325_DACCLKEN_R_SFT) +#define NAU8325_DACEN_R_SFT 8 +#define NAU8325_DACEN_R_MASK (0x3 << NAU8325_DACEN_R_SFT) +#define NAU8325_DACCLKEN_SFT 6 +#define NAU8325_DACCLKEN_MASK (0x3 << NAU8325_DACCLKEN_SFT) +#define NAU8325_DACEN_SFT 4 +#define NAU8325_DACEN_MASK (0x3 << NAU8325_DACEN_SFT) +#define NAU8325_BIASEN_SFT 2 +#define NAU8325_BIASEN_MASK (0x3 << NAU8325_BIASEN_SFT) +#define NAU8325_VMIDEN_MASK 0x3 + +/* ANALOG_CONTROL_2 (0x62) */ +#define NAU8325_PWMMOD_SFT 14 +#define NAU8325_PWMMOD_MASK (0x1 << NAU8325_PWMMOD_SFT) +#define NAU8325_DACTEST_SFT 6 +#define NAU8325_DACTEST_MASK (0x3 << NAU8325_DACTEST_SFT) +#define NAU8325_DACREFCAP_SFT 4 +#define NAU8325_DACREFCAP_MASK (0x3 << NAU8325_DACREFCAP_SFT) + +/* ANALOG_CONTROL_3 (0x63) */ +#define NAU8325_POWER_DOWN_L_SFT 12 +#define NAU8325_POWER_DOWN_L_MASK (0x3 << NAU8325_POWER_DOWN_L_SFT) +#define NAU8325_POWER_DOWN_R_SFT 11 +#define NAU8325_POWER_DOWN_R_MASK (0x3 << NAU8325_DACREFCAP_SFT) +#define NAU8325_CLASSD_FINE_SFT 5 +#define NAU8325_CLASSD_FINE_MASK (0x3 << NAU8325_CLASSD_FINE_SFT) +#define NAU8325_CLASSD_COARSE_GAIN_MASK 0xf + +/* ANALOG_CONTROL_4 (0x64) */ +#define NAU8325_CLASSD_OCPN_SFT 12 +#define NAU8325_CLASSD_OCPN_MASK (0xf << NAU8325_CLASSD_OCPN_SFT) +#define NAU8325_CLASSD_OCPP_SFT 8 +#define NAU8325_CLASSD_OCPP_MASK (0xf << NAU8325_CLASSD_OCPP_SFT) +#define NAU8325_CLASSD_SLEWN_MASK 0xff + +/* ANALOG_CONTROL_5 (0x65) */ +#define NAU8325_MCLK_RANGE_SFT 2 +#define NAU8325_MCLK_RANGE_EN (0x1 << NAU8325_MCLK_RANGE_SFT) +#define NAU8325_MCLK8XEN_SFT 1 +#define NAU8325_MCLK8XEN_EN (0x1 << NAU8325_MCLK8XEN_SFT) +#define NAU8325_MCLK4XEN_EN 0x1 + +/* ANALOG_CONTROL_6 (0x66) */ +#define NAU8325_VBATLOW_SFT 4 +#define NAU8325_VBATLOW_MASK (0x1 << NAU8325_VBATLOW_SFT) +#define NAU8325_VDDSPK_LIM_SFT 3 +#define NAU8325_VDDSPK_LIM_EN (0x1 << NAU8325_VDDSPK_LIM_SFT) +#define NAU8325_VDDSPK_LIM_MASK 0x7 + +/* CLIP_CTRL (0x69)*/ +#define NAU8325_ANTI_CLIP_SFT 4 +#define NAU8325_ANTI_CLIP_EN (0x1 << NAU8325_ANTI_CLIP_SFT) + +/* RDAC (0x73) */ +#define NAU8325_CLK_DAC_DELAY_SFT 4 +#define NAU8325_CLK_DAC_DELAY_EN (0x7 << NAU8325_CLK_DAC_DELAY_SFT) +#define NAU8325_DACVREFSEL_SFT 2 +#define NAU8325_DACVREFSEL_MASK (0x3 << NAU8325_DACVREFSEL_SFT) + +#define NAU8325_CODEC_DAI "nau8325-hifi" + +struct nau8325 { + struct device *dev; + struct regmap *regmap; + int mclk; + int fs; + int vref_impedance_ohms; + int dac_vref_microvolt; + int clock_detection; + int clock_det_data; + int alc_enable; +}; + +struct nau8325_src_attr { + int param; + unsigned int val; +}; + +enum { + NAU8325_MCLK_FS_RATIO_256, + NAU8325_MCLK_FS_RATIO_400, + NAU8325_MCLK_FS_RATIO_500, + NAU8325_MCLK_FS_RATIO_NUM, +}; + +struct nau8325_srate_attr { + int fs; + int range; + bool max; + unsigned int mclk_src[NAU8325_MCLK_FS_RATIO_NUM]; +}; + +struct nau8325_osr_attr { + unsigned int osr; + unsigned int clk_src; +}; + +#endif /* __NAU8325_H__ */ From 9b21531866fced634bcc7fca33667c7ef258433f Mon Sep 17 00:00:00 2001 From: Frank Li Date: Thu, 28 Mar 2024 10:51:33 -0400 Subject: [PATCH 131/603] ASoC: dt-bindings: fsl,imx-asrc/spdif: Add power-domains property Add power-domains property for asrc and spdif since fsl,imx8qm-asrc/spdif and fsl,imx8qxp-asrc/spdif require 'power-domains'. Set 'power-domains' as required property for compatible string fsl,imx8qm-asrc/spdif and fsl,imx8qxp-asrc/spdif. Acked-by: Conor Dooley Signed-off-by: Frank Li Link: https://msgid.link/r/20240328-asrc_8qxp-v8-1-801cd6bb5be2@nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,imx-asrc.yaml | 14 ++++++++++++++ .../devicetree/bindings/sound/fsl,spdif.yaml | 15 +++++++++++++++ 2 files changed, 29 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml index bfef2fcb75b14..76aa1f2484883 100644 --- a/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml @@ -74,6 +74,9 @@ properties: - const: asrck_f - const: spba + power-domains: + maxItems: 1 + fsl,asrc-rate: $ref: /schemas/types.yaml#/definitions/uint32 description: The mutual sample rate used by DPCM Back Ends @@ -131,6 +134,17 @@ allOf: properties: fsl,asrc-clk-map: false + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-asrc + - fsl,imx8qxp-asrc + then: + required: + - power-domains + additionalProperties: false examples: diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 1d64e8337aa4b..56f8c0c8afdea 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -86,6 +86,9 @@ properties: registers. Set this flag for HCDs with big endian descriptors and big endian registers. + power-domains: + maxItems: 1 + required: - compatible - reg @@ -97,6 +100,18 @@ required: additionalProperties: false +allOf: + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + then: + required: + - power-domains + examples: - | spdif@2004000 { From 82330327ce03889f20072cef5167499d51cdf260 Mon Sep 17 00:00:00 2001 From: Frank Li Date: Thu, 28 Mar 2024 10:51:34 -0400 Subject: [PATCH 132/603] ASoC: dt-bindings: fsl,imx-asrc: update max interrupt numbers fsl,imx8qxp-spdif and fsl,imx8qm-spdif have 2 interrupts. Other platforms have 1 interrupt. Increase max interrupt number to 2 and add restriction for platforms except i.MX8QXP and i.MX8QM. Reviewed-by: Rob Herring Signed-off-by: Frank Li Link: https://msgid.link/r/20240328-asrc_8qxp-v8-2-801cd6bb5be2@nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,spdif.yaml | 20 ++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml index 56f8c0c8afdea..204f361cea27a 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.yaml @@ -31,7 +31,10 @@ properties: maxItems: 1 interrupts: - maxItems: 1 + minItems: 1 + items: + - description: Combined or receive interrupt + - description: Transmit interrupt dmas: items: @@ -101,6 +104,21 @@ required: additionalProperties: false allOf: + - if: + properties: + compatible: + enum: + - fsl,imx8qm-spdif + - fsl,imx8qxp-spdif + then: + properties: + interrupts: + minItems: 2 + else: + properties: + interrupts: + maxItems: 1 + - if: properties: compatible: From 7b5f2072657a9041cbaf4ba139f672be11694ca3 Mon Sep 17 00:00:00 2001 From: Frank Li Date: Thu, 28 Mar 2024 10:51:35 -0400 Subject: [PATCH 133/603] ASoC: dt-bindings: fsl-sai: allow only one dma-names Some sai only connect one direction dma (rx/tx) in SOC. For example: imx8qxp sai5 only connect tx dma channel. So allow only one "rx" or "tx" for dma-names. Remove description under dmas because no user use index to get dma channel. All user use 'dma-names' to get correct dma channel. dma-names already in 'required' list. Reviewed-by: Rob Herring Signed-off-by: Frank Li Link: https://msgid.link/r/20240328-asrc_8qxp-v8-3-801cd6bb5be2@nxp.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl,sai.yaml | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml index 2456d958adeef..a5d9c246cc476 100644 --- a/Documentation/devicetree/bindings/sound/fsl,sai.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -81,14 +81,12 @@ properties: dmas: minItems: 1 - items: - - description: DMA controller phandle and request line for RX - - description: DMA controller phandle and request line for TX + maxItems: 2 dma-names: minItems: 1 items: - - const: rx + - enum: [ rx, tx ] - const: tx interrupts: From ea60ab95723f5738e7737b56dda95e6feefa5b50 Mon Sep 17 00:00:00 2001 From: Aleksandr Mishin Date: Thu, 28 Mar 2024 20:33:37 +0300 Subject: [PATCH 134/603] ASoC: kirkwood: Fix potential NULL dereference In kirkwood_dma_hw_params() mv_mbus_dram_info() returns NULL if CONFIG_PLAT_ORION macro is not defined. Fix this bug by adding NULL check. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: bb6a40fc5a83 ("ASoC: kirkwood: Fix reference to PCM buffer address") Signed-off-by: Aleksandr Mishin Link: https://msgid.link/r/20240328173337.21406-1-amishin@t-argos.ru Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index dd2f806526c10..ef00792e1d49a 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -182,6 +182,9 @@ static int kirkwood_dma_hw_params(struct snd_soc_component *component, const struct mbus_dram_target_info *dram = mv_mbus_dram_info(); unsigned long addr = substream->runtime->dma_addr; + if (!dram) + return 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_PLAYBACK_WIN, addr, dram); From 08ea486a61451189b190c7b89e406b889cf693fa Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 1 Apr 2024 10:02:08 +0000 Subject: [PATCH 135/603] ALSA: control: Introduce snd_ctl_find_id_mixer_locked() Adds wrapper function snd_ctl_find_id_mixer_locked(). This is identical to snd_ctl_find_id_mixer() except that it can be called from code that is already holding controls_rwsem. Signed-off-by: Richard Fitzgerald Reviewed-by: Takashi Iwai Link: https://msgid.link/r/20240401100210.61277-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/control.h | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/include/sound/control.h b/include/sound/control.h index 9a4f4f7138da8..c1659036c4a77 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -167,6 +167,29 @@ snd_ctl_find_id_mixer(struct snd_card *card, const char *name) return snd_ctl_find_id(card, &id); } +/** + * snd_ctl_find_id_mixer_locked - find the control instance with the given name string + * @card: the card instance + * @name: the name string + * + * Finds the control instance with the given name and + * @SNDRV_CTL_ELEM_IFACE_MIXER. Other fields are set to zero. + * + * This is merely a wrapper to snd_ctl_find_id_locked(). + * The caller must down card->controls_rwsem before calling this function. + * + * Return: The pointer of the instance if found, or %NULL if not. + */ +static inline struct snd_kcontrol * +snd_ctl_find_id_mixer_locked(struct snd_card *card, const char *name) +{ + struct snd_ctl_elem_id id = {}; + + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strscpy(id.name, name, sizeof(id.name)); + return snd_ctl_find_id_locked(card, &id); +} + int snd_ctl_create(struct snd_card *card); int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn); From 897cc72b08374c1224a9ded03c82dfc8e41f80c2 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 1 Apr 2024 10:02:09 +0000 Subject: [PATCH 136/603] ASoC: soc-card: Use snd_ctl_find_id_mixer() instead of open-coding Use the snd_ctl_find_id_mixer[_locked]() wrapper in snd_soc_card_get_kcontrol[_locked]() instead of open-coding a custom list walk of the card controls list. Signed-off-by: Richard Fitzgerald Reviewed-by: Takashi Iwai Link: https://msgid.link/r/20240401100210.61277-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-card.c | 21 ++++----------------- 1 file changed, 4 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c index 8a2f163da6bc9..0a3104d4ad235 100644 --- a/sound/soc/soc-card.c +++ b/sound/soc/soc-card.c @@ -32,33 +32,20 @@ static inline int _soc_card_ret(struct snd_soc_card *card, struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, const char *name) { - struct snd_card *card = soc_card->snd_card; - struct snd_kcontrol *kctl; - - /* must be held read or write */ - lockdep_assert_held(&card->controls_rwsem); - if (unlikely(!name)) return NULL; - list_for_each_entry(kctl, &card->controls, list) - if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) - return kctl; - return NULL; + return snd_ctl_find_id_mixer_locked(soc_card->snd_card, name); } EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol_locked); struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name) { - struct snd_card *card = soc_card->snd_card; - struct snd_kcontrol *kctl; - - down_read(&card->controls_rwsem); - kctl = snd_soc_card_get_kcontrol_locked(soc_card, name); - up_read(&card->controls_rwsem); + if (unlikely(!name)) + return NULL; - return kctl; + return snd_ctl_find_id_mixer(soc_card->snd_card, name); } EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol); From ef7784e41db73f3d31ce545227ebba4483479a26 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 1 Apr 2024 10:02:10 +0000 Subject: [PATCH 137/603] ASoC: soc-card: Add KUnit test case for snd_soc_card_get_kcontrol Add a new snd-soc-card KUnit test with a simple test case for snd_soc_card_get_kcontrol() and snd_soc_card_get_kcontrol_locked(). Signed-off-by: Richard Fitzgerald Reviewed-by: Takashi Iwai Link: https://msgid.link/r/20240401100210.61277-4-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/Kconfig | 8 ++ sound/soc/Makefile | 4 + sound/soc/soc-card-test.c | 184 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 196 insertions(+) create mode 100644 sound/soc/soc-card-test.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 439fa631c342a..a52afb423b46b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -66,6 +66,14 @@ config SND_SOC_TOPOLOGY_KUNIT_TEST userspace applications such as pulseaudio, to prevent unnecessary problems. +config SND_SOC_CARD_KUNIT_TEST + tristate "KUnit tests for SoC card" + depends on KUNIT + default KUNIT_ALL_TESTS + help + If you want to perform tests on ALSA SoC card functions say Y here. + If unsure, say N. + config SND_SOC_UTILS_KUNIT_TEST tristate "KUnit tests for SoC utils" depends on KUNIT diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8376fdb217ed1..f90f5300b36e5 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -12,6 +12,10 @@ ifneq ($(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST),) obj-$(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST) += soc-topology-test.o endif +ifneq ($(CONFIG_SND_SOC_CARD_KUNIT_TEST),) +obj-$(CONFIG_SND_SOC_CARD_KUNIT_TEST) += soc-card-test.o +endif + ifneq ($(CONFIG_SND_SOC_UTILS_KUNIT_TEST),) # snd-soc-test-objs := soc-utils-test.o obj-$(CONFIG_SND_SOC_UTILS_KUNIT_TEST) += soc-utils-test.o diff --git a/sound/soc/soc-card-test.c b/sound/soc/soc-card-test.c new file mode 100644 index 0000000000000..075c52fe82e51 --- /dev/null +++ b/sound/soc/soc-card-test.c @@ -0,0 +1,184 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (C) 2024 Cirrus Logic, Inc. and +// Cirrus Logic International Semiconductor Ltd. + +#include +#include +#include +#include +#include +#include + +struct soc_card_test_priv { + struct device *card_dev; + struct snd_soc_card *card; +}; + +static const struct snd_kcontrol_new test_card_controls[] = { + SOC_SINGLE("Fee", SND_SOC_NOPM, 0, 1, 0), + SOC_SINGLE("Fi", SND_SOC_NOPM, 1, 1, 0), + SOC_SINGLE("Fo", SND_SOC_NOPM, 2, 1, 0), + SOC_SINGLE("Fum", SND_SOC_NOPM, 3, 1, 0), + SOC_SINGLE("Left Fee", SND_SOC_NOPM, 4, 1, 0), + SOC_SINGLE("Right Fee", SND_SOC_NOPM, 5, 1, 0), + SOC_SINGLE("Left Fi", SND_SOC_NOPM, 6, 1, 0), + SOC_SINGLE("Right Fi", SND_SOC_NOPM, 7, 1, 0), + SOC_SINGLE("Left Fo", SND_SOC_NOPM, 8, 1, 0), + SOC_SINGLE("Right Fo", SND_SOC_NOPM, 9, 1, 0), + SOC_SINGLE("Left Fum", SND_SOC_NOPM, 10, 1, 0), + SOC_SINGLE("Right Fum", SND_SOC_NOPM, 11, 1, 0), +}; + +static void test_snd_soc_card_get_kcontrol(struct kunit *test) +{ + struct soc_card_test_priv *priv = test->priv; + struct snd_soc_card *card = priv->card; + struct snd_kcontrol *kc; + struct soc_mixer_control *mc; + int i, ret; + + ret = snd_soc_add_card_controls(card, test_card_controls, ARRAY_SIZE(test_card_controls)); + KUNIT_ASSERT_EQ(test, ret, 0); + + /* Look up every control */ + for (i = 0; i < ARRAY_SIZE(test_card_controls); ++i) { + kc = snd_soc_card_get_kcontrol(card, test_card_controls[i].name); + KUNIT_EXPECT_NOT_ERR_OR_NULL_MSG(test, kc, "Failed to find '%s'\n", + test_card_controls[i].name); + if (!kc) + continue; + + /* Test that it is the correct control */ + mc = (struct soc_mixer_control *)kc->private_value; + KUNIT_EXPECT_EQ_MSG(test, mc->shift, i, "For '%s'\n", test_card_controls[i].name); + } + + /* Test some names that should not be found */ + kc = snd_soc_card_get_kcontrol(card, "None"); + KUNIT_EXPECT_NULL(test, kc); + + kc = snd_soc_card_get_kcontrol(card, "Left None"); + KUNIT_EXPECT_NULL(test, kc); + + kc = snd_soc_card_get_kcontrol(card, "Left"); + KUNIT_EXPECT_NULL(test, kc); + + kc = snd_soc_card_get_kcontrol(card, NULL); + KUNIT_EXPECT_NULL(test, kc); +} + +static void test_snd_soc_card_get_kcontrol_locked(struct kunit *test) +{ + struct soc_card_test_priv *priv = test->priv; + struct snd_soc_card *card = priv->card; + struct snd_kcontrol *kc, *kcw; + struct soc_mixer_control *mc; + int i, ret; + + ret = snd_soc_add_card_controls(card, test_card_controls, ARRAY_SIZE(test_card_controls)); + KUNIT_ASSERT_EQ(test, ret, 0); + + /* Look up every control */ + for (i = 0; i < ARRAY_SIZE(test_card_controls); ++i) { + down_read(&card->snd_card->controls_rwsem); + kc = snd_soc_card_get_kcontrol_locked(card, test_card_controls[i].name); + up_read(&card->snd_card->controls_rwsem); + KUNIT_EXPECT_NOT_ERR_OR_NULL_MSG(test, kc, "Failed to find '%s'\n", + test_card_controls[i].name); + if (!kc) + continue; + + /* Test that it is the correct control */ + mc = (struct soc_mixer_control *)kc->private_value; + KUNIT_EXPECT_EQ_MSG(test, mc->shift, i, "For '%s'\n", test_card_controls[i].name); + + down_write(&card->snd_card->controls_rwsem); + kcw = snd_soc_card_get_kcontrol_locked(card, test_card_controls[i].name); + up_write(&card->snd_card->controls_rwsem); + KUNIT_EXPECT_NOT_ERR_OR_NULL_MSG(test, kcw, "Failed to find '%s'\n", + test_card_controls[i].name); + + KUNIT_EXPECT_PTR_EQ(test, kc, kcw); + } + + /* Test some names that should not be found */ + down_read(&card->snd_card->controls_rwsem); + kc = snd_soc_card_get_kcontrol_locked(card, "None"); + up_read(&card->snd_card->controls_rwsem); + KUNIT_EXPECT_NULL(test, kc); + + down_read(&card->snd_card->controls_rwsem); + kc = snd_soc_card_get_kcontrol_locked(card, "Left None"); + up_read(&card->snd_card->controls_rwsem); + KUNIT_EXPECT_NULL(test, kc); + + down_read(&card->snd_card->controls_rwsem); + kc = snd_soc_card_get_kcontrol_locked(card, "Left"); + up_read(&card->snd_card->controls_rwsem); + KUNIT_EXPECT_NULL(test, kc); + + down_read(&card->snd_card->controls_rwsem); + kc = snd_soc_card_get_kcontrol_locked(card, NULL); + up_read(&card->snd_card->controls_rwsem); + KUNIT_EXPECT_NULL(test, kc); +} + +static int soc_card_test_case_init(struct kunit *test) +{ + struct soc_card_test_priv *priv; + int ret; + + priv = kunit_kzalloc(test, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + test->priv = priv; + + priv->card_dev = kunit_device_register(test, "sound-soc-card-test"); + priv->card_dev = get_device(priv->card_dev); + if (!priv->card_dev) + return -ENODEV; + + priv->card = kunit_kzalloc(test, sizeof(*priv->card), GFP_KERNEL); + if (!priv->card) + return -ENOMEM; + + priv->card->name = "soc-card-test"; + priv->card->dev = priv->card_dev; + priv->card->owner = THIS_MODULE; + + ret = snd_soc_register_card(priv->card); + if (!ret) + return ret; + + return 0; +} + +static void soc_card_test_case_exit(struct kunit *test) +{ + struct soc_card_test_priv *priv = test->priv; + + if (priv->card) + snd_soc_unregister_card(priv->card); + + if (priv->card_dev) + put_device(priv->card_dev); +} + +static struct kunit_case soc_card_test_cases[] = { + KUNIT_CASE(test_snd_soc_card_get_kcontrol), + KUNIT_CASE(test_snd_soc_card_get_kcontrol_locked), + {} +}; + +static struct kunit_suite soc_card_test_suite = { + .name = "soc-card", + .test_cases = soc_card_test_cases, + .init = soc_card_test_case_init, + .exit = soc_card_test_case_exit, +}; + +kunit_test_suites(&soc_card_test_suite); + +MODULE_DESCRIPTION("ASoC soc-card KUnit test"); +MODULE_LICENSE("GPL"); From 35aaed1bc3c32c52b7fda525cd7bd19998035518 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 2 Apr 2024 02:23:54 +0000 Subject: [PATCH 138/603] ASoC: rsnd: cleanup regmap table R-Car Sound is using almost same register mapping on each generation, therefore we can share it. Here indicates some memo/comments. SSI/SSIU/SCU have multi channels (SSI0 - SSI9 etc), but ADG is single, so, we don't need to have multi channel settings for it - rsnd_gen_regmap_init(..., 10, ..ADG, ...); + rsnd_gen_regmap_init(..., 1, ..ADG, ...); Gen4 added new SSI_BUSIF, but it is not used Gen4 calls rsnd_gen_regmap_init() for SDMC, but the address is calculated by rsnd_gen4_dma_addr(). It needs physical address, but other info are not needed. use NULL conf for it. - rsnd_gen_regmap_init(..., 10, ..SDMC, conf_sdmc); + rsnd_gen_regmap_init(..., 1, ..SDMC, conf_null); Gen4 have only SSI0/SSIU0, we don't need to have multi channel settings. - rsnd_gen_regmap_init(..., 10, ..SSI, ...); + rsnd_gen_regmap_init(..., 1, ..SSI, ...); SSI_MODE2 address is different between Gen2/3 and Gen4, but it is not used on Gen4. Thus we can share same regmap table. Gen2/3 : 0x808 Gen4 : 0xa0c Signed-off-by: Kuninori Morimoto Tested-by: Khanh Le Link: https://msgid.link/r/871q7obiyt.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/gen.c | 517 +++++++++++++++++---------------------- sound/soc/sh/rcar/rsnd.h | 1 - 2 files changed, 226 insertions(+), 292 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 86bdecc24956b..e566712e52346 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -215,70 +215,220 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, return 0; } +/* + * (A) : Gen4 is 0xa0c, but it is not used. + * see + * rsnd_ssiu_init() + */ +static const struct rsnd_regmap_field_conf conf_common_ssiu[] = { + RSND_GEN_S_REG(SSI_MODE0, 0x800), + RSND_GEN_S_REG(SSI_MODE1, 0x804), + RSND_GEN_S_REG(SSI_MODE2, 0x808), // (A) + RSND_GEN_S_REG(SSI_CONTROL, 0x810), + RSND_GEN_S_REG(SSI_SYS_STATUS0, 0x840), + RSND_GEN_S_REG(SSI_SYS_STATUS1, 0x844), + RSND_GEN_S_REG(SSI_SYS_STATUS2, 0x848), + RSND_GEN_S_REG(SSI_SYS_STATUS3, 0x84c), + RSND_GEN_S_REG(SSI_SYS_STATUS4, 0x880), + RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884), + RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888), + RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c), + RSND_GEN_S_REG(HDMI0_SEL, 0x9e0), + RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), + RSND_GEN_M_REG(SSI_BUSIF0_MODE, 0x0, 0x80), + RSND_GEN_M_REG(SSI_BUSIF0_ADINR, 0x4, 0x80), + RSND_GEN_M_REG(SSI_BUSIF0_DALIGN, 0x8, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80), + RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80), + RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80), + RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80), + RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80), + RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80), + RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80), + RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80), + RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), + RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), + RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), + RSND_GEN_S_REG(SSI9_BUSIF0_MODE, 0x48c), + RSND_GEN_S_REG(SSI9_BUSIF0_ADINR, 0x484), + RSND_GEN_S_REG(SSI9_BUSIF0_DALIGN, 0x488), + RSND_GEN_S_REG(SSI9_BUSIF1_MODE, 0x4a0), + RSND_GEN_S_REG(SSI9_BUSIF1_ADINR, 0x4a4), + RSND_GEN_S_REG(SSI9_BUSIF1_DALIGN, 0x4a8), + RSND_GEN_S_REG(SSI9_BUSIF2_MODE, 0x4c0), + RSND_GEN_S_REG(SSI9_BUSIF2_ADINR, 0x4c4), + RSND_GEN_S_REG(SSI9_BUSIF2_DALIGN, 0x4c8), + RSND_GEN_S_REG(SSI9_BUSIF3_MODE, 0x4e0), + RSND_GEN_S_REG(SSI9_BUSIF3_ADINR, 0x4e4), + RSND_GEN_S_REG(SSI9_BUSIF3_DALIGN, 0x4e8), + RSND_GEN_S_REG(SSI9_BUSIF4_MODE, 0xd80), + RSND_GEN_S_REG(SSI9_BUSIF4_ADINR, 0xd84), + RSND_GEN_S_REG(SSI9_BUSIF4_DALIGN, 0xd88), + RSND_GEN_S_REG(SSI9_BUSIF5_MODE, 0xda0), + RSND_GEN_S_REG(SSI9_BUSIF5_ADINR, 0xda4), + RSND_GEN_S_REG(SSI9_BUSIF5_DALIGN, 0xda8), + RSND_GEN_S_REG(SSI9_BUSIF6_MODE, 0xdc0), + RSND_GEN_S_REG(SSI9_BUSIF6_ADINR, 0xdc4), + RSND_GEN_S_REG(SSI9_BUSIF6_DALIGN, 0xdc8), + RSND_GEN_S_REG(SSI9_BUSIF7_MODE, 0xde0), + RSND_GEN_S_REG(SSI9_BUSIF7_ADINR, 0xde4), + RSND_GEN_S_REG(SSI9_BUSIF7_DALIGN, 0xde8), +}; + +static const struct rsnd_regmap_field_conf conf_common_scu[] = { + RSND_GEN_M_REG(SRC_I_BUSIF_MODE, 0x0, 0x20), + RSND_GEN_M_REG(SRC_O_BUSIF_MODE, 0x4, 0x20), + RSND_GEN_M_REG(SRC_BUSIF_DALIGN, 0x8, 0x20), + RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20), + RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20), + RSND_GEN_M_REG(SRC_INT_ENABLE0, 0x18, 0x20), + RSND_GEN_M_REG(CMD_BUSIF_MODE, 0x184, 0x20), + RSND_GEN_M_REG(CMD_BUSIF_DALIGN, 0x188, 0x20), + RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20), + RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20), + RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8), + RSND_GEN_S_REG(SCU_SYS_INT_EN0, 0x1cc), + RSND_GEN_S_REG(SCU_SYS_STATUS1, 0x1d0), + RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1d4), + RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40), + RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40), + RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40), + RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40), + RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40), + RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40), + RSND_GEN_M_REG(SRC_BSDSR, 0x22c, 0x40), + RSND_GEN_M_REG(SRC_BSISR, 0x238, 0x40), + RSND_GEN_M_REG(CTU_SWRSR, 0x500, 0x100), + RSND_GEN_M_REG(CTU_CTUIR, 0x504, 0x100), + RSND_GEN_M_REG(CTU_ADINR, 0x508, 0x100), + RSND_GEN_M_REG(CTU_CPMDR, 0x510, 0x100), + RSND_GEN_M_REG(CTU_SCMDR, 0x514, 0x100), + RSND_GEN_M_REG(CTU_SV00R, 0x518, 0x100), + RSND_GEN_M_REG(CTU_SV01R, 0x51c, 0x100), + RSND_GEN_M_REG(CTU_SV02R, 0x520, 0x100), + RSND_GEN_M_REG(CTU_SV03R, 0x524, 0x100), + RSND_GEN_M_REG(CTU_SV04R, 0x528, 0x100), + RSND_GEN_M_REG(CTU_SV05R, 0x52c, 0x100), + RSND_GEN_M_REG(CTU_SV06R, 0x530, 0x100), + RSND_GEN_M_REG(CTU_SV07R, 0x534, 0x100), + RSND_GEN_M_REG(CTU_SV10R, 0x538, 0x100), + RSND_GEN_M_REG(CTU_SV11R, 0x53c, 0x100), + RSND_GEN_M_REG(CTU_SV12R, 0x540, 0x100), + RSND_GEN_M_REG(CTU_SV13R, 0x544, 0x100), + RSND_GEN_M_REG(CTU_SV14R, 0x548, 0x100), + RSND_GEN_M_REG(CTU_SV15R, 0x54c, 0x100), + RSND_GEN_M_REG(CTU_SV16R, 0x550, 0x100), + RSND_GEN_M_REG(CTU_SV17R, 0x554, 0x100), + RSND_GEN_M_REG(CTU_SV20R, 0x558, 0x100), + RSND_GEN_M_REG(CTU_SV21R, 0x55c, 0x100), + RSND_GEN_M_REG(CTU_SV22R, 0x560, 0x100), + RSND_GEN_M_REG(CTU_SV23R, 0x564, 0x100), + RSND_GEN_M_REG(CTU_SV24R, 0x568, 0x100), + RSND_GEN_M_REG(CTU_SV25R, 0x56c, 0x100), + RSND_GEN_M_REG(CTU_SV26R, 0x570, 0x100), + RSND_GEN_M_REG(CTU_SV27R, 0x574, 0x100), + RSND_GEN_M_REG(CTU_SV30R, 0x578, 0x100), + RSND_GEN_M_REG(CTU_SV31R, 0x57c, 0x100), + RSND_GEN_M_REG(CTU_SV32R, 0x580, 0x100), + RSND_GEN_M_REG(CTU_SV33R, 0x584, 0x100), + RSND_GEN_M_REG(CTU_SV34R, 0x588, 0x100), + RSND_GEN_M_REG(CTU_SV35R, 0x58c, 0x100), + RSND_GEN_M_REG(CTU_SV36R, 0x590, 0x100), + RSND_GEN_M_REG(CTU_SV37R, 0x594, 0x100), + RSND_GEN_M_REG(MIX_SWRSR, 0xd00, 0x40), + RSND_GEN_M_REG(MIX_MIXIR, 0xd04, 0x40), + RSND_GEN_M_REG(MIX_ADINR, 0xd08, 0x40), + RSND_GEN_M_REG(MIX_MIXMR, 0xd10, 0x40), + RSND_GEN_M_REG(MIX_MVPDR, 0xd14, 0x40), + RSND_GEN_M_REG(MIX_MDBAR, 0xd18, 0x40), + RSND_GEN_M_REG(MIX_MDBBR, 0xd1c, 0x40), + RSND_GEN_M_REG(MIX_MDBCR, 0xd20, 0x40), + RSND_GEN_M_REG(MIX_MDBDR, 0xd24, 0x40), + RSND_GEN_M_REG(MIX_MDBER, 0xd28, 0x40), + RSND_GEN_M_REG(DVC_SWRSR, 0xe00, 0x100), + RSND_GEN_M_REG(DVC_DVUIR, 0xe04, 0x100), + RSND_GEN_M_REG(DVC_ADINR, 0xe08, 0x100), + RSND_GEN_M_REG(DVC_DVUCR, 0xe10, 0x100), + RSND_GEN_M_REG(DVC_ZCMCR, 0xe14, 0x100), + RSND_GEN_M_REG(DVC_VRCTR, 0xe18, 0x100), + RSND_GEN_M_REG(DVC_VRPDR, 0xe1c, 0x100), + RSND_GEN_M_REG(DVC_VRDBR, 0xe20, 0x100), + RSND_GEN_M_REG(DVC_VOL0R, 0xe28, 0x100), + RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100), + RSND_GEN_M_REG(DVC_VOL2R, 0xe30, 0x100), + RSND_GEN_M_REG(DVC_VOL3R, 0xe34, 0x100), + RSND_GEN_M_REG(DVC_VOL4R, 0xe38, 0x100), + RSND_GEN_M_REG(DVC_VOL5R, 0xe3c, 0x100), + RSND_GEN_M_REG(DVC_VOL6R, 0xe40, 0x100), + RSND_GEN_M_REG(DVC_VOL7R, 0xe44, 0x100), + RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100), +}; + +static const struct rsnd_regmap_field_conf conf_common_adg[] = { + RSND_GEN_S_REG(BRRA, 0x00), + RSND_GEN_S_REG(BRRB, 0x04), + RSND_GEN_S_REG(BRGCKR, 0x08), + RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), + RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10), + RSND_GEN_S_REG(AUDIO_CLK_SEL2, 0x14), + RSND_GEN_S_REG(DIV_EN, 0x30), + RSND_GEN_S_REG(SRCIN_TIMSEL0, 0x34), + RSND_GEN_S_REG(SRCIN_TIMSEL1, 0x38), + RSND_GEN_S_REG(SRCIN_TIMSEL2, 0x3c), + RSND_GEN_S_REG(SRCIN_TIMSEL3, 0x40), + RSND_GEN_S_REG(SRCIN_TIMSEL4, 0x44), + RSND_GEN_S_REG(SRCOUT_TIMSEL0, 0x48), + RSND_GEN_S_REG(SRCOUT_TIMSEL1, 0x4c), + RSND_GEN_S_REG(SRCOUT_TIMSEL2, 0x50), + RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), + RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), + RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), +}; + +static const struct rsnd_regmap_field_conf conf_common_ssi[] = { + RSND_GEN_M_REG(SSICR, 0x00, 0x40), + RSND_GEN_M_REG(SSISR, 0x04, 0x40), + RSND_GEN_M_REG(SSITDR, 0x08, 0x40), + RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40), + RSND_GEN_M_REG(SSIWSR, 0x20, 0x40), +}; + /* * Gen4 */ static int rsnd_gen4_probe(struct rsnd_priv *priv) { - static const struct rsnd_regmap_field_conf conf_ssiu[] = { - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898), - RSND_GEN_S_REG(SSI_SYS_STATUS0, 0x840), - RSND_GEN_S_REG(SSI_SYS_STATUS2, 0x848), - RSND_GEN_S_REG(SSI_SYS_STATUS4, 0x880), - RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888), - - RSND_GEN_S_REG(SSI_BUSIF0_MODE, 0x0), - RSND_GEN_S_REG(SSI_BUSIF0_ADINR, 0x4), - RSND_GEN_S_REG(SSI_BUSIF0_DALIGN, 0x8), - RSND_GEN_S_REG(SSI_BUSIF1_MODE, 0x20), - RSND_GEN_S_REG(SSI_BUSIF1_ADINR, 0x24), - RSND_GEN_S_REG(SSI_BUSIF1_DALIGN, 0x28), - RSND_GEN_S_REG(SSI_BUSIF2_MODE, 0x40), - RSND_GEN_S_REG(SSI_BUSIF2_ADINR, 0x44), - RSND_GEN_S_REG(SSI_BUSIF2_DALIGN, 0x48), - RSND_GEN_S_REG(SSI_BUSIF3_MODE, 0x60), - RSND_GEN_S_REG(SSI_BUSIF3_ADINR, 0x64), - RSND_GEN_S_REG(SSI_BUSIF3_DALIGN, 0x68), - RSND_GEN_S_REG(SSI_BUSIF4_MODE, 0x500), - RSND_GEN_S_REG(SSI_BUSIF4_ADINR, 0x504), - RSND_GEN_S_REG(SSI_BUSIF4_DALIGN, 0x508), - RSND_GEN_S_REG(SSI_BUSIF5_MODE, 0x520), - RSND_GEN_S_REG(SSI_BUSIF5_ADINR, 0x524), - RSND_GEN_S_REG(SSI_BUSIF5_DALIGN, 0x528), - RSND_GEN_S_REG(SSI_BUSIF6_MODE, 0x540), - RSND_GEN_S_REG(SSI_BUSIF6_ADINR, 0x544), - RSND_GEN_S_REG(SSI_BUSIF6_DALIGN, 0x548), - RSND_GEN_S_REG(SSI_BUSIF7_MODE, 0x560), - RSND_GEN_S_REG(SSI_BUSIF7_ADINR, 0x564), - RSND_GEN_S_REG(SSI_BUSIF7_DALIGN, 0x568), - RSND_GEN_S_REG(SSI_CTRL, 0x010), - RSND_GEN_S_REG(SSI_INT_ENABLE, 0x018), - RSND_GEN_S_REG(SSI_MODE, 0x00c), - RSND_GEN_S_REG(SSI_MODE2, 0xa0c), - }; - static const struct rsnd_regmap_field_conf conf_adg[] = { - RSND_GEN_S_REG(BRRA, 0x00), - RSND_GEN_S_REG(BRRB, 0x04), - RSND_GEN_S_REG(BRGCKR, 0x08), - RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), - }; - static const struct rsnd_regmap_field_conf conf_ssi[] = { - RSND_GEN_S_REG(SSICR, 0x00), - RSND_GEN_S_REG(SSISR, 0x04), - RSND_GEN_S_REG(SSITDR, 0x08), - RSND_GEN_S_REG(SSIRDR, 0x0c), - RSND_GEN_S_REG(SSIWSR, 0x20), - }; - static const struct rsnd_regmap_field_conf conf_sdmc[] = { - RSND_GEN_M_REG(SSI_BUSIF, 0x0, 0x8000), - }; - int ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN4_ADG, "adg", conf_adg); - int ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN4_SSIU, "ssiu", conf_ssiu); - int ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN4_SSI, "ssi", conf_ssi); - int ret_sdmc = rsnd_gen_regmap_init(priv, 10, RSND_GEN4_SDMC, "sdmc", conf_sdmc); + struct rsnd_regmap_field_conf conf_null[] = { }; + + /* + * ssiu: SSIU0 + * ssi : SSI0 + */ + int ret_ssiu = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SSIU, "ssiu", conf_common_ssiu); + int ret_ssi = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SSI, "ssi", conf_common_ssi); + int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_ADG, "adg", conf_common_adg); + int ret_sdmc = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SDMC, "sdmc", conf_null); return ret_adg | ret_ssiu | ret_ssi | ret_sdmc; } @@ -288,215 +438,17 @@ static int rsnd_gen4_probe(struct rsnd_priv *priv) */ static int rsnd_gen2_probe(struct rsnd_priv *priv) { - static const struct rsnd_regmap_field_conf conf_ssiu[] = { - RSND_GEN_S_REG(SSI_MODE0, 0x800), - RSND_GEN_S_REG(SSI_MODE1, 0x804), - RSND_GEN_S_REG(SSI_MODE2, 0x808), - RSND_GEN_S_REG(SSI_CONTROL, 0x810), - RSND_GEN_S_REG(SSI_SYS_STATUS0, 0x840), - RSND_GEN_S_REG(SSI_SYS_STATUS1, 0x844), - RSND_GEN_S_REG(SSI_SYS_STATUS2, 0x848), - RSND_GEN_S_REG(SSI_SYS_STATUS3, 0x84c), - RSND_GEN_S_REG(SSI_SYS_STATUS4, 0x880), - RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884), - RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888), - RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898), - RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c), - RSND_GEN_S_REG(HDMI0_SEL, 0x9e0), - RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), - - /* FIXME: it needs SSI_MODE2/3 in the future */ - RSND_GEN_M_REG(SSI_BUSIF0_MODE, 0x0, 0x80), - RSND_GEN_M_REG(SSI_BUSIF0_ADINR, 0x4, 0x80), - RSND_GEN_M_REG(SSI_BUSIF0_DALIGN, 0x8, 0x80), - RSND_GEN_M_REG(SSI_BUSIF1_MODE, 0x20, 0x80), - RSND_GEN_M_REG(SSI_BUSIF1_ADINR, 0x24, 0x80), - RSND_GEN_M_REG(SSI_BUSIF1_DALIGN, 0x28, 0x80), - RSND_GEN_M_REG(SSI_BUSIF2_MODE, 0x40, 0x80), - RSND_GEN_M_REG(SSI_BUSIF2_ADINR, 0x44, 0x80), - RSND_GEN_M_REG(SSI_BUSIF2_DALIGN, 0x48, 0x80), - RSND_GEN_M_REG(SSI_BUSIF3_MODE, 0x60, 0x80), - RSND_GEN_M_REG(SSI_BUSIF3_ADINR, 0x64, 0x80), - RSND_GEN_M_REG(SSI_BUSIF3_DALIGN, 0x68, 0x80), - RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80), - RSND_GEN_M_REG(SSI_BUSIF4_ADINR, 0x504, 0x80), - RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80), - RSND_GEN_M_REG(SSI_BUSIF5_MODE, 0x520, 0x80), - RSND_GEN_M_REG(SSI_BUSIF5_ADINR, 0x524, 0x80), - RSND_GEN_M_REG(SSI_BUSIF5_DALIGN, 0x528, 0x80), - RSND_GEN_M_REG(SSI_BUSIF6_MODE, 0x540, 0x80), - RSND_GEN_M_REG(SSI_BUSIF6_ADINR, 0x544, 0x80), - RSND_GEN_M_REG(SSI_BUSIF6_DALIGN, 0x548, 0x80), - RSND_GEN_M_REG(SSI_BUSIF7_MODE, 0x560, 0x80), - RSND_GEN_M_REG(SSI_BUSIF7_ADINR, 0x564, 0x80), - RSND_GEN_M_REG(SSI_BUSIF7_DALIGN, 0x568, 0x80), - RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), - RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), - RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), - RSND_GEN_S_REG(SSI9_BUSIF0_MODE, 0x48c), - RSND_GEN_S_REG(SSI9_BUSIF0_ADINR, 0x484), - RSND_GEN_S_REG(SSI9_BUSIF0_DALIGN, 0x488), - RSND_GEN_S_REG(SSI9_BUSIF1_MODE, 0x4a0), - RSND_GEN_S_REG(SSI9_BUSIF1_ADINR, 0x4a4), - RSND_GEN_S_REG(SSI9_BUSIF1_DALIGN, 0x4a8), - RSND_GEN_S_REG(SSI9_BUSIF2_MODE, 0x4c0), - RSND_GEN_S_REG(SSI9_BUSIF2_ADINR, 0x4c4), - RSND_GEN_S_REG(SSI9_BUSIF2_DALIGN, 0x4c8), - RSND_GEN_S_REG(SSI9_BUSIF3_MODE, 0x4e0), - RSND_GEN_S_REG(SSI9_BUSIF3_ADINR, 0x4e4), - RSND_GEN_S_REG(SSI9_BUSIF3_DALIGN, 0x4e8), - RSND_GEN_S_REG(SSI9_BUSIF4_MODE, 0xd80), - RSND_GEN_S_REG(SSI9_BUSIF4_ADINR, 0xd84), - RSND_GEN_S_REG(SSI9_BUSIF4_DALIGN, 0xd88), - RSND_GEN_S_REG(SSI9_BUSIF5_MODE, 0xda0), - RSND_GEN_S_REG(SSI9_BUSIF5_ADINR, 0xda4), - RSND_GEN_S_REG(SSI9_BUSIF5_DALIGN, 0xda8), - RSND_GEN_S_REG(SSI9_BUSIF6_MODE, 0xdc0), - RSND_GEN_S_REG(SSI9_BUSIF6_ADINR, 0xdc4), - RSND_GEN_S_REG(SSI9_BUSIF6_DALIGN, 0xdc8), - RSND_GEN_S_REG(SSI9_BUSIF7_MODE, 0xde0), - RSND_GEN_S_REG(SSI9_BUSIF7_ADINR, 0xde4), - RSND_GEN_S_REG(SSI9_BUSIF7_DALIGN, 0xde8), - }; - - static const struct rsnd_regmap_field_conf conf_scu[] = { - RSND_GEN_M_REG(SRC_I_BUSIF_MODE,0x0, 0x20), - RSND_GEN_M_REG(SRC_O_BUSIF_MODE,0x4, 0x20), - RSND_GEN_M_REG(SRC_BUSIF_DALIGN,0x8, 0x20), - RSND_GEN_M_REG(SRC_ROUTE_MODE0, 0xc, 0x20), - RSND_GEN_M_REG(SRC_CTRL, 0x10, 0x20), - RSND_GEN_M_REG(SRC_INT_ENABLE0, 0x18, 0x20), - RSND_GEN_M_REG(CMD_BUSIF_MODE, 0x184, 0x20), - RSND_GEN_M_REG(CMD_BUSIF_DALIGN,0x188, 0x20), - RSND_GEN_M_REG(CMD_ROUTE_SLCT, 0x18c, 0x20), - RSND_GEN_M_REG(CMD_CTRL, 0x190, 0x20), - RSND_GEN_S_REG(SCU_SYS_STATUS0, 0x1c8), - RSND_GEN_S_REG(SCU_SYS_INT_EN0, 0x1cc), - RSND_GEN_S_REG(SCU_SYS_STATUS1, 0x1d0), - RSND_GEN_S_REG(SCU_SYS_INT_EN1, 0x1d4), - RSND_GEN_M_REG(SRC_SWRSR, 0x200, 0x40), - RSND_GEN_M_REG(SRC_SRCIR, 0x204, 0x40), - RSND_GEN_M_REG(SRC_ADINR, 0x214, 0x40), - RSND_GEN_M_REG(SRC_IFSCR, 0x21c, 0x40), - RSND_GEN_M_REG(SRC_IFSVR, 0x220, 0x40), - RSND_GEN_M_REG(SRC_SRCCR, 0x224, 0x40), - RSND_GEN_M_REG(SRC_BSDSR, 0x22c, 0x40), - RSND_GEN_M_REG(SRC_BSISR, 0x238, 0x40), - RSND_GEN_M_REG(CTU_SWRSR, 0x500, 0x100), - RSND_GEN_M_REG(CTU_CTUIR, 0x504, 0x100), - RSND_GEN_M_REG(CTU_ADINR, 0x508, 0x100), - RSND_GEN_M_REG(CTU_CPMDR, 0x510, 0x100), - RSND_GEN_M_REG(CTU_SCMDR, 0x514, 0x100), - RSND_GEN_M_REG(CTU_SV00R, 0x518, 0x100), - RSND_GEN_M_REG(CTU_SV01R, 0x51c, 0x100), - RSND_GEN_M_REG(CTU_SV02R, 0x520, 0x100), - RSND_GEN_M_REG(CTU_SV03R, 0x524, 0x100), - RSND_GEN_M_REG(CTU_SV04R, 0x528, 0x100), - RSND_GEN_M_REG(CTU_SV05R, 0x52c, 0x100), - RSND_GEN_M_REG(CTU_SV06R, 0x530, 0x100), - RSND_GEN_M_REG(CTU_SV07R, 0x534, 0x100), - RSND_GEN_M_REG(CTU_SV10R, 0x538, 0x100), - RSND_GEN_M_REG(CTU_SV11R, 0x53c, 0x100), - RSND_GEN_M_REG(CTU_SV12R, 0x540, 0x100), - RSND_GEN_M_REG(CTU_SV13R, 0x544, 0x100), - RSND_GEN_M_REG(CTU_SV14R, 0x548, 0x100), - RSND_GEN_M_REG(CTU_SV15R, 0x54c, 0x100), - RSND_GEN_M_REG(CTU_SV16R, 0x550, 0x100), - RSND_GEN_M_REG(CTU_SV17R, 0x554, 0x100), - RSND_GEN_M_REG(CTU_SV20R, 0x558, 0x100), - RSND_GEN_M_REG(CTU_SV21R, 0x55c, 0x100), - RSND_GEN_M_REG(CTU_SV22R, 0x560, 0x100), - RSND_GEN_M_REG(CTU_SV23R, 0x564, 0x100), - RSND_GEN_M_REG(CTU_SV24R, 0x568, 0x100), - RSND_GEN_M_REG(CTU_SV25R, 0x56c, 0x100), - RSND_GEN_M_REG(CTU_SV26R, 0x570, 0x100), - RSND_GEN_M_REG(CTU_SV27R, 0x574, 0x100), - RSND_GEN_M_REG(CTU_SV30R, 0x578, 0x100), - RSND_GEN_M_REG(CTU_SV31R, 0x57c, 0x100), - RSND_GEN_M_REG(CTU_SV32R, 0x580, 0x100), - RSND_GEN_M_REG(CTU_SV33R, 0x584, 0x100), - RSND_GEN_M_REG(CTU_SV34R, 0x588, 0x100), - RSND_GEN_M_REG(CTU_SV35R, 0x58c, 0x100), - RSND_GEN_M_REG(CTU_SV36R, 0x590, 0x100), - RSND_GEN_M_REG(CTU_SV37R, 0x594, 0x100), - RSND_GEN_M_REG(MIX_SWRSR, 0xd00, 0x40), - RSND_GEN_M_REG(MIX_MIXIR, 0xd04, 0x40), - RSND_GEN_M_REG(MIX_ADINR, 0xd08, 0x40), - RSND_GEN_M_REG(MIX_MIXMR, 0xd10, 0x40), - RSND_GEN_M_REG(MIX_MVPDR, 0xd14, 0x40), - RSND_GEN_M_REG(MIX_MDBAR, 0xd18, 0x40), - RSND_GEN_M_REG(MIX_MDBBR, 0xd1c, 0x40), - RSND_GEN_M_REG(MIX_MDBCR, 0xd20, 0x40), - RSND_GEN_M_REG(MIX_MDBDR, 0xd24, 0x40), - RSND_GEN_M_REG(MIX_MDBER, 0xd28, 0x40), - RSND_GEN_M_REG(DVC_SWRSR, 0xe00, 0x100), - RSND_GEN_M_REG(DVC_DVUIR, 0xe04, 0x100), - RSND_GEN_M_REG(DVC_ADINR, 0xe08, 0x100), - RSND_GEN_M_REG(DVC_DVUCR, 0xe10, 0x100), - RSND_GEN_M_REG(DVC_ZCMCR, 0xe14, 0x100), - RSND_GEN_M_REG(DVC_VRCTR, 0xe18, 0x100), - RSND_GEN_M_REG(DVC_VRPDR, 0xe1c, 0x100), - RSND_GEN_M_REG(DVC_VRDBR, 0xe20, 0x100), - RSND_GEN_M_REG(DVC_VOL0R, 0xe28, 0x100), - RSND_GEN_M_REG(DVC_VOL1R, 0xe2c, 0x100), - RSND_GEN_M_REG(DVC_VOL2R, 0xe30, 0x100), - RSND_GEN_M_REG(DVC_VOL3R, 0xe34, 0x100), - RSND_GEN_M_REG(DVC_VOL4R, 0xe38, 0x100), - RSND_GEN_M_REG(DVC_VOL5R, 0xe3c, 0x100), - RSND_GEN_M_REG(DVC_VOL6R, 0xe40, 0x100), - RSND_GEN_M_REG(DVC_VOL7R, 0xe44, 0x100), - RSND_GEN_M_REG(DVC_DVUER, 0xe48, 0x100), - }; - static const struct rsnd_regmap_field_conf conf_adg[] = { - RSND_GEN_S_REG(BRRA, 0x00), - RSND_GEN_S_REG(BRRB, 0x04), - RSND_GEN_S_REG(BRGCKR, 0x08), - RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), - RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10), - RSND_GEN_S_REG(AUDIO_CLK_SEL2, 0x14), - RSND_GEN_S_REG(DIV_EN, 0x30), - RSND_GEN_S_REG(SRCIN_TIMSEL0, 0x34), - RSND_GEN_S_REG(SRCIN_TIMSEL1, 0x38), - RSND_GEN_S_REG(SRCIN_TIMSEL2, 0x3c), - RSND_GEN_S_REG(SRCIN_TIMSEL3, 0x40), - RSND_GEN_S_REG(SRCIN_TIMSEL4, 0x44), - RSND_GEN_S_REG(SRCOUT_TIMSEL0, 0x48), - RSND_GEN_S_REG(SRCOUT_TIMSEL1, 0x4c), - RSND_GEN_S_REG(SRCOUT_TIMSEL2, 0x50), - RSND_GEN_S_REG(SRCOUT_TIMSEL3, 0x54), - RSND_GEN_S_REG(SRCOUT_TIMSEL4, 0x58), - RSND_GEN_S_REG(CMDOUT_TIMSEL, 0x5c), - }; - static const struct rsnd_regmap_field_conf conf_ssi[] = { - RSND_GEN_M_REG(SSICR, 0x00, 0x40), - RSND_GEN_M_REG(SSISR, 0x04, 0x40), - RSND_GEN_M_REG(SSITDR, 0x08, 0x40), - RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40), - RSND_GEN_M_REG(SSIWSR, 0x20, 0x40), - }; - int ret_ssiu; - int ret_scu; - int ret_adg; - int ret_ssi; - - ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, "ssiu", conf_ssiu); - ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, "scu", conf_scu); - ret_adg = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_ADG, "adg", conf_adg); - ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, "ssi", conf_ssi); - if (ret_ssiu < 0 || - ret_scu < 0 || - ret_adg < 0 || - ret_ssi < 0) - return ret_ssiu | ret_scu | ret_adg | ret_ssi; - - return 0; + /* + * ssi : SSI0 - SSI9 + * ssiu: SSIU0 - SSIU9 + * scu : SRC0 - SRC9 etc + */ + int ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, "ssi", conf_common_ssi); + int ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, "ssiu", conf_common_ssiu); + int ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, "scu", conf_common_scu); + int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN2_ADG, "adg", conf_common_adg); + + return ret_ssi | ret_ssiu | ret_scu | ret_adg; } /* @@ -505,30 +457,13 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) static int rsnd_gen1_probe(struct rsnd_priv *priv) { - static const struct rsnd_regmap_field_conf conf_adg[] = { - RSND_GEN_S_REG(BRRA, 0x00), - RSND_GEN_S_REG(BRRB, 0x04), - RSND_GEN_S_REG(BRGCKR, 0x08), - RSND_GEN_S_REG(AUDIO_CLK_SEL0, 0x0c), - RSND_GEN_S_REG(AUDIO_CLK_SEL1, 0x10), - }; - static const struct rsnd_regmap_field_conf conf_ssi[] = { - RSND_GEN_M_REG(SSICR, 0x00, 0x40), - RSND_GEN_M_REG(SSISR, 0x04, 0x40), - RSND_GEN_M_REG(SSITDR, 0x08, 0x40), - RSND_GEN_M_REG(SSIRDR, 0x0c, 0x40), - RSND_GEN_M_REG(SSIWSR, 0x20, 0x40), - }; - int ret_adg; - int ret_ssi; - - ret_adg = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_ADG, "adg", conf_adg); - ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, "ssi", conf_ssi); - if (ret_adg < 0 || - ret_ssi < 0) - return ret_adg | ret_ssi; + /* + * ssi : SSI0 - SSI8 + */ + int ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, "ssi", conf_common_ssi); + int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN1_ADG, "adg", conf_common_adg); - return 0; + return ret_adg | ret_ssi; } /* diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index da716b1f52e48..e063286cc3287 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -200,7 +200,6 @@ enum rsnd_reg { SSI_SYS_INT_ENABLE5, SSI_SYS_INT_ENABLE6, SSI_SYS_INT_ENABLE7, - SSI_BUSIF, HDMI0_SEL, HDMI1_SEL, SSI9_BUSIF0_MODE, From 6e4e5432942a57f4c3e6a5a4a97a4d1a164dca61 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 2 Apr 2024 02:24:01 +0000 Subject: [PATCH 139/603] ASoC: rsnd: don't get resource from ID All SoC has reg-name, no need to support getting resource from ID any more. Remove it. To get physical address for DMA settings, it still need to know the index, but it is no longer fixed. This patch tidyup it. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87zfuca4e6.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/cmd.c | 2 +- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sh/rcar/dma.c | 6 +++--- sound/soc/sh/rcar/dvc.c | 2 +- sound/soc/sh/rcar/gen.c | 22 ++++++++++------------ sound/soc/sh/rcar/mix.c | 2 +- sound/soc/sh/rcar/rsnd.h | 19 +++++-------------- sound/soc/sh/rcar/src.c | 6 +++--- sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/sh/rcar/ssiu.c | 2 +- 10 files changed, 27 insertions(+), 38 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index 329e6ab1b2220..ec086d8e4d443 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -119,7 +119,7 @@ static void rsnd_cmd_debug_info(struct seq_file *m, struct rsnd_dai_stream *io, struct rsnd_mod *mod) { - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU, 0x180 + rsnd_mod_id_raw(mod) * 0x20, 0x30); } #define DEBUG_INFO .debug_info = rsnd_cmd_debug_info diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index e39eb2ac7e955..a35fc5ef87703 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -284,7 +284,7 @@ static void rsnd_ctu_debug_info(struct seq_file *m, struct rsnd_dai_stream *io, struct rsnd_mod *mod) { - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU, 0x500 + rsnd_mod_id_raw(mod) * 0x100, 0x100); } #define DEBUG_INFO .debug_info = rsnd_ctu_debug_info diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 1c494e521463c..7b499eee50806 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -585,8 +585,8 @@ rsnd_gen2_dma_addr(struct rsnd_dai_stream *io, { struct rsnd_priv *priv = rsnd_io_to_priv(io); struct device *dev = rsnd_priv_to_dev(priv); - phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SSI); - phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_GEN2_SCU); + phys_addr_t ssi_reg = rsnd_gen_get_phy_addr(priv, RSND_BASE_SSI); + phys_addr_t src_reg = rsnd_gen_get_phy_addr(priv, RSND_BASE_SCU); int is_ssi = !!(rsnd_io_to_mod_ssi(io) == mod) || !!(rsnd_io_to_mod_ssiu(io) == mod); int use_src = !!rsnd_io_to_mod_src(io); @@ -666,7 +666,7 @@ rsnd_gen4_dma_addr(struct rsnd_dai_stream *io, struct rsnd_mod *mod, int is_play, int is_from) { struct rsnd_priv *priv = rsnd_io_to_priv(io); - phys_addr_t addr = rsnd_gen_get_phy_addr(priv, RSND_GEN4_SDMC); + phys_addr_t addr = rsnd_gen_get_phy_addr(priv, RSND_BASE_SDMC); int id = rsnd_mod_id(mod); int busif = rsnd_mod_id_sub(mod); diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 16befcbc312cb..f349d6ab9fe55 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -294,7 +294,7 @@ static void rsnd_dvc_debug_info(struct seq_file *m, struct rsnd_dai_stream *io, struct rsnd_mod *mod) { - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU, 0xe00 + rsnd_mod_id(mod) * 0x100, 0x60); } #define DEBUG_INFO .debug_info = rsnd_dvc_debug_info diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index e566712e52346..d1f20cde66be1 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -177,8 +177,6 @@ static int _rsnd_gen_regmap_init(struct rsnd_priv *priv, regc.name = name; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, name); - if (!res) - res = platform_get_resource(pdev, IORESOURCE_MEM, reg_id); if (!res) return -ENODEV; @@ -425,10 +423,10 @@ static int rsnd_gen4_probe(struct rsnd_priv *priv) * ssiu: SSIU0 * ssi : SSI0 */ - int ret_ssiu = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SSIU, "ssiu", conf_common_ssiu); - int ret_ssi = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SSI, "ssi", conf_common_ssi); - int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_ADG, "adg", conf_common_adg); - int ret_sdmc = rsnd_gen_regmap_init(priv, 1, RSND_GEN4_SDMC, "sdmc", conf_null); + int ret_ssiu = rsnd_gen_regmap_init(priv, 1, RSND_BASE_SSIU, "ssiu", conf_common_ssiu); + int ret_ssi = rsnd_gen_regmap_init(priv, 1, RSND_BASE_SSI, "ssi", conf_common_ssi); + int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_BASE_ADG, "adg", conf_common_adg); + int ret_sdmc = rsnd_gen_regmap_init(priv, 1, RSND_BASE_SDMC, "sdmc", conf_null); return ret_adg | ret_ssiu | ret_ssi | ret_sdmc; } @@ -443,10 +441,10 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) * ssiu: SSIU0 - SSIU9 * scu : SRC0 - SRC9 etc */ - int ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSI, "ssi", conf_common_ssi); - int ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SSIU, "ssiu", conf_common_ssiu); - int ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_GEN2_SCU, "scu", conf_common_scu); - int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN2_ADG, "adg", conf_common_adg); + int ret_ssi = rsnd_gen_regmap_init(priv, 10, RSND_BASE_SSI, "ssi", conf_common_ssi); + int ret_ssiu = rsnd_gen_regmap_init(priv, 10, RSND_BASE_SSIU, "ssiu", conf_common_ssiu); + int ret_scu = rsnd_gen_regmap_init(priv, 10, RSND_BASE_SCU, "scu", conf_common_scu); + int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_BASE_ADG, "adg", conf_common_adg); return ret_ssi | ret_ssiu | ret_scu | ret_adg; } @@ -460,8 +458,8 @@ static int rsnd_gen1_probe(struct rsnd_priv *priv) /* * ssi : SSI0 - SSI8 */ - int ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_GEN1_SSI, "ssi", conf_common_ssi); - int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_GEN1_ADG, "adg", conf_common_adg); + int ret_ssi = rsnd_gen_regmap_init(priv, 9, RSND_BASE_SSI, "ssi", conf_common_ssi); + int ret_adg = rsnd_gen_regmap_init(priv, 1, RSND_BASE_ADG, "adg", conf_common_adg); return ret_adg | ret_ssi; } diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index 1de0e085804cc..e724103a2e8de 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -259,7 +259,7 @@ static void rsnd_mix_debug_info(struct seq_file *m, struct rsnd_dai_stream *io, struct rsnd_mod *mod) { - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU, 0xd00 + rsnd_mod_id(mod) * 0x40, 0x30); } #define DEBUG_INFO .debug_info = rsnd_mix_debug_info diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e063286cc3287..45cf213202805 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -20,20 +20,11 @@ #include #include -#define RSND_GEN1_SRU 0 -#define RSND_GEN1_ADG 1 -#define RSND_GEN1_SSI 2 - -#define RSND_GEN2_SCU 0 -#define RSND_GEN2_ADG 1 -#define RSND_GEN2_SSIU 2 -#define RSND_GEN2_SSI 3 - -#define RSND_GEN4_ADG 0 -#define RSND_GEN4_SSIU 1 -#define RSND_GEN4_SSI 2 -#define RSND_GEN4_SDMC 3 - +#define RSND_BASE_ADG 0 +#define RSND_BASE_SSI 1 +#define RSND_BASE_SSIU 2 +#define RSND_BASE_SCU 3 // for Gen2/Gen3 +#define RSND_BASE_SDMC 3 // for Gen4 reuse #define RSND_BASE_MAX 4 /* diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 3241a1bdc9eaf..8822d50b6d86c 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -606,13 +606,13 @@ static void rsnd_src_debug_info(struct seq_file *m, struct rsnd_dai_stream *io, struct rsnd_mod *mod) { - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU, rsnd_mod_id(mod) * 0x20, 0x20); seq_puts(m, "\n"); - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU, 0x1c0, 0x20); seq_puts(m, "\n"); - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SCU, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SCU, 0x200 + rsnd_mod_id(mod) * 0x40, 0x40); } #define DEBUG_INFO .debug_info = rsnd_src_debug_info diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 0a46aa1975fa8..8d2a86383ae01 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1049,7 +1049,7 @@ static void rsnd_ssi_debug_info(struct seq_file *m, seq_printf(m, "chan: %d\n", ssi->chan); seq_printf(m, "user: %d\n", ssi->usrcnt); - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SSI, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SSI, rsnd_mod_id(mod) * 0x40, 0x40); } #define DEBUG_INFO .debug_info = rsnd_ssi_debug_info diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 17bd8cc86dd0a..665e8b2db579e 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -413,7 +413,7 @@ static void rsnd_ssiu_debug_info(struct seq_file *m, struct rsnd_dai_stream *io, struct rsnd_mod *mod) { - rsnd_debugfs_mod_reg_show(m, mod, RSND_GEN2_SSIU, + rsnd_debugfs_mod_reg_show(m, mod, RSND_BASE_SSIU, rsnd_mod_id(mod) * 0x80, 0x80); } #define DEBUG_INFO .debug_info = rsnd_ssiu_debug_info From 0b8ef53e120981c218ba037dddda98f414af2207 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 2 Apr 2024 02:24:09 +0000 Subject: [PATCH 140/603] ASoC: rsnd: rename rsnd_is_e3() to rsnd_is_gen3_e3() Renesas Sound driver is using rsnd_is_genX() macro to handling difference. We can use "grep rsnd_is_gen" to find-out difference for each SoC except rsnd_is_e3(). Let's put same rule for E3 as well. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87y19wa4dy.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 45cf213202805..ff294aa2d6407 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -703,7 +703,7 @@ struct rsnd_priv { #define rsnd_is_gen2(priv) (((priv)->flags & RSND_GEN_MASK) == RSND_GEN2) #define rsnd_is_gen3(priv) (((priv)->flags & RSND_GEN_MASK) == RSND_GEN3) #define rsnd_is_gen4(priv) (((priv)->flags & RSND_GEN_MASK) == RSND_GEN4) -#define rsnd_is_e3(priv) (((priv)->flags & \ +#define rsnd_is_gen3_e3(priv) (((priv)->flags & \ (RSND_GEN_MASK | RSND_SOC_MASK)) == \ (RSND_GEN3 | RSND_SOC_E)) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 8822d50b6d86c..431e6d195b096 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -310,7 +310,7 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, /* * E3 need to overwrite */ - if (rsnd_is_e3(priv)) + if (rsnd_is_gen3_e3(priv)) switch (rsnd_mod_id(mod)) { case 0: case 4: From 5be0e7f7f275aa5f05d8708db021cef0fed749d1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 2 Apr 2024 02:24:15 +0000 Subject: [PATCH 141/603] ASoC: rsnd: R-Car Gen1/Gen2 exception Renesas Sound driver is using rsnd_is_genX() macro to handling difference. We can use "grep rsnd_is_gen" to find-out difference for each SoC today. In general, driver uses flag to handling SoC difference. But in Renesas Sound case, the differences are very specific, if you try to make it general, the code will become be very complicated. Some of them can use flag to handling, but we can find-out all diffence by using "grep rsnd_is_gen" today. It is not so good, but not so bad so far. So we will keep current style for a while. This patch cares Gen1/Gen2 exception by using negative handling to makes no more update for this part when we support Gen5 or later. - if (rsnd_is_gen3() || rsnd_is_gen4() || rsnd_is_gen5() || ...) + if (!rsnd_is_gen1() && !rsnd_is_gen2()) Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87wmpga4ds.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 0b1aa23c11897..6bc7027ed4dbf 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1512,7 +1512,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) continue; for_each_endpoint_of_node(ports, dai_np) { __rsnd_dai_probe(priv, dai_np, dai_np, 0, dai_i); - if (rsnd_is_gen3(priv) || rsnd_is_gen4(priv)) { + if (!rsnd_is_gen1(priv) && !rsnd_is_gen2(priv)) { rdai = rsnd_rdai_get(priv, dai_i); rsnd_parse_connect_graph(priv, &rdai->playback, dai_np); @@ -1531,7 +1531,7 @@ static int rsnd_dai_probe(struct rsnd_priv *priv) for_each_child_of_node(node, dai_np) { __rsnd_dai_probe(priv, dai_np, np, dai_i, dai_i); - if (rsnd_is_gen3(priv) || rsnd_is_gen4(priv)) { + if (!rsnd_is_gen1(priv) && !rsnd_is_gen2(priv)) { rdai = rsnd_rdai_get(priv, dai_i); rsnd_parse_connect_simple(priv, &rdai->playback, dai_np); From 07f6232ff1c9909cd87e42020c91b265a58918da Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 2 Apr 2024 02:24:22 +0000 Subject: [PATCH 142/603] ASoC: rsnd: no exception for SCU R-Car Gen1 and Gen4 doesn't have SCU, and current driver checks Gen1 (only) on each devices (A). But these drivers are assuming it might be not enabled after that (B). /* This driver doesn't support Gen1 at this point */ (A) if (rsnd_is_gen1(priv)) return 0; (B) node = rsnd_src_of_node(priv); if (!node) return 0; /* not used is not error */ Gen4 DT doesn't have these device settings. (A) check doesn't work for it, but (B) check handling it. Gen1 DT doesn't have these device settings either, and (A) is handling it, but (B) only is very enough. (A) is no longer needed. This patch removes (A) Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/87v850a4dl.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/cmd.c | 4 ---- sound/soc/sh/rcar/ctu.c | 4 ---- sound/soc/sh/rcar/dvc.c | 4 ---- sound/soc/sh/rcar/mix.c | 4 ---- sound/soc/sh/rcar/src.c | 4 ---- 5 files changed, 20 deletions(-) diff --git a/sound/soc/sh/rcar/cmd.c b/sound/soc/sh/rcar/cmd.c index ec086d8e4d443..8d9a1e345a22c 100644 --- a/sound/soc/sh/rcar/cmd.c +++ b/sound/soc/sh/rcar/cmd.c @@ -157,10 +157,6 @@ int rsnd_cmd_probe(struct rsnd_priv *priv) struct rsnd_cmd *cmd; int i, nr; - /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) - return 0; - /* same number as DVC */ nr = priv->dvc_nr; if (!nr) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index a35fc5ef87703..a26ec7b780cd6 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -323,10 +323,6 @@ int rsnd_ctu_probe(struct rsnd_priv *priv) char name[CTU_NAME_SIZE]; int i, nr, ret; - /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) - return 0; - node = rsnd_ctu_of_node(priv); if (!node) return 0; /* not used is not error */ diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index f349d6ab9fe55..da91dd301aabe 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -331,10 +331,6 @@ int rsnd_dvc_probe(struct rsnd_priv *priv) char name[RSND_DVC_NAME_SIZE]; int i, nr, ret; - /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) - return 0; - node = rsnd_dvc_of_node(priv); if (!node) return 0; /* not used is not error */ diff --git a/sound/soc/sh/rcar/mix.c b/sound/soc/sh/rcar/mix.c index e724103a2e8de..024d91cc87484 100644 --- a/sound/soc/sh/rcar/mix.c +++ b/sound/soc/sh/rcar/mix.c @@ -295,10 +295,6 @@ int rsnd_mix_probe(struct rsnd_priv *priv) char name[MIX_NAME_SIZE]; int i, nr, ret; - /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) - return 0; - node = rsnd_mix_of_node(priv); if (!node) return 0; /* not used is not error */ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 431e6d195b096..e7f86db0d94c3 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -652,10 +652,6 @@ int rsnd_src_probe(struct rsnd_priv *priv) char name[RSND_SRC_NAME_SIZE]; int i, nr, ret; - /* This driver doesn't support Gen1 at this point */ - if (rsnd_is_gen1(priv)) - return 0; - node = rsnd_src_of_node(priv); if (!node) return 0; /* not used is not error */ From 91ebd32eee6cd7714b49ca4463a0f315bc26ce0e Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Mon, 1 Apr 2024 18:47:53 +0800 Subject: [PATCH 143/603] ASoC: nau8325: Revise soc_enum structure definition. This patch is modified nau8325_dac_oversampl_enum enumerated. Use SOC_VALUE_ENUM instead to hide empty (presumably invalid) options. Signed-off-by: Seven Lee Link: https://msgid.link/r/20240401104753.3432507-1-wtli@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8325.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/nau8325.c b/sound/soc/codecs/nau8325.c index a00a30364c897..d65f731445977 100644 --- a/sound/soc/codecs/nau8325.c +++ b/sound/soc/codecs/nau8325.c @@ -174,13 +174,20 @@ static bool nau8325_volatile_reg(struct device *dev, unsigned int reg) } } -static const char * const nau8325_dac_oversampl[] = { - "64", "256", "128", "", "32" }; +static const char * const nau8325_dac_oversampl_texts[] = { + "64", "256", "128", "32", +}; + +static const unsigned int nau8325_dac_oversampl_values[] = { + 0, 1, 2, 4, +}; static const struct soc_enum nau8325_dac_oversampl_enum = - SOC_ENUM_SINGLE(NAU8325_R29_DAC_CTRL1, NAU8325_DAC_OVERSAMPLE_SFT, - ARRAY_SIZE(nau8325_dac_oversampl), - nau8325_dac_oversampl); + SOC_VALUE_ENUM_SINGLE(NAU8325_R29_DAC_CTRL1, + NAU8325_DAC_OVERSAMPLE_SFT, 0x7, + ARRAY_SIZE(nau8325_dac_oversampl_texts), + nau8325_dac_oversampl_texts, + nau8325_dac_oversampl_values); static const DECLARE_TLV_DB_MINMAX_MUTE(dac_vol_tlv, -8000, 600); From f690cdcc01a5b549715fefe22a98962e7672516b Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Tue, 2 Apr 2024 09:59:56 -0500 Subject: [PATCH 144/603] ASoC: SOF: ipc4-loader: save FW version info to debugfs To check loaded FW version: $ hexdump -C /sys/kernel/debug/sof/fw_version 00000000 02 00 07 00 63 00 01 00 00 00 00 00 00 00 00 00 |....c...........| Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Signed-off-by: Yong Zhi Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402145959.172619-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-loader.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index c79479afa8d0d..641c4f24cca98 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -80,6 +80,14 @@ static ssize_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev, dev_dbg(sdev->dev, "Header length: %u, module count: %u\n", fw_header->len, fw_header->num_module_entries); + /* copy the fw_version of basefw into debugfs at first boot */ + if (fw == sdev->basefw.fw) { + sdev->fw_version.major = fw_header->major_version; + sdev->fw_version.minor = fw_header->minor_version; + sdev->fw_version.micro = fw_header->hotfix_version; + sdev->fw_version.build = fw_header->build_version; + } + fw_lib->modules = devm_kmalloc_array(sdev->dev, fw_header->num_module_entries, sizeof(*fw_module), GFP_KERNEL); if (!fw_lib->modules) From dbb6ca68b55ddf23d0b6de782c7641624a285fc2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 09:59:57 -0500 Subject: [PATCH 145/603] ASoC: SOF: amd: acp-loader: abort firmware download on write error cppcheck reports this issue: sound/soc/sof/amd/acp-loader.c:299:6: style: Variable 'ret' is reassigned a value before the old one has been used. [redundantAssignment] ret = request_firmware(&adata->fw_dbin, fw_filename, sdev->dev); ^ sound/soc/sof/amd/acp-loader.c:289:6: note: ret is assigned ret = snd_sof_dsp_block_write(sdev, SOF_FW_BLK_TYPE_IRAM, 0, ^ sound/soc/sof/amd/acp-loader.c:299:6: note: ret is overwritten ret = request_firmware(&adata->fw_dbin, fw_filename, sdev->dev); ^ This behavior is probably unintentional, there's no reason to return an error for the DRAM but not the IRAM. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240402145959.172619-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-loader.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/amd/acp-loader.c b/sound/soc/sof/amd/acp-loader.c index aad904839b817..2d5e58846499d 100644 --- a/sound/soc/sof/amd/acp-loader.c +++ b/sound/soc/sof/amd/acp-loader.c @@ -289,6 +289,8 @@ int acp_sof_load_signed_firmware(struct snd_sof_dev *sdev) ret = snd_sof_dsp_block_write(sdev, SOF_FW_BLK_TYPE_IRAM, 0, (void *)sdev->basefw.fw->data, sdev->basefw.fw->size); + if (ret < 0) + return ret; fw_filename = kasprintf(GFP_KERNEL, "%s/%s", plat_data->fw_filename_prefix, From 458e3870507f7ebd26a2f5c7e925d5b31a873114 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 09:59:58 -0500 Subject: [PATCH 146/603] ASoC: SOF: ipc4-priv: align prototype and function declaration Somehow instance_id and id were mixed. Align on instance_id for consistency. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240402145959.172619-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-priv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index f3b908b093f95..85e8cf91aab9e 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -112,7 +112,7 @@ extern const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops; extern const struct sof_ipc_pcm_ops ipc4_pcm_ops; extern const struct sof_ipc_fw_tracing_ops ipc4_mtrace_ops; -int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 id, u32 state); +int sof_ipc4_set_pipeline_state(struct snd_sof_dev *sdev, u32 instance_id, u32 state); int sof_ipc4_mtrace_update_pos(struct snd_sof_dev *sdev, int core); int sof_ipc4_query_fw_configuration(struct snd_sof_dev *sdev); From c143cfe4f87070f11d7550b38f72625b51bf229f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 09:59:59 -0500 Subject: [PATCH 147/603] ASoC: SOF: ipc4-topology: remove shadowed variable Cppcheck reports this: sound/soc/sof/ipc4-topology.c:569:23: style: Local variable 'sdev' shadows outer variable [shadowVariable] struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); ^ sound/soc/sof/ipc4-topology.c:512:22: note: Shadowed declaration struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); ^ sound/soc/sof/ipc4-topology.c:569:23: note: Shadow variable struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); ^ Remove shadowed variable. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240402145959.172619-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index da4a83afb87a8..651bff2122a3b 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -566,7 +566,6 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget) switch (ipc4_copier->dai_type) { case SOF_DAI_INTEL_ALH: { - struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct sof_ipc4_alh_configuration_blob *blob; struct snd_soc_dapm_path *p; struct snd_sof_widget *w; From 831045513c8a2ef14c3cf39b33d1ccedf588c4a8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:12 -0500 Subject: [PATCH 148/603] ASoC: SOF: Intel: hda-dai: fix channel map configuration for aggregated dailink The existing code derives the channel map used to program the HDaudio link DMA from the hw_params, but that is not quite right in the case of aggregation. The code in soc-pcm.c splits the hw_params depending on the codec_ch_map, and we need to reconstruct the channel-map to insert the data in the right places. This issue is seen only on amplifier feedback capture where the data from the second amplifier was replaced by that of the first amplifier. Note that the loop iterator of the macro for_each_rtd_cpu_dais() is reused in a following loop. This is different to all existing usages of that macro, hence the use of a boolean flag to avoid an access to an uninitialized variable. Fixes: 2960ee5c4814 ("ASoC: SOF: Intel: hda-dai: add helpers for SoundWire callbacks") Reviewed-by: Bard Liao Reviewed-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c1682bcdb5a66..6a39ca632f55e 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -439,10 +439,17 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, int link_id) { struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); const struct hda_dai_widget_dma_ops *ops; + struct snd_soc_dai_link_ch_map *ch_maps; struct hdac_ext_stream *hext_stream; + struct snd_soc_dai *dai; struct snd_sof_dev *sdev; + bool cpu_dai_found = false; + int cpu_dai_id; + int ch_mask; int ret; + int j; ret = non_hda_dai_hw_params(substream, params, cpu_dai); if (ret < 0) { @@ -457,9 +464,29 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, if (!hext_stream) return -ENODEV; - /* in the case of SoundWire we need to program the PCMSyCM registers */ + /* + * in the case of SoundWire we need to program the PCMSyCM registers. In case + * of aggregated devices, we need to define the channel mask for each sublink + * by reconstructing the split done in soc-pcm.c + */ + for_each_rtd_cpu_dais(rtd, cpu_dai_id, dai) { + if (dai == cpu_dai) { + cpu_dai_found = true; + break; + } + } + + if (!cpu_dai_found) + return -ENODEV; + + ch_mask = 0; + for_each_link_ch_maps(rtd->dai_link, j, ch_maps) { + if (ch_maps->cpu == cpu_dai_id) + ch_mask |= ch_maps->ch_mask; + } + ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id, - GENMASK(params_channels(params) - 1, 0), + ch_mask, hdac_stream(hext_stream)->stream_tag, substream->stream); if (ret < 0) { From 24b1f93df400e1ab1731e7bcb320e693a6a73792 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 2 Apr 2024 10:18:13 -0500 Subject: [PATCH 149/603] Revert "ASoC: SOF: Intel: hda-dai-ops: reset device count for SoundWire DAIs" This reverts commit 699e146d9ebf42ee2a5d4e4e28f7a49c4aef0105. Don't reset device_count as we will use the multi-gateway firmware configuration. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 24 +----------------------- 1 file changed, 1 insertion(+), 23 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index c50ca9e72d373..d3e168ad11804 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -435,28 +435,6 @@ static int hda_ipc4_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *c return ret; } -static struct hdac_ext_stream *sdw_hda_ipc4_get_hext_stream(struct snd_sof_dev *sdev, - struct snd_soc_dai *cpu_dai, - struct snd_pcm_substream *substream) -{ - struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); - struct snd_sof_widget *swidget = w->dobj.private; - struct snd_sof_dai *dai = swidget->private; - struct sof_ipc4_copier *ipc4_copier = dai->private; - struct sof_ipc4_alh_configuration_blob *blob; - - blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; - - /* - * Starting with ACE_2_0, re-setting the device_count is mandatory to avoid using - * the multi-gateway firmware configuration. The DMA hardware can take care of - * multiple links without needing any firmware assistance - */ - blob->alh_cfg.device_count = 1; - - return hda_ipc4_get_hext_stream(sdev, cpu_dai, substream); -} - static const struct hda_dai_widget_dma_ops hda_ipc4_dma_ops = { .get_hext_stream = hda_ipc4_get_hext_stream, .assign_hext_stream = hda_assign_hext_stream, @@ -498,7 +476,7 @@ static const struct hda_dai_widget_dma_ops dmic_ipc4_dma_ops = { }; static const struct hda_dai_widget_dma_ops sdw_ipc4_dma_ops = { - .get_hext_stream = sdw_hda_ipc4_get_hext_stream, + .get_hext_stream = hda_ipc4_get_hext_stream, .assign_hext_stream = hda_assign_hext_stream, .release_hext_stream = hda_release_hext_stream, .setup_hext_stream = hda_setup_hext_stream, From 1d0fb3d0c30749779cdd88be98761b17ebfe5590 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 2 Apr 2024 10:18:14 -0500 Subject: [PATCH 150/603] Revert "ASoC: SOF: Intel: hda-dai-ops: only allocate/release streams for first CPU DAI" This reverts commit f8ba62ac863c33fc0d8ac3f1270985c2b77f4377. The SoundWire aggregated solution was to use one DMA on multiple links. But, the solution changed to use one DMA for each link. It means that we should assign HDaudio stream_tag for each cpu_dai. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai-ops.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index d3e168ad11804..1afdb06499a37 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -145,17 +145,9 @@ static struct hdac_ext_stream *hda_assign_hext_stream(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; struct hdac_ext_stream *hext_stream; - /* only allocate a stream_tag for the first DAI in the dailink */ - dai = snd_soc_rtd_to_cpu(rtd, 0); - if (dai == cpu_dai) - hext_stream = hda_link_stream_assign(sof_to_bus(sdev), substream); - else - hext_stream = snd_soc_dai_get_dma_data(dai, substream); - + hext_stream = hda_link_stream_assign(sof_to_bus(sdev), substream); if (!hext_stream) return NULL; @@ -168,14 +160,9 @@ static void hda_release_hext_stream(struct snd_sof_dev *sdev, struct snd_soc_dai struct snd_pcm_substream *substream) { struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream); - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - /* only release a stream_tag for the first DAI in the dailink */ - dai = snd_soc_rtd_to_cpu(rtd, 0); - if (dai == cpu_dai) - snd_hdac_ext_stream_release(hext_stream, HDAC_EXT_STREAM_TYPE_LINK); snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); + snd_hdac_ext_stream_release(hext_stream, HDAC_EXT_STREAM_TYPE_LINK); } static void hda_setup_hext_stream(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream, From e9c6b118de1afc1d32a4eb3bc9f3d114d4fe0f1a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 2 Apr 2024 10:18:15 -0500 Subject: [PATCH 151/603] ASoC: SOF: make dma_config_tlv be an array Each stream needs a dma_config_tlv. We will handle multi dma_config_tlv in the follow up commits. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 2 +- sound/soc/sof/ipc4-topology.c | 25 ++++++++++++------------- sound/soc/sof/ipc4-topology.h | 2 +- 3 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 6a39ca632f55e..01c544b7e0467 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -392,7 +392,7 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, /* configure TLV */ ipc4_copier = widget_to_copier(w); - dma_config_tlv = &ipc4_copier->dma_config_tlv; + dma_config_tlv = &ipc4_copier->dma_config_tlv[0]; dma_config_tlv->type = SOF_IPC4_GTW_DMA_CONFIG_ID; /* dma_config_priv_size is zero */ dma_config_tlv->length = sizeof(dma_config_tlv->dma_config); diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index da4a83afb87a8..1e9276b9b35ce 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1454,6 +1454,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, u32 deep_buffer_dma_ms = 0; int output_fmt_index; bool single_output_format; + int i; dev_dbg(sdev->dev, "copier %s, type %d", swidget->widget->name, swidget->id); @@ -1679,7 +1680,6 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, u32 ch_map; u32 step; u32 mask; - int i; blob = (struct sof_ipc4_alh_configuration_blob *)ipc4_copier->copier_config; @@ -1789,19 +1789,18 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, gtw_cfg_config_length = copier_data->gtw_cfg.config_length * 4; ipc_size = sizeof(*copier_data) + gtw_cfg_config_length; - if (ipc4_copier->dma_config_tlv.type == SOF_IPC4_GTW_DMA_CONFIG_ID && - ipc4_copier->dma_config_tlv.length) { - dma_config_tlv_size = sizeof(ipc4_copier->dma_config_tlv) + - ipc4_copier->dma_config_tlv.dma_config.dma_priv_config_size; - - /* paranoia check on TLV size/length */ - if (dma_config_tlv_size != ipc4_copier->dma_config_tlv.length + - sizeof(uint32_t) * 2) { - dev_err(sdev->dev, "Invalid configuration, TLV size %d length %d\n", - dma_config_tlv_size, ipc4_copier->dma_config_tlv.length); - return -EINVAL; - } + dma_config_tlv_size = 0; + for (i = 0; i < SOF_IPC4_DMA_DEVICE_MAX_COUNT; i++) { + if (ipc4_copier->dma_config_tlv[i].type != SOF_IPC4_GTW_DMA_CONFIG_ID) + continue; + dma_config_tlv_size += ipc4_copier->dma_config_tlv[i].length; + dma_config_tlv_size += + ipc4_copier->dma_config_tlv[i].dma_config.dma_priv_config_size; + dma_config_tlv_size += (sizeof(ipc4_copier->dma_config_tlv[i]) - + sizeof(ipc4_copier->dma_config_tlv[i].dma_config)); + } + if (dma_config_tlv_size) { ipc_size += dma_config_tlv_size; /* we also need to increase the size at the gtw level */ diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index dce174a190ddc..aa5122c3721d3 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -313,7 +313,7 @@ struct sof_ipc4_copier { struct sof_ipc4_gtw_attributes *gtw_attr; u32 dai_type; int dai_index; - struct sof_ipc4_dma_config_tlv dma_config_tlv; + struct sof_ipc4_dma_config_tlv dma_config_tlv[SOF_IPC4_DMA_DEVICE_MAX_COUNT]; }; /** From 8fa10a243600ca8bd92fdc871100deb308fca5f1 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 2 Apr 2024 10:18:16 -0500 Subject: [PATCH 152/603] ASoC: SOF: Intel: hda-dai: set lowest N bits in ch_mask We always use the lowest N channels of stream. So, set ch_mask to GENMASK(params_channels(params) - 1, 0). Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 01c544b7e0467..810d2997794f0 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -441,7 +441,6 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); const struct hda_dai_widget_dma_ops *ops; - struct snd_soc_dai_link_ch_map *ch_maps; struct hdac_ext_stream *hext_stream; struct snd_soc_dai *dai; struct snd_sof_dev *sdev; @@ -449,7 +448,6 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, int cpu_dai_id; int ch_mask; int ret; - int j; ret = non_hda_dai_hw_params(substream, params, cpu_dai); if (ret < 0) { @@ -479,11 +477,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, if (!cpu_dai_found) return -ENODEV; - ch_mask = 0; - for_each_link_ch_maps(rtd->dai_link, j, ch_maps) { - if (ch_maps->cpu == cpu_dai_id) - ch_mask |= ch_maps->ch_mask; - } + ch_mask = GENMASK(params_channels(params) - 1, 0); ret = hdac_bus_eml_sdw_map_stream_ch(sof_to_bus(sdev), link_id, cpu_dai->id, ch_mask, From 17386cb1b48b0d85f69b21ae13d5408d67180e30 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 2 Apr 2024 10:18:17 -0500 Subject: [PATCH 153/603] ASoC: SOF: Intel: hda-dai: set dma_stream_channel_map device sof_ipc4_dma_config_tlv{} is required for ACE2.x. The patch follow the convention to set the dma_stream_channel_map.mapping device as "link_id << 8 | pdi_id". And the mapping in sof_ipc4_alh_configuration_blob{} should be the same as dma_stream_channel_map.mapping in sof_ipc4_dma_config{}. The purposes of device id is to map DMA tlv. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 36 +++++++++++++++++++++++++++++++++-- sound/soc/sof/ipc4-topology.c | 13 +++++++++++++ 2 files changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 810d2997794f0..de71e1595a782 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -346,6 +346,7 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_dma_config_tlv *dma_config_tlv; const struct hda_dai_widget_dma_ops *ops; struct sof_ipc4_dma_config *dma_config; @@ -353,6 +354,8 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream; struct hdac_stream *hstream; struct snd_sof_dev *sdev; + struct snd_soc_dai *dai; + int cpu_dai_id; int stream_id; int ret; @@ -392,7 +395,12 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, /* configure TLV */ ipc4_copier = widget_to_copier(w); - dma_config_tlv = &ipc4_copier->dma_config_tlv[0]; + for_each_rtd_cpu_dais(rtd, cpu_dai_id, dai) { + if (dai == cpu_dai) + break; + } + + dma_config_tlv = &ipc4_copier->dma_config_tlv[cpu_dai_id]; dma_config_tlv->type = SOF_IPC4_GTW_DMA_CONFIG_ID; /* dma_config_priv_size is zero */ dma_config_tlv->length = sizeof(dma_config_tlv->dma_config); @@ -403,7 +411,11 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, dma_config->pre_allocated_by_host = 1; dma_config->dma_channel_id = stream_id - 1; dma_config->stream_id = stream_id; - dma_config->dma_stream_channel_map.device_count = 0; /* mapping not used */ + /* + * Currently we use a DMA for each device in ALH blob. The device will + * be copied in sof_ipc4_prepare_copier_module. + */ + dma_config->dma_stream_channel_map.device_count = 1; dma_config->dma_priv_config_size = 0; skip_tlv: @@ -440,7 +452,10 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct sof_ipc4_dma_config_tlv *dma_config_tlv; const struct hda_dai_widget_dma_ops *ops; + struct sof_ipc4_dma_config *dma_config; + struct sof_ipc4_copier *ipc4_copier; struct hdac_ext_stream *hext_stream; struct snd_soc_dai *dai; struct snd_sof_dev *sdev; @@ -448,6 +463,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, int cpu_dai_id; int ch_mask; int ret; + int i; ret = non_hda_dai_hw_params(substream, params, cpu_dai); if (ret < 0) { @@ -489,6 +505,22 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, return ret; } + ipc4_copier = widget_to_copier(w); + dma_config_tlv = &ipc4_copier->dma_config_tlv[cpu_dai_id]; + dma_config = &dma_config_tlv->dma_config; + dma_config->dma_stream_channel_map.mapping[0].device = link_id << 8 | cpu_dai->id; + dma_config->dma_stream_channel_map.mapping[0].channel_mask = ch_mask; + + /* + * copy the dma_config_tlv to all ipc4_copier in the same link. Because only one copier + * will be handled in sof_ipc4_prepare_copier_module. + */ + for_each_rtd_cpu_dais(rtd, i, dai) { + w = snd_soc_dai_get_widget(dai, substream->stream); + ipc4_copier = widget_to_copier(w); + memcpy(&ipc4_copier->dma_config_tlv[cpu_dai_id], dma_config_tlv, + sizeof(*dma_config_tlv)); + } return 0; } diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 1e9276b9b35ce..cca5d43e5fd8b 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1672,6 +1672,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, */ if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { struct sof_ipc4_alh_configuration_blob *blob; + struct sof_ipc4_dma_config *dma_config; struct sof_ipc4_copier_data *alh_data; struct sof_ipc4_copier *alh_copier; struct snd_sof_widget *w; @@ -1711,6 +1712,18 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, alh_copier = (struct sof_ipc4_copier *)dai->private; alh_data = &alh_copier->data; blob->alh_cfg.mapping[i].device = alh_data->gtw_cfg.node_id; + + /* + * The mapping[i] device in ALH blob should be the same as the + * dma_config_tlv[i] mapping device if a dma_config_tlv is present. + * The device id will be used for DMA tlv mapping purposes. + */ + if (ipc4_copier->dma_config_tlv[i].length) { + dma_config = &ipc4_copier->dma_config_tlv[i].dma_config; + blob->alh_cfg.mapping[i].device = + dma_config->dma_stream_channel_map.mapping[0].device; + } + /* * Set the same channel mask for playback as the audio data is * duplicated for all speakers. For capture, split the channels From 58f32cb7011a8a15b18f35e4c0ee044aa98e365b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:18 -0500 Subject: [PATCH 154/603] ASoC: SOF: Intel: hda-dai: add helpers to set dai config We need to be able to set the dai config differently for SoundWire. Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 51 ++++++++++++++++++++++++++--------- 1 file changed, 38 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index de71e1595a782..c48ac931753cb 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -221,15 +221,15 @@ static int __maybe_unused hda_dai_hw_free(struct snd_pcm_substream *substream, return hda_link_dma_cleanup(substream, hext_stream, cpu_dai); } -static int __maybe_unused hda_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int __maybe_unused hda_dai_hw_params_data(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai, + struct snd_sof_dai_config_data *data, + unsigned int flags) { struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, substream->stream); const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, dai); struct hdac_ext_stream *hext_stream; - struct snd_sof_dai_config_data data = { 0 }; - unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS; struct snd_sof_dev *sdev = widget_to_sdev(w); int ret; @@ -249,9 +249,19 @@ static int __maybe_unused hda_dai_hw_params(struct snd_pcm_substream *substream, hext_stream = ops->get_hext_stream(sdev, dai, substream); flags |= SOF_DAI_CONFIG_FLAGS_2_STEP_STOP << SOF_DAI_CONFIG_FLAGS_QUIRK_SHIFT; - data.dai_data = hdac_stream(hext_stream)->stream_tag - 1; + data->dai_data = hdac_stream(hext_stream)->stream_tag - 1; + + return hda_dai_config(w, flags, data); +} + +static int __maybe_unused hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_sof_dai_config_data data = { 0 }; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS; - return hda_dai_config(w, flags, &data); + return hda_dai_hw_params_data(substream, params, dai, &data, flags); } /* @@ -341,9 +351,11 @@ static struct sof_ipc4_copier *widget_to_copier(struct snd_soc_dapm_widget *w) return ipc4_copier; } -static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai, + struct snd_sof_dai_config_data *data, + unsigned int flags) { struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); @@ -366,9 +378,9 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, } /* use HDaudio stream handling */ - ret = hda_dai_hw_params(substream, params, cpu_dai); + ret = hda_dai_hw_params_data(substream, params, cpu_dai, data, flags); if (ret < 0) { - dev_err(cpu_dai->dev, "%s: hda_dai_hw_params failed: %d\n", __func__, ret); + dev_err(cpu_dai->dev, "%s: hda_dai_hw_params_data failed: %d\n", __func__, ret); return ret; } @@ -422,6 +434,16 @@ static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, return 0; } +static int non_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct snd_sof_dai_config_data data = { 0 }; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS; + + return non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags); +} + static int non_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -453,6 +475,8 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sof_ipc4_dma_config_tlv *dma_config_tlv; + struct snd_sof_dai_config_data data = { 0 }; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS; const struct hda_dai_widget_dma_ops *ops; struct sof_ipc4_dma_config *dma_config; struct sof_ipc4_copier *ipc4_copier; @@ -465,7 +489,8 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, int ret; int i; - ret = non_hda_dai_hw_params(substream, params, cpu_dai); + data.dai_index = (link_id << 8) | cpu_dai->id; + ret = non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags); if (ret < 0) { dev_err(cpu_dai->dev, "%s: non_hda_dai_hw_params failed %d\n", __func__, ret); return ret; From 219271481e8965e80ee425cdc2db85230a333a97 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:19 -0500 Subject: [PATCH 155/603] ASoC: SOF: Intel: set the DMA TLV device as dai_index We've already defined the value for dai_index, let's use it instead of open-coding the same thing. No functionality change. Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c48ac931753cb..86efcbe8f0d84 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -533,7 +533,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, ipc4_copier = widget_to_copier(w); dma_config_tlv = &ipc4_copier->dma_config_tlv[cpu_dai_id]; dma_config = &dma_config_tlv->dma_config; - dma_config->dma_stream_channel_map.mapping[0].device = link_id << 8 | cpu_dai->id; + dma_config->dma_stream_channel_map.mapping[0].device = data.dai_index; dma_config->dma_stream_channel_map.mapping[0].channel_mask = ch_mask; /* From 8bc3b56cac748f6ef6a4b96c906007a546e7fb5a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:20 -0500 Subject: [PATCH 156/603] ASoC: SOF: Intel: hda: extend signature of sdw_hda_dai_hw_params() Add intel_alh_id to set the expected gateway node_id in a follow-up patch. Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 3 ++- sound/soc/sof/intel/hda.c | 3 ++- sound/soc/sof/intel/hda.h | 3 ++- 3 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 86efcbe8f0d84..5e3229c8fe132 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -470,7 +470,8 @@ static const struct snd_soc_dai_ops dmic_dai_ops = { int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai, - int link_id) + int link_id, + int intel_alh_id) { struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index e26b8fd682e5e..63f1cf3b915f4 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -159,7 +159,8 @@ static int sdw_ace2x_params_stream(struct device *dev, return sdw_hda_dai_hw_params(params_data->substream, params_data->hw_params, params_data->dai, - params_data->link_id); + params_data->link_id, + params_data->alh_stream_id); } static int sdw_ace2x_free_stream(struct device *dev, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index b36eb7c789133..3bf7427dc918b 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -844,7 +844,8 @@ static inline bool hda_common_check_sdw_irq(struct snd_sof_dev *sdev) int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai, - int link_id); + int link_id, + int intel_alh_id); int sdw_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai, From a936456d4bce27edc1a18dab270c657e9c07590c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:21 -0500 Subject: [PATCH 157/603] ASoC: SOF: IPC4: extend dai_data with node_id The node_id value needs to be handled specifically for ALH. Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 1 + sound/soc/sof/intel/hda.c | 1 + sound/soc/sof/ipc4-topology.c | 8 ++++++-- sound/soc/sof/sof-audio.h | 1 + 4 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 5e3229c8fe132..86c2325e59491 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -491,6 +491,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, int i; data.dai_index = (link_id << 8) | cpu_dai->id; + data.dai_node_id = intel_alh_id; ret = non_hda_dai_hw_params_data(substream, params, cpu_dai, &data, flags); if (ret < 0) { dev_err(cpu_dai->dev, "%s: non_hda_dai_hw_params failed %d\n", __func__, ret); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 63f1cf3b915f4..ae1a38f20bdbb 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -145,6 +145,7 @@ static int sdw_params_stream(struct device *dev, data.dai_index = (params_data->link_id << 8) | d->id; data.dai_data = params_data->alh_stream_id; + data.dai_node_id = data.dai_data; return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_PARAMS, &data); } diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index cca5d43e5fd8b..0368ef6d0807b 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -2825,7 +2825,11 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * case SOF_DAI_INTEL_HDA: gtw_attr = ipc4_copier->gtw_attr; gtw_attr->lp_buffer_alloc = pipeline->lp_mode; - fallthrough; + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + } + break; case SOF_DAI_INTEL_ALH: /* * Do not clear the node ID when this op is invoked with @@ -2834,7 +2838,7 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * */ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; - copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_node_id); } break; case SOF_DAI_INTEL_DMIC: diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 9ea2ac5adac79..fd664d5586f0d 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -91,6 +91,7 @@ struct snd_sof_pcm; struct snd_sof_dai_config_data { int dai_index; int dai_data; /* contains DAI-specific information */ + int dai_node_id; /* contains DAI-specific information for Gateway configuration */ }; /** From 2ac9e09ba0e874deeba13c3259dc18f22b622311 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:22 -0500 Subject: [PATCH 158/603] ASoC: SOF: Intel: hda: move helper to static inline MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To allow using widget_to_sdev() in other files, move it as static inline in shared header file. Reviewed-by: Rander Wang Reviewed-by: Péter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dai.c | 8 -------- sound/soc/sof/intel/hda.h | 8 ++++++++ 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 86c2325e59491..3f2fd84907d2f 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -29,14 +29,6 @@ static bool hda_use_tplg_nhlt; module_param_named(sof_use_tplg_nhlt, hda_use_tplg_nhlt, bool, 0444); MODULE_PARM_DESC(sof_use_tplg_nhlt, "SOF topology nhlt override"); -static struct snd_sof_dev *widget_to_sdev(struct snd_soc_dapm_widget *w) -{ - struct snd_sof_widget *swidget = w->dobj.private; - struct snd_soc_component *component = swidget->scomp; - - return snd_soc_component_get_drvdata(component); -} - int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, struct snd_sof_dai_config_data *data) { diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3bf7427dc918b..f530a05cfc928 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -1000,4 +1000,12 @@ int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, struct snd_soc_dai *cpu_dai); +static inline struct snd_sof_dev *widget_to_sdev(struct snd_soc_dapm_widget *w) +{ + struct snd_sof_widget *swidget = w->dobj.private; + struct snd_soc_component *component = swidget->scomp; + + return snd_soc_component_get_drvdata(component); +} + #endif From bfe9225455c032c9dd5637047760cf59562e599f Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 2 Apr 2024 10:18:23 -0500 Subject: [PATCH 159/603] ASoC: SOF: Intel: hda: Clear Soundwire node ID during BE DAI hw_free MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When an xrun happens, the BE DAI hw_params doesn't get invoked before the stream restarts with a prepare. In this case, clearing the node ID when the DAI widget is freed and unprepared will result in an error when it is re-initialized. In order to avoid this, move the code to clear the node ID to the BE DAI hw_free op to keep it balanced with the BE DAI hw_params. Reviewed-by: Rander Wang Reviewed-by: Péter Ujfalusi Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 24 ++++++++++++++++++++++++ sound/soc/sof/ipc4-topology.c | 4 ---- 2 files changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ae1a38f20bdbb..2c64c25d6f3b1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -31,6 +31,7 @@ #include "../sof-audio.h" #include "../sof-pci-dev.h" #include "../ops.h" +#include "../ipc4-topology.h" #include "hda.h" #include "telemetry.h" @@ -150,8 +151,31 @@ static int sdw_params_stream(struct device *dev, return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_PARAMS, &data); } +static int sdw_params_free(struct device *dev, struct sdw_intel_stream_free_data *free_data) +{ + struct snd_soc_dai *d = free_data->dai; + struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(d, free_data->substream->stream); + struct snd_sof_dev *sdev = widget_to_sdev(w); + + if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { + struct snd_sof_widget *swidget = w->dobj.private; + struct snd_sof_dai *dai = swidget->private; + struct sof_ipc4_copier_data *copier_data; + struct sof_ipc4_copier *ipc4_copier; + + ipc4_copier = dai->private; + copier_data = &ipc4_copier->data; + + /* clear the node ID */ + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + } + + return 0; +} + struct sdw_intel_ops sdw_callback = { .params_stream = sdw_params_stream, + .free_stream = sdw_params_free, }; static int sdw_ace2x_params_stream(struct device *dev, diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 0368ef6d0807b..e8a5e9fbd7968 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1277,7 +1277,6 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) } if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { - struct sof_ipc4_copier_data *copier_data = &ipc4_copier->data; struct sof_ipc4_alh_configuration_blob *blob; unsigned int group_id; @@ -1287,9 +1286,6 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) ALH_MULTI_GTW_BASE; ida_free(&alh_group_ida, group_id); } - - /* clear the node ID */ - copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; } } From a6f2b279d22894e81b23464620d03da6429d9ab5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:24 -0500 Subject: [PATCH 160/603] ASoC: SOF: sof-audio: revisit sof_pcm_stream_free() error handling and logs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason the existing code stops on the first error, which potentially leaves the DMA and widgets in a weird state. Change to free-up all resources even in case of errors. Also add a more consistent error handling and logs, with the first error code returned to the caller. Reviewed-by: Rander Wang Reviewed-by: Péter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 28 ++++++++++++++++++++-------- 1 file changed, 20 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index e693dcb475e4d..b5ca2861edbd5 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -834,17 +834,21 @@ int sof_pcm_stream_free(struct snd_sof_dev *sdev, struct snd_pcm_substream *subs { const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm); int ret; + int err = 0; if (spcm->prepared[substream->stream]) { /* stop DMA first if needed */ if (pcm_ops && pcm_ops->platform_stop_during_hw_free) snd_sof_pcm_platform_trigger(sdev, substream, SNDRV_PCM_TRIGGER_STOP); - /* Send PCM_FREE IPC to reset pipeline */ + /* free PCM in the DSP */ if (pcm_ops && pcm_ops->hw_free) { ret = pcm_ops->hw_free(sdev->component, substream); - if (ret < 0) - return ret; + if (ret < 0) { + dev_err(sdev->dev, "%s: pcm_ops hw_free failed %d\n", + __func__, ret); + err = ret; + } } spcm->prepared[substream->stream] = false; @@ -852,17 +856,25 @@ int sof_pcm_stream_free(struct snd_sof_dev *sdev, struct snd_pcm_substream *subs /* reset the DMA */ ret = snd_sof_pcm_platform_hw_free(sdev, substream); - if (ret < 0) - return ret; + if (ret < 0) { + dev_err(sdev->dev, "%s: platform hw free failed %d\n", + __func__, ret); + if (!err) + err = ret; + } /* free widget list */ if (free_widget_list) { ret = sof_widget_list_free(sdev, spcm, dir); - if (ret < 0) - dev_err(sdev->dev, "failed to free widgets during suspend\n"); + if (ret < 0) { + dev_err(sdev->dev, "%s: sof_widget_list_free failed %d\n", + __func__, ret); + if (!err) + err = ret; + } } - return ret; + return err; } /* From bb83ae04d9158276d17640f50c2a1e049100acb6 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:25 -0500 Subject: [PATCH 161/603] ASoC: SOF: pcm: simplify sof_pcm_hw_free() with helper MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The same sequence is used twice, use common helper. Reviewed-by: Rander Wang Reviewed-by: Péter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-15-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 32 +++----------------------------- 1 file changed, 3 insertions(+), 29 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 33d576b176478..7b732f31f9740 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -196,9 +196,8 @@ static int sof_pcm_hw_free(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); - const struct sof_ipc_pcm_ops *pcm_ops = sof_ipc_get_ops(sdev, pcm); struct snd_sof_pcm *spcm; - int ret, err = 0; + int ret; /* nothing to do for BE */ if (rtd->dai_link->no_pcm) @@ -211,36 +210,11 @@ static int sof_pcm_hw_free(struct snd_soc_component *component, dev_dbg(component->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); - if (spcm->prepared[substream->stream]) { - /* stop DMA first if needed */ - if (pcm_ops && pcm_ops->platform_stop_during_hw_free) - snd_sof_pcm_platform_trigger(sdev, substream, SNDRV_PCM_TRIGGER_STOP); - - /* free PCM in the DSP */ - if (pcm_ops && pcm_ops->hw_free) { - ret = pcm_ops->hw_free(component, substream); - if (ret < 0) - err = ret; - } - - spcm->prepared[substream->stream] = false; - } - - /* reset DMA */ - ret = snd_sof_pcm_platform_hw_free(sdev, substream); - if (ret < 0) { - dev_err(component->dev, "error: platform hw free failed\n"); - err = ret; - } - - /* free the DAPM widget list */ - ret = sof_widget_list_free(sdev, spcm, substream->stream); - if (ret < 0) - err = ret; + ret = sof_pcm_stream_free(sdev, substream, spcm, substream->stream, true); cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work); - return err; + return ret; } static int sof_pcm_prepare(struct snd_soc_component *component, From dbc78bce74f5f9057ba02bdc8d1549d24c573900 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:26 -0500 Subject: [PATCH 162/603] ASoC: SOF: pcm: add pending_stop state variable MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add a state variable to keep track of delayed stops, in case pcm_ops->platform_stop_during_hw_free is set. This patch should be iso-functionality, possibly removing no-op cases. The main purpose of this new state variable is to prepare a follow-up patch to reset all PCM and DMAs in case of stop/prepare xrun sequences. Reviewed-by: Rander Wang Reviewed-by: Péter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-16-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 11 +++++++++++ sound/soc/sof/sof-audio.c | 1 + sound/soc/sof/sof-audio.h | 1 + 3 files changed, 13 insertions(+) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 7b732f31f9740..2e8782dddc807 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -276,6 +276,8 @@ static int sof_pcm_trigger(struct snd_soc_component *component, dev_dbg(component->dev, "pcm: trigger stream %d dir %d cmd %d\n", spcm->pcm.pcm_id, substream->stream, cmd); + spcm->pending_stop[substream->stream] = false; + switch (cmd) { case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ipc_first = true; @@ -345,6 +347,15 @@ static int sof_pcm_trigger(struct snd_soc_component *component, /* invoke platform trigger to stop DMA even if pcm_ops isn't set or if it failed */ if (!pcm_ops || !pcm_ops->platform_stop_during_hw_free) snd_sof_pcm_platform_trigger(sdev, substream, cmd); + + /* + * set the pending_stop flag to indicate that pipeline stop has been delayed. + * This will be used later to stop the pipelines during prepare when recovering + * from xruns. + */ + if (pcm_ops && pcm_ops->platform_stop_during_hw_free && + cmd == SNDRV_PCM_TRIGGER_STOP) + spcm->pending_stop[substream->stream] = true; break; default: break; diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index b5ca2861edbd5..32fef64ef10dd 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -852,6 +852,7 @@ int sof_pcm_stream_free(struct snd_sof_dev *sdev, struct snd_pcm_substream *subs } spcm->prepared[substream->stream] = false; + spcm->pending_stop[substream->stream] = false; } /* reset the DMA */ diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index fd664d5586f0d..80a5bd69ef1c7 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -342,6 +342,7 @@ struct snd_sof_pcm { struct list_head list; /* list in sdev pcm list */ struct snd_pcm_hw_params params[2]; bool prepared[2]; /* PCM_PARAMS set successfully */ + bool pending_stop[2]; /* only used if (!pcm_ops->platform_stop_during_hw_free) */ }; struct snd_sof_led_control { From ebd3b3014eebdd490f2c509d79e719fbcc680963 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 2 Apr 2024 10:18:27 -0500 Subject: [PATCH 163/603] ASoC: SOF: pcm: reset all PCM sources in case of xruns MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With the delayed stops, the xrun handling is problematic: the applications expects everything to be reset but the firmware and DMA are still in a PAUSED state. This patch makes sure the prepare while pending_stop is set is special-cased. Reviewed-by: Rander Wang Reviewed-by: Péter Ujfalusi Co-developed-by: Ranjani Sridharan Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-17-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 2e8782dddc807..7b88e24b7701e 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -221,6 +221,7 @@ static int sof_pcm_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_sof_pcm *spcm; int ret; @@ -232,8 +233,18 @@ static int sof_pcm_prepare(struct snd_soc_component *component, if (!spcm) return -EINVAL; - if (spcm->prepared[substream->stream]) - return 0; + if (spcm->prepared[substream->stream]) { + if (!spcm->pending_stop[substream->stream]) + return 0; + + /* + * this case should be reached in case of xruns where we absolutely + * want to free-up and reset all PCM/DMA resources + */ + ret = sof_pcm_stream_free(sdev, substream, spcm, substream->stream, true); + if (ret < 0) + return ret; + } dev_dbg(component->dev, "pcm: prepare stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); From f0caa4fc244ca739ce6d12168aa588c412c81190 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Tue, 2 Apr 2024 10:18:28 -0500 Subject: [PATCH 164/603] ASoC: SOF: ipc4-topology: Save the ALH DAI index during hw_params MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The node_id for multi-gateway ALH DAI's get overwritten with the group_id during the DAI copier's ipc_prepare op. So, save the ALH dai_index during the BE DAI hw_params in the dai_index field of struct ipc4_copier and use that to set the device ID in the configuration blob. This will avoid errors during copier init after an xrun. Note that the dai_index is typically set in topology for DMIC/SSP, but it's not used for ALH. Reclaiming this dai_index field to store the node_id does not generate a conflict with topology-defined values. Reviewed-by: Rander Wang Reviewed-by: Péter Ujfalusi Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240402151828.175002-18-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 1 + sound/soc/sof/ipc4-topology.c | 12 ++++++++++-- sound/soc/sof/ipc4-topology.h | 1 + 3 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 2c64c25d6f3b1..d38dc43c2f1c9 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -164,6 +164,7 @@ static int sdw_params_free(struct device *dev, struct sdw_intel_stream_free_data struct sof_ipc4_copier *ipc4_copier; ipc4_copier = dai->private; + ipc4_copier->dai_index = 0; copier_data = &ipc4_copier->data; /* clear the node ID */ diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index e8a5e9fbd7968..793bca09bbf41 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1700,6 +1700,8 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, */ i = 0; list_for_each_entry(w, &sdev->widget_list, list) { + u32 node_type; + if (w->widget->sname && strcmp(w->widget->sname, swidget->widget->sname)) continue; @@ -1707,7 +1709,10 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, dai = w->private; alh_copier = (struct sof_ipc4_copier *)dai->private; alh_data = &alh_copier->data; - blob->alh_cfg.mapping[i].device = alh_data->gtw_cfg.node_id; + node_type = SOF_IPC4_GET_NODE_TYPE(alh_data->gtw_cfg.node_id); + blob->alh_cfg.mapping[i].device = SOF_IPC4_NODE_TYPE(node_type); + blob->alh_cfg.mapping[i].device |= + SOF_IPC4_NODE_INDEX(alh_copier->dai_index); /* * The mapping[i] device in ALH blob should be the same as the @@ -2830,12 +2835,15 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * /* * Do not clear the node ID when this op is invoked with * SOF_DAI_CONFIG_FLAGS_HW_FREE. It is needed to free the group_ida during - * unprepare. + * unprepare. The node_id for multi-gateway DAI's will be overwritten with the + * group_id during copier's ipc_prepare op. */ if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + ipc4_copier->dai_index = data->dai_node_id; copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_node_id); } + break; case SOF_DAI_INTEL_DMIC: case SOF_DAI_INTEL_SSP: diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index aa5122c3721d3..6e33208a357f5 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -45,6 +45,7 @@ #define SOF_IPC4_NODE_INDEX_MASK 0xFF #define SOF_IPC4_NODE_INDEX(x) ((x) & SOF_IPC4_NODE_INDEX_MASK) #define SOF_IPC4_NODE_TYPE(x) ((x) << 8) +#define SOF_IPC4_GET_NODE_TYPE(node_id) ((node_id) >> 8) /* Node ID for SSP type DAI copiers */ #define SOF_IPC4_NODE_INDEX_INTEL_SSP(x) (((x) & 0xf) << 4) From 4189b54220e5af15e948a48524b45d5ea2e5660d Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 1 Apr 2024 20:54:16 +0800 Subject: [PATCH 165/603] ASoC: dt-bindings: fsl-asoc-card: convert to YAML Convert the fsl-asoc-card binding to YAML. When testing dtbs_check, found below compatible strings are not listed in document: fsl,imx-sgtl5000 fsl,imx53-cpuvo-sgtl5000 fsl,imx51-babbage-sgtl5000 fsl,imx53-m53evk-sgtl5000 fsl,imx53-qsb-sgtl5000 fsl,imx53-voipac-sgtl5000 fsl,imx6-armadeus-sgtl5000 fsl,imx6-rex-sgtl5000 fsl,imx6-sabreauto-cs42888 fsl,imx6-wandboard-sgtl5000 fsl,imx6dl-nit6xlite-sgtl5000 fsl,imx6q-ba16-sgtl5000 fsl,imx6q-nitrogen6_max-sgtl5000 fsl,imx6q-nitrogen6_som2-sgtl5000 fsl,imx6q-nitrogen6x-sgtl5000 fsl,imx6q-sabrelite-sgtl5000 fsl,imx6q-sabresd-wm8962 fsl,imx6q-udoo-ac97 fsl,imx6q-ventana-sgtl5000 fsl,imx6sl-evk-wm8962 fsl,imx6sx-sdb-mqs fsl,imx6sx-sdb-wm8962 fsl,imx7d-evk-wm8960 karo,tx53-audio-sgtl5000 tq,imx53-mba53-sgtl5000 So add them in yaml file to pass the test. Also correct the 'dai-format' to 'format' in document. For 'audio-routing', the items are not listed. Because this fsl-asoc-card is generic driver, which supports several codecs, if list all the items, there will be a long list. Signed-off-by: Shengjiu Wang Reviewed-by: Rob Herring Link: https://msgid.link/r/1711976056-19884-3-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- .../bindings/sound/fsl-asoc-card.txt | 117 ----------- .../bindings/sound/fsl-asoc-card.yaml | 195 ++++++++++++++++++ 2 files changed, 195 insertions(+), 117 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl-asoc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt deleted file mode 100644 index 4e8dbc5abfd1d..0000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ /dev/null @@ -1,117 +0,0 @@ -Freescale Generic ASoC Sound Card with ASRC support - -The Freescale Generic ASoC Sound Card can be used, ideally, for all Freescale -SoCs connecting with external CODECs. - -The idea of this generic sound card is a bit like ASoC Simple Card. However, -for Freescale SoCs (especially those released in recent years), most of them -have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And -this is a specific feature that might be painstakingly controlled and merged -into the Simple Card. - -So having this generic sound card allows all Freescale SoC users to benefit -from the simplification of a new card support and the capability of the wide -sample rates support through ASRC. - -Note: The card is initially designed for those sound cards who use AC'97, I2S - and PCM DAI formats. However, it'll be also possible to support those non - AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as - long as the driver has been properly upgraded. - - -The compatible list for this generic sound card currently: - "fsl,imx-audio-ac97" - - "fsl,imx-audio-cs42888" - - "fsl,imx-audio-cs427x" - (compatible with CS4271 and CS4272) - - "fsl,imx-audio-wm8962" - - "fsl,imx-audio-sgtl5000" - (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt) - - "fsl,imx-audio-wm8960" - - "fsl,imx-audio-mqs" - - "fsl,imx-audio-wm8524" - - "fsl,imx-audio-tlv320aic32x4" - - "fsl,imx-audio-tlv320aic31xx" - - "fsl,imx-audio-si476x" - - "fsl,imx-audio-wm8958" - - "fsl,imx-audio-nau8822" - -Required properties: - - - compatible : Contains one of entries in the compatible list. - - - model : The user-visible name of this sound complex - - - audio-cpu : The phandle of an CPU DAI controller - - - audio-codec : The phandle of an audio codec - -Optional properties: - - - audio-asrc : The phandle of ASRC. It can be absent if there's no - need to add ASRC support via DPCM. - - - audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. There're a few pre-designed board connectors: - * Line Out Jack - * Line In Jack - * Headphone Jack - * Mic Jack - * Ext Spk - * AMIC (stands for Analog Microphone Jack) - * DMIC (stands for Digital Microphone Jack) - - Note: The "Mic Jack" and "AMIC" are redundant while - coexisting in order to support the old bindings - of wm8962 and sgtl5000. - - - hp-det-gpio : The GPIO that detect headphones are plugged in - - mic-det-gpio : The GPIO that detect microphones are plugged in - - bitclock-master : Indicates dai-link bit clock master; for details see simple-card.yaml. - - frame-master : Indicates dai-link frame master; for details see simple-card.yaml. - - dai-format : audio format, for details see simple-card.yaml. - - frame-inversion : dai-link uses frame clock inversion, for details see simple-card.yaml. - - bitclock-inversion : dai-link uses bit clock inversion, for details see simple-card.yaml. - - mclk-id : main clock id, specific for each card configuration. - -Optional unless SSI is selected as a CPU DAI: - - - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) - - - mux-ext-port : The external port of the i.MX audio muxer - -Example: -sound-cs42888 { - compatible = "fsl,imx-audio-cs42888"; - model = "cs42888-audio"; - audio-cpu = <&esai>; - audio-asrc = <&asrc>; - audio-codec = <&cs42888>; - audio-routing = - "Line Out Jack", "AOUT1L", - "Line Out Jack", "AOUT1R", - "Line Out Jack", "AOUT2L", - "Line Out Jack", "AOUT2R", - "Line Out Jack", "AOUT3L", - "Line Out Jack", "AOUT3R", - "Line Out Jack", "AOUT4L", - "Line Out Jack", "AOUT4R", - "AIN1L", "Line In Jack", - "AIN1R", "Line In Jack", - "AIN2L", "Line In Jack", - "AIN2R", "Line In Jack"; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml new file mode 100644 index 0000000000000..42ca39eebd49a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -0,0 +1,195 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl-asoc-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Generic ASoC Sound Card with ASRC support + +description: + The Freescale Generic ASoC Sound Card can be used, ideally, + for all Freescale SoCs connecting with external CODECs. + + The idea of this generic sound card is a bit like ASoC Simple Card. + However, for Freescale SoCs (especially those released in recent years), + most of them have ASRC inside. And this is a specific feature that might + be painstakingly controlled and merged into the Simple Card. + + So having this generic sound card allows all Freescale SoC users to + benefit from the simplification of a new card support and the capability + of the wide sample rates support through ASRC. + + Note, The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. + +maintainers: + - Shengjiu Wang + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx-sgtl5000 + - fsl,imx53-cpuvo-sgtl5000 + - fsl,imx51-babbage-sgtl5000 + - fsl,imx53-m53evk-sgtl5000 + - fsl,imx53-qsb-sgtl5000 + - fsl,imx53-voipac-sgtl5000 + - fsl,imx6-armadeus-sgtl5000 + - fsl,imx6-rex-sgtl5000 + - fsl,imx6-sabreauto-cs42888 + - fsl,imx6-wandboard-sgtl5000 + - fsl,imx6dl-nit6xlite-sgtl5000 + - fsl,imx6q-ba16-sgtl5000 + - fsl,imx6q-nitrogen6_max-sgtl5000 + - fsl,imx6q-nitrogen6_som2-sgtl5000 + - fsl,imx6q-nitrogen6x-sgtl5000 + - fsl,imx6q-sabrelite-sgtl5000 + - fsl,imx6q-sabresd-wm8962 + - fsl,imx6q-udoo-ac97 + - fsl,imx6q-ventana-sgtl5000 + - fsl,imx6sl-evk-wm8962 + - fsl,imx6sx-sdb-mqs + - fsl,imx6sx-sdb-wm8962 + - fsl,imx7d-evk-wm8960 + - karo,tx53-audio-sgtl5000 + - tq,imx53-mba53-sgtl5000 + - enum: + - fsl,imx-audio-ac97 + - fsl,imx-audio-cs42888 + - fsl,imx-audio-mqs + - fsl,imx-audio-sgtl5000 + - fsl,imx-audio-wm8960 + - fsl,imx-audio-wm8962 + - items: + - enum: + - fsl,imx-audio-ac97 + - fsl,imx-audio-cs42888 + - fsl,imx-audio-cs427x + - fsl,imx-audio-mqs + - fsl,imx-audio-nau8822 + - fsl,imx-audio-sgtl5000 + - fsl,imx-audio-si476x + - fsl,imx-audio-tlv320aic31xx + - fsl,imx-audio-tlv320aic32x4 + - fsl,imx-audio-wm8524 + - fsl,imx-audio-wm8960 + - fsl,imx-audio-wm8962 + - fsl,imx-audio-wm8958 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex + + audio-asrc: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of ASRC. It can be absent if there's no + need to add ASRC support via DPCM. + + audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an audio codec + + audio-cpu: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an CPU DAI controller + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. There're a few pre-designed board + connectors. "AMIC" stands for Analog Microphone Jack. + "DMIC" stands for Digital Microphone Jack. The "Mic Jack" and "AMIC" + are redundant while coexisting in order to support the old bindings + of wm8962 and sgtl5000. + + hp-det-gpio: + deprecated: true + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + mic-det-gpio: + deprecated: true + maxItems: 1 + description: The GPIO that detect microphones are plugged in + + mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphones are plugged in + + bitclock-master: + $ref: simple-card.yaml#/definitions/bitclock-master + description: Indicates dai-link bit clock master. + + frame-master: + $ref: simple-card.yaml#/definitions/frame-master + description: Indicates dai-link frame master. + + format: + $ref: simple-card.yaml#/definitions/format + description: audio format. + + frame-inversion: + $ref: simple-card.yaml#/definitions/frame-inversion + description: dai-link uses frame clock inversion. + + bitclock-inversion: + $ref: simple-card.yaml#/definitions/bitclock-inversion + description: dai-link uses bit clock inversion. + + mclk-id: + $ref: /schemas/types.yaml#/definitions/uint32 + description: main clock id, specific for each card configuration. + + mux-int-port: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [1, 2, 7] + description: The internal port of the i.MX audio muxer (AUDMUX) + + mux-ext-port: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [3, 4, 5, 6] + description: The external port of the i.MX audio muxer + + ssi-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of an CPU DAI controller + +required: + - compatible + - model + +unevaluatedProperties: false + +examples: + - | + sound-cs42888 { + compatible = "fsl,imx-audio-cs42888"; + model = "cs42888-audio"; + audio-cpu = <&esai>; + audio-asrc = <&asrc>; + audio-codec = <&cs42888>; + audio-routing = + "Line Out Jack", "AOUT1L", + "Line Out Jack", "AOUT1R", + "Line Out Jack", "AOUT2L", + "Line Out Jack", "AOUT2R", + "Line Out Jack", "AOUT3L", + "Line Out Jack", "AOUT3R", + "Line Out Jack", "AOUT4L", + "Line Out Jack", "AOUT4R", + "AIN1L", "Line In Jack", + "AIN1R", "Line In Jack", + "AIN2L", "Line In Jack", + "AIN2R", "Line In Jack"; + }; From 27aa58180473f81990f35238dc8aec40d34c778d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:36 +0200 Subject: [PATCH 166/603] ASoC: Intel: avs: hdaudio: Use devm_kstrdup_const MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit HDA codec name is constant and we just duplicate it, use const variant of devm_kstrdup to possibly save a bit of memory. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/hdaudio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c index 79b4aca413338..2de703b909c30 100644 --- a/sound/soc/intel/avs/boards/hdaudio.c +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -54,7 +54,7 @@ static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int if (!dl[i].cpus->dai_name) return -ENOMEM; - dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL); + dl[i].codecs->name = devm_kstrdup_const(dev, cname, GFP_KERNEL); if (!dl[i].codecs->name) return -ENOMEM; @@ -191,7 +191,7 @@ static int avs_hdaudio_probe(struct platform_device *pdev) if (!binder->platforms || !binder->codecs) return -ENOMEM; - binder->codecs->name = devm_kstrdup(dev, dev_name(&codec->core.dev), GFP_KERNEL); + binder->codecs->name = devm_kstrdup_const(dev, dev_name(&codec->core.dev), GFP_KERNEL); if (!binder->codecs->name) return -ENOMEM; From 68a71af3aee2b776a0030e1393864c9501997cbb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:37 +0200 Subject: [PATCH 167/603] ASoC: Intel: avs: Use devm_kstrdup_const MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit FW name is constant and we just duplicate it, use const variant of devm_kstrdup to possibly save a bit of memory. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/avs.h | 2 +- sound/soc/intel/avs/utils.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index f80f79415344a..9a02e2b528bc5 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -107,7 +107,7 @@ struct avs_spec { }; struct avs_fw_entry { - char *name; + const char *name; const struct firmware *fw; struct list_head node; diff --git a/sound/soc/intel/avs/utils.c b/sound/soc/intel/avs/utils.c index 82416b86662d8..8100c2fa0a7ed 100644 --- a/sound/soc/intel/avs/utils.c +++ b/sound/soc/intel/avs/utils.c @@ -250,7 +250,7 @@ int avs_request_firmware(struct avs_dev *adev, const struct firmware **fw_p, con if (!entry) return -ENOMEM; - entry->name = kstrdup(name, GFP_KERNEL); + entry->name = kstrdup_const(name, GFP_KERNEL); if (!entry->name) { kfree(entry); return -ENOMEM; @@ -258,7 +258,7 @@ int avs_request_firmware(struct avs_dev *adev, const struct firmware **fw_p, con ret = request_firmware(&entry->fw, name, adev->dev); if (ret < 0) { - kfree(entry->name); + kfree_const(entry->name); kfree(entry); return ret; } @@ -282,7 +282,7 @@ void avs_release_last_firmware(struct avs_dev *adev) list_del(&entry->node); release_firmware(entry->fw); - kfree(entry->name); + kfree_const(entry->name); kfree(entry); } @@ -296,7 +296,7 @@ void avs_release_firmwares(struct avs_dev *adev) list_for_each_entry_safe(entry, tmp, &adev->fw_list, node) { list_del(&entry->node); release_firmware(entry->fw); - kfree(entry->name); + kfree_const(entry->name); kfree(entry); } } From a3c95efc432ddd18882cfab420cbea14937fb950 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:38 +0200 Subject: [PATCH 168/603] ASoC: Intel: avs: hdaudio: Constify probing_link MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit probing_link is passed to devm_kmemdup, and is never modified, may as well mark it const. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/hdaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c index 2de703b909c30..4433175814f80 100644 --- a/sound/soc/intel/avs/boards/hdaudio.c +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -155,7 +155,7 @@ static int avs_probing_link_init(struct snd_soc_pcm_runtime *rtm) return 0; } -static struct snd_soc_dai_link probing_link = { +static const struct snd_soc_dai_link probing_link = { .name = "probing-LINK", .id = -1, .nonatomic = 1, From 2dbe7c809841e645a6b9d6d51801d3c594739d93 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:39 +0200 Subject: [PATCH 169/603] ASoC: Intel: avs: es8336: Constify card_headset_pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit card_headset_pins is never modified, mark it const. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-5-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/es8336.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 1090082e7d5bf..0171806094b4a 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -85,7 +85,7 @@ static const struct snd_kcontrol_new card_controls[] = { SOC_DAPM_PIN_SWITCH("Internal Mic"), }; -static struct snd_soc_jack_pin card_headset_pins[] = { +static const struct snd_soc_jack_pin card_headset_pins[] = { { .pin = "Headphone", .mask = SND_JACK_HEADPHONE, From bf400f16a03788ccdb1de1d9bfe9458082f24ef7 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:40 +0200 Subject: [PATCH 170/603] ASoC: Intel: avs: nau8825: Constify card_headset_pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit card_headset_pins is never modified, mark it const. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-6-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 1c1e2083f474d..cd9e6a6f92122 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -67,7 +67,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = { { "Headset Mic", NULL, "Platform Clock" }, }; -static struct snd_soc_jack_pin card_headset_pins[] = { +static const struct snd_soc_jack_pin card_headset_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, From 130c953a75cc62dd5c73c295bae14197dce46b85 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:41 +0200 Subject: [PATCH 171/603] ASoC: Intel: avs: rt274: Constify card_headset_pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit card_headset_pins is never modified, mark it const. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-7-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt274.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index bfcb8845fd15d..4ddc86db43dc2 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -75,7 +75,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = { {"MIC", NULL, "Platform Clock"}, }; -static struct snd_soc_jack_pin card_headset_pins[] = { +static const struct snd_soc_jack_pin card_headset_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, From 682bedc8b0bae52fc5fc95cb3dba6e3d590da653 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:42 +0200 Subject: [PATCH 172/603] ASoC: Intel: avs: rt286: Constify card_headset_pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit card_headset_pins is never modified, mark it const. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-8-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index 28d7d86b1cc99..88cc30dc429cc 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -38,7 +38,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = { {"Speaker", NULL, "SPOL"}, }; -static struct snd_soc_jack_pin card_headset_pins[] = { +static const struct snd_soc_jack_pin card_headset_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, From 33a886eee711298e799f1f52432e2f5e234cd015 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:43 +0200 Subject: [PATCH 173/603] ASoC: Intel: avs: rt298: Constify card_headset_pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit card_headset_pins is never modified, mark it const. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-9-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt298.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 80f490b9e1184..99ec84920433a 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -49,7 +49,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = { {"Speaker", NULL, "SPOL"}, }; -static struct snd_soc_jack_pin card_headset_pins[] = { +static const struct snd_soc_jack_pin card_headset_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, From a055674868fa9d36f290dba906a76dd45801a37e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:44 +0200 Subject: [PATCH 174/603] ASoC: Intel: avs: rt5663: Constify card_headset_pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit card_headset_pins is never modified, mark it const. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-10-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt5663.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c index b4762c2a7bf2d..4a1be5b229c17 100644 --- a/sound/soc/intel/avs/boards/rt5663.c +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -43,7 +43,7 @@ static const struct snd_soc_dapm_route card_routes[] = { { "IN1N", NULL, "Headset Mic" }, }; -static struct snd_soc_jack_pin card_headset_pins[] = { +static const struct snd_soc_jack_pin card_headset_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, From 5faf65b5d1e36ec0a1eb3ba8ba2a8b7a779575ce Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Wed, 3 Apr 2024 11:31:45 +0200 Subject: [PATCH 175/603] ASoC: Intel: avs: rt5682: Constify card_headset_pins MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit card_headset_pins is never modified, mark it const. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://msgid.link/r/20240403093145.3375857-11-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 243f979fda98a..a672578aa4d65 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -80,7 +80,7 @@ static const struct snd_soc_dapm_route card_base_routes[] = { { "IN1P", NULL, "Headset Mic" }, }; -static struct snd_soc_jack_pin card_jack_pins[] = { +static const struct snd_soc_jack_pin card_jack_pins[] = { { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, From bbdf9af261adca039de29e7fc1faff367bf7e9a0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:04 +0300 Subject: [PATCH 176/603] ASoC: SOF: Intel: hda: Create debugfs file to force a clean DSP boot When IMR boot is supported on a platform it is always going to be used to boot the DSP unless some catastrophic event happens. There is no way for a developer to force a clean DSP boot without removing and re-inserting the modules. Create a 'skip_imr_boot' debugfs file which can be used to force the next DSP boot as clean (prune) boot. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240403105210.17949-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 7 ++++++- sound/soc/sof/intel/hda.c | 1 + sound/soc/sof/intel/lnl.c | 7 ++++++- sound/soc/sof/intel/mtl.c | 7 ++++++- 4 files changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index b81f231abee3e..d5b9209beb5a6 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -15,6 +15,7 @@ * Hardware interface for HDA DSP code loader */ +#include #include #include #include @@ -643,8 +644,12 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) /* Check if IMR boot is usable */ if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) && (sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT || - sdev->pdata->ipc_type == SOF_IPC_TYPE_4)) + sdev->pdata->ipc_type == SOF_IPC_TYPE_4)) { hdev->imrboot_supported = true; + debugfs_create_bool("skip_imr_boot", + 0644, sdev->debugfs_root, + &hdev->skip_imr_boot); + } } hda_sdw_int_enable(sdev, true); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index d38dc43c2f1c9..33721e817ef42 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -19,6 +19,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 7ae017a00184e..1365f86b7fb72 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -6,6 +6,7 @@ * Hardware interface for audio DSP on LunarLake. */ +#include #include #include #include @@ -83,8 +84,12 @@ static int lnl_dsp_post_fw_run(struct snd_sof_dev *sdev) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; /* Check if IMR boot is usable */ - if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT)) + if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT)) { hda->imrboot_supported = true; + debugfs_create_bool("skip_imr_boot", + 0644, sdev->debugfs_root, + &hda->skip_imr_boot); + } } return 0; diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index df05dc77b8d5e..8a55d5a2703e4 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -9,6 +9,7 @@ * Hardware interface for audio DSP on Meteorlake. */ +#include #include #include #include @@ -294,8 +295,12 @@ int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev) } /* Check if IMR boot is usable */ - if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT)) + if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT)) { hdev->imrboot_supported = true; + debugfs_create_bool("skip_imr_boot", + 0644, sdev->debugfs_root, + &hdev->skip_imr_boot); + } } hda_sdw_int_enable(sdev, true); From 1f1b820dc3c65b6883da3130ba3b8624dcbf87db Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:05 +0300 Subject: [PATCH 177/603] ASoC: SOF: Intel: mtl: Correct rom_status_reg ACE1 architecture changed the place where the ROM updates the status code from the shared SRAM window to HFFLGP1QW0 register for the status and HFFLGP1QW0 + 4 for the error code. The rom_status_reg is not used on MTL because it was wrongly assigned based on older platform convention (SRAM window) and it was giving inconsistent readings. Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)") Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 4 ++-- sound/soc/sof/intel/mtl.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 8a55d5a2703e4..34a0707086c5a 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -746,7 +746,7 @@ const struct sof_intel_dsp_desc mtl_chip_info = { .ipc_ack = MTL_DSP_REG_HFIPCXIDA, .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, - .rom_status_reg = MTL_DSP_ROM_STS, + .rom_status_reg = MTL_DSP_REG_HFFLGPXQWY, .rom_init_timeout = 300, .ssp_count = MTL_SSP_COUNT, .ssp_base_offset = CNL_SSP_BASE_OFFSET, @@ -774,7 +774,7 @@ const struct sof_intel_dsp_desc arl_s_chip_info = { .ipc_ack = MTL_DSP_REG_HFIPCXIDA, .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, - .rom_status_reg = MTL_DSP_ROM_STS, + .rom_status_reg = MTL_DSP_REG_HFFLGPXQWY, .rom_init_timeout = 300, .ssp_count = MTL_SSP_COUNT, .ssp_base_offset = CNL_SSP_BASE_OFFSET, diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index cc5a1f46fd095..82dd6b8c48592 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -76,8 +76,8 @@ #define MTL_DSP_ROM_STS MTL_SRAM_WINDOW_OFFSET(0) /* ROM status */ #define MTL_DSP_ROM_ERROR (MTL_SRAM_WINDOW_OFFSET(0) + 0x4) /* ROM error code */ -#define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* ROM debug status */ -#define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* ROM debug error code */ +#define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* DSP core0 status */ +#define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* DSP core0 error */ #define MTL_DSP_REG_HfIMRIS1 0x162088 #define MTL_DSP_REG_HfIMRIS1_IU_MASK BIT(0) From b852574c671a9983dd51c81582c8c5085f3dc382 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:06 +0300 Subject: [PATCH 178/603] ASoC: SOF: Intel: lnl: Correct rom_status_reg ACE2 architecture changed the place where the ROM updates the status code from the shared SRAM window (and HFFLGP1QW0 in ACE1) to HFDSC register for the status and HFDEC (HFDSC + 4) for the error code. The rom_status_reg is not used on LNL because it was wrongly assigned based on older platform convention (SRAM window) and it was giving inconsistent readings. Add new header file for lnl specific register definitions. Fixes: 64a63d9914a5 ("ASoC: SOF: Intel: LNL: Add support for Lunarlake platform") Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 3 ++- sound/soc/sof/intel/lnl.h | 15 +++++++++++++++ 2 files changed, 17 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/intel/lnl.h diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 1365f86b7fb72..3a50a299dd334 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -17,6 +17,7 @@ #include "hda-ipc.h" #include "../sof-audio.h" #include "mtl.h" +#include "lnl.h" #include /* LunarLake ops */ @@ -199,7 +200,7 @@ const struct sof_intel_dsp_desc lnl_chip_info = { .ipc_ack = MTL_DSP_REG_HFIPCXIDA, .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, - .rom_status_reg = MTL_DSP_ROM_STS, + .rom_status_reg = LNL_DSP_REG_HFDSC, .rom_init_timeout = 300, .ssp_count = MTL_SSP_COUNT, .d0i3_offset = MTL_HDA_VS_D0I3C, diff --git a/sound/soc/sof/intel/lnl.h b/sound/soc/sof/intel/lnl.h new file mode 100644 index 0000000000000..4f4734fe7e089 --- /dev/null +++ b/sound/soc/sof/intel/lnl.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2024 Intel Corporation. All rights reserved. + */ + +#ifndef __SOF_INTEL_LNL_H +#define __SOF_INTEL_LNL_H + +#define LNL_DSP_REG_HFDSC 0x160200 /* DSP core0 status */ +#define LNL_DSP_REG_HFDEC 0x160204 /* DSP core0 error */ + +#endif /* __SOF_INTEL_LNL_H */ From 26187f44aabdf3df7609b7c78724a059c230a2ad Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:07 +0300 Subject: [PATCH 179/603] ASoC: SOF: Intel: mtl: Disable interrupts when firmware boot failed In case of error during the firmware boot we need to disable the interrupts which were enabled as part of the boot sequence. Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)") Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 34a0707086c5a..044e1cdd46e5c 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -508,6 +508,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) dump_msg = kasprintf(GFP_KERNEL, "Boot iteration failed: %d/%d", hda->boot_iteration, HDA_FW_BOOT_ATTEMPTS); snd_sof_dsp_dbg_dump(sdev, dump_msg, flags); + mtl_enable_interrupts(sdev, false); mtl_dsp_core_power_down(sdev, SOF_DSP_PRIMARY_CORE); kfree(dump_msg); From 6b1c1c47e76f0161bda2b1ac2e86a219fe70244f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:08 +0300 Subject: [PATCH 180/603] ASoC: SOF: Intel: mtl: Implement firmware boot state check With the corrected rom_status_reg values we can now add a check for target boot status for firmware booting. With the check now we can identify failed firmware boots (IMR boots) and we can use the fallback to purge boot the DSP. Fixes: 064520e8aeaa ("ASoC: SOF: Intel: Add support for MeteorLake (MTL)") Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 37 ++++++++++++++++++++++++++++++++----- 1 file changed, 32 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 044e1cdd46e5c..4c3cfa719571c 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -444,7 +444,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; - unsigned int status; + unsigned int status, target_status; u32 ipc_hdr, flags; char *dump_msg; int ret; @@ -490,13 +490,40 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) mtl_enable_ipc_interrupts(sdev); + if (chip->rom_status_reg == MTL_DSP_ROM_STS) { + /* + * Workaround: when the ROM status register is pointing to + * the SRAM window (MTL_DSP_ROM_STS) the platform cannot catch + * ROM_INIT_DONE because of a very short timing window. + * Follow the recommendations and skip target state waiting. + */ + return 0; + } + /* - * ACE workaround: don't wait for ROM INIT. - * The platform cannot catch ROM_INIT_DONE because of a very short - * timing window. Follow the recommendations and skip this part. + * step 7: + * - Cold/Full boot: wait for ROM init to proceed to download the firmware + * - IMR boot: wait for ROM firmware entered (firmware booted up from IMR) */ + if (imr_boot) + target_status = FSR_STATE_FW_ENTERED; + else + target_status = FSR_STATE_INIT_DONE; - return 0; + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, + chip->rom_status_reg, status, + (FSR_TO_STATE_CODE(status) == target_status), + HDA_DSP_REG_POLL_INTERVAL_US, + chip->rom_init_timeout * + USEC_PER_MSEC); + + if (!ret) + return 0; + + if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) + dev_err(sdev->dev, + "%s: timeout with rom_status_reg (%#x) read\n", + __func__, chip->rom_status_reg); err: flags = SOF_DBG_DUMP_PCI | SOF_DBG_DUMP_MBOX | SOF_DBG_DUMP_OPTIONAL; From 3dc2682870ea8f5a7749c069dfc4e0040e69cb5d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:09 +0300 Subject: [PATCH 181/603] ASoC: SOF: Intel: hda-dsp/mtl: Add support for ACE ROM state codes The ROM state codes differ between CAVS and ACE architecture, there is a slight overlap. Add the ACE related state defines to mtl.h, introduce new table and use it on case the function is called when running on ACE architecture. Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 72 +++++++++++++++++++++++++++++++++++---- sound/soc/sof/intel/hda.h | 2 ++ sound/soc/sof/intel/mtl.h | 44 ++++++++++++++++++++++++ 3 files changed, 112 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 33721e817ef42..2fc10bec7331c 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -35,6 +35,7 @@ #include "../ipc4-topology.h" #include "hda.h" #include "telemetry.h" +#include "mtl.h" #define CREATE_TRACE_POINTS #include @@ -597,7 +598,7 @@ static const struct hda_dsp_msg_code hda_dsp_rom_fw_error_texts[] = { }; #define FSR_ROM_STATE_ENTRY(state) {FSR_STATE_ROM_##state, #state} -static const struct hda_dsp_msg_code fsr_rom_state_names[] = { +static const struct hda_dsp_msg_code cavs_fsr_rom_state_names[] = { FSR_ROM_STATE_ENTRY(INIT), FSR_ROM_STATE_ENTRY(INIT_DONE), FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED), @@ -620,6 +621,58 @@ static const struct hda_dsp_msg_code fsr_rom_state_names[] = { FSR_ROM_STATE_ENTRY(CSE_IPC_DOWN), }; +static const struct hda_dsp_msg_code ace_fsr_rom_state_names[] = { + FSR_ROM_STATE_ENTRY(INIT), + FSR_ROM_STATE_ENTRY(INIT_DONE), + FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED), + FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED), + FSR_ROM_STATE_ENTRY(FW_FW_LOADED), + FSR_ROM_STATE_ENTRY(FW_ENTERED), + FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK), + FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET), + FSR_ROM_STATE_ENTRY(RESET_VECTOR_DONE), + FSR_ROM_STATE_ENTRY(PURGE_BOOT), + FSR_ROM_STATE_ENTRY(RESTORE_BOOT), + FSR_ROM_STATE_ENTRY(FW_ENTRY_POINT), + FSR_ROM_STATE_ENTRY(VALIDATE_PUB_KEY), + FSR_ROM_STATE_ENTRY(POWER_DOWN_HPSRAM), + FSR_ROM_STATE_ENTRY(POWER_DOWN_ULPSRAM), + FSR_ROM_STATE_ENTRY(POWER_UP_ULPSRAM_STACK), + FSR_ROM_STATE_ENTRY(POWER_UP_HPSRAM_DMA), + FSR_ROM_STATE_ENTRY(BEFORE_EP_POINTER_READ), + FSR_ROM_STATE_ENTRY(VALIDATE_MANIFEST), + FSR_ROM_STATE_ENTRY(VALIDATE_FW_MODULE), + FSR_ROM_STATE_ENTRY(PROTECT_IMR_REGION), + FSR_ROM_STATE_ENTRY(PUSH_MODEL_ROUTINE), + FSR_ROM_STATE_ENTRY(PULL_MODEL_ROUTINE), + FSR_ROM_STATE_ENTRY(VALIDATE_PKG_DIR), + FSR_ROM_STATE_ENTRY(VALIDATE_CPD), + FSR_ROM_STATE_ENTRY(VALIDATE_CSS_MAN_HEADER), + FSR_ROM_STATE_ENTRY(VALIDATE_BLOB_SVN), + FSR_ROM_STATE_ENTRY(VERIFY_IFWI_PARTITION), + FSR_ROM_STATE_ENTRY(REMOVE_ACCESS_CONTROL), + FSR_ROM_STATE_ENTRY(AUTH_BYPASS), + FSR_ROM_STATE_ENTRY(AUTH_ENABLED), + FSR_ROM_STATE_ENTRY(INIT_DMA), + FSR_ROM_STATE_ENTRY(PURGE_FW_ENTRY), + FSR_ROM_STATE_ENTRY(PURGE_FW_END), + FSR_ROM_STATE_ENTRY(CLEAN_UP_BSS_DONE), + FSR_ROM_STATE_ENTRY(IMR_RESTORE_ENTRY), + FSR_ROM_STATE_ENTRY(IMR_RESTORE_END), + FSR_ROM_STATE_ENTRY(FW_MANIFEST_IN_DMA_BUFF), + FSR_ROM_STATE_ENTRY(LOAD_CSE_MAN_TO_IMR), + FSR_ROM_STATE_ENTRY(LOAD_FW_MAN_TO_IMR), + FSR_ROM_STATE_ENTRY(LOAD_FW_CODE_TO_IMR), + FSR_ROM_STATE_ENTRY(FW_LOADING_DONE), + FSR_ROM_STATE_ENTRY(FW_CODE_LOADED), + FSR_ROM_STATE_ENTRY(VERIFY_IMAGE_TYPE), + FSR_ROM_STATE_ENTRY(AUTH_API_INIT), + FSR_ROM_STATE_ENTRY(AUTH_API_PROC), + FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_BUSY), + FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_RESULT), + FSR_ROM_STATE_ENTRY(AUTH_API_CLEANUP), +}; + #define FSR_BRINGUP_STATE_ENTRY(state) {FSR_STATE_BRINGUP_##state, #state} static const struct hda_dsp_msg_code fsr_bringup_state_names[] = { FSR_BRINGUP_STATE_ENTRY(INIT), @@ -664,7 +717,7 @@ hda_dsp_get_state_text(u32 code, const struct hda_dsp_msg_code *msg_code, return NULL; } -static void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level) +void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level) { const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); const char *state_text, *error_text, *module_text; @@ -680,12 +733,19 @@ static void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level) else module_text = fsr_module_names[module]; - if (module == FSR_MOD_BRNGUP) + if (module == FSR_MOD_BRNGUP) { state_text = hda_dsp_get_state_text(state, fsr_bringup_state_names, ARRAY_SIZE(fsr_bringup_state_names)); - else - state_text = hda_dsp_get_state_text(state, fsr_rom_state_names, - ARRAY_SIZE(fsr_rom_state_names)); + } else { + if (chip->hw_ip_version < SOF_INTEL_ACE_1_0) + state_text = hda_dsp_get_state_text(state, + cavs_fsr_rom_state_names, + ARRAY_SIZE(cavs_fsr_rom_state_names)); + else + state_text = hda_dsp_get_state_text(state, + ace_fsr_rom_state_names, + ARRAY_SIZE(ace_fsr_rom_state_names)); + } /* not for us, must be generic sof message */ if (!state_text) { diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index f530a05cfc928..88936bff33192 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -689,6 +689,8 @@ int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id); irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context); int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir); +void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level); + /* * DSP Code loader. */ diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index 82dd6b8c48592..9a36484cc6212 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -78,6 +78,50 @@ #define MTL_DSP_REG_HFFLGPXQWY 0x163200 /* DSP core0 status */ #define MTL_DSP_REG_HFFLGPXQWY_ERROR 0x163204 /* DSP core0 error */ + +/* FSR status codes */ +#define FSR_STATE_ROM_RESET_VECTOR_DONE 0x8 +#define FSR_STATE_ROM_PURGE_BOOT 0x9 +#define FSR_STATE_ROM_RESTORE_BOOT 0xA +#define FSR_STATE_ROM_FW_ENTRY_POINT 0xB +#define FSR_STATE_ROM_VALIDATE_PUB_KEY 0xC +#define FSR_STATE_ROM_POWER_DOWN_HPSRAM 0xD +#define FSR_STATE_ROM_POWER_DOWN_ULPSRAM 0xE +#define FSR_STATE_ROM_POWER_UP_ULPSRAM_STACK 0xF +#define FSR_STATE_ROM_POWER_UP_HPSRAM_DMA 0x10 +#define FSR_STATE_ROM_BEFORE_EP_POINTER_READ 0x11 +#define FSR_STATE_ROM_VALIDATE_MANIFEST 0x12 +#define FSR_STATE_ROM_VALIDATE_FW_MODULE 0x13 +#define FSR_STATE_ROM_PROTECT_IMR_REGION 0x14 +#define FSR_STATE_ROM_PUSH_MODEL_ROUTINE 0x15 +#define FSR_STATE_ROM_PULL_MODEL_ROUTINE 0x16 +#define FSR_STATE_ROM_VALIDATE_PKG_DIR 0x17 +#define FSR_STATE_ROM_VALIDATE_CPD 0x18 +#define FSR_STATE_ROM_VALIDATE_CSS_MAN_HEADER 0x19 +#define FSR_STATE_ROM_VALIDATE_BLOB_SVN 0x1A +#define FSR_STATE_ROM_VERIFY_IFWI_PARTITION 0x1B +#define FSR_STATE_ROM_REMOVE_ACCESS_CONTROL 0x1C +#define FSR_STATE_ROM_AUTH_BYPASS 0x1D +#define FSR_STATE_ROM_AUTH_ENABLED 0x1E +#define FSR_STATE_ROM_INIT_DMA 0x1F +#define FSR_STATE_ROM_PURGE_FW_ENTRY 0x20 +#define FSR_STATE_ROM_PURGE_FW_END 0x21 +#define FSR_STATE_ROM_CLEAN_UP_BSS_DONE 0x22 +#define FSR_STATE_ROM_IMR_RESTORE_ENTRY 0x23 +#define FSR_STATE_ROM_IMR_RESTORE_END 0x24 +#define FSR_STATE_ROM_FW_MANIFEST_IN_DMA_BUFF 0x25 +#define FSR_STATE_ROM_LOAD_CSE_MAN_TO_IMR 0x26 +#define FSR_STATE_ROM_LOAD_FW_MAN_TO_IMR 0x27 +#define FSR_STATE_ROM_LOAD_FW_CODE_TO_IMR 0x28 +#define FSR_STATE_ROM_FW_LOADING_DONE 0x29 +#define FSR_STATE_ROM_FW_CODE_LOADED 0x2A +#define FSR_STATE_ROM_VERIFY_IMAGE_TYPE 0x2B +#define FSR_STATE_ROM_AUTH_API_INIT 0x2C +#define FSR_STATE_ROM_AUTH_API_PROC 0x2D +#define FSR_STATE_ROM_AUTH_API_FIRST_BUSY 0x2E +#define FSR_STATE_ROM_AUTH_API_FIRST_RESULT 0x2F +#define FSR_STATE_ROM_AUTH_API_CLEANUP 0x30 + #define MTL_DSP_REG_HfIMRIS1 0x162088 #define MTL_DSP_REG_HfIMRIS1_IU_MASK BIT(0) From 40bdf121a3ed91281196068e50789888e4b1d2d2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 3 Apr 2024 13:52:10 +0300 Subject: [PATCH 182/603] ASoC: SOF: Intel: mtl: Correct the mtl_dsp_dump output The ROM/firmware state handling has changed between CAVS and ACE architecture: CAVS: ROM and firmware uses the SRAM window for the state and status/error code reporting ACE: ROM code is using two registers to report the state and error while the firmware is using the SRAM window to report states and status/error codes. Use the generic hda_dsp_get_state() to decode ROM state and error codes and print out the firmware state and status/error code only if the SRAM window is accessible - the firmware is booted and the Status readout is not 0xffffffff. Signed-off-by: Peter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Kai Vehmanen Reviewed-by: Pierre-Louis Bossart Reviewed-by: Liam Girdwood Link: https://msgid.link/r/20240403105210.17949-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 4c3cfa719571c..8ff864ae98333 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -310,22 +310,16 @@ int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev) void mtl_dsp_dump(struct snd_sof_dev *sdev, u32 flags) { char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR; - u32 romdbgsts; - u32 romdbgerr; u32 fwsts; u32 fwlec; + hda_dsp_get_state(sdev, level); fwsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_ROM_STS); fwlec = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_ROM_ERROR); - romdbgsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY); - romdbgerr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY_ERROR); - - dev_err(sdev->dev, "ROM status: %#x, ROM error: %#x\n", fwsts, fwlec); - dev_err(sdev->dev, "ROM debug status: %#x, ROM debug error: %#x\n", romdbgsts, - romdbgerr); - romdbgsts = snd_sof_dsp_read(sdev, HDA_DSP_BAR, MTL_DSP_REG_HFFLGPXQWY + 0x8 * 3); - dev_printk(level, sdev->dev, "ROM feature bit%s enabled\n", - romdbgsts & BIT(24) ? "" : " not"); + + if (fwsts != 0xffffffff) + dev_err(sdev->dev, "Firmware state: %#x, status/error code: %#x\n", + fwsts, fwlec); sof_ipc4_intel_dump_telemetry_state(sdev, flags); } From 38d5387b7660476fd7e8e07d16ee436819e3544d Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Wed, 3 Apr 2024 13:03:18 +0200 Subject: [PATCH 183/603] ASoC: codecs: rk3308: fix "defined but not used" warning on !OF Building with CONFIG_OF=n triggers: warning: 'rk3308_codec_of_match' defined but not used [-Wunused-const-variable=] warning: unused variable 'rk3308_codec_of_match' [-Wunused-const-variable] Even though OF is needed for probing, fix by declaring as __maybe_unused to still allow building on non-OF configurations for build testing. Fixes: 9fdd7b45da18 ("arm64: defconfig: enable Rockchip RK3308 internal audio codec driver") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202403271905.BYbGJiPi-lkp@intel.com/ Closes: https://lore.kernel.org/oe-kbuild-all/202403271907.0z0uuG5I-lkp@intel.com/ Signed-off-by: Luca Ceresoli Link: https://msgid.link/r/20240403-rk3308-audio-codec-fix-warning-v2-1-816bae4c1dc5@bootlin.com Signed-off-by: Mark Brown --- sound/soc/codecs/rk3308_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk3308_codec.c b/sound/soc/codecs/rk3308_codec.c index 9d3e4691a7b53..74c3836995b15 100644 --- a/sound/soc/codecs/rk3308_codec.c +++ b/sound/soc/codecs/rk3308_codec.c @@ -953,7 +953,7 @@ static int rk3308_codec_platform_probe(struct platform_device *pdev) return 0; } -static const struct of_device_id rk3308_codec_of_match[] = { +static const struct of_device_id __maybe_unused rk3308_codec_of_match[] = { { .compatible = "rockchip,rk3308-codec", }, {}, }; From 03246ade4646653bbc98fa6fa506891a51983259 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Wed, 3 Apr 2024 13:03:19 +0200 Subject: [PATCH 184/603] ASoC: codecs: rk3308: depend on ARM64 || COMPILE_TEST This codec is only known to exist in the RK3308 ARM64 SoC, so depend on it except for compile test cases. Note that the driver won't probe without CONFIG_OF, but ARM64 selects OF already so it is not needed. Suggested-by: Mark Brown Signed-off-by: Luca Ceresoli Link: https://msgid.link/r/20240403-rk3308-audio-codec-fix-warning-v2-2-816bae4c1dc5@bootlin.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1752814fffdda..4e47bb6850aac 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1436,6 +1436,7 @@ config SND_SOC_PEB2466 config SND_SOC_RK3308 tristate "Rockchip RK3308 audio CODEC" + depends on ARM64 || COMPILE_TEST select REGMAP_MMIO help This is a device driver for the audio codec embedded in the From a358f67d16942f3bbc4d6b0e4b65efd5dda1c304 Mon Sep 17 00:00:00 2001 From: Laurentiu Mihalcea Date: Fri, 29 Mar 2024 00:12:00 +0200 Subject: [PATCH 185/603] ASoC: SOF: imx: drop usage of the imx8_*_clocks API Currently, the driver has to keep track of all the clocks it uses via an array of "struct clk_bulk_data", which doesn't scale well and is unnecessary. As such, replace the usage of the imx8_*_clocks with "devm_clk_bulk_get_all()" and friends. Signed-off-by: Laurentiu Mihalcea Reviewed-by: Iuliana Prodan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Daniel Baluta Link: https://msgid.link/r/20240328221201.24722-2-laurentiumihalcea111@gmail.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8.c | 41 +++++++++++++++--------------------- sound/soc/sof/imx/imx8m.c | 40 +++++++++++++++-------------------- sound/soc/sof/imx/imx8ulp.c | 42 +++++++++++++++++-------------------- 3 files changed, 53 insertions(+), 70 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 07f51489d6c9d..d65e09a927f62 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -41,13 +41,6 @@ #define MBOX_OFFSET 0x800000 #define MBOX_SIZE 0x1000 -/* DSP clocks */ -static struct clk_bulk_data imx8_dsp_clks[] = { - { .id = "ipg" }, - { .id = "ocram" }, - { .id = "core" }, -}; - struct imx8_priv { struct device *dev; struct snd_sof_dev *sdev; @@ -64,7 +57,8 @@ struct imx8_priv { struct device **pd_dev; struct device_link **link; - struct imx_clocks *clks; + struct clk_bulk_data *clks; + int clk_num; }; static int imx8_get_mailbox_offset(struct snd_sof_dev *sdev) @@ -196,10 +190,6 @@ static int imx8_probe(struct snd_sof_dev *sdev) if (!priv) return -ENOMEM; - priv->clks = devm_kzalloc(&pdev->dev, sizeof(*priv->clks), GFP_KERNEL); - if (!priv->clks) - return -ENOMEM; - sdev->num_cores = 1; sdev->pdata->hw_pdata = priv; priv->dev = sdev->dev; @@ -313,17 +303,18 @@ static int imx8_probe(struct snd_sof_dev *sdev) /* set default mailbox offset for FW ready message */ sdev->dsp_box.offset = MBOX_OFFSET; - /* init clocks info */ - priv->clks->dsp_clks = imx8_dsp_clks; - priv->clks->num_dsp_clks = ARRAY_SIZE(imx8_dsp_clks); - - ret = imx8_parse_clocks(sdev, priv->clks); - if (ret < 0) + ret = devm_clk_bulk_get_all(sdev->dev, &priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to fetch clocks: %d\n", ret); goto exit_pdev_unregister; + } + priv->clk_num = ret; - ret = imx8_enable_clocks(sdev, priv->clks); - if (ret < 0) + ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable clocks: %d\n", ret); goto exit_pdev_unregister; + } return 0; @@ -343,7 +334,7 @@ static void imx8_remove(struct snd_sof_dev *sdev) struct imx8_priv *priv = sdev->pdata->hw_pdata; int i; - imx8_disable_clocks(sdev, priv->clks); + clk_bulk_disable_unprepare(priv->clk_num, priv->clks); platform_device_unregister(priv->ipc_dev); for (i = 0; i < priv->num_domains; i++) { @@ -373,7 +364,7 @@ static void imx8_suspend(struct snd_sof_dev *sdev) for (i = 0; i < DSP_MU_CHAN_NUM; i++) imx_dsp_free_channel(priv->dsp_ipc, i); - imx8_disable_clocks(sdev, priv->clks); + clk_bulk_disable_unprepare(priv->clk_num, priv->clks); } static int imx8_resume(struct snd_sof_dev *sdev) @@ -382,9 +373,11 @@ static int imx8_resume(struct snd_sof_dev *sdev) int ret; int i; - ret = imx8_enable_clocks(sdev, priv->clks); - if (ret < 0) + ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable clocks: %d\n", ret); return ret; + } for (i = 0; i < DSP_MU_CHAN_NUM; i++) imx_dsp_request_channel(priv->dsp_ipc, i); diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 222cd1467da67..3e0b88002b4f6 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -26,12 +26,6 @@ #define MBOX_OFFSET 0x800000 #define MBOX_SIZE 0x1000 -static struct clk_bulk_data imx8m_dsp_clks[] = { - { .id = "ipg" }, - { .id = "ocram" }, - { .id = "core" }, -}; - /* DAP registers */ #define IMX8M_DAP_DEBUG 0x28800000 #define IMX8M_DAP_DEBUG_SIZE (64 * 1024) @@ -54,7 +48,8 @@ struct imx8m_priv { struct imx_dsp_ipc *dsp_ipc; struct platform_device *ipc_dev; - struct imx_clocks *clks; + struct clk_bulk_data *clks; + int clk_num; void __iomem *dap; struct regmap *regmap; @@ -163,10 +158,6 @@ static int imx8m_probe(struct snd_sof_dev *sdev) if (!priv) return -ENOMEM; - priv->clks = devm_kzalloc(&pdev->dev, sizeof(*priv->clks), GFP_KERNEL); - if (!priv->clks) - return -ENOMEM; - sdev->num_cores = 1; sdev->pdata->hw_pdata = priv; priv->dev = sdev->dev; @@ -250,17 +241,18 @@ static int imx8m_probe(struct snd_sof_dev *sdev) goto exit_pdev_unregister; } - /* init clocks info */ - priv->clks->dsp_clks = imx8m_dsp_clks; - priv->clks->num_dsp_clks = ARRAY_SIZE(imx8m_dsp_clks); - - ret = imx8_parse_clocks(sdev, priv->clks); - if (ret < 0) + ret = devm_clk_bulk_get_all(sdev->dev, &priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to fetch clocks: %d\n", ret); goto exit_pdev_unregister; + } + priv->clk_num = ret; - ret = imx8_enable_clocks(sdev, priv->clks); - if (ret < 0) + ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable clocks: %d\n", ret); goto exit_pdev_unregister; + } return 0; @@ -273,7 +265,7 @@ static void imx8m_remove(struct snd_sof_dev *sdev) { struct imx8m_priv *priv = sdev->pdata->hw_pdata; - imx8_disable_clocks(sdev, priv->clks); + clk_bulk_disable_unprepare(priv->clk_num, priv->clks); platform_device_unregister(priv->ipc_dev); } @@ -336,9 +328,11 @@ static int imx8m_resume(struct snd_sof_dev *sdev) int ret; int i; - ret = imx8_enable_clocks(sdev, priv->clks); - if (ret < 0) + ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable clocks: %d\n", ret); return ret; + } for (i = 0; i < DSP_MU_CHAN_NUM; i++) imx_dsp_request_channel(priv->dsp_ipc, i); @@ -354,7 +348,7 @@ static void imx8m_suspend(struct snd_sof_dev *sdev) for (i = 0; i < DSP_MU_CHAN_NUM; i++) imx_dsp_free_channel(priv->dsp_ipc, i); - imx8_disable_clocks(sdev, priv->clks); + clk_bulk_disable_unprepare(priv->clk_num, priv->clks); } static int imx8m_dsp_runtime_resume(struct snd_sof_dev *sdev) diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 7b527ffde488f..296fbf11f6d01 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -40,13 +40,6 @@ #define MBOX_OFFSET 0x800000 #define MBOX_SIZE 0x1000 -static struct clk_bulk_data imx8ulp_dsp_clks[] = { - { .id = "core" }, - { .id = "ipg" }, - { .id = "ocram" }, - { .id = "mu" }, -}; - struct imx8ulp_priv { struct device *dev; struct snd_sof_dev *sdev; @@ -56,7 +49,8 @@ struct imx8ulp_priv { struct platform_device *ipc_dev; struct regmap *regmap; - struct imx_clocks *clks; + struct clk_bulk_data *clks; + int clk_num; }; static void imx8ulp_sim_lpav_start(struct imx8ulp_priv *priv) @@ -175,10 +169,6 @@ static int imx8ulp_probe(struct snd_sof_dev *sdev) if (!priv) return -ENOMEM; - priv->clks = devm_kzalloc(&pdev->dev, sizeof(*priv->clks), GFP_KERNEL); - if (!priv->clks) - return -ENOMEM; - sdev->num_cores = 1; sdev->pdata->hw_pdata = priv; priv->dev = sdev->dev; @@ -259,16 +249,18 @@ static int imx8ulp_probe(struct snd_sof_dev *sdev) goto exit_pdev_unregister; } - priv->clks->dsp_clks = imx8ulp_dsp_clks; - priv->clks->num_dsp_clks = ARRAY_SIZE(imx8ulp_dsp_clks); - - ret = imx8_parse_clocks(sdev, priv->clks); - if (ret < 0) + ret = devm_clk_bulk_get_all(sdev->dev, &priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to fetch clocks: %d\n", ret); goto exit_pdev_unregister; + } + priv->clk_num = ret; - ret = imx8_enable_clocks(sdev, priv->clks); - if (ret < 0) + ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable clocks: %d\n", ret); goto exit_pdev_unregister; + } return 0; @@ -282,7 +274,7 @@ static void imx8ulp_remove(struct snd_sof_dev *sdev) { struct imx8ulp_priv *priv = sdev->pdata->hw_pdata; - imx8_disable_clocks(sdev, priv->clks); + clk_bulk_disable_unprepare(priv->clk_num, priv->clks); platform_device_unregister(priv->ipc_dev); } @@ -303,7 +295,7 @@ static int imx8ulp_suspend(struct snd_sof_dev *sdev) for (i = 0; i < DSP_MU_CHAN_NUM; i++) imx_dsp_free_channel(priv->dsp_ipc, i); - imx8_disable_clocks(sdev, priv->clks); + clk_bulk_disable_unprepare(priv->clk_num, priv->clks); return 0; } @@ -311,9 +303,13 @@ static int imx8ulp_suspend(struct snd_sof_dev *sdev) static int imx8ulp_resume(struct snd_sof_dev *sdev) { struct imx8ulp_priv *priv = (struct imx8ulp_priv *)sdev->pdata->hw_pdata; - int i; + int i, ret; - imx8_enable_clocks(sdev, priv->clks); + ret = clk_bulk_prepare_enable(priv->clk_num, priv->clks); + if (ret < 0) { + dev_err(sdev->dev, "failed to enable clocks: %d\n", ret); + return ret; + } for (i = 0; i < DSP_MU_CHAN_NUM; i++) imx_dsp_request_channel(priv->dsp_ipc, i); From e618b8b37aa1dac42f50c87373d71ecc68ead97d Mon Sep 17 00:00:00 2001 From: Laurentiu Mihalcea Date: Fri, 29 Mar 2024 00:12:01 +0200 Subject: [PATCH 186/603] ASoC: SOF: imx: remove imx8_*_clocks API Since the i.MX drivers no longer use the imx8_*_clocks API this can be removed. Signed-off-by: Laurentiu Mihalcea Reviewed-by: Iuliana Prodan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Daniel Baluta Link: https://msgid.link/r/20240328221201.24722-3-laurentiumihalcea111@gmail.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx-common.c | 24 ------------------------ sound/soc/sof/imx/imx-common.h | 9 --------- 2 files changed, 33 deletions(-) diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c index 36e3d414a18f4..2981aea123d97 100644 --- a/sound/soc/sof/imx/imx-common.c +++ b/sound/soc/sof/imx/imx-common.c @@ -74,28 +74,4 @@ void imx8_dump(struct snd_sof_dev *sdev, u32 flags) } EXPORT_SYMBOL(imx8_dump); -int imx8_parse_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks) -{ - int ret; - - ret = devm_clk_bulk_get(sdev->dev, clks->num_dsp_clks, clks->dsp_clks); - if (ret) - dev_err(sdev->dev, "Failed to request DSP clocks\n"); - - return ret; -} -EXPORT_SYMBOL(imx8_parse_clocks); - -int imx8_enable_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks) -{ - return clk_bulk_prepare_enable(clks->num_dsp_clks, clks->dsp_clks); -} -EXPORT_SYMBOL(imx8_enable_clocks); - -void imx8_disable_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks) -{ - clk_bulk_disable_unprepare(clks->num_dsp_clks, clks->dsp_clks); -} -EXPORT_SYMBOL(imx8_disable_clocks); - MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/imx/imx-common.h b/sound/soc/sof/imx/imx-common.h index ec4b3a5c7496a..13d7f3ef675e0 100644 --- a/sound/soc/sof/imx/imx-common.h +++ b/sound/soc/sof/imx/imx-common.h @@ -15,13 +15,4 @@ void imx8_get_registers(struct snd_sof_dev *sdev, void imx8_dump(struct snd_sof_dev *sdev, u32 flags); -struct imx_clocks { - struct clk_bulk_data *dsp_clks; - int num_dsp_clks; -}; - -int imx8_parse_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks); -int imx8_enable_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks); -void imx8_disable_clocks(struct snd_sof_dev *sdev, struct imx_clocks *clks); - #endif From 4d0333798ebbfa1683cc3bc056d1b25b8c24344c Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 11:31:23 +0000 Subject: [PATCH 187/603] firmware: cs_dsp: Add locked wrappers for coeff read and write It is a common pattern for functions to take and release the DSP pwr_lock over the cs_dsp calls to read and write firmware controls. Add wrapper functions to do this sequence so that the calling code can be simplified to a single function call.. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Reviewed-by: Takashi Iwai Link: https://msgid.link/r/20240325113127.112783-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- drivers/firmware/cirrus/cs_dsp.c | 54 ++++++++++++++++++++++++++ include/linux/firmware/cirrus/cs_dsp.h | 4 ++ 2 files changed, 58 insertions(+) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 79d4254d1f9bc..1263382e0d100 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -819,6 +819,33 @@ int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, } EXPORT_SYMBOL_NS_GPL(cs_dsp_coeff_write_ctrl, FW_CS_DSP); +/** + * cs_dsp_coeff_lock_and_write_ctrl() - Writes the given buffer to the given coefficient control + * @ctl: pointer to coefficient control + * @off: word offset at which data should be written + * @buf: the buffer to write to the given control + * @len: the length of the buffer in bytes + * + * Same as cs_dsp_coeff_write_ctrl() but takes pwr_lock. + * + * Return: A negative number on error, 1 when the control value changed and 0 when it has not. + */ +int cs_dsp_coeff_lock_and_write_ctrl(struct cs_dsp_coeff_ctl *ctl, + unsigned int off, const void *buf, size_t len) +{ + struct cs_dsp *dsp = ctl->dsp; + int ret; + + lockdep_assert_not_held(&dsp->pwr_lock); + + mutex_lock(&dsp->pwr_lock); + ret = cs_dsp_coeff_write_ctrl(ctl, off, buf, len); + mutex_unlock(&dsp->pwr_lock); + + return ret; +} +EXPORT_SYMBOL_GPL(cs_dsp_coeff_lock_and_write_ctrl); + static int cs_dsp_coeff_read_ctrl_raw(struct cs_dsp_coeff_ctl *ctl, unsigned int off, void *buf, size_t len) { @@ -891,6 +918,33 @@ int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl, } EXPORT_SYMBOL_NS_GPL(cs_dsp_coeff_read_ctrl, FW_CS_DSP); +/** + * cs_dsp_coeff_lock_and_read_ctrl() - Reads the given coefficient control into the given buffer + * @ctl: pointer to coefficient control + * @off: word offset at which data should be read + * @buf: the buffer to store to the given control + * @len: the length of the buffer in bytes + * + * Same as cs_dsp_coeff_read_ctrl() but takes pwr_lock. + * + * Return: Zero for success, a negative number on error. + */ +int cs_dsp_coeff_lock_and_read_ctrl(struct cs_dsp_coeff_ctl *ctl, + unsigned int off, void *buf, size_t len) +{ + struct cs_dsp *dsp = ctl->dsp; + int ret; + + lockdep_assert_not_held(&dsp->pwr_lock); + + mutex_lock(&dsp->pwr_lock); + ret = cs_dsp_coeff_read_ctrl(ctl, off, buf, len); + mutex_unlock(&dsp->pwr_lock); + + return ret; +} +EXPORT_SYMBOL_GPL(cs_dsp_coeff_lock_and_read_ctrl); + static int cs_dsp_coeff_init_control_caches(struct cs_dsp *dsp) { struct cs_dsp_coeff_ctl *ctl; diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 29cd11d5a3cfd..6097b1f701f12 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -239,8 +239,12 @@ void cs_dsp_cleanup_debugfs(struct cs_dsp *dsp); int cs_dsp_coeff_write_acked_control(struct cs_dsp_coeff_ctl *ctl, unsigned int event_id); int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off, const void *buf, size_t len); +int cs_dsp_coeff_lock_and_write_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off, + const void *buf, size_t len); int cs_dsp_coeff_read_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off, void *buf, size_t len); +int cs_dsp_coeff_lock_and_read_ctrl(struct cs_dsp_coeff_ctl *ctl, unsigned int off, + void *buf, size_t len); struct cs_dsp_coeff_ctl *cs_dsp_get_ctl(struct cs_dsp *dsp, const char *name, int type, unsigned int alg); From 3802a9969bd365749f5e34928082cff96ed7769b Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 11:31:24 +0000 Subject: [PATCH 188/603] ASoC: wm_adsp: Use cs_dsp_coeff_lock_and_[read|write]_ctrl() Using the cs_dsp_coeff_lock_and_[read|write]_ctrl() wrappers tidies the calling functions as it does not need to manage the DSP pwr_lock. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Reviewed-by: Takashi Iwai Link: https://msgid.link/r/20240325113127.112783-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 24 ++++++------------------ 1 file changed, 6 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9cb9068c0462a..f41d668e4c01a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -403,13 +403,8 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; char *p = ucontrol->value.bytes.data; - int ret = 0; - - mutex_lock(&cs_ctl->dsp->pwr_lock); - ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, p, cs_ctl->len); - mutex_unlock(&cs_ctl->dsp->pwr_lock); - return ret; + return cs_dsp_coeff_lock_and_write_ctrl(cs_ctl, 0, p, cs_ctl->len); } static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, @@ -426,13 +421,11 @@ static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, if (!scratch) return -ENOMEM; - if (copy_from_user(scratch, bytes, size)) { + if (copy_from_user(scratch, bytes, size)) ret = -EFAULT; - } else { - mutex_lock(&cs_ctl->dsp->pwr_lock); - ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, scratch, size); - mutex_unlock(&cs_ctl->dsp->pwr_lock); - } + else + ret = cs_dsp_coeff_lock_and_write_ctrl(cs_ctl, 0, scratch, size); + vfree(scratch); return ret; @@ -474,13 +467,8 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; char *p = ucontrol->value.bytes.data; - int ret; - mutex_lock(&cs_ctl->dsp->pwr_lock); - ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, p, cs_ctl->len); - mutex_unlock(&cs_ctl->dsp->pwr_lock); - - return ret; + return cs_dsp_coeff_lock_and_read_ctrl(cs_ctl, 0, p, cs_ctl->len); } static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, From 62daf3df8a6b1920f7613e478935443a8f449708 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 11:31:25 +0000 Subject: [PATCH 189/603] ALSA: hda: hda_cs_dsp_ctl: Use cs_dsp_coeff_lock_and_[read|write]_ctrl() Using the cs_dsp_coeff_lock_and_[read|write]_ctrl() wrappers tidies the calling functions as it does not need to manage the DSP pwr_lock. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Reviewed-by: Takashi Iwai Link: https://msgid.link/r/20240325113127.112783-4-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/pci/hda/hda_cs_dsp_ctl.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 463ca06036bfe..725544d819410 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -51,13 +51,8 @@ static int hda_cs_dsp_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_v struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; char *p = ucontrol->value.bytes.data; - int ret = 0; - - mutex_lock(&cs_ctl->dsp->pwr_lock); - ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, p, cs_ctl->len); - mutex_unlock(&cs_ctl->dsp->pwr_lock); - return ret; + return cs_dsp_coeff_lock_and_write_ctrl(cs_ctl, 0, p, cs_ctl->len); } static int hda_cs_dsp_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) @@ -65,13 +60,8 @@ static int hda_cs_dsp_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_v struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; char *p = ucontrol->value.bytes.data; - int ret; - - mutex_lock(&cs_ctl->dsp->pwr_lock); - ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, p, cs_ctl->len); - mutex_unlock(&cs_ctl->dsp->pwr_lock); - return ret; + return cs_dsp_coeff_lock_and_read_ctrl(cs_ctl, 0, p, cs_ctl->len); } static unsigned int wmfw_convert_flags(unsigned int in) From e81f5c9f7d06a69dc505fa6ad351df6cc86a6c2d Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 11:31:26 +0000 Subject: [PATCH 190/603] ASoC: wm_adsp: Remove notification of driver write Any control that the driver is updating should be marked as SYSTEM and therefore will not have an ALSA control to notify. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Reviewed-by: Takashi Iwai Link: https://msgid.link/r/20240325113127.112783-5-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f41d668e4c01a..517f46fb9130d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -672,7 +672,6 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { struct cs_dsp_coeff_ctl *cs_ctl; - struct wm_coeff_ctl *ctl; int ret; mutex_lock(&dsp->cs_dsp.pwr_lock); @@ -683,12 +682,7 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, if (ret < 0) return ret; - if (ret == 0 || (cs_ctl->flags & WMFW_CTL_FLAG_SYS)) - return 0; - - ctl = cs_ctl->priv; - - return snd_soc_component_notify_control(dsp->component, ctl->name); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp_write_ctl); From d641def12ec929af6c4f9b1b28efcd3e5dff21b4 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 25 Mar 2024 11:31:27 +0000 Subject: [PATCH 191/603] ALSA: hda: hda_cs_dsp_ctl: Remove notification of driver write Any control that the driver is updating should be marked as SYSTEM and therefore will not have an ALSA control to notify. Signed-off-by: Simon Trimmer Signed-off-by: Richard Fitzgerald Reviewed-by: Takashi Iwai Link: https://msgid.link/r/20240325113127.112783-6-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/pci/hda/hda_cs_dsp_ctl.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 725544d819410..7f2d35cf245b1 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -201,7 +201,6 @@ int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, unsigned int alg, const void *buf, size_t len) { struct cs_dsp_coeff_ctl *cs_ctl; - struct hda_cs_dsp_coeff_ctl *ctl; int ret; mutex_lock(&dsp->pwr_lock); @@ -211,13 +210,6 @@ int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, if (ret < 0) return ret; - if (ret == 0 || (cs_ctl->flags & WMFW_CTL_FLAG_SYS)) - return 0; - - ctl = cs_ctl->priv; - - snd_ctl_notify(ctl->card, SNDRV_CTL_EVENT_MASK_VALUE, &ctl->kctl->id); - return 0; } EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_write_ctl, SND_HDA_CS_DSP_CONTROLS); From 2ba2d17170f9e415fc31d7978214f0f26e8afc64 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 4 Apr 2024 11:57:34 +0200 Subject: [PATCH 192/603] ASoC: codecs: rk3308: fix building without CONFIG_OF The newly added codec has an extraneous of_match_ptr() annotation for the ID table: sound/soc/codecs/rk3308_codec.c:956:34: error: 'rk3308_codec_of_match' defined but not used [-Werror=unused-const-variable=] 956 | static const struct of_device_id rk3308_codec_of_match[] = { Remove it to avoid introducing a warning when -Wunused-const-variable gets enabled by default: Fixes: 4ed0915f5bc4 ("ASoC: codecs: Add RK3308 internal audio codec driver") Signed-off-by: Arnd Bergmann Link: https://msgid.link/r/20240404095755.650364-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/rk3308_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk3308_codec.c b/sound/soc/codecs/rk3308_codec.c index 74c3836995b15..8b51e87a17115 100644 --- a/sound/soc/codecs/rk3308_codec.c +++ b/sound/soc/codecs/rk3308_codec.c @@ -962,7 +962,7 @@ MODULE_DEVICE_TABLE(of, rk3308_codec_of_match); static struct platform_driver rk3308_codec_driver = { .driver = { .name = "rk3308-acodec", - .of_match_table = of_match_ptr(rk3308_codec_of_match), + .of_match_table = rk3308_codec_of_match, }, .probe = rk3308_codec_platform_probe, }; From 9f1aeef4cbcb0e8c616e29cdce3232f9f102c2c1 Mon Sep 17 00:00:00 2001 From: Francesco Dolcini Date: Thu, 4 Apr 2024 13:19:53 +0200 Subject: [PATCH 193/603] ASoC: nau8822: Remove unused member from struct Remove unused mclk_idx struct member. Signed-off-by: Francesco Dolcini Link: https://msgid.link/r/20240404111953.99287-1-francesco@dolcini.it Signed-off-by: Mark Brown --- sound/soc/codecs/nau8822.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 646f6bb64bc5b..6ecd46e459230 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -215,7 +215,6 @@ struct nau8822_pll { struct nau8822 { struct device *dev; struct regmap *regmap; - int mclk_idx; struct nau8822_pll pll; int sysclk; int div_id; From 251ea652050e715886b272f831cf5016150a60c8 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 3 Apr 2024 11:16:26 +0200 Subject: [PATCH 194/603] ASoC: Intel: Skylake: Remove soc-topology ABI v4 support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The only known users are Chromebook configurations. Starting from kernel v5.4, all of them are making use of soc-topology ABI v5. Cc: Curtis Malainey Cc: Łukasz Majczak Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240403091629.647267-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- include/uapi/sound/skl-tplg-interface.h | 74 ----------- sound/soc/intel/skylake/skl-topology.c | 169 ------------------------ 2 files changed, 243 deletions(-) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index 4bf9c4f9add8a..940c4269322bd 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -165,78 +165,4 @@ enum skl_tuple_type { SKL_TYPE_DATA }; -/* v4 configuration data */ - -struct skl_dfw_v4_module_pin { - __u16 module_id; - __u16 instance_id; -} __packed; - -struct skl_dfw_v4_module_fmt { - __u32 channels; - __u32 freq; - __u32 bit_depth; - __u32 valid_bit_depth; - __u32 ch_cfg; - __u32 interleaving_style; - __u32 sample_type; - __u32 ch_map; -} __packed; - -struct skl_dfw_v4_module_caps { - __u32 set_params:2; - __u32 rsvd:30; - __u32 param_id; - __u32 caps_size; - __u32 caps[HDA_SST_CFG_MAX]; -} __packed; - -struct skl_dfw_v4_pipe { - __u8 pipe_id; - __u8 pipe_priority; - __u16 conn_type:4; - __u16 rsvd:4; - __u16 memory_pages:8; -} __packed; - -struct skl_dfw_v4_module { - char uuid[SKL_UUID_STR_SZ]; - - __u16 module_id; - __u16 instance_id; - __u32 max_mcps; - __u32 mem_pages; - __u32 obs; - __u32 ibs; - __u32 vbus_id; - - __u32 max_in_queue:8; - __u32 max_out_queue:8; - __u32 time_slot:8; - __u32 core_id:4; - __u32 rsvd1:4; - - __u32 module_type:8; - __u32 conn_type:4; - __u32 dev_type:4; - __u32 hw_conn_type:4; - __u32 rsvd2:12; - - __u32 params_fixup:8; - __u32 converter:8; - __u32 input_pin_type:1; - __u32 output_pin_type:1; - __u32 is_dynamic_in_pin:1; - __u32 is_dynamic_out_pin:1; - __u32 is_loadable:1; - __u32 rsvd3:11; - - struct skl_dfw_v4_pipe pipe; - struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE]; - struct skl_dfw_v4_module_fmt out_fmt[MAX_OUT_QUEUE]; - struct skl_dfw_v4_module_pin in_pin[MAX_IN_QUEUE]; - struct skl_dfw_v4_module_pin out_pin[MAX_OUT_QUEUE]; - struct skl_dfw_v4_module_caps caps; -} __packed; - #endif diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 96cfebded0724..e27f0fc3d8979 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2682,168 +2682,6 @@ static int skl_tplg_get_desc_blocks(struct device *dev, return -EINVAL; } -/* Functions to parse private data from configuration file format v4 */ - -/* - * Add pipeline from topology binary into driver pipeline list - * - * If already added we return that instance - * Otherwise we create a new instance and add into driver list - */ -static int skl_tplg_add_pipe_v4(struct device *dev, - struct skl_module_cfg *mconfig, struct skl_dev *skl, - struct skl_dfw_v4_pipe *dfw_pipe) -{ - struct skl_pipeline *ppl; - struct skl_pipe *pipe; - struct skl_pipe_params *params; - - list_for_each_entry(ppl, &skl->ppl_list, node) { - if (ppl->pipe->ppl_id == dfw_pipe->pipe_id) { - mconfig->pipe = ppl->pipe; - return 0; - } - } - - ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL); - if (!ppl) - return -ENOMEM; - - pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL); - if (!pipe) - return -ENOMEM; - - params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL); - if (!params) - return -ENOMEM; - - pipe->ppl_id = dfw_pipe->pipe_id; - pipe->memory_pages = dfw_pipe->memory_pages; - pipe->pipe_priority = dfw_pipe->pipe_priority; - pipe->conn_type = dfw_pipe->conn_type; - pipe->state = SKL_PIPE_INVALID; - pipe->p_params = params; - INIT_LIST_HEAD(&pipe->w_list); - - ppl->pipe = pipe; - list_add(&ppl->node, &skl->ppl_list); - - mconfig->pipe = pipe; - - return 0; -} - -static void skl_fill_module_pin_info_v4(struct skl_dfw_v4_module_pin *dfw_pin, - struct skl_module_pin *m_pin, - bool is_dynamic, int max_pin) -{ - int i; - - for (i = 0; i < max_pin; i++) { - m_pin[i].id.module_id = dfw_pin[i].module_id; - m_pin[i].id.instance_id = dfw_pin[i].instance_id; - m_pin[i].in_use = false; - m_pin[i].is_dynamic = is_dynamic; - m_pin[i].pin_state = SKL_PIN_UNBIND; - } -} - -static void skl_tplg_fill_fmt_v4(struct skl_module_pin_fmt *dst_fmt, - struct skl_dfw_v4_module_fmt *src_fmt, - int pins) -{ - int i; - - for (i = 0; i < pins; i++) { - dst_fmt[i].fmt.channels = src_fmt[i].channels; - dst_fmt[i].fmt.s_freq = src_fmt[i].freq; - dst_fmt[i].fmt.bit_depth = src_fmt[i].bit_depth; - dst_fmt[i].fmt.valid_bit_depth = src_fmt[i].valid_bit_depth; - dst_fmt[i].fmt.ch_cfg = src_fmt[i].ch_cfg; - dst_fmt[i].fmt.ch_map = src_fmt[i].ch_map; - dst_fmt[i].fmt.interleaving_style = - src_fmt[i].interleaving_style; - dst_fmt[i].fmt.sample_type = src_fmt[i].sample_type; - } -} - -static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w, - struct skl_dev *skl, struct device *dev, - struct skl_module_cfg *mconfig) -{ - struct skl_dfw_v4_module *dfw = - (struct skl_dfw_v4_module *)tplg_w->priv.data; - int ret; - int idx = mconfig->fmt_cfg_idx; - - dev_dbg(dev, "Parsing Skylake v4 widget topology data\n"); - - ret = guid_parse(dfw->uuid, (guid_t *)mconfig->guid); - if (ret) - return ret; - mconfig->id.module_id = -1; - mconfig->id.instance_id = dfw->instance_id; - mconfig->module->resources[0].cpc = dfw->max_mcps / 1000; - mconfig->module->resources[0].ibs = dfw->ibs; - mconfig->module->resources[0].obs = dfw->obs; - mconfig->core_id = dfw->core_id; - mconfig->module->max_input_pins = dfw->max_in_queue; - mconfig->module->max_output_pins = dfw->max_out_queue; - mconfig->module->loadable = dfw->is_loadable; - skl_tplg_fill_fmt_v4(mconfig->module->formats[0].inputs, dfw->in_fmt, - MAX_IN_QUEUE); - skl_tplg_fill_fmt_v4(mconfig->module->formats[0].outputs, dfw->out_fmt, - MAX_OUT_QUEUE); - - mconfig->params_fixup = dfw->params_fixup; - mconfig->converter = dfw->converter; - mconfig->m_type = dfw->module_type; - mconfig->vbus_id = dfw->vbus_id; - mconfig->module->resources[0].is_pages = dfw->mem_pages; - - ret = skl_tplg_add_pipe_v4(dev, mconfig, skl, &dfw->pipe); - if (ret) - return ret; - - mconfig->dev_type = dfw->dev_type; - mconfig->hw_conn_type = dfw->hw_conn_type; - mconfig->time_slot = dfw->time_slot; - mconfig->formats_config[idx].caps_size = dfw->caps.caps_size; - - mconfig->m_in_pin = devm_kcalloc(dev, - MAX_IN_QUEUE, sizeof(*mconfig->m_in_pin), - GFP_KERNEL); - if (!mconfig->m_in_pin) - return -ENOMEM; - - mconfig->m_out_pin = devm_kcalloc(dev, - MAX_OUT_QUEUE, sizeof(*mconfig->m_out_pin), - GFP_KERNEL); - if (!mconfig->m_out_pin) - return -ENOMEM; - - skl_fill_module_pin_info_v4(dfw->in_pin, mconfig->m_in_pin, - dfw->is_dynamic_in_pin, - mconfig->module->max_input_pins); - skl_fill_module_pin_info_v4(dfw->out_pin, mconfig->m_out_pin, - dfw->is_dynamic_out_pin, - mconfig->module->max_output_pins); - - if (mconfig->formats_config[idx].caps_size) { - mconfig->formats_config[idx].set_params = dfw->caps.set_params; - mconfig->formats_config[idx].param_id = dfw->caps.param_id; - mconfig->formats_config[idx].caps = - devm_kzalloc(dev, mconfig->formats_config[idx].caps_size, - GFP_KERNEL); - if (!mconfig->formats_config[idx].caps) - return -ENOMEM; - memcpy(mconfig->formats_config[idx].caps, dfw->caps.caps, - dfw->caps.caps_size); - } - - return 0; -} - static int skl_tplg_get_caps_data(struct device *dev, char *data, struct skl_module_cfg *mconfig) { @@ -2877,13 +2715,6 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, char *data; int ret; - /* - * v4 configuration files have a valid UUID at the start of - * the widget's private data. - */ - if (uuid_is_valid((char *)tplg_w->priv.data)) - return skl_tplg_get_pvt_data_v4(tplg_w, skl, dev, mconfig); - /* Read the NUM_DATA_BLOCKS descriptor */ array = (struct snd_soc_tplg_vendor_array *)tplg_w->priv.data; ret = skl_tplg_get_desc_blocks(dev, array); From c57468dc1f80a4a3bf6dd271688837d3c1f26e75 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 3 Apr 2024 11:16:27 +0200 Subject: [PATCH 195/603] ASoC: topology: Remove ABI v4 support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There are no users of soc-topology ABI v4 since kernel v5.4 so remove all v4 -> v5 converters. Cc: Curtis Malainey Cc: Łukasz Majczak Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240403091629.647267-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 210 +++------------------------------------ 1 file changed, 16 insertions(+), 194 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index ba4890991f0d7..75a7481a4f182 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1761,75 +1761,6 @@ static int soc_tplg_pcm_create(struct soc_tplg *tplg, return soc_tplg_fe_link_create(tplg, pcm); } -/* copy stream caps from the old version 4 of source */ -static void stream_caps_new_ver(struct snd_soc_tplg_stream_caps *dest, - struct snd_soc_tplg_stream_caps_v4 *src) -{ - dest->size = cpu_to_le32(sizeof(*dest)); - memcpy(dest->name, src->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); - dest->formats = src->formats; - dest->rates = src->rates; - dest->rate_min = src->rate_min; - dest->rate_max = src->rate_max; - dest->channels_min = src->channels_min; - dest->channels_max = src->channels_max; - dest->periods_min = src->periods_min; - dest->periods_max = src->periods_max; - dest->period_size_min = src->period_size_min; - dest->period_size_max = src->period_size_max; - dest->buffer_size_min = src->buffer_size_min; - dest->buffer_size_max = src->buffer_size_max; -} - -/** - * pcm_new_ver - Create the new version of PCM from the old version. - * @tplg: topology context - * @src: older version of pcm as a source - * @pcm: latest version of pcm created from the source - * - * Support from version 4. User should free the returned pcm manually. - */ -static int pcm_new_ver(struct soc_tplg *tplg, - struct snd_soc_tplg_pcm *src, - struct snd_soc_tplg_pcm **pcm) -{ - struct snd_soc_tplg_pcm *dest; - struct snd_soc_tplg_pcm_v4 *src_v4; - int i; - - *pcm = NULL; - - if (le32_to_cpu(src->size) != sizeof(*src_v4)) { - dev_err(tplg->dev, "ASoC: invalid PCM size\n"); - return -EINVAL; - } - - dev_warn(tplg->dev, "ASoC: old version of PCM\n"); - src_v4 = (struct snd_soc_tplg_pcm_v4 *)src; - dest = kzalloc(sizeof(*dest), GFP_KERNEL); - if (!dest) - return -ENOMEM; - - dest->size = cpu_to_le32(sizeof(*dest)); /* size of latest abi version */ - memcpy(dest->pcm_name, src_v4->pcm_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); - memcpy(dest->dai_name, src_v4->dai_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); - dest->pcm_id = src_v4->pcm_id; - dest->dai_id = src_v4->dai_id; - dest->playback = src_v4->playback; - dest->capture = src_v4->capture; - dest->compress = src_v4->compress; - dest->num_streams = src_v4->num_streams; - for (i = 0; i < le32_to_cpu(dest->num_streams); i++) - memcpy(&dest->stream[i], &src_v4->stream[i], - sizeof(struct snd_soc_tplg_stream)); - - for (i = 0; i < 2; i++) - stream_caps_new_ver(&dest->caps[i], &src_v4->caps[i]); - - *pcm = dest; - return 0; -} - static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { @@ -1845,8 +1776,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, /* check the element size and count */ pcm = (struct snd_soc_tplg_pcm *)tplg->pos; size = le32_to_cpu(pcm->size); - if (size > sizeof(struct snd_soc_tplg_pcm) - || size < sizeof(struct snd_soc_tplg_pcm_v4)) { + if (size > sizeof(struct snd_soc_tplg_pcm)) { dev_err(tplg->dev, "ASoC: invalid size %d for PCM elems\n", size); return -EINVAL; @@ -1865,15 +1795,11 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, /* check ABI version by size, create a new version of pcm * if abi not match. */ - if (size == sizeof(*pcm)) { - abi_match = true; - _pcm = pcm; - } else { - abi_match = false; - ret = pcm_new_ver(tplg, pcm, &_pcm); - if (ret < 0) - return ret; - } + if (size != sizeof(*pcm)) + return -EINVAL; + + abi_match = true; + _pcm = pcm; /* create the FE DAIs and DAI links */ ret = soc_tplg_pcm_create(tplg, _pcm); @@ -1965,49 +1891,6 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, } } -/** - * link_new_ver - Create a new physical link config from the old - * version of source. - * @tplg: topology context - * @src: old version of phyical link config as a source - * @link: latest version of physical link config created from the source - * - * Support from version 4. User need free the returned link config manually. - */ -static int link_new_ver(struct soc_tplg *tplg, - struct snd_soc_tplg_link_config *src, - struct snd_soc_tplg_link_config **link) -{ - struct snd_soc_tplg_link_config *dest; - struct snd_soc_tplg_link_config_v4 *src_v4; - int i; - - *link = NULL; - - if (le32_to_cpu(src->size) != - sizeof(struct snd_soc_tplg_link_config_v4)) { - dev_err(tplg->dev, "ASoC: invalid physical link config size\n"); - return -EINVAL; - } - - dev_warn(tplg->dev, "ASoC: old version of physical link config\n"); - - src_v4 = (struct snd_soc_tplg_link_config_v4 *)src; - dest = kzalloc(sizeof(*dest), GFP_KERNEL); - if (!dest) - return -ENOMEM; - - dest->size = cpu_to_le32(sizeof(*dest)); - dest->id = src_v4->id; - dest->num_streams = src_v4->num_streams; - for (i = 0; i < le32_to_cpu(dest->num_streams); i++) - memcpy(&dest->stream[i], &src_v4->stream[i], - sizeof(struct snd_soc_tplg_stream)); - - *link = dest; - return 0; -} - /** * snd_soc_find_dai_link - Find a DAI link * @@ -2124,8 +2007,7 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, /* check the element size and count */ link = (struct snd_soc_tplg_link_config *)tplg->pos; size = le32_to_cpu(link->size); - if (size > sizeof(struct snd_soc_tplg_link_config) - || size < sizeof(struct snd_soc_tplg_link_config_v4)) { + if (size > sizeof(struct snd_soc_tplg_link_config)) { dev_err(tplg->dev, "ASoC: invalid size %d for physical link elems\n", size); return -EINVAL; @@ -2140,15 +2022,11 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { link = (struct snd_soc_tplg_link_config *)tplg->pos; size = le32_to_cpu(link->size); - if (size == sizeof(*link)) { - abi_match = true; - _link = link; - } else { - abi_match = false; - ret = link_new_ver(tplg, link, &_link); - if (ret < 0) - return ret; - } + if (size != sizeof(*link)) + return -EINVAL; + + abi_match = true; + _link = link; ret = soc_tplg_link_config(tplg, _link); if (ret < 0) { @@ -2273,57 +2151,6 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, return 0; } -/** - * manifest_new_ver - Create a new version of manifest from the old version - * of source. - * @tplg: topology context - * @src: old version of manifest as a source - * @manifest: latest version of manifest created from the source - * - * Support from version 4. Users need free the returned manifest manually. - */ -static int manifest_new_ver(struct soc_tplg *tplg, - struct snd_soc_tplg_manifest *src, - struct snd_soc_tplg_manifest **manifest) -{ - struct snd_soc_tplg_manifest *dest; - struct snd_soc_tplg_manifest_v4 *src_v4; - int size; - - *manifest = NULL; - - size = le32_to_cpu(src->size); - if (size != sizeof(*src_v4)) { - dev_warn(tplg->dev, "ASoC: invalid manifest size %d\n", - size); - if (size) - return -EINVAL; - src->size = cpu_to_le32(sizeof(*src_v4)); - } - - dev_warn(tplg->dev, "ASoC: old version of manifest\n"); - - src_v4 = (struct snd_soc_tplg_manifest_v4 *)src; - dest = kzalloc(sizeof(*dest) + le32_to_cpu(src_v4->priv.size), - GFP_KERNEL); - if (!dest) - return -ENOMEM; - - dest->size = cpu_to_le32(sizeof(*dest)); /* size of latest abi version */ - dest->control_elems = src_v4->control_elems; - dest->widget_elems = src_v4->widget_elems; - dest->graph_elems = src_v4->graph_elems; - dest->pcm_elems = src_v4->pcm_elems; - dest->dai_link_elems = src_v4->dai_link_elems; - dest->priv.size = src_v4->priv.size; - if (dest->priv.size) - memcpy(dest->priv.data, src_v4->priv.data, - le32_to_cpu(src_v4->priv.size)); - - *manifest = dest; - return 0; -} - static int soc_tplg_manifest_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { @@ -2334,16 +2161,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, manifest = (struct snd_soc_tplg_manifest *)tplg->pos; /* check ABI version by size, create a new manifest if abi not match */ - if (le32_to_cpu(manifest->size) == sizeof(*manifest)) { - abi_match = true; - _manifest = manifest; - } else { - abi_match = false; + if (le32_to_cpu(manifest->size) != sizeof(*manifest)) + return -EINVAL; - ret = manifest_new_ver(tplg, manifest, &_manifest); - if (ret < 0) - return ret; - } + abi_match = true; + _manifest = manifest; /* pass control to component driver for optional further init */ if (tplg->ops && tplg->ops->manifest) From 82c192540b242a8bf3c3d4140905650df9f327d4 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 3 Apr 2024 11:16:28 +0200 Subject: [PATCH 196/603] ASoC: topology: Cleanup after ABI v4 support removal MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Flag 'abi_match' along with several other local variables has no purpose with v4 related members removed. Cc: Curtis Malainey Cc: Łukasz Majczak Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240403091629.647267-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 47 +++++++++------------------------------- 1 file changed, 10 insertions(+), 37 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 75a7481a4f182..d6d3688372356 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1764,11 +1764,10 @@ static int soc_tplg_pcm_create(struct soc_tplg *tplg, static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { - struct snd_soc_tplg_pcm *pcm, *_pcm; + struct snd_soc_tplg_pcm *pcm; int count; int size; int i; - bool abi_match; int ret; count = le32_to_cpu(hdr->count); @@ -1798,24 +1797,15 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, if (size != sizeof(*pcm)) return -EINVAL; - abi_match = true; - _pcm = pcm; - /* create the FE DAIs and DAI links */ - ret = soc_tplg_pcm_create(tplg, _pcm); - if (ret < 0) { - if (!abi_match) - kfree(_pcm); + ret = soc_tplg_pcm_create(tplg, pcm); + if (ret < 0) return ret; - } /* offset by version-specific struct size and * real priv data size */ - tplg->pos += size + le32_to_cpu(_pcm->priv.size); - - if (!abi_match) - kfree(_pcm); /* free the duplicated one */ + tplg->pos += size + le32_to_cpu(pcm->priv.size); } dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count); @@ -1996,11 +1986,10 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, static int soc_tplg_link_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { - struct snd_soc_tplg_link_config *link, *_link; + struct snd_soc_tplg_link_config *link; int count; int size; int i, ret; - bool abi_match; count = le32_to_cpu(hdr->count); @@ -2025,23 +2014,14 @@ static int soc_tplg_link_elems_load(struct soc_tplg *tplg, if (size != sizeof(*link)) return -EINVAL; - abi_match = true; - _link = link; - - ret = soc_tplg_link_config(tplg, _link); - if (ret < 0) { - if (!abi_match) - kfree(_link); + ret = soc_tplg_link_config(tplg, link); + if (ret < 0) return ret; - } /* offset by version-specific struct size and * real priv data size */ - tplg->pos += size + le32_to_cpu(_link->priv.size); - - if (!abi_match) - kfree(_link); /* free the duplicated one */ + tplg->pos += size + le32_to_cpu(link->priv.size); } return 0; @@ -2154,8 +2134,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg, static int soc_tplg_manifest_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { - struct snd_soc_tplg_manifest *manifest, *_manifest; - bool abi_match; + struct snd_soc_tplg_manifest *manifest; int ret = 0; manifest = (struct snd_soc_tplg_manifest *)tplg->pos; @@ -2164,15 +2143,9 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, if (le32_to_cpu(manifest->size) != sizeof(*manifest)) return -EINVAL; - abi_match = true; - _manifest = manifest; - /* pass control to component driver for optional further init */ if (tplg->ops && tplg->ops->manifest) - ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest); - - if (!abi_match) /* free the duplicated one */ - kfree(_manifest); + ret = tplg->ops->manifest(tplg->comp, tplg->index, manifest); return ret; } From 4ba509bf3988f36b536b49b9e7022a6358f87f56 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 3 Apr 2024 11:16:29 +0200 Subject: [PATCH 197/603] ASoC: topology: Remove obsolete ABI v4 structs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit There are no users of soc-topology ABI v4 since kernel v5.4 and no kernel code makes use of them. Cc: Curtis Malainey Cc: Łukasz Majczak Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240403091629.647267-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 56 --------------------------------------- 1 file changed, 56 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 10851bca71740..99333cbd3114e 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -576,60 +576,4 @@ struct snd_soc_tplg_dai { struct snd_soc_tplg_private priv; } __attribute__((packed)); -/* - * Old version of ABI structs, supported for backward compatibility. - */ - -/* Manifest v4 */ -struct snd_soc_tplg_manifest_v4 { - __le32 size; /* in bytes of this structure */ - __le32 control_elems; /* number of control elements */ - __le32 widget_elems; /* number of widget elements */ - __le32 graph_elems; /* number of graph elements */ - __le32 pcm_elems; /* number of PCM elements */ - __le32 dai_link_elems; /* number of DAI link elements */ - struct snd_soc_tplg_private priv; -} __packed; - -/* Stream Capabilities v4 */ -struct snd_soc_tplg_stream_caps_v4 { - __le32 size; /* in bytes of this structure */ - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le64 formats; /* supported formats SNDRV_PCM_FMTBIT_* */ - __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ - __le32 rate_min; /* min rate */ - __le32 rate_max; /* max rate */ - __le32 channels_min; /* min channels */ - __le32 channels_max; /* max channels */ - __le32 periods_min; /* min number of periods */ - __le32 periods_max; /* max number of periods */ - __le32 period_size_min; /* min period size bytes */ - __le32 period_size_max; /* max period size bytes */ - __le32 buffer_size_min; /* min buffer size bytes */ - __le32 buffer_size_max; /* max buffer size bytes */ -} __packed; - -/* PCM v4 */ -struct snd_soc_tplg_pcm_v4 { - __le32 size; /* in bytes of this structure */ - char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le32 pcm_id; /* unique ID - used to match with DAI link */ - __le32 dai_id; /* unique ID - used to match */ - __le32 playback; /* supports playback mode */ - __le32 capture; /* supports capture mode */ - __le32 compress; /* 1 = compressed; 0 = PCM */ - struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */ - __le32 num_streams; /* number of streams */ - struct snd_soc_tplg_stream_caps_v4 caps[2]; /* playback and capture for DAI */ -} __packed; - -/* Physical link config v4 */ -struct snd_soc_tplg_link_config_v4 { - __le32 size; /* in bytes of this structure */ - __le32 id; /* unique ID - used to match */ - struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */ - __le32 num_streams; /* number of streams */ -} __packed; - #endif From 84ae7d9cfa5f6ec284efccedcb7baf7c075774d4 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 4 Apr 2024 10:38:45 +0300 Subject: [PATCH 198/603] ASoC: SOF: Disable pointless writes to debugfs file The permissions on this debugfs file are 0444 so it can't be written to. And writing to the file hasn't done anything since commit 6e9548cdb30e ("ASoC: SOF: Convert the generic IPC flood test into SOF client"). Delete the write function. Signed-off-by: Dan Carpenter Acked-by: Peter Ujfalusi Link: https://msgid.link/r/9d2477cf-25aa-4d12-818f-fdafc9aaa28a@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/sof/debug.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 7c8aafca8fdef..937e51b02a245 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -19,24 +19,6 @@ #include "sof-priv.h" #include "ops.h" -static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, - size_t count, loff_t *ppos) -{ - size_t size; - char *string; - int ret; - - string = kzalloc(count+1, GFP_KERNEL); - if (!string) - return -ENOMEM; - - size = simple_write_to_buffer(string, count, ppos, buffer, count); - ret = size; - - kfree(string); - return ret; -} - static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer, size_t count, loff_t *ppos) { @@ -126,7 +108,6 @@ static const struct file_operations sof_dfs_fops = { .open = simple_open, .read = sof_dfsentry_read, .llseek = default_llseek, - .write = sof_dfsentry_write, }; /* create FS entry for debug files that can expose DSP memories, registers */ From 2cfa0eea2675ce569dc80d2c0d2d644a452b77b3 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 4 Apr 2024 10:34:53 +0300 Subject: [PATCH 199/603] ASoC: SOF: Clean up sof_ipc_flood_dfs_write() This function doesn't support partial writes so using simple_write_to_buffer() doesn't really make sense. It's better to just use copy_from_user(). Signed-off-by: Dan Carpenter Acked-by: Peter Ujfalusi Link: https://msgid.link/r/a35dded2-392b-4ccb-9dbb-d782ac9b6547@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/sof/sof-client-ipc-flood-test.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c index c0d6723aed59c..1b2e9e25a8366 100644 --- a/sound/soc/sof/sof-client-ipc-flood-test.c +++ b/sound/soc/sof/sof-client-ipc-flood-test.c @@ -160,15 +160,20 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf unsigned long ipc_count = 0; struct dentry *dentry; int err; - size_t size; char *string; int ret; + if (*ppos != 0) + return -EINVAL; + string = kzalloc(count + 1, GFP_KERNEL); if (!string) return -ENOMEM; - size = simple_write_to_buffer(string, count, ppos, buffer, count); + if (copy_from_user(string, buffer, count)) { + ret = -EFAULT; + goto out; + } /* * write op is only supported for ipc_flood_count or @@ -198,7 +203,7 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf /* limit max duration/ipc count for flood test */ if (flood_duration_test) { if (!ipc_duration_ms) { - ret = size; + ret = count; goto out; } @@ -207,7 +212,7 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf ipc_duration_ms = MAX_IPC_FLOOD_DURATION_MS; } else { if (!ipc_count) { - ret = size; + ret = count; goto out; } @@ -231,9 +236,9 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf if (err < 0) dev_err_ratelimited(dev, "debugfs write failed to idle %d\n", err); - /* return size if test is successful */ + /* return count if test is successful */ if (ret >= 0) - ret = size; + ret = count; out: kfree(string); return ret; From e9ae5adce99e27ef29ea9ccb01afe260bf1c1408 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2024 13:54:45 -0500 Subject: [PATCH 200/603] ASoC: SOF: Intel: hda-loader: add flag to tag ICCMAX stream MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The existing code conflates capture stream and ICCMAX stream. This isn't going to be true any longer when we add support for SDW BPT RX streams. Add a boolean tag to flag ICCMAX streams. No functionality change, only future-proofing change. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240404185448.136157-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 10 +++++----- sound/soc/sof/intel/hda.h | 2 +- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index d5b9209beb5a6..dafb6b3ebb20c 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -46,7 +46,7 @@ static void hda_ssp_set_cbp_cfp(struct snd_sof_dev *sdev) struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, unsigned int size, struct snd_dma_buffer *dmab, - int direction) + int direction, bool is_iccmax) { struct hdac_ext_stream *hext_stream; struct hdac_stream *hstream; @@ -73,7 +73,7 @@ struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned hstream->format_val = format; hstream->bufsize = size; - if (direction == SNDRV_PCM_STREAM_CAPTURE) { + if (is_iccmax) { ret = hda_dsp_iccmax_stream_hw_params(sdev, hext_stream, dmab, NULL); if (ret < 0) { dev_err(sdev->dev, "error: iccmax stream prepare failed: %d\n", ret); @@ -335,7 +335,7 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) * the data, so use a buffer of PAGE_SIZE for receiving. */ iccmax_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, PAGE_SIZE, - &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE); + &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE, true); if (IS_ERR(iccmax_stream)) { dev_err(sdev->dev, "error: dma prepare for ICCMAX stream failed\n"); return PTR_ERR(iccmax_stream); @@ -421,7 +421,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) /* prepare DMA for code loader stream */ hext_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, stripped_firmware.size, - &dmab, SNDRV_PCM_STREAM_PLAYBACK); + &dmab, SNDRV_PCM_STREAM_PLAYBACK, false); if (IS_ERR(hext_stream)) { dev_err(sdev->dev, "error: dma prepare for fw loading failed\n"); return PTR_ERR(hext_stream); @@ -538,7 +538,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, /* prepare DMA for code loader stream */ hext_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, stripped_firmware.size, - &dmab, SNDRV_PCM_STREAM_PLAYBACK); + &dmab, SNDRV_PCM_STREAM_PLAYBACK, false); if (IS_ERR(hext_stream)) { dev_err(sdev->dev, "%s: DMA prepare failed\n", __func__); return PTR_ERR(hext_stream); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 16140ae22c905..30a185d4c7ab8 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -705,7 +705,7 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev); int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream); struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, unsigned int size, struct snd_dma_buffer *dmab, - int direction); + int direction, bool is_iccmax); int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct hdac_ext_stream *hext_stream); int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot); From 6cbf086143cf9674c7f029e1cf435c65a537066a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2024 13:54:46 -0500 Subject: [PATCH 201/603] ASoC: SOF: Intel: hda-loader: change signature of code loader helpers MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We need to reuse cl_prepare, cl_trigger and cl_cleanup helpers from a SoundWire context where only the device information is available. Rather than pass the 'sdev' argument, use get_drvdata() to retrieve the required structure. For consistency, rename hda_cl_stream_prepare() as hda_cl_prepare(). These three helpers are also exported so that they can be referenced from another module. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240404185448.136157-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 50 +++++++++++++++++--------------- sound/soc/sof/intel/hda.h | 11 ++++--- 2 files changed, 34 insertions(+), 27 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index dafb6b3ebb20c..5716772a7a910 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -44,13 +44,13 @@ static void hda_ssp_set_cbp_cfp(struct snd_sof_dev *sdev) } } -struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, - unsigned int size, struct snd_dma_buffer *dmab, - int direction, bool is_iccmax) +struct hdac_ext_stream *hda_cl_prepare(struct device *dev, unsigned int format, + unsigned int size, struct snd_dma_buffer *dmab, + int direction, bool is_iccmax) { + struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct hdac_ext_stream *hext_stream; struct hdac_stream *hstream; - struct pci_dev *pci = to_pci_dev(sdev->dev); int ret; hext_stream = hda_dsp_stream_get(sdev, direction, 0); @@ -63,7 +63,7 @@ struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned hstream->substream = NULL; /* allocate DMA buffer */ - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab); + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, dev, size, dmab); if (ret < 0) { dev_err(sdev->dev, "error: memory alloc failed: %d\n", ret); goto out_put; @@ -96,6 +96,7 @@ struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned hda_dsp_stream_put(sdev, direction, hstream->stream_tag); return ERR_PTR(ret); } +EXPORT_SYMBOL_NS(hda_cl_prepare, SND_SOC_SOF_INTEL_HDA_COMMON); /* * first boot sequence has some extra steps. @@ -220,9 +221,9 @@ int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) return ret; } -static int cl_trigger(struct snd_sof_dev *sdev, - struct hdac_ext_stream *hext_stream, int cmd) +int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int cmd) { + struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct hdac_stream *hstream = &hext_stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); @@ -246,10 +247,12 @@ static int cl_trigger(struct snd_sof_dev *sdev, return hda_dsp_stream_trigger(sdev, hext_stream, cmd); } } +EXPORT_SYMBOL_NS(hda_cl_trigger, SND_SOC_SOF_INTEL_HDA_COMMON); -int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, +int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab, struct hdac_ext_stream *hext_stream) { + struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct hdac_stream *hstream = &hext_stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); int ret = 0; @@ -278,6 +281,7 @@ int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, return ret; } +EXPORT_SYMBOL_NS(hda_cl_cleanup, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream) { @@ -286,7 +290,7 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream unsigned int reg; int ret, status; - ret = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_START); + ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_START); if (ret < 0) { dev_err(sdev->dev, "error: DMA trigger start failed\n"); return ret; @@ -309,7 +313,7 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream __func__, chip->rom_status_reg); } - ret = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_STOP); + ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_STOP); if (ret < 0) { dev_err(sdev->dev, "error: DMA trigger stop failed\n"); if (!status) @@ -334,8 +338,8 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) * Prepare capture stream for ICCMAX. We do not need to store * the data, so use a buffer of PAGE_SIZE for receiving. */ - iccmax_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, PAGE_SIZE, - &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE, true); + iccmax_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT, PAGE_SIZE, + &dmab_bdl, SNDRV_PCM_STREAM_CAPTURE, true); if (IS_ERR(iccmax_stream)) { dev_err(sdev->dev, "error: dma prepare for ICCMAX stream failed\n"); return PTR_ERR(iccmax_stream); @@ -347,7 +351,7 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) * Perform iccmax stream cleanup. This should be done even if firmware loading fails. * If the cleanup also fails, we return the initial error */ - ret1 = hda_cl_cleanup(sdev, &dmab_bdl, iccmax_stream); + ret1 = hda_cl_cleanup(sdev->dev, &dmab_bdl, iccmax_stream); if (ret1 < 0) { dev_err(sdev->dev, "error: ICCMAX stream cleanup failed\n"); @@ -419,9 +423,9 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) init_waitqueue_head(&sdev->boot_wait); /* prepare DMA for code loader stream */ - hext_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, - stripped_firmware.size, - &dmab, SNDRV_PCM_STREAM_PLAYBACK, false); + hext_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT, + stripped_firmware.size, + &dmab, SNDRV_PCM_STREAM_PLAYBACK, false); if (IS_ERR(hext_stream)) { dev_err(sdev->dev, "error: dma prepare for fw loading failed\n"); return PTR_ERR(hext_stream); @@ -494,7 +498,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) * This should be done even if firmware loading fails. * If the cleanup also fails, we return the initial error */ - ret1 = hda_cl_cleanup(sdev, &dmab, hext_stream); + ret1 = hda_cl_cleanup(sdev->dev, &dmab, hext_stream); if (ret1 < 0) { dev_err(sdev->dev, "error: Code loader DSP cleanup failed\n"); @@ -536,9 +540,9 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, stripped_firmware.size = fw_lib->sof_fw.fw->size - fw_lib->sof_fw.payload_offset; /* prepare DMA for code loader stream */ - hext_stream = hda_cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, - stripped_firmware.size, - &dmab, SNDRV_PCM_STREAM_PLAYBACK, false); + hext_stream = hda_cl_prepare(sdev->dev, HDA_CL_STREAM_FORMAT, + stripped_firmware.size, + &dmab, SNDRV_PCM_STREAM_PLAYBACK, false); if (IS_ERR(hext_stream)) { dev_err(sdev->dev, "%s: DMA prepare failed\n", __func__); return PTR_ERR(hext_stream); @@ -581,7 +585,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, goto cleanup; } - ret = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_START); + ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_START); if (ret < 0) { dev_err(sdev->dev, "%s: DMA trigger start failed\n", __func__); goto cleanup; @@ -598,7 +602,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, ret = sof_ipc_tx_message_no_reply(sdev->ipc, &msg, 0); /* Stop the DMA channel */ - ret1 = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_STOP); + ret1 = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_STOP); if (ret1 < 0) { dev_err(sdev->dev, "%s: DMA trigger stop failed\n", __func__); if (!ret) @@ -607,7 +611,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, cleanup: /* clean up even in case of error and return the first error */ - ret1 = hda_cl_cleanup(sdev, &dmab, hext_stream); + ret1 = hda_cl_cleanup(sdev->dev, &dmab, hext_stream); if (ret1 < 0) { dev_err(sdev->dev, "%s: Code loader DSP cleanup failed\n", __func__); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 30a185d4c7ab8..b4b037758fcbb 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -703,10 +703,13 @@ void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level); int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev); int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev); int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream); -struct hdac_ext_stream *hda_cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, - unsigned int size, struct snd_dma_buffer *dmab, - int direction, bool is_iccmax); -int hda_cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, + +struct hdac_ext_stream *hda_cl_prepare(struct device *dev, unsigned int format, + unsigned int size, struct snd_dma_buffer *dmab, + int direction, bool is_iccmax); +int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int cmd); + +int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab, struct hdac_ext_stream *hext_stream); int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot); #define HDA_CL_STREAM_FORMAT 0x40 From d5263dbbd8af026159b16a08a94bedfe51b5f67b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2024 13:54:47 -0500 Subject: [PATCH 202/603] ASoC: SOF: Intel: don't ignore IOC interrupts for non-audio transfers MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The HDaudio stream interrupts are ignored unless the stream is PCM or compressed audio. For alternate non-audio usages, such as code loader or SoundWire BPT case, the IOC interrupt on the last buffer transferred is silently ignored. This patch adds a 'struct completion' for each HDaudio stream. This capability helps detect if the non-audio data transfers completed. There is no performance impact for audio streams. In the code loader case, the code currently starts the DMA and directly checks if the firmware status changes, without checking if the DMA succeeded. With a first pass waiting for the DMA to complete, system validation engineers can gather more precise timing information on firmware boot time or root-cause boot failures more accurately. A timeout of 500ms was selected for the code loader DMA. This is an experimental value which should be more than enough - higher values would certainly be problematic from a usage/latency perspective. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240404185448.136157-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 28 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda-stream.c | 22 +++++++++++++++++++--- sound/soc/sof/intel/hda.h | 2 ++ 3 files changed, 49 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 5716772a7a910..dc88b7ea599ea 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -226,10 +226,15 @@ int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct hdac_stream *hstream = &hext_stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); + struct sof_intel_hda_stream *hda_stream; /* code loader is special case that reuses stream ops */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: + hda_stream = container_of(hext_stream, struct sof_intel_hda_stream, + hext_stream); + reinit_completion(&hda_stream->ioc); + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, 1 << hstream->index, 1 << hstream->index); @@ -283,19 +288,38 @@ int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab, } EXPORT_SYMBOL_NS(hda_cl_cleanup, SND_SOC_SOF_INTEL_HDA_COMMON); +#define HDA_CL_DMA_IOC_TIMEOUT_MS 500 + int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; + struct sof_intel_hda_stream *hda_stream; + unsigned long time_left; unsigned int reg; int ret, status; + hda_stream = container_of(hext_stream, struct sof_intel_hda_stream, + hext_stream); + + dev_dbg(sdev->dev, "Code loader DMA starting\n"); + ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_START); if (ret < 0) { dev_err(sdev->dev, "error: DMA trigger start failed\n"); return ret; } + /* Wait for completion of transfer */ + time_left = wait_for_completion_timeout(&hda_stream->ioc, + msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS)); + + if (!time_left) { + dev_err(sdev->dev, "Code loader DMA did not complete\n"); + return -ETIMEDOUT; + } + dev_dbg(sdev->dev, "Code loader DMA done, waiting for FW_ENTERED status\n"); + status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, chip->rom_status_reg, reg, (FSR_TO_STATE_CODE(reg) == FSR_STATE_FW_ENTERED), @@ -311,6 +335,8 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream dev_err(sdev->dev, "%s: timeout with rom_status_reg (%#x) read\n", __func__, chip->rom_status_reg); + } else { + dev_dbg(sdev->dev, "Code loader FW_ENTERED status\n"); } ret = hda_cl_trigger(sdev->dev, hext_stream, SNDRV_PCM_TRIGGER_STOP); @@ -318,6 +344,8 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream dev_err(sdev->dev, "error: DMA trigger stop failed\n"); if (!status) status = ret; + } else { + dev_dbg(sdev->dev, "Code loader DMA stopped\n"); } return status; diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 0c189d3b19c1a..76c33795ade45 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -765,10 +765,25 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) writeb(sd_status, s->sd_addr + SOF_HDA_ADSP_REG_SD_STS); active = true; - if ((!s->substream && !s->cstream) || - !s->running || - (sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0) + if (!s->running) continue; + if ((sd_status & SOF_HDA_CL_DMA_SD_INT_COMPLETE) == 0) + continue; + if (!s->substream && !s->cstream) { + /* + * when no substream is found, the DMA may used for code loading + * or data transfers which can rely on wait_for_completion() + */ + struct sof_intel_hda_stream *hda_stream; + struct hdac_ext_stream *hext_stream; + + hext_stream = stream_to_hdac_ext_stream(s); + hda_stream = container_of(hext_stream, struct sof_intel_hda_stream, + hext_stream); + + complete(&hda_stream->ioc); + continue; + } /* Inform ALSA only in case not do that with IPC */ if (s->substream && sof_hda->no_ipc_position) { @@ -880,6 +895,7 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) return -ENOMEM; hda_stream->sdev = sdev; + init_completion(&hda_stream->ioc); hext_stream = &hda_stream->hext_stream; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index b4b037758fcbb..b59d1a572bce3 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -11,6 +11,7 @@ #ifndef __SOF_INTEL_HDA_H #define __SOF_INTEL_HDA_H +#include #include #include #include @@ -559,6 +560,7 @@ struct sof_intel_hda_stream { struct sof_intel_stream sof_intel_stream; int host_reserved; /* reserve host DMA channel */ u32 flags; + struct completion ioc; }; #define hstream_to_sof_hda_stream(hstream) \ From 716778b3d71fe1c799a4f5cf84faf4979f80b267 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2024 13:54:48 -0500 Subject: [PATCH 203/603] ASoC: SOF: Intel: hda-stream: clarify comment MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The Yoda grammar and multiple negatives are unclear. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Rander Wang Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240404185448.136157-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 76c33795ade45..91b1371745e66 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -785,7 +785,7 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) continue; } - /* Inform ALSA only in case not do that with IPC */ + /* Inform ALSA only if the IPC position is not used */ if (s->substream && sof_hda->no_ipc_position) { snd_sof_pcm_period_elapsed(s->substream); } else if (s->cstream) { From bd5863f9ef12e26c2cfdce9c0adbf0222c731a3c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2024 14:03:54 -0500 Subject: [PATCH 204/603] ASoC: SOF: Intel: hda-ctrl: add missing WAKE_STS clear MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason, the programming sequences in the SOF driver do not include a clear of the WAKE_STS bits before resetting the controller. This clear is not formally required by the HDaudio specification, but was added to harden the snd-hda-reset back in 2007. Adding this sequence back avoids an issue reported by the Intel CI. Closes: https://github.com/thesofproject/linux/issues/4889 Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240404190357.138073-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ctrl.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 84bf01bd360a3..b4f0756e21f6c 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -184,6 +184,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev) struct hdac_bus *bus = sof_to_bus(sdev); struct hdac_stream *stream; int sd_offset, ret = 0; + u32 gctl; if (bus->chip_init) return 0; @@ -192,6 +193,12 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev) hda_dsp_ctrl_misc_clock_gating(sdev, false); + /* clear WAKE_STS if not in reset */ + gctl = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_GCTL); + if (gctl & SOF_HDA_GCTL_RESET) + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_WAKESTS, SOF_HDA_WAKESTS_INT_MASK); + /* reset HDA controller */ ret = hda_dsp_ctrl_link_reset(sdev, true); if (ret < 0) { From ab9182441ee5a94dd6f47743ed1b7b6b07b63cb2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2024 14:03:55 -0500 Subject: [PATCH 205/603] ASoC: SOF: Intel: lnl: add helper to detect SoundWire wakes The global STATESTS register will provide information on all links. Rather than iterate on all possible links, the helpers only filters the range of possible bits for a quick lookup. The process_wakeen() helper will walk through all the links and deal with wakes. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Link: https://msgid.link/r/20240404190357.138073-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 8e71933443410..8aa5430e95767 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -189,6 +189,23 @@ static int lnl_dsp_disable_interrupts(struct snd_sof_dev *sdev) return mtl_enable_interrupts(sdev, false); } +static bool lnl_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) +{ + struct hdac_bus *bus = sof_to_bus(sdev); + u16 wake_sts; + + /* + * we need to use the global HDaudio WAKEEN/STS to be able to + * detect wakes in low-power modes. The link-specific information + * is handled in the process_wakeen() helper, this helper only + * detects a SoundWire wake without identifying the link. + */ + wake_sts = snd_hdac_chip_readw(bus, STATESTS); + + /* filter out the range of SDIs that can be set for SoundWire */ + return wake_sts & GENMASK(SDW_MAX_DEVICES, SDW_INTEL_DEV_NUM_IDA_MIN); +} + const struct sof_intel_dsp_desc lnl_chip_info = { .cores_num = 5, .init_core_mask = BIT(0), @@ -205,6 +222,7 @@ const struct sof_intel_dsp_desc lnl_chip_info = { .read_sdw_lcount = hda_sdw_check_lcount_ext, .enable_sdw_irq = lnl_enable_sdw_irq, .check_sdw_irq = lnl_dsp_check_sdw_irq, + .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, From b69480edf4eb71c6f754e3c4020e8dd72a330558 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2024 14:03:56 -0500 Subject: [PATCH 206/603] ASoC: SOF: Intel: hda-codec: preserve WAKEEN values Since LunarLake, we use the HDadio WAKEEN/WAKESTS to detect wakes for SoundWire codecs. Unfortunately, the existing code in hda_codec_jack_wake_enable() unconditionally resets the WAKEEN bits. This patch changes the initialization to preserve SoundWire WAKEEN bits. For HDAudio codecs the same strategy is used, WAKEEN is only set when the jacktbl.used property is set. Closes: https://github.com/thesofproject/linux/issues/4687 Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Reviewed-by: Keqiao Zhang Reviewed-by: Bard Liao Link: https://msgid.link/r/20240404190357.138073-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 9f84b0d287a5f..6a13f38a8d210 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -79,18 +79,27 @@ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev, bool enable) struct hdac_bus *bus = sof_to_bus(sdev); struct hda_codec *codec; unsigned int mask = 0; + unsigned int val = 0; if (IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC_DEBUG_SUPPORT) && sof_debug_check_flag(SOF_DBG_FORCE_NOCODEC)) return; if (enable) { - list_for_each_codec(codec, hbus) + list_for_each_codec(codec, hbus) { + /* only set WAKEEN when needed for HDaudio codecs */ + mask |= BIT(codec->core.addr); if (codec->jacktbl.used) - mask |= BIT(codec->core.addr); + val |= BIT(codec->core.addr); + } + } else { + list_for_each_codec(codec, hbus) { + /* reset WAKEEN only HDaudio codecs */ + mask |= BIT(codec->core.addr); + } } - snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, mask); + snd_hdac_chip_updatew(bus, WAKEEN, mask & STATESTS_INT_MASK, val); } EXPORT_SYMBOL_NS_GPL(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC); From 35b5806e2edee1741f6bd2de2a5c149a876c4a60 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 4 Apr 2024 14:03:57 -0500 Subject: [PATCH 207/603] ASoC: SOF: Intel: hda-ctrl: only clear WAKESTS for HDaudio codecs When a PME wake happens due to a SoundWire wake, we currently clear all WAKESTS bits during the resume operation initiated by the PCI subsystem. As a result, we are unable to identify which SoundWire links need to be resumed and don't properly handle jack detection. This patch only clears the WAKESTS bits for the HDaudio codecs detected earlier. Note that we still clear all WAKESTS bits unconditionally in hda_dsp_ctrl_stop_chip(). The existing behavior is potentially racy if e.g. a jack event happens during a suspend routine, but there's a risk of breaking shutdown or reboot sequences so the code is left as is for now. Closes: https://github.com/thesofproject/linux/issues/4687 Co-developed-by: Bard Liao Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Reviewed-by: Keqiao Zhang Reviewed-by: Bard Liao Link: https://msgid.link/r/20240404190357.138073-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index b4f0756e21f6c..6d941209847f5 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -228,7 +228,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev) /* clear WAKESTS */ snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, SOF_HDA_WAKESTS, - SOF_HDA_WAKESTS_INT_MASK); + bus->codec_mask); hda_codec_rirb_status_clear(sdev); From 680507581e025d16a0b6d3782603ca8c598fbe2b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 5 Apr 2024 11:09:17 +0200 Subject: [PATCH 208/603] ASoC: Intel: avs: Restore stream decoupling on prepare MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Revert changes from commit b87b8f43afd5 ("ASoC: Intel: avs: Drop superfluous stream decoupling") to restore working streaming during S3. Fixes: b87b8f43afd5 ("ASoC: Intel: avs: Drop superfluous stream decoupling") Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 2cafbc392cdbe..72f1bc3b7b1fe 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -356,6 +356,7 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn stream_info->sig_bits); format_val = snd_hdac_stream_format(runtime->channels, bits, runtime->rate); + snd_hdac_ext_stream_decouple(bus, link_stream, true); snd_hdac_ext_stream_reset(link_stream); snd_hdac_ext_stream_setup(link_stream, format_val); @@ -611,6 +612,7 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so struct avs_dev *adev = to_avs_dev(dai->dev); struct hdac_ext_stream *host_stream; unsigned int format_val; + struct hdac_bus *bus; unsigned int bits; int ret; @@ -620,6 +622,8 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so if (hdac_stream(host_stream)->prepared) return 0; + bus = hdac_stream(host_stream)->bus; + snd_hdac_ext_stream_decouple(bus, data->host_stream, true); snd_hdac_stream_reset(hdac_stream(host_stream)); stream_info = snd_soc_dai_get_pcm_stream(dai, substream->stream); From c91b692781c1839fcc389b2a9120e46593c6424b Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:18 +0200 Subject: [PATCH 209/603] ASoC: Intel: avs: Fix debug-slot offset calculation For resources with ID other than 0 the current calculus is incorrect. Fixes: 275b583d047a ("ASoC: Intel: avs: ICL-based platforms support") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/icl.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index 9d9921e1cd4de..3e0716160f5a6 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -66,7 +66,7 @@ struct avs_icl_memwnd2 { struct avs_icl_memwnd2_desc slot_desc[AVS_ICL_MEMWND2_SLOTS_COUNT]; u8 rsvd[PAGE_SIZE]; }; - u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][PAGE_SIZE]; + u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][SZ_4K]; } __packed; #define AVS_ICL_SLOT_UNUSED \ @@ -89,8 +89,7 @@ static int avs_icl_slot_offset(struct avs_dev *adev, union avs_icl_memwnd2_slot_ for (i = 0; i < AVS_ICL_MEMWND2_SLOTS_COUNT; i++) if (desc[i].slot_id.val == slot_type.val) - return offsetof(struct avs_icl_memwnd2, slot_array) + - avs_skl_log_buffer_offset(adev, i); + return offsetof(struct avs_icl_memwnd2, slot_array) + i * SZ_4K; return -ENXIO; } From 6dd68c2da44d92c25b323bcc2603421463437a64 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:19 +0200 Subject: [PATCH 210/603] ASoC: Intel: avs: Silence false-positive memcpy() warnings Commit df8fc4e934c1 ("kbuild: Enable -fstrict-flex-arrays=3") enforced strict flex array declarations. This generates false-positive in form of: "memcpy: detected field-spanning write". Avoid it by utilizing the DECLARE_FLEX_ARRAY() macro. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/messages.h | 4 ++-- sound/soc/intel/avs/path.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h index 4e609a08863c8..007bc4fb6d99d 100644 --- a/sound/soc/intel/avs/messages.h +++ b/sound/soc/intel/avs/messages.h @@ -752,9 +752,9 @@ struct avs_copier_gtw_cfg { union avs_connector_node_id node_id; u32 dma_buffer_size; u32 config_length; - struct { + union { union avs_gtw_attributes attrs; - u32 blob[]; + DECLARE_FLEX_ARRAY(u32, blob); } config; } __packed; diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index e785fc2a7008f..5944865a1193d 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -254,7 +254,7 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod) /* config_length in DWORDs */ cfg->gtw_cfg.config_length = DIV_ROUND_UP(data_size, 4); if (data) - memcpy(&cfg->gtw_cfg.config, data, data_size); + memcpy(&cfg->gtw_cfg.config.blob, data, data_size); mod->gtw_attrs = cfg->gtw_cfg.config.attrs; From beeeee9686affef32ee04d8ae30db8c53cbc7aee Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:20 +0200 Subject: [PATCH 211/603] ASoC: Intel: avs: Fix config_length for config-less copiers Copier's config_length shall always be at least one even if there is no configuration payload to carry. While the firmware treats config_length=0 or 1 in the same manner, the driver shall initialize the module properly. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/path.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 5944865a1193d..9bd03a44fcb97 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -148,11 +148,12 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod) struct avs_copier_cfg *cfg; struct nhlt_specific_cfg *ep_blob; union avs_connector_node_id node_id = {0}; - size_t cfg_size, data_size = 0; + size_t cfg_size, data_size; void *data = NULL; u32 dma_type; int ret; + data_size = sizeof(cfg->gtw_cfg.config); dma_type = t->cfg_ext->copier.dma_type; node_id.dma_type = dma_type; @@ -233,10 +234,7 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod) break; } - cfg_size = sizeof(*cfg) + data_size; - /* Every config-BLOB contains gateway attributes. */ - if (data_size) - cfg_size -= sizeof(cfg->gtw_cfg.config.attrs); + cfg_size = offsetof(struct avs_copier_cfg, gtw_cfg.config) + data_size; if (cfg_size > AVS_MAILBOX_SIZE) return -EINVAL; From 9d2e26f31c7cc3fa495c423af9b4902ec0dc7be3 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:21 +0200 Subject: [PATCH 212/603] ASoC: Intel: avs: Fix ASRC module initialization The ASRC module configuration consists of several reserved fields. Zero them out when initializing the module to avoid sending invalid data. Fixes: 274d79e51875 ("ASoC: Intel: avs: Configure modules according to their type") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/path.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 9bd03a44fcb97..8dfd90587427f 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -365,6 +365,7 @@ static int avs_asrc_create(struct avs_dev *adev, struct avs_path_module *mod) struct avs_tplg_module *t = mod->template; struct avs_asrc_cfg cfg; + memset(&cfg, 0, sizeof(cfg)); cfg.base.cpc = t->cfg_base->cpc; cfg.base.ibs = t->cfg_base->ibs; cfg.base.obs = t->cfg_base->obs; From 47714847592b574ff594ceca7ebe0ada70dbac3c Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:22 +0200 Subject: [PATCH 213/603] ASoC: Intel: avs: Replace risky functions with safer variants strscpy() and snprintf() are the recommended equivalents of their riskier friends. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/loader.c | 4 ++-- sound/soc/intel/avs/pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c index 8e34d3536082a..57370f0905dc3 100644 --- a/sound/soc/intel/avs/loader.c +++ b/sound/soc/intel/avs/loader.c @@ -535,7 +535,7 @@ int avs_dsp_load_libraries(struct avs_dev *adev, struct avs_tplg_library *libs, if (ret) return ret; - strncpy(adev->lib_names[id], man->name, AVS_LIB_NAME_SIZE); + strscpy(adev->lib_names[id], man->name, AVS_LIB_NAME_SIZE); id++; next_lib: i++; @@ -698,7 +698,7 @@ int avs_dsp_first_boot_firmware(struct avs_dev *adev) } /* basefw always occupies slot 0 */ - strcpy(&adev->lib_names[0][0], "BASEFW"); + strscpy(adev->lib_names[0], "BASEFW", AVS_LIB_NAME_SIZE); ida_init(&adev->ppl_ida); diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 72f1bc3b7b1fe..405de1d581786 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -1420,7 +1420,7 @@ static void avs_component_hda_unregister_dais(struct snd_soc_component *componen mach = dev_get_platdata(component->card->dev); codec = mach->pdata; - sprintf(name, "%s-cpu", dev_name(&codec->core.dev)); + snprintf(name, sizeof(name), "%s-cpu", dev_name(&codec->core.dev)); for_each_component_dais_safe(component, dai, save) { int stream; From c7e832cabe635df47c2bf6df7801e97bf3045b1e Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:23 +0200 Subject: [PATCH 214/603] ASoC: Intel: avs: Fix potential integer overflow While stream_tag for CLDMA on SKL-based platforms is always 1, function hda_cldma_setup() uses AZX_SD_CTL_STRM() macro which does: stream_tag << 20 what combined with stream_tag type of 'unsigned int' generates a potential overflow issue. Update the field type to fix that. Fixes: 45864e49a05a ("ASoC: Intel: avs: Implement CLDMA transfer") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/cldma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c index d7a9390b5e483..585579840b646 100644 --- a/sound/soc/intel/avs/cldma.c +++ b/sound/soc/intel/avs/cldma.c @@ -35,7 +35,7 @@ struct hda_cldma { unsigned int buffer_size; unsigned int num_periods; - unsigned int stream_tag; + unsigned char stream_tag; void __iomem *sd_addr; struct snd_dma_buffer dmab_data; From 41bf4525fadb3d8df3860420d6ac9025c51a3bac Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:24 +0200 Subject: [PATCH 215/603] ASoC: Intel: avs: Test result of avs_get_module_entry() While PROBE_MOD_UUID is always part of the base AudioDSP firmware manifest, from maintenance point of view it is better to check the result. Fixes: dab8d000e25c ("ASoC: Intel: avs: Add data probing requests") Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/probes.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c index 817e543036f29..7e781a3156909 100644 --- a/sound/soc/intel/avs/probes.c +++ b/sound/soc/intel/avs/probes.c @@ -19,8 +19,11 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id struct avs_probe_cfg cfg = {{0}}; struct avs_module_entry mentry; u8 dummy; + int ret; - avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); + ret = avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); + if (ret) + return ret; /* * Probe module uses no cycles, audio data format and input and output @@ -39,11 +42,12 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id static void avs_dsp_delete_probe(struct avs_dev *adev) { struct avs_module_entry mentry; + int ret; - avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); - - /* There is only ever one probe module instance. */ - avs_dsp_delete_module(adev, mentry.module_id, 0, INVALID_PIPELINE_ID, 0); + ret = avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); + if (!ret) + /* There is only ever one probe module instance. */ + avs_dsp_delete_module(adev, mentry.module_id, 0, INVALID_PIPELINE_ID, 0); } static inline struct hdac_ext_stream *avs_compr_get_host_stream(struct snd_compr_stream *cstream) From d58275f474b4a27b4e97839ffe8d9fe55c0cc40a Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:25 +0200 Subject: [PATCH 216/603] ASoC: Intel: avs: Remove dead code The result of list_next_entry()/list_last_entry() is never null. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/path.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 8dfd90587427f..fa3fec339548c 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -709,8 +709,6 @@ static int avs_path_pipeline_arm(struct avs_dev *adev, /* bind current module to next module on list */ source = mod; sink = list_next_entry(mod, node); - if (!source || !sink) - return -EINVAL; ret = avs_ipc_bind(adev, source->module_id, source->instance_id, sink->module_id, sink->instance_id, 0, 0); From 30df76bbcb59254ce646477e3e05f00021a10117 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:26 +0200 Subject: [PATCH 217/603] ASoC: Intel: avs: Wake from D0ix when starting streaming It is recommended to keep the DSP domain in full-power when starting DMA engines. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/icl.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index 3e0716160f5a6..d279ec1e0ad0e 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -109,6 +109,10 @@ int avs_icl_log_buffer_offset(struct avs_dev *adev, u32 core) bool avs_icl_d0ix_toggle(struct avs_dev *adev, struct avs_ipc_msg *tx, bool wake) { + /* Full-power when starting DMA engines. */ + if (tx->glb.set_ppl_state.state == AVS_PPL_STATE_RUNNING) + return true; + /* Payload-less IPCs do not take part in d0ix toggling. */ return tx->size; } From ff0aefe2d217ce6fec6487b225737b2019eb88c0 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:27 +0200 Subject: [PATCH 218/603] ASoC: Intel: avs: Init debugfs before booting firmware When bringing up setups it's vital to have access to debug functionality even if firmware boot fails. As order of probe()ing operations is changed, update remove() procedure accordingly. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-12-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index d7f8940099cec..76782a0f32bc3 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -209,6 +209,7 @@ static void avs_hda_probe_work(struct work_struct *work) snd_hdac_ext_bus_ppcap_enable(bus, true); snd_hdac_ext_bus_ppcap_int_enable(bus, true); + avs_debugfs_init(adev); ret = avs_dsp_first_boot_firmware(adev); if (ret < 0) @@ -217,7 +218,6 @@ static void avs_hda_probe_work(struct work_struct *work) adev->nhlt = intel_nhlt_init(adev->dev); if (!adev->nhlt) dev_info(bus->dev, "platform has no NHLT\n"); - avs_debugfs_init(adev); avs_register_all_boards(adev); @@ -548,9 +548,9 @@ static void avs_pci_remove(struct pci_dev *pci) avs_unregister_all_boards(adev); - avs_debugfs_exit(adev); if (adev->nhlt) intel_nhlt_free(adev->nhlt); + avs_debugfs_exit(adev); if (avs_platattr_test(adev, CLDMA)) hda_cldma_free(&code_loader); From c2b10acb62c195db2c976d614d9d8092ad6339ae Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 5 Apr 2024 11:09:28 +0200 Subject: [PATCH 219/603] ASoC: Intel: avs: Add assert_static to guarantee ABI sizes MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In order to make sure that IPC interface is stable use assert_static to check union and struct sizes that describe communication interface. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-13-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/avs.h | 1 + sound/soc/intel/avs/icl.c | 3 +++ sound/soc/intel/avs/loader.c | 2 ++ sound/soc/intel/avs/messages.h | 43 ++++++++++++++++++++++++++++++++++ 4 files changed, 49 insertions(+) diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index f80f79415344a..39bdec591d582 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -381,6 +381,7 @@ struct avs_apl_log_buffer_layout { u32 write_ptr; u8 buffer[]; } __packed; +static_assert(sizeof(struct avs_apl_log_buffer_layout) == 8); #define avs_apl_log_payload_size(adev) \ (avs_log_buffer_size(adev) - sizeof(struct avs_apl_log_buffer_layout)) diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index d279ec1e0ad0e..e8b4983e03e9f 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -52,12 +52,14 @@ union avs_icl_memwnd2_slot_type { u32 type:24; }; } __packed; +static_assert(sizeof(union avs_icl_memwnd2_slot_type) == 4); struct avs_icl_memwnd2_desc { u32 resource_id; union avs_icl_memwnd2_slot_type slot_id; u32 vma; } __packed; +static_assert(sizeof(struct avs_icl_memwnd2_desc) == 12); #define AVS_ICL_MEMWND2_SLOTS_COUNT 15 @@ -68,6 +70,7 @@ struct avs_icl_memwnd2 { }; u8 slot_array[AVS_ICL_MEMWND2_SLOTS_COUNT][SZ_4K]; } __packed; +static_assert(sizeof(struct avs_icl_memwnd2) == 65536); #define AVS_ICL_SLOT_UNUSED \ ((union avs_icl_memwnd2_slot_type) { 0x00000000U }) diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c index 57370f0905dc3..c255c898b7a86 100644 --- a/sound/soc/intel/avs/loader.c +++ b/sound/soc/intel/avs/loader.c @@ -56,6 +56,7 @@ struct avs_fw_manifest { u32 feature_mask; struct avs_fw_version version; } __packed; +static_assert(sizeof(struct avs_fw_manifest) == 36); struct avs_fw_ext_manifest { u32 id; @@ -64,6 +65,7 @@ struct avs_fw_ext_manifest { u16 version_minor; u32 entries; } __packed; +static_assert(sizeof(struct avs_fw_ext_manifest) == 16); static int avs_fw_ext_manifest_strip(struct firmware *fw) { diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h index 007bc4fb6d99d..285d89607b6a8 100644 --- a/sound/soc/intel/avs/messages.h +++ b/sound/soc/intel/avs/messages.h @@ -93,12 +93,14 @@ union avs_global_msg { } ext; }; } __packed; +static_assert(sizeof(union avs_global_msg) == 8); struct avs_tlv { u32 type; u32 length; u32 value[]; } __packed; +static_assert(sizeof(struct avs_tlv) == 8); enum avs_module_msg_type { AVS_MOD_INIT_INSTANCE = 0, @@ -155,6 +157,7 @@ union avs_module_msg { } ext; }; } __packed; +static_assert(sizeof(union avs_module_msg) == 8); #define AVS_IPC_NOT_SUPPORTED 15 @@ -190,6 +193,7 @@ union avs_reply_msg { } ext; }; } __packed; +static_assert(sizeof(union avs_reply_msg) == 8); enum avs_notify_msg_type { AVS_NOTIFY_PHRASE_DETECTED = 4, @@ -226,6 +230,7 @@ union avs_notify_msg { } ext; }; } __packed; +static_assert(sizeof(union avs_notify_msg) == 8); #define AVS_MSG(hdr) { .val = hdr } @@ -264,6 +269,7 @@ struct avs_notify_voice_data { u16 kpd_score; u16 reserved; } __packed; +static_assert(sizeof(struct avs_notify_voice_data) == 4); struct avs_notify_res_data { u32 resource_type; @@ -272,6 +278,7 @@ struct avs_notify_res_data { u32 reserved; u32 data[6]; } __packed; +static_assert(sizeof(struct avs_notify_res_data) == 40); struct avs_notify_mod_data { u32 module_instance_id; @@ -279,6 +286,7 @@ struct avs_notify_mod_data { u32 data_size; u32 data[]; } __packed; +static_assert(sizeof(struct avs_notify_mod_data) == 12); /* ROM messages */ enum avs_rom_control_msg_type { @@ -332,6 +340,7 @@ struct avs_dxstate_info { u32 core_mask; /* which cores are subject for power transition */ u32 dx_mask; /* bit[n]=1 core n goes to D0, bit[n]=0 it goes to D3 */ } __packed; +static_assert(sizeof(struct avs_dxstate_info) == 8); int avs_ipc_set_dx(struct avs_dev *adev, u32 core_mask, bool powerup); int avs_ipc_set_d0ix(struct avs_dev *adev, bool enable_pg, bool streaming); @@ -367,11 +376,13 @@ struct avs_skl_log_state { u32 enable; u32 min_priority; } __packed; +static_assert(sizeof(struct avs_skl_log_state) == 8); struct avs_skl_log_state_info { u32 core_mask; struct avs_skl_log_state logs_core[]; } __packed; +static_assert(sizeof(struct avs_skl_log_state_info) == 4); struct avs_apl_log_state_info { u32 aging_timer_period; @@ -379,6 +390,7 @@ struct avs_apl_log_state_info { u32 core_mask; struct avs_skl_log_state logs_core[]; } __packed; +static_assert(sizeof(struct avs_apl_log_state_info) == 12); enum avs_icl_log_priority { AVS_ICL_LOG_CRITICAL = 0, @@ -403,6 +415,7 @@ struct avs_icl_log_state_info { u32 enable; u32 logs_priorities_mask[]; } __packed; +static_assert(sizeof(struct avs_icl_log_state_info) == 12); int avs_ipc_set_enable_logs(struct avs_dev *adev, u8 *log_info, size_t size); @@ -521,6 +534,7 @@ struct avs_module_type { u32 lib_code:1; u32 rsvd:24; } __packed; +static_assert(sizeof(struct avs_module_type) == 4); union avs_segment_flags { u32 ul; @@ -537,12 +551,14 @@ union avs_segment_flags { u32 length:16; }; } __packed; +static_assert(sizeof(union avs_segment_flags) == 4); struct avs_segment_desc { union avs_segment_flags flags; u32 v_base_addr; u32 file_offset; } __packed; +static_assert(sizeof(struct avs_segment_desc) == 12); struct avs_module_entry { u16 module_id; @@ -559,11 +575,13 @@ struct avs_module_entry { u16 instance_bss_size; struct avs_segment_desc segments[3]; } __packed; +static_assert(sizeof(struct avs_module_entry) == 116); struct avs_mods_info { u32 count; struct avs_module_entry entries[]; } __packed; +static_assert(sizeof(struct avs_mods_info) == 4); static inline bool avs_module_entry_is_loaded(struct avs_module_entry *mentry) { @@ -577,6 +595,7 @@ struct avs_sys_time { u32 val_l; u32 val_u; } __packed; +static_assert(sizeof(struct avs_sys_time) == 8); int avs_ipc_set_system_time(struct avs_dev *adev); @@ -680,6 +699,7 @@ struct avs_audio_format { u32 sample_type:8; u32 reserved:8; } __packed; +static_assert(sizeof(struct avs_audio_format) == 24); struct avs_modcfg_base { u32 cpc; @@ -688,12 +708,14 @@ struct avs_modcfg_base { u32 is_pages; struct avs_audio_format audio_fmt; } __packed; +static_assert(sizeof(struct avs_modcfg_base) == 40); struct avs_pin_format { u32 pin_index; u32 iobs; struct avs_audio_format audio_fmt; } __packed; +static_assert(sizeof(struct avs_pin_format) == 32); struct avs_modcfg_ext { struct avs_modcfg_base base; @@ -703,6 +725,7 @@ struct avs_modcfg_ext { /* input pin formats followed by output ones */ struct avs_pin_format pin_fmts[]; } __packed; +static_assert(sizeof(struct avs_modcfg_ext) == 56); enum avs_dma_type { AVS_DMA_HDA_HOST_OUTPUT = 0, @@ -726,6 +749,7 @@ union avs_virtual_index { u8 instance:3; } dmic; } __packed; +static_assert(sizeof(union avs_virtual_index) == 1); union avs_connector_node_id { u32 val; @@ -735,6 +759,7 @@ union avs_connector_node_id { u32 rsvd:19; }; } __packed; +static_assert(sizeof(union avs_connector_node_id) == 4); #define INVALID_PIPELINE_ID 0xFF #define INVALID_NODE_ID \ @@ -747,6 +772,7 @@ union avs_gtw_attributes { u32 rsvd:31; }; } __packed; +static_assert(sizeof(union avs_gtw_attributes) == 4); struct avs_copier_gtw_cfg { union avs_connector_node_id node_id; @@ -757,6 +783,7 @@ struct avs_copier_gtw_cfg { DECLARE_FLEX_ARRAY(u32, blob); } config; } __packed; +static_assert(sizeof(struct avs_copier_gtw_cfg) == 16); struct avs_copier_cfg { struct avs_modcfg_base base; @@ -764,6 +791,7 @@ struct avs_copier_cfg { u32 feature_mask; struct avs_copier_gtw_cfg gtw_cfg; } __packed; +static_assert(sizeof(struct avs_copier_cfg) == 84); struct avs_volume_cfg { u32 channel_id; @@ -772,22 +800,26 @@ struct avs_volume_cfg { u32 reserved; /* alignment */ u64 curve_duration; } __packed; +static_assert(sizeof(struct avs_volume_cfg) == 24); struct avs_peakvol_cfg { struct avs_modcfg_base base; struct avs_volume_cfg vols[]; } __packed; +static_assert(sizeof(struct avs_peakvol_cfg) == 40); struct avs_micsel_cfg { struct avs_modcfg_base base; struct avs_audio_format out_fmt; } __packed; +static_assert(sizeof(struct avs_micsel_cfg) == 64); struct avs_mux_cfg { struct avs_modcfg_base base; struct avs_audio_format ref_fmt; struct avs_audio_format out_fmt; } __packed; +static_assert(sizeof(struct avs_mux_cfg) == 88); struct avs_updown_mixer_cfg { struct avs_modcfg_base base; @@ -796,21 +828,25 @@ struct avs_updown_mixer_cfg { s32 coefficients[AVS_CHANNELS_MAX]; u32 channel_map; } __packed; +static_assert(sizeof(struct avs_updown_mixer_cfg) == 84); struct avs_src_cfg { struct avs_modcfg_base base; u32 out_freq; } __packed; +static_assert(sizeof(struct avs_src_cfg) == 44); struct avs_probe_gtw_cfg { union avs_connector_node_id node_id; u32 dma_buffer_size; } __packed; +static_assert(sizeof(struct avs_probe_gtw_cfg) == 8); struct avs_probe_cfg { struct avs_modcfg_base base; struct avs_probe_gtw_cfg gtw_cfg; } __packed; +static_assert(sizeof(struct avs_probe_cfg) == 48); struct avs_aec_cfg { struct avs_modcfg_base base; @@ -818,6 +854,7 @@ struct avs_aec_cfg { struct avs_audio_format out_fmt; u32 cpc_lp_mode; } __packed; +static_assert(sizeof(struct avs_aec_cfg) == 92); struct avs_asrc_cfg { struct avs_modcfg_base base; @@ -828,11 +865,13 @@ struct avs_asrc_cfg { u32 disable_jitter_buffer:1; u32 rsvd3:27; } __packed; +static_assert(sizeof(struct avs_asrc_cfg) == 48); struct avs_wov_cfg { struct avs_modcfg_base base; u32 cpc_lp_mode; } __packed; +static_assert(sizeof(struct avs_wov_cfg) == 44); /* Module runtime parameters */ @@ -845,6 +884,7 @@ struct avs_copier_sink_format { struct avs_audio_format src_fmt; struct avs_audio_format sink_fmt; } __packed; +static_assert(sizeof(struct avs_copier_sink_format) == 52); int avs_ipc_copier_set_sink_format(struct avs_dev *adev, u16 module_id, u8 instance_id, u32 sink_id, @@ -878,6 +918,7 @@ struct avs_probe_dma { union avs_connector_node_id node_id; u32 dma_buffer_size; } __packed; +static_assert(sizeof(struct avs_probe_dma) == 8); enum avs_probe_type { AVS_PROBE_TYPE_INPUT = 0, @@ -894,6 +935,7 @@ union avs_probe_point_id { u32 index:6; } id; } __packed; +static_assert(sizeof(union avs_probe_point_id) == 4); enum avs_connection_purpose { AVS_CONNECTION_PURPOSE_EXTRACT = 0, @@ -906,6 +948,7 @@ struct avs_probe_point_desc { u32 purpose; union avs_connector_node_id node_id; } __packed; +static_assert(sizeof(struct avs_probe_point_desc) == 12); int avs_ipc_probe_get_dma(struct avs_dev *adev, struct avs_probe_dma **dmas, size_t *num_dmas); int avs_ipc_probe_attach_dma(struct avs_dev *adev, struct avs_probe_dma *dmas, size_t num_dmas); From 9a385993504e47a0fd6fd34b5384827b4abdee60 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 5 Apr 2024 11:09:29 +0200 Subject: [PATCH 220/603] ASoC: Intel: avs: Rule invalid buffer and period sizes out While HDAudio controller supports buffer packets up to 128 bytes low, audio format shall be taken into consideration when calculating buffer and period sizes to avoid undesired xruns. As *_size in ALSA terms means frames (channels times bit-depth-bytes), hw_rules can calculate minimal buffer and period sizes solely from sample rate and the number of milliseconds commonly used on the AudioDSP firmware side. Signed-off-by: Cezary Rojewski Link: https://msgid.link/r/20240405090929.1184068-14-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 405de1d581786..77a7e8f939515 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -457,6 +457,26 @@ static const struct snd_pcm_hw_constraint_list hw_rates = { const struct snd_soc_dai_ops avs_dai_fe_ops; +static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *interval = hw_param_interval(params, rule->var); + struct snd_interval to; + + snd_interval_any(&to); + to.integer = interval->integer; + to.max = interval->max; + /* + * Commonly 2ms buffer size is used in HDA scenarios whereas 4ms is used + * when streaming through GPDMA. Align to the latter to account for both. + */ + to.min = params_rate(params) / 1000 * 4; + + if (rule->var == SNDRV_PCM_HW_PARAM_PERIOD_SIZE) + to.min /= params_periods(params); + + return snd_interval_refine(interval, &to); +} + static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -492,6 +512,14 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so if (ret < 0) goto err; + /* Adjust buffer and period size based on the audio format. */ + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, hw_rule_param_size, NULL, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, hw_rule_param_size, NULL, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_set_sync(substream); dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p", From 72829b98ff3a22efb66e5b618bd0219111db1811 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:14 +0200 Subject: [PATCH 221/603] ALSA: emux: fix /proc teardown at module unload We forgot to remember the wavetable /proc entry, so we'd fail to free it at module unload. This matters only when only the synth module is unloaded, as unloading the card driver would tear down the sub-entry anyway. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-2-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_proc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c index 7993e6a01e548..820351f52551e 100644 --- a/sound/synth/emux/emux_proc.c +++ b/sound/synth/emux/emux_proc.c @@ -102,6 +102,7 @@ void snd_emux_proc_init(struct snd_emux *emu, struct snd_card *card, int device) entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; entry->c.text.read = snd_emux_proc_info_read; + emu->proc = entry; } void snd_emux_proc_free(struct snd_emux *emu) From 3f3e0dfc83d586fe9204936fccae771754a9dbc2 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:15 +0200 Subject: [PATCH 222/603] ALSA: emux: prune unused parameter from snd_soundfont_load_guspatch() The `client` parameter was not used, so eliminate it from the call chain. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-3-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- include/sound/soundfont.h | 2 +- sound/synth/emux/emux_hwdep.c | 3 +-- sound/synth/emux/emux_oss.c | 3 +-- sound/synth/emux/soundfont.c | 7 +++---- 4 files changed, 6 insertions(+), 9 deletions(-) diff --git a/include/sound/soundfont.h b/include/sound/soundfont.h index e445688a4f4f9..98ed98d89d6de 100644 --- a/include/sound/soundfont.h +++ b/include/sound/soundfont.h @@ -89,7 +89,7 @@ struct snd_sf_list { int snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, long count, int client); int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data, - long count, int client); + long count); int snd_soundfont_close_check(struct snd_sf_list *sflist, int client); struct snd_sf_list *snd_sf_new(struct snd_sf_callback *callback, diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index 81719bfb8ed73..fd8f978cde1c6 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -27,8 +27,7 @@ snd_emux_hwdep_load_patch(struct snd_emux *emu, void __user *arg) if (patch.key == GUS_PATCH) return snd_soundfont_load_guspatch(emu->sflist, arg, - patch.len + sizeof(patch), - TMP_CLIENT_ID); + patch.len + sizeof(patch)); if (patch.type >= SNDRV_SFNT_LOAD_INFO && patch.type <= SNDRV_SFNT_PROBE_DATA) { diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index d8d32671f7038..04df46b269d37 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -205,8 +205,7 @@ snd_emux_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, return -ENXIO; if (format == GUS_PATCH) - rc = snd_soundfont_load_guspatch(emu->sflist, buf, count, - SF_CLIENT_NO(p->chset.port)); + rc = snd_soundfont_load_guspatch(emu->sflist, buf, count); else if (format == SNDRV_OSS_SOUNDFONT_PATCH) { struct soundfont_patch_info patch; if (count < (int)sizeof(patch)) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 16f00097cb95a..e1e47518ac92c 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -941,8 +941,7 @@ int snd_sf_vol_table[128] = { /* load GUS patch */ static int -load_guspatch(struct snd_sf_list *sflist, const char __user *data, - long count, int client) +load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) { struct patch_info patch; struct snd_soundfont *sf; @@ -1122,11 +1121,11 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, /* load GUS patch */ int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data, - long count, int client) + long count) { int rc; lock_preset(sflist); - rc = load_guspatch(sflist, data, count, client); + rc = load_guspatch(sflist, data, count); unlock_preset(sflist); return rc; } From 19061f35b3eaf4925960be44d870244b99df8d1d Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:16 +0200 Subject: [PATCH 223/603] ALSA: emux: fix validation of snd_emux.num_ports Both bounds had off-by-one errors. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-4-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_seq.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index b227c7e0bc2ac..1adaa75df2f6a 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -65,11 +65,11 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) return -ENODEV; } - if (emu->num_ports < 0) { + if (emu->num_ports <= 0) { snd_printk(KERN_WARNING "seqports must be greater than zero\n"); emu->num_ports = 1; - } else if (emu->num_ports >= SNDRV_EMUX_MAX_PORTS) { - snd_printk(KERN_WARNING "too many ports." + } else if (emu->num_ports > SNDRV_EMUX_MAX_PORTS) { + snd_printk(KERN_WARNING "too many ports. " "limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS); emu->num_ports = SNDRV_EMUX_MAX_PORTS; } From 877d1e81c7a4c47c69a098cd8b87756b2809e885 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:17 +0200 Subject: [PATCH 224/603] ALSA: emux: fix init of patch_info.truesize in load_data() The field is explicitly documented to be initialized by the driver (which it actually is). Also, using patch_info.size would be actually wrong for 16-bit data, as one field counts samples, while the other counts bytes. load_guspatch() already did it right. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-5-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/soundfont.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index e1e47518ac92c..ad0231d7a39db 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -735,7 +735,7 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) sp->v = sample_info; sp->v.sf_id = sf->id; sp->v.dummy = 0; - sp->v.truesize = sp->v.size; + sp->v.truesize = 0; /* * If there is wave data then load it. From 1edeac6555e9df008b1729ca445868c1177baa8b Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:18 +0200 Subject: [PATCH 225/603] ALSA: emu10k1: prune vestiges of SNDRV_SFNT_SAMPLE_{BIDIR,REVERSE}_LOOP support This is required only to implement WAVE_BIDIR_LOOP and WAVE_LOOP_BACK in the GUS patch loader. It has not worked on emu10k1 since before ALSA hit mainline, yet nobody appears to have complained. And as it isn't super easy to implement, just admit defeat and clean up the code. If somebody wanted to resurrect the feature, the emu8k driver could serve as a template, but the code would be quite different. But arguably, this should be done in user space in the first place, as this doesn't represent a hardware feature (somewhat ironically, the actual GUS driver has no synth support, and therefore no GUS patch loader). Note that instead of properly rejecting affected samples, we continue to just pretend that the feature wasn't requested. This is extremely questionable behavior, but avoids that possibly unused instruments suddenly prevent loading the entire file, which would break backwards compatibility. But at least we log a warning now. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-6-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 73 ++++--------------------------- 1 file changed, 8 insertions(+), 65 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 89890f24509f0..49214c2268084 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -28,8 +28,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, { int offset; int truesize, size, blocksize; - __maybe_unused int loopsize; - int loopend, sampleend; unsigned int start_addr; struct snd_emu10k1 *emu; @@ -43,32 +41,24 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return 0; } + if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP | SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) { + /* should instead return -ENOTSUPP; but compatibility */ + printk(KERN_WARNING "Emu10k1 wavetable patch %d with unsupported loop feature\n", + sp->v.sample); + } + /* recalculate address offset */ sp->v.end -= sp->v.start; sp->v.loopstart -= sp->v.start; sp->v.loopend -= sp->v.start; sp->v.start = 0; - /* some samples have invalid data. the addresses are corrected in voice info */ - sampleend = sp->v.end; - if (sampleend > sp->v.size) - sampleend = sp->v.size; - loopend = sp->v.loopend; - if (loopend > sampleend) - loopend = sampleend; - /* be sure loop points start < end */ if (sp->v.loopstart >= sp->v.loopend) swap(sp->v.loopstart, sp->v.loopend); /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; - loopsize = 0; -#if 0 /* not supported */ - if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) - loopsize = sp->v.loopend - sp->v.loopstart; - truesize += loopsize; -#endif if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) truesize += BLANK_LOOP_SIZE; @@ -96,8 +86,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, snd_emu10k1_synth_bzero(emu, sp->block, offset, size); offset += size; - /* copy start->loopend */ - size = loopend; + /* copy provided samples */ + size = sp->v.size; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; if (offset + size > blocksize) @@ -108,53 +98,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return -EFAULT; } offset += size; - data += size; - -#if 0 /* not supported yet */ - /* handle reverse (or bidirectional) loop */ - if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) { - /* copy loop in reverse */ - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) { - int woffset; - unsigned short *wblock = (unsigned short*)block; - woffset = offset / 2; - if (offset + loopsize * 2 > blocksize) - return -EINVAL; - for (i = 0; i < loopsize; i++) - wblock[woffset + i] = wblock[woffset - i -1]; - offset += loopsize * 2; - } else { - if (offset + loopsize > blocksize) - return -EINVAL; - for (i = 0; i < loopsize; i++) - block[offset + i] = block[offset - i -1]; - offset += loopsize; - } - - /* modify loop pointers */ - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_BIDIR_LOOP) { - sp->v.loopend += loopsize; - } else { - sp->v.loopstart += loopsize; - sp->v.loopend += loopsize; - } - /* add sample pointer */ - sp->v.end += loopsize; - } -#endif - - /* loopend -> sample end */ - size = sp->v.size - loopend; - if (size < 0) - return -EINVAL; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - size *= 2; - if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) { - snd_emu10k1_synth_free(emu, sp->block); - sp->block = NULL; - return -EFAULT; - } - offset += size; /* clear rest of samples (if any) */ if (offset < blocksize) From de67aab120d4d5ba7d9e94ee5b25464ae0d1bd0e Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:19 +0200 Subject: [PATCH 226/603] ALSA: emux: centralize & improve patch info validation This does several closely related things: - Move the code from the drivers into the SoundFont loader, which de-duplicates it. - Sort of explain the weird "recalculate address offset" feature. Note that I don't think it actually makes any sense - the calling user space code should do that. The background is certainly that the source data (the SoundFont format) uses pointers into a single wave block (and the API allows doing the same for on-board ROM), but the API expects the wave data from user space to be pre-chopped into individual patches anyway. - Make sure that the specified offsets actually lie within the supplied wave data. Note that we don't validate ROM offsets, so one can play back anything within the sound card's address space. - In load_guspatch(), don't call the sample_new callback anymore when the patch size is zero, as was already the case in load_data(). The callbacks would instantly return in that case anyway; these checks are now removed. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-7-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_patch.c | 13 ----------- sound/pci/emu10k1/emu10k1_patch.c | 16 ------------- sound/synth/emux/soundfont.c | 37 ++++++++++++++++++++++++++++++- 3 files changed, 36 insertions(+), 30 deletions(-) diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index 8c1e7f2bfc34a..ab4f988f080dc 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -148,13 +148,6 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (snd_BUG_ON(!sp)) return -EINVAL; - if (sp->v.size == 0) - return 0; - - /* be sure loop points start < end */ - if (sp->v.loopstart > sp->v.loopend) - swap(sp->v.loopstart, sp->v.loopend); - /* compute true data size to be loaded */ truesize = sp->v.size; if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP|SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) @@ -177,12 +170,6 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, return -EFAULT; } - /* recalculate address offset */ - sp->v.end -= sp->v.start; - sp->v.loopstart -= sp->v.start; - sp->v.loopend -= sp->v.start; - sp->v.start = 0; - /* dram position (in word) -- mem_offset is byte */ dram_offset = EMU8000_DRAM_OFFSET + (sp->block->offset >> 1); dram_start = dram_offset; diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 49214c2268084..47d69a0e44bc6 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -35,28 +35,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (snd_BUG_ON(!sp || !hdr)) return -EINVAL; - if (sp->v.size == 0) { - dev_dbg(emu->card->dev, - "emu: rom font for sample %d\n", sp->v.sample); - return 0; - } - if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP | SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) { /* should instead return -ENOTSUPP; but compatibility */ printk(KERN_WARNING "Emu10k1 wavetable patch %d with unsupported loop feature\n", sp->v.sample); } - /* recalculate address offset */ - sp->v.end -= sp->v.start; - sp->v.loopstart -= sp->v.start; - sp->v.loopend -= sp->v.start; - sp->v.start = 0; - - /* be sure loop points start < end */ - if (sp->v.loopstart >= sp->v.loopend) - swap(sp->v.loopstart, sp->v.loopend); - /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index ad0231d7a39db..6d6f0102ed5b9 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -689,6 +689,21 @@ find_sample(struct snd_soundfont *sf, int sample_id) } +static int +validate_sample_info(struct soundfont_sample_info *si) +{ + if (si->end < 0 || si->end > si->size) + return -EINVAL; + if (si->loopstart < 0 || si->loopstart > si->end) + return -EINVAL; + if (si->loopend < 0 || si->loopend > si->end) + return -EINVAL; + /* be sure loop points start < end */ + if (si->loopstart > si->loopend) + swap(si->loopstart, si->loopend); + return 0; +} + /* * Load sample information, this can include data to be loaded onto * the soundcard. It can also just be a pointer into soundcard ROM. @@ -727,6 +742,21 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) return -EINVAL; } + if (sample_info.size > 0) { + if (sample_info.start < 0) + return -EINVAL; + + // Here we "rebase out" the start address, because the + // real start is the start of the provided sample data. + sample_info.end -= sample_info.start; + sample_info.loopstart -= sample_info.start; + sample_info.loopend -= sample_info.start; + sample_info.start = 0; + + if (validate_sample_info(&sample_info) < 0) + return -EINVAL; + } + /* Allocate a new sample structure */ sp = sf_sample_new(sflist, sf); if (!sp) @@ -974,6 +1004,11 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) smp->v.loopend = patch.loop_end; smp->v.size = patch.len; + if (validate_sample_info(&smp->v) < 0) { + sf_sample_delete(sflist, sf, smp); + return -EINVAL; + } + /* set up mode flags */ smp->v.mode_flags = 0; if (!(patch.mode & WAVE_16_BITS)) @@ -1011,7 +1046,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) /* * load wave data */ - if (sflist->callback.sample_new) { + if (smp->v.size > 0 && sflist->callback.sample_new) { rc = sflist->callback.sample_new (sflist->callback.private_data, smp, sflist->memhdr, data, count); From 89b32ccb12ae67e630c6453d778ec30a592a212f Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:20 +0200 Subject: [PATCH 227/603] ALSA: emux: improve patch ioctl data validation In load_data(), make the validation of and skipping over the main info block match that in load_guspatch(). In load_guspatch(), add checking that the specified patch length matches the actually supplied data, like load_data() already did. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-8-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/soundfont.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 6d6f0102ed5b9..4edc693da8e74 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -716,7 +716,6 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) struct snd_soundfont *sf; struct soundfont_sample_info sample_info; struct snd_sf_sample *sp; - long off; /* patch must be opened */ sf = sflist->currsf; @@ -726,12 +725,16 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) if (is_special_type(sf->type)) return -EINVAL; + if (count < (long)sizeof(sample_info)) { + return -EINVAL; + } if (copy_from_user(&sample_info, data, sizeof(sample_info))) return -EFAULT; + data += sizeof(sample_info); + count -= sizeof(sample_info); - off = sizeof(sample_info); - - if (sample_info.size != (count-off)/2) + // SoundFont uses S16LE samples. + if (sample_info.size * 2 != count) return -EINVAL; /* Check for dup */ @@ -774,7 +777,7 @@ load_data(struct snd_sf_list *sflist, const void __user *data, long count) int rc; rc = sflist->callback.sample_new (sflist->callback.private_data, sp, sflist->memhdr, - data + off, count - off); + data, count); if (rc < 0) { sf_sample_delete(sflist, sf, sp); return rc; @@ -986,10 +989,12 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) } if (copy_from_user(&patch, data, sizeof(patch))) return -EFAULT; - count -= sizeof(patch); data += sizeof(patch); + if ((patch.len << (patch.mode & WAVE_16_BITS ? 1 : 0)) != count) + return -EINVAL; + sf = newsf(sflist, SNDRV_SFNT_PAT_TYPE_GUS|SNDRV_SFNT_PAT_SHARED, NULL); if (sf == NULL) return -ENOMEM; From 6e36d4c2744e143625cd2fcbf9d38ff76cda5e2a Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:21 +0200 Subject: [PATCH 228/603] ALSA: emu10k1: move patch loader assertions into low-level functions Convert some checks in snd_emu10k1_sample_new() back into assertions (as they were prior to da3cec35dd (ALSA: Kill snd_assert() in sound/pci/*, 2008-08-08)), and move them into the low-level memory access functions they protect. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-9-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 4 ---- sound/pci/emu10k1/memory.c | 6 ++++++ 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 47d69a0e44bc6..55bb60d31fe45 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -65,8 +65,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, size = BLANK_HEAD_SIZE; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; - if (offset + size > blocksize) - return -EINVAL; snd_emu10k1_synth_bzero(emu, sp->block, offset, size); offset += size; @@ -74,8 +72,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, size = sp->v.size; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; - if (offset + size > blocksize) - return -EINVAL; if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) { snd_emu10k1_synth_free(emu, sp->block); sp->block = NULL; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 20b07117574b4..fc94444041516 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -574,6 +574,9 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk void *ptr; struct snd_emu10k1_memblk *p = (struct snd_emu10k1_memblk *)blk; + if (snd_BUG_ON(offset + size > p->mem.size)) + return -EFAULT; + offset += blk->offset & (PAGE_SIZE - 1); end_offset = offset + size; page = get_aligned_page(offset); @@ -604,6 +607,9 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me void *ptr; struct snd_emu10k1_memblk *p = (struct snd_emu10k1_memblk *)blk; + if (snd_BUG_ON(offset + size > p->mem.size)) + return -EFAULT; + offset += blk->offset & (PAGE_SIZE - 1); end_offset = offset + size; page = get_aligned_page(offset); From 38fc804a776ea66ca8ac8113022e445c587f5e01 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:22 +0200 Subject: [PATCH 229/603] ALSA: emu10k1: fix sample signedness issues in wavetable loader The hardware supports S16LE and U8 samples, while U16LE and S8 (which the driver implicitly claims to support) require sign flipping. Note that this matters only for the GUS patch loader, as the implemented SoundFont v2.01 spec is limited to S16LE. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-10-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 4 +-- sound/pci/emu10k1/emu10k1_patch.c | 30 ++++++++----------- sound/pci/emu10k1/memory.c | 49 +++++++++++++++++++++++++------ 3 files changed, 55 insertions(+), 28 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 1af9e68193920..9e3bd4f814609 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1882,8 +1882,8 @@ int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size, struct snd_dma_buffer *dmab); struct snd_util_memblk *snd_emu10k1_synth_alloc(struct snd_emu10k1 *emu, unsigned int size); int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *blk); -int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size); -int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size); +int snd_emu10k1_synth_memset(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size, u8 value); +int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, const char __user *data, int size, u32 xor); int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk); /* voice allocation */ diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 55bb60d31fe45..eb3d1ef8a33a9 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -26,6 +26,8 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, struct snd_util_memhdr *hdr, const void __user *data, long count) { + u8 fill; + u32 xor; int offset; int truesize, size, blocksize; unsigned int start_addr; @@ -41,6 +43,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, sp->v.sample); } + if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) { + fill = 0x80; + xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0 : 0x80808080; + } else { + fill = 0; + xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0x80008000 : 0; + } + /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) @@ -65,14 +75,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, size = BLANK_HEAD_SIZE; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; - snd_emu10k1_synth_bzero(emu, sp->block, offset, size); + snd_emu10k1_synth_memset(emu, sp->block, offset, size, fill); offset += size; /* copy provided samples */ size = sp->v.size; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) size *= 2; - if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size)) { + if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) { snd_emu10k1_synth_free(emu, sp->block); sp->block = NULL; return -EFAULT; @@ -81,7 +91,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* clear rest of samples (if any) */ if (offset < blocksize) - snd_emu10k1_synth_bzero(emu, sp->block, offset, blocksize - offset); + snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill); if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) { /* if no blank loop is attached in the sample, add it */ @@ -91,20 +101,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } } -#if 0 /* not supported yet */ - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) { - /* unsigned -> signed */ - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) { - unsigned short *wblock = (unsigned short*)block; - for (i = 0; i < truesize; i++) - wblock[i] ^= 0x8000; - } else { - for (i = 0; i < truesize; i++) - block[i] ^= 0x80; - } - } -#endif - /* recalculate offset */ start_addr = BLANK_HEAD_SIZE * 2; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index fc94444041516..d29711777161c 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -565,10 +565,10 @@ static inline void *offset_ptr(struct snd_emu10k1 *emu, int page, int offset) } /* - * bzero(blk + offset, size) + * memset(blk + offset, value, size) */ -int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, - int offset, int size) +int snd_emu10k1_synth_memset(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, + int offset, int size, u8 value) { int page, nextofs, end_offset, temp, temp1; void *ptr; @@ -588,20 +588,47 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk temp = temp1; ptr = offset_ptr(emu, page + p->first_page, offset); if (ptr) - memset(ptr, 0, temp); + memset(ptr, value, temp); offset = nextofs; page++; } while (offset < end_offset); return 0; } -EXPORT_SYMBOL(snd_emu10k1_synth_bzero); +EXPORT_SYMBOL(snd_emu10k1_synth_memset); + +// Note that the value is assumed to be suitably repetitive. +static void xor_range(void *ptr, int size, u32 value) +{ + if ((long)ptr & 1) { + *(u8 *)ptr ^= (u8)value; + ptr++; + size--; + } + if (size > 1 && ((long)ptr & 2)) { + *(u16 *)ptr ^= (u16)value; + ptr += 2; + size -= 2; + } + while (size > 3) { + *(u32 *)ptr ^= value; + ptr += 4; + size -= 4; + } + if (size > 1) { + *(u16 *)ptr ^= (u16)value; + ptr += 2; + size -= 2; + } + if (size > 0) + *(u8 *)ptr ^= (u8)value; +} /* - * copy_from_user(blk + offset, data, size) + * copy_from_user(blk + offset, data, size) ^ xor */ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, - int offset, const char __user *data, int size) + int offset, const char __user *data, int size, u32 xor) { int page, nextofs, end_offset, temp, temp1; void *ptr; @@ -620,8 +647,12 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me if (temp1 < temp) temp = temp1; ptr = offset_ptr(emu, page + p->first_page, offset); - if (ptr && copy_from_user(ptr, data, temp)) - return -EFAULT; + if (ptr) { + if (copy_from_user(ptr, data, temp)) + return -EFAULT; + if (xor) + xor_range(ptr, temp, xor); + } offset = nextofs; data += temp; page++; From bca5174b437307c9315e25768ed2b2bfcf6f561c Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:23 +0200 Subject: [PATCH 230/603] ALSA: emu10k1: fix playback of 8-bit wavetable samples Samples are byte-sized in this mode, and thus the offset calculation needs no shifting. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-11-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_callback.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 941bfbf812ed3..5f6c47cbb809b 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -310,6 +310,7 @@ start_voice(struct snd_emux_voice *vp) { unsigned int temp; int ch; + bool w_16; u32 psst, dsl, map, ccca, vtarget; unsigned int addr, mapped_offset; struct snd_midi_channel *chan; @@ -321,6 +322,7 @@ start_voice(struct snd_emux_voice *vp) if (snd_BUG_ON(ch < 0)) return -EINVAL; chan = vp->chan; + w_16 = !(vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS); emem = (struct snd_emu10k1_memblk *)vp->block; if (emem == NULL) @@ -330,7 +332,7 @@ start_voice(struct snd_emux_voice *vp) /* dev_err(hw->card->devK, "emu: cannot map!\n"); */ return -ENOMEM; } - mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1; + mapped_offset = snd_emu10k1_memblk_offset(emem) >> w_16; vp->reg.start += mapped_offset; vp->reg.end += mapped_offset; vp->reg.loopstart += mapped_offset; @@ -371,7 +373,7 @@ start_voice(struct snd_emux_voice *vp) unsigned int shift = (vp->apitch - 0xe000) >> 10; ccca |= shift << 25; } - if (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_8BITS) + if (!w_16) ccca |= CCCA_8BITSELECT; vtarget = (unsigned int)vp->vtarget << 16; From 93fd86a47de3097488611ffbfe12b4940933670d Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:24 +0200 Subject: [PATCH 231/603] ALSA: emu10k1: merge conditions in patch loader This de-duplicates the code slightly. But the real reason is that it moves the code up, which the next patch will depend on. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-12-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index eb3d1ef8a33a9..281881f7d0a44 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -53,8 +53,14 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* compute true data size to be loaded */ truesize = sp->v.size + BLANK_HEAD_SIZE; - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) + if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) { truesize += BLANK_LOOP_SIZE; + /* if no blank loop is attached in the sample, add it */ + if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT) { + sp->v.loopstart = sp->v.end + BLANK_LOOP_START; + sp->v.loopend = sp->v.end + BLANK_LOOP_END; + } + } /* try to allocate a memory block */ blocksize = truesize; @@ -93,14 +99,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (offset < blocksize) snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill); - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_NO_BLANK) { - /* if no blank loop is attached in the sample, add it */ - if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_SINGLESHOT) { - sp->v.loopstart = sp->v.end + BLANK_LOOP_START; - sp->v.loopend = sp->v.end + BLANK_LOOP_END; - } - } - /* recalculate offset */ start_addr = BLANK_HEAD_SIZE * 2; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) From 392925791a5b6f41806d445ea71319a116e32295 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:25 +0200 Subject: [PATCH 232/603] ALSA: emu10k1: fix wavetable offset recalculation The offsets are counted in samples, not in bytes. While the code block is being rewritten, also move it up a bit, to avoid churn in a subsequent patch. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-13-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 281881f7d0a44..ad16de99b8001 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -30,7 +30,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, u32 xor; int offset; int truesize, size, blocksize; - unsigned int start_addr; struct snd_emu10k1 *emu; emu = rec->hw; @@ -62,6 +61,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } } + /* recalculate offset */ + sp->v.start += BLANK_HEAD_SIZE; + sp->v.end += BLANK_HEAD_SIZE; + sp->v.loopstart += BLANK_HEAD_SIZE; + sp->v.loopend += BLANK_HEAD_SIZE; + /* try to allocate a memory block */ blocksize = truesize; if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) @@ -99,15 +104,6 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (offset < blocksize) snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill); - /* recalculate offset */ - start_addr = BLANK_HEAD_SIZE * 2; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - start_addr >>= 1; - sp->v.start += start_addr; - sp->v.end += start_addr; - sp->v.loopstart += start_addr; - sp->v.loopend += start_addr; - return 0; } From 80d7c3cccd546c16da2ef9d2e88eaf215498c1e1 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:26 +0200 Subject: [PATCH 233/603] ALSA: emu10k1: de-duplicate size calculations for 16-bit samples Instead of repeatedly checking the sample width, assign a size shift centrally. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-14-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index ad16de99b8001..481fe03fef4de 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -28,6 +28,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, { u8 fill; u32 xor; + int shift; int offset; int truesize, size, blocksize; struct snd_emu10k1 *emu; @@ -43,9 +44,11 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) { + shift = 0; fill = 0x80; xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0 : 0x80808080; } else { + shift = 1; fill = 0; xor = (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_UNSIGNED) ? 0x80008000 : 0; } @@ -68,9 +71,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, sp->v.loopend += BLANK_HEAD_SIZE; /* try to allocate a memory block */ - blocksize = truesize; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - blocksize *= 2; + blocksize = truesize << shift; sp->block = snd_emu10k1_synth_alloc(emu, blocksize); if (sp->block == NULL) { dev_dbg(emu->card->dev, @@ -83,16 +84,12 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, /* write blank samples at head */ offset = 0; - size = BLANK_HEAD_SIZE; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - size *= 2; + size = BLANK_HEAD_SIZE << shift; snd_emu10k1_synth_memset(emu, sp->block, offset, size, fill); offset += size; /* copy provided samples */ - size = sp->v.size; - if (! (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS)) - size *= 2; + size = sp->v.size << shift; if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) { snd_emu10k1_synth_free(emu, sp->block); sp->block = NULL; From 65db949667b0b74f4534e96f762aff0e6687dc51 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:27 +0200 Subject: [PATCH 234/603] ALSA: emu10k1: improve cache behavior documentation Resulting from more reverse engineering in the course of debugging. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-15-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 28 ++++++++++++++++++---------- 1 file changed, 18 insertions(+), 10 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 9e3bd4f814609..12c7dc760724a 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -598,17 +598,25 @@ SUB_REG(PEFE, FILTERAMOUNT, 0x000000ff) /* Filter envlope amount */ // In stereo mode, the two channels' caches are concatenated into one, // and hold the interleaved frames. // The cache holds 64 frames, so the upper half is not used in 8-bit mode. -// All registers mentioned below count in frames. -// The cache is a ring buffer; CCR_READADDRESS operates modulo 64. -// The cache is filled from (CCCA_CURRADDR - CCR_CACHEINVALIDSIZE) -// into (CCR_READADDRESS - CCR_CACHEINVALIDSIZE). +// All registers mentioned below count in frames. Shortcuts: +// CA = CCCA_CURRADDR, CRA = CCR_READADDRESS, +// CLA = CCR_CACHELOOPADDRHI:CLP_CACHELOOPADDR, +// CIS = CCR_CACHEINVALIDSIZE, LIS = CCR_LOOPINVALSIZE, +// CLF = CCR_CACHELOOPFLAG, LF = CCR_LOOPFLAG +// The cache is a ring buffer; CRA operates modulo 64. +// The cache is filled from (CA - CIS) into (CRA - CIS). // The engine has a fetch threshold of 32 bytes, so it tries to keep -// CCR_CACHEINVALIDSIZE below 8 (16-bit stereo), 16 (16-bit mono, -// 8-bit stereo), or 32 (8-bit mono). The actual transfers are pretty -// unpredictable, especially if several voices are running. -// Frames are consumed at CCR_READADDRESS, which is incremented afterwards, -// along with CCCA_CURRADDR and CCR_CACHEINVALIDSIZE. This implies that the -// actual playback position always lags CCCA_CURRADDR by exactly 64 frames. +// CIS below 8 (16-bit stereo), 16 (16-bit mono, 8-bit stereo), or +// 32 (8-bit mono). The actual transfers are pretty unpredictable, +// especially if several voices are running. +// Frames are consumed at CRA, which is incremented afterwards, +// along with CA and CIS. This implies that the actual playback +// position always lags CA by exactly 64 frames. +// When CA reaches DSL_LOOPENDADDR, LF is set for one frame's time. +// LF's rising edge causes the current values of CA and CIS to be +// copied into CLA and LIS, resp., and CLF to be set. +// If CLF is set, the first LIS of the CIS frames are instead +// filled from (CLA - LIS), and CLF is subsequently reset. #define CD0 0x20 /* Cache data registers 0 .. 0x1f */ #define PTB 0x40 /* Page table base register */ From d0440680a197bef3cfd725b0982518f5d05079a5 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:28 +0200 Subject: [PATCH 235/603] ALSA: emu10k1: fix wavetable playback position and caching, take 2 Compensate for the cache lag of 64 frames, and actually populate the cache. Without these, the playback would start with garbage (which would be (mostly?) masqueraded by the note's attack phase). Note that we set the starting address only 61 frames ahead, to compensate for the interpolator's epsilon. Unlike for PCM playback, we don't even need to manually silence-fill the first frames in the cache, because we insert some silence in front of each sample anyway. A challenge are extremely short samples with a loop end below the cache size, because a) we'd have to wrap the current address to be within the loop and b) automatic pre-filling of the cache with the right data does not work in this case. We could pre-fill the cache manually, but that's slow, requires additional code for each sample width, and is made even more complex by the driver's virtual address space having no contiguous mapping for the CPU. We could have the engine fill the cache piece-wise (which is really what happens when playback is running), but that would also be complex, and we'd need to wait for the engine to handle each piece, so it wouldn't be that much faster than the manual fill. For the case of requiring only one loop iteration prior to reaching the cache size, we could leverage the engine's looping mechanism around CCR_CACHELOOPFLAG, but this special case doesn't seem worth the complexity. So we just unroll the loop as far as necessary to be able to play back the sample without any fiddling. Pedantically, this would be incorrect for loop-until-release samples with a low loop end which are released very quickly, but that would be relatively harmless, is not a plausible use case in the first place, and SoundFont sample mode 3 isn't actually implemented anyway (it's conflated with mode 1, infinite looping). Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-16-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_callback.c | 7 ++-- sound/pci/emu10k1/emu10k1_patch.c | 53 +++++++++++++++++++++++++--- 2 files changed, 53 insertions(+), 7 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 5f6c47cbb809b..ef26e4d3e2a3b 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -255,7 +255,7 @@ lookup_voices(struct snd_emux *emu, struct snd_emu10k1 *hw, /* check if sample is finished playing (non-looping only) */ if (bp != best + V_OFF && bp != best + V_FREE && (vp->reg.sample_mode & SNDRV_SFNT_SAMPLE_SINGLESHOT)) { - val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch); + val = snd_emu10k1_ptr_read(hw, CCCA_CURRADDR, vp->ch) - 64 + 3; if (val >= vp->reg.loopstart) bp = best + V_OFF; } @@ -364,7 +364,7 @@ start_voice(struct snd_emux_voice *vp) map = (hw->silent_page.addr << hw->address_mode) | (hw->address_mode ? MAP_PTI_MASK1 : MAP_PTI_MASK0); - addr = vp->reg.start; + addr = vp->reg.start + 64 - 3; temp = vp->reg.parm.filterQ; ccca = (temp << 28) | addr; if (vp->apitch < 0xe400) @@ -432,6 +432,9 @@ start_voice(struct snd_emux_voice *vp) /* Q & current address (Q 4bit value, MSB) */ CCCA, ccca, + /* cache */ + CCR, REG_VAL_PUT(CCR_CACHEINVALIDSIZE, 64), + /* reset volume */ VTFT, vtarget | vp->ftarget, CVCF, vtarget | CVCF_CURRENTFILTER_MASK, diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 481fe03fef4de..2a13fb32c1d20 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -31,6 +31,7 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, int shift; int offset; int truesize, size, blocksize; + int loop_start, loop_end, loop_size, data_end, unroll; struct snd_emu10k1 *emu; emu = rec->hw; @@ -64,12 +65,35 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, } } + loop_start = sp->v.loopstart; + loop_end = sp->v.loopend; + loop_size = loop_end - loop_start; + if (!loop_size) + return -EINVAL; + data_end = sp->v.end; + /* recalculate offset */ sp->v.start += BLANK_HEAD_SIZE; sp->v.end += BLANK_HEAD_SIZE; sp->v.loopstart += BLANK_HEAD_SIZE; sp->v.loopend += BLANK_HEAD_SIZE; + // Automatic pre-filling of the cache does not work in the presence + // of loops (*), and we don't want to fill it manually, as that is + // fiddly and slow. So we unroll the loop until the loop end is + // beyond the cache size. + // (*) Strictly speaking, a single iteration is supported (that's + // how it works when the playback engine runs), but handling this + // special case is not worth it. + unroll = 0; + while (sp->v.loopend < 64) { + truesize += loop_size; + sp->v.loopstart += loop_size; + sp->v.loopend += loop_size; + sp->v.end += loop_size; + unroll++; + } + /* try to allocate a memory block */ blocksize = truesize << shift; sp->block = snd_emu10k1_synth_alloc(emu, blocksize); @@ -89,12 +113,26 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, offset += size; /* copy provided samples */ - size = sp->v.size << shift; - if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) { - snd_emu10k1_synth_free(emu, sp->block); - sp->block = NULL; - return -EFAULT; + if (unroll && loop_end <= data_end) { + size = loop_end << shift; + if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) + goto faulty; + offset += size; + + data += loop_start << shift; + while (--unroll > 0) { + size = loop_size << shift; + if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) + goto faulty; + offset += size; + } + + size = (data_end - loop_start) << shift; + } else { + size = data_end << shift; } + if (snd_emu10k1_synth_copy_from_user(emu, sp->block, offset, data, size, xor)) + goto faulty; offset += size; /* clear rest of samples (if any) */ @@ -102,6 +140,11 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, snd_emu10k1_synth_memset(emu, sp->block, offset, blocksize - offset, fill); return 0; + +faulty: + snd_emu10k1_synth_free(emu, sp->block); + sp->block = NULL; + return -EFAULT; } /* From 62001ad1b4ee412c915120ee6ef2cfdc924bd007 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:29 +0200 Subject: [PATCH 236/603] ALSA: emu10k1: shrink blank space in front of wavetable samples There is no need for it to be 32 samples - 3 will do just fine (which is the interpolator's epsilon). The old size was presumably meant to compensate for the cache's presence, but we're now handling that properly. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-17-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_patch.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index 2a13fb32c1d20..dbfa89435ac2a 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -16,7 +16,7 @@ #define BLANK_LOOP_START 4 #define BLANK_LOOP_END 8 #define BLANK_LOOP_SIZE 12 -#define BLANK_HEAD_SIZE 32 +#define BLANK_HEAD_SIZE 3 /* * allocate a sample block and copy data from userspace From 4c4cbe66828f185903d2127aed6ac9b7302e9d3a Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sat, 6 Apr 2024 08:48:30 +0200 Subject: [PATCH 237/603] ALSA: emux: simplify snd_sf_list.callback handling Both drivers provide both sample_new and sample_free, and it makes no sense to pretend that they could not. In fact, load_data() would already crash if sample_new was null. So remove the remaining null checks. Contrary to that, the emu10k1 driver actually has a null sample_reset, though I'm not convinced that this inconsistency is justified. Signed-off-by: Oswald Buddenhagen Message-ID: <20240406064830.1029573-18-oswald.buddenhagen@gmx.de> Signed-off-by: Takashi Iwai --- sound/synth/emux/emux.c | 6 ++---- sound/synth/emux/soundfont.c | 12 +++++------- 2 files changed, 7 insertions(+), 11 deletions(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index a82af93748526..01444fc960d08 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -94,10 +94,8 @@ int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, ch /* create soundfont list */ memset(&sf_cb, 0, sizeof(sf_cb)); sf_cb.private_data = emu; - if (emu->ops.sample_new) - sf_cb.sample_new = sf_sample_new; - if (emu->ops.sample_free) - sf_cb.sample_free = sf_sample_free; + sf_cb.sample_new = sf_sample_new; + sf_cb.sample_free = sf_sample_free; if (emu->ops.sample_reset) sf_cb.sample_reset = sf_sample_reset; emu->sflist = snd_sf_new(&sf_cb, emu->memhdr); diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 4edc693da8e74..2373ed580bf8c 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -1051,7 +1051,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) /* * load wave data */ - if (smp->v.size > 0 && sflist->callback.sample_new) { + if (smp->v.size > 0) { rc = sflist->callback.sample_new (sflist->callback.private_data, smp, sflist->memhdr, data, count); @@ -1416,9 +1416,8 @@ snd_sf_clear(struct snd_sf_list *sflist) } for (sp = sf->samples; sp; sp = nextsp) { nextsp = sp->next; - if (sflist->callback.sample_free) - sflist->callback.sample_free(sflist->callback.private_data, - sp, sflist->memhdr); + sflist->callback.sample_free(sflist->callback.private_data, + sp, sflist->memhdr); kfree(sp); } kfree(sf); @@ -1520,9 +1519,8 @@ snd_soundfont_remove_unlocked(struct snd_sf_list *sflist) nextsp = sp->next; sf->samples = nextsp; sflist->mem_used -= sp->v.truesize; - if (sflist->callback.sample_free) - sflist->callback.sample_free(sflist->callback.private_data, - sp, sflist->memhdr); + sflist->callback.sample_free(sflist->callback.private_data, + sp, sflist->memhdr); kfree(sp); } } From c64c4e3f789123e82ffae3404d108cf826c9599a Mon Sep 17 00:00:00 2001 From: Mohammad Rafi Shaik Date: Mon, 8 Apr 2024 09:53:30 +0530 Subject: [PATCH 238/603] ASoC: dt-bindings: qcom,sm8250: Add QCM6490 snd QCS6490 sound card Document the bindings for the Qualcomm QCM6490 IDP and QCS6490 RB3Gen2 board specific sound card. The bindings are the same as for other newer Qualcomm ADSP sound cards, thus keep them in existing qcom,sm8250.yaml file, even though Linux driver is separate. Signed-off-by: Mohammad Rafi Shaik Acked-by: Krzysztof Kozlowski Reviewed-by: Srinivas Kandagatla Link: https://msgid.link/r/20240408042331.403103-2-quic_mohs@quicinc.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 2ab6871e89e5e..b2e15ebbd1bc9 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -29,6 +29,8 @@ properties: - enum: - qcom,apq8016-sbc-sndcard - qcom,msm8916-qdsp6-sndcard + - qcom,qcm6490-idp-sndcard + - qcom,qcs6490-rb3gen2-sndcard - qcom,qrb5165-rb5-sndcard - qcom,sc7180-qdsp6-sndcard - qcom,sc8280xp-sndcard From 5485c3fa96f78314207a30e81dbe270424d70ede Mon Sep 17 00:00:00 2001 From: Mohammad Rafi Shaik Date: Mon, 8 Apr 2024 09:53:31 +0530 Subject: [PATCH 239/603] ASoC: qcom: sc8280xp: Add support for QCM6490 and QCS6490 Add compatibles for sound card on Qualcomm QCM6490 IDP and QCS6490 RB3Gen2 boards. Signed-off-by: Mohammad Rafi Shaik Reviewed-by: Srinivas Kandagatla Link: https://msgid.link/r/20240408042331.403103-3-quic_mohs@quicinc.com Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index b7fd503a16666..878bd50ad4a73 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -169,6 +169,8 @@ static int sc8280xp_platform_probe(struct platform_device *pdev) } static const struct of_device_id snd_sc8280xp_dt_match[] = { + {.compatible = "qcom,qcm6490-idp-sndcard", "qcm6490"}, + {.compatible = "qcom,qcs6490-rb3gen2-sndcard", "qcs6490"}, {.compatible = "qcom,sc8280xp-sndcard", "sc8280xp"}, {.compatible = "qcom,sm8450-sndcard", "sm8450"}, {.compatible = "qcom,sm8550-sndcard", "sm8550"}, From 8167bd1c8a45c289316417cf938bcd8ceec24fd5 Mon Sep 17 00:00:00 2001 From: Animesh Agarwal Date: Sat, 6 Apr 2024 19:44:49 +0530 Subject: [PATCH 240/603] ASoC: dt-bindings: ti,pcm1681: Convert to dtschema Convert the Texas Instruments PCM1681 bindings to DT schema. Make bindings complete by adding #sound-dai-cells. Signed-off-by: Animesh Agarwal Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/20240406141454.45529-1-animeshagarwal28@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ti,pcm1681.txt | 15 ------- .../devicetree/bindings/sound/ti,pcm1681.yaml | 43 +++++++++++++++++++ 2 files changed, 43 insertions(+), 15 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt b/Documentation/devicetree/bindings/sound/ti,pcm1681.txt deleted file mode 100644 index 4df17185ab807..0000000000000 --- a/Documentation/devicetree/bindings/sound/ti,pcm1681.txt +++ /dev/null @@ -1,15 +0,0 @@ -Texas Instruments PCM1681 8-channel PWM Processor - -Required properties: - - - compatible: Should contain "ti,pcm1681". - - reg: The i2c address. Should contain <0x4c>. - -Examples: - - i2c_bus { - pcm1681@4c { - compatible = "ti,pcm1681"; - reg = <0x4c>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml new file mode 100644 index 0000000000000..5aa00617291c9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml @@ -0,0 +1,43 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM1681 8-channel PWM Processor + +maintainers: + - Shenghao Ding + - Kevin Lu + - Baojun Xu + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: ti,pcm1681 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + pcm1681: audio-codec@4c { + compatible = "ti,pcm1681"; + reg = <0x4c>; + }; + }; From 9cb83ed19b35e6c596b4e6644708ac4e011f32ab Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 8 Apr 2024 10:35:59 +0300 Subject: [PATCH 241/603] ASoC: Intel: sof_rt5682: Fix uninitialized variable in probe Initialize "is_legacy_cpu" to false to prevent an uninitialized variable bug. Fixes: 8efcd4864652 ("ASoC: Intel: sof_rt5682: use common module for sof_card_private initialization") Signed-off-by: Dan Carpenter Acked-by: Pierre-Louis Bossart Link: https://msgid.link/r/6e16433f-1897-46a2-b3ee-e177e7d846b8@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index aadd341a202cc..966bcd3e8bdc5 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -606,7 +606,7 @@ static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct sof_card_private *ctx; - bool is_legacy_cpu; + bool is_legacy_cpu = false; int ret; if (pdev->id_entry && pdev->id_entry->driver_data) From 25ab9c40e77ed4d73995fb6eef514e2e9c421dab Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:33:47 +0300 Subject: [PATCH 242/603] ASoC: SOF: Intel: tgl: Add fw_regs area to debugfs map for IPC4 Expose the firmware registers via debugfs. it can be of great help while debugging complex issues. The area is only available with IPC4. Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240409113349.21623-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/tgl.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index c2bb04c89b9d6..6cda47a78118a 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -22,6 +22,13 @@ static const struct snd_sof_debugfs_map tgl_dsp_debugfs[] = { {"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS}, }; +static const struct snd_sof_debugfs_map tgl_ipc4_dsp_debugfs[] = { + {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"fw_regs", HDA_DSP_BAR, SRAM_WINDOW_OFFSET(0), 0x1000, SOF_DEBUGFS_ACCESS_D0_ONLY}, +}; + static int tgl_dsp_core_get(struct snd_sof_dev *sdev, int core) { const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; @@ -75,6 +82,8 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev) /* debug */ sof_tgl_ops.ipc_dump = cnl_ipc_dump; + sof_tgl_ops.debug_map = tgl_dsp_debugfs; + sof_tgl_ops.debug_map_count = ARRAY_SIZE(tgl_dsp_debugfs); sof_tgl_ops.set_power_state = hda_dsp_set_power_state_ipc3; } @@ -105,6 +114,8 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev) /* debug */ sof_tgl_ops.ipc_dump = cnl_ipc4_dump; sof_tgl_ops.dbg_dump = hda_ipc4_dsp_dump; + sof_tgl_ops.debug_map = tgl_ipc4_dsp_debugfs; + sof_tgl_ops.debug_map_count = ARRAY_SIZE(tgl_ipc4_dsp_debugfs); sof_tgl_ops.set_power_state = hda_dsp_set_power_state_ipc4; } @@ -112,10 +123,6 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev) /* set DAI driver ops */ hda_set_dai_drv_ops(sdev, &sof_tgl_ops); - /* debug */ - sof_tgl_ops.debug_map = tgl_dsp_debugfs; - sof_tgl_ops.debug_map_count = ARRAY_SIZE(tgl_dsp_debugfs); - /* pre/post fw run */ sof_tgl_ops.post_fw_run = hda_dsp_post_fw_run; From d33a369ddad522eff40c6b9687db1488e19729d6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:33:48 +0300 Subject: [PATCH 243/603] ASoC: SOF: Intel: mtl: Add fw_regs area to debugfs map Expose the firmware registers via debugfs. it can be of great help while debugging complex issues. Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240409113349.21623-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index dc203505536c5..5d3b2d4be78ae 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -25,6 +25,7 @@ static const struct snd_sof_debugfs_map mtl_dsp_debugfs[] = { {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS}, {"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS}, {"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"fw_regs", HDA_DSP_BAR, MTL_SRAM_WINDOW_OFFSET(0), 0x1000, SOF_DEBUGFS_ACCESS_D0_ONLY}, }; static void mtl_ipc_host_done(struct snd_sof_dev *sdev) From 61faefa220262de60d652dd6b91fdcb7ecae2258 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 14:33:49 +0300 Subject: [PATCH 244/603] ASoC: SOF: Intel: lnl: Add fw_regs area to debugfs map Expose the firmware registers via debugfs. it can be of great help while debugging complex issues. Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://msgid.link/r/20240409113349.21623-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 8aa5430e95767..ac64a9f9cf9d1 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -28,6 +28,7 @@ static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = { {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS}, {"pp", HDA_DSP_PP_BAR, 0, 0x1000, SOF_DEBUGFS_ACCESS_ALWAYS}, {"dsp", HDA_DSP_BAR, 0, 0x10000, SOF_DEBUGFS_ACCESS_ALWAYS}, + {"fw_regs", HDA_DSP_BAR, MTL_SRAM_WINDOW_OFFSET(0), 0x1000, SOF_DEBUGFS_ACCESS_D0_ONLY}, }; /* this helps allows the DSP to setup DMIC/SSP */ From 9fd60615ad2835d8e0081fd2458721c8b994f28f Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Tue, 9 Apr 2024 17:34:56 +0200 Subject: [PATCH 245/603] ASoC: codecs: Rockchip on-SoC codecs should depend on ARCH_ROCKCHIP The various Rockchip embedded audio codecs are only present on Rockchip SoCs. Hence add dependencies on ARCH_ROCKCHIP, to prevent asking the user about these drivers when configuring a kernel without Rockchip SoC support. Signed-off-by: Geert Uytterhoeven Link: https://msgid.link/r/6cdbaf4afcf4d2059b257f6cb3a8a61bf5e17688.1712676714.git.geert+renesas@glider.be Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4e47bb6850aac..995eab954dd5d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1173,6 +1173,7 @@ config SND_SOC_IDT821034 config SND_SOC_INNO_RK3036 tristate "Inno codec driver for RK3036 SoC" + depends on ARCH_ROCKCHIP || COMPILE_TEST select REGMAP_MMIO config SND_SOC_ISABELLE @@ -1437,6 +1438,7 @@ config SND_SOC_PEB2466 config SND_SOC_RK3308 tristate "Rockchip RK3308 audio CODEC" depends on ARM64 || COMPILE_TEST + depends on ARCH_ROCKCHIP || COMPILE_TEST select REGMAP_MMIO help This is a device driver for the audio codec embedded in the @@ -1447,6 +1449,7 @@ config SND_SOC_RK3308 config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" + depends on ARCH_ROCKCHIP || COMPILE_TEST select REGMAP_MMIO config SND_SOC_RK817 From 22a1dd652de329394ca81dd2fe046444920c96dc Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:01 +0200 Subject: [PATCH 246/603] ASoC: dt-bindings: davinci-mcbsp: convert McBSP bindings to yaml schema Convert the binding for McBSP controllers for TI SoCs from txt to YAML schema. Add properties 'clocks', 'clock-names', 'power-domains' and '#sound-dai-cells' which were missing from the txt file. Add '#sound-dai-cells' and 'clocks' in the example which were missing from the txt file. Signed-off-by: Bastien Curutchet Reviewed-by: Rob Herring Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-2-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- .../bindings/sound/davinci-mcbsp.txt | 50 ---------- .../bindings/sound/davinci-mcbsp.yaml | 95 +++++++++++++++++++ 2 files changed, 95 insertions(+), 50 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt deleted file mode 100644 index 3ffc2562fb31a..0000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt +++ /dev/null @@ -1,50 +0,0 @@ -Texas Instruments DaVinci McBSP module -~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - -This binding describes the "Multi-channel Buffered Serial Port" (McBSP) -audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x. - - -Required properties: -~~~~~~~~~~~~~~~~~~~~ -- compatible : - "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms - -- reg : physical base address and length of the controller memory mapped - region(s). -- reg-names : Should contain: - * "mpu" for the main registers (required). - * "dat" for the data FIFO (optional). - -- dmas: three element list of DMA controller phandles, DMA request line and - TC channel ordered triplets. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: -~~~~~~~~~~~~~~~~~~~~ -- interrupts : Interrupt numbers for McBSP -- interrupt-names : Known interrupt names are "rx" and "tx" - -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt - -Example (AM1808): -~~~~~~~~~~~~~~~~~ - -mcbsp0: mcbsp@1d10000 { - compatible = "ti,da850-mcbsp"; - pinctrl-names = "default"; - pinctrl-0 = <&mcbsp0_pins>; - - reg = <0x00110000 0x1000>, - <0x00310000 0x1000>; - reg-names = "mpu", "dat"; - interrupts = <97 98>; - interrupt-names = "rx", "tx"; - dmas = <&edma0 3 1 - &edma0 2 1>; - dma-names = "tx", "rx"; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml new file mode 100644 index 0000000000000..139b594dd192e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml @@ -0,0 +1,95 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcbsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McBSP Controller for TI SoCs + +maintainers: + - Bastien Curutchet + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,da850-mcbsp + + reg: + minItems: 1 + items: + - description: CFG registers + - description: data registers + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + dmas: + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + items: + - const: tx + - const: rx + + interrupts: + items: + - description: RX interrupt + - description: TX interrupt + + interrupt-names: + items: + - const: rx + - const: tx + + clocks: + items: + - description: functional clock + + clock-names: + items: + - const: fck + + power-domains: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - "#sound-dai-cells" + - compatible + - reg + - reg-names + - dmas + - dma-names + - clocks + +unevaluatedProperties: false + +examples: + - | + mcbsp0@1d10000 { + #sound-dai-cells = <0>; + compatible = "ti,da850-mcbsp"; + pinctrl-names = "default"; + pinctrl-0 = <&mcbsp0_pins>; + + reg = <0x111000 0x1000>, + <0x311000 0x1000>; + reg-names = "mpu", "dat"; + interrupts = <97>, <98>; + interrupt-names = "rx", "tx"; + dmas = <&edma0 3 1>, + <&edma0 2 1>; + dma-names = "tx", "rx"; + + clocks = <&psc1 14>; + }; From 6a4b6b062a5917d611c1bde7189c5147cf0ca832 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:02 +0200 Subject: [PATCH 247/603] ASoC: dt-bindings: davinci-mcbsp: Add optional clock The McBSP uses an internal sample rate generator to provide bit clock or frame clock. This sample rate generator can be programmed to be driven by McBSP's internal clock source or by an external clock source (located on CLKS pin). The external clock source is not described in the bindings. Add an optional clock item that allows to select an external clock as sample rate generator's input. Signed-off-by: Bastien Curutchet Reviewed-by: Rob Herring Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-3-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml index 139b594dd192e..0c2b1936c6a12 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml +++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml @@ -50,12 +50,16 @@ properties: - const: tx clocks: + minItems: 1 items: - description: functional clock + - description: external input clock for sample rate generator. clock-names: + minItems: 1 items: - const: fck + - const: clks power-domains: maxItems: 1 From 904fb8f843a99ae7473d184412b6c7bd46a51593 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:03 +0200 Subject: [PATCH 248/603] ASoC: ti: davinci-i2s: Remove the unused clk_input_pin attribute The clk_input_pin attribute of davinci_mcbsp_dev struct is not set since commit 257ade78b601 ("ASoC: davinci-i2s: Convert to use edma-pcm"). Remove the attribute. Keep the behaviour of the MCBSP_CLKR case as MCBSP_CLKR == 0. I can't test the BC_FP format so I added back the initial comment that was removed by commit ec6375533748 ("ASoC: DaVinci: Added selection of clk input pin for McBSP"). This was the last dependency to linux/platform_data/davinci_asp.h so it is not included anymore. Remove the enum mcbsp_clk_input_pin from davinci_asp.h as it is not used anywhere else. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-4-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- include/linux/platform_data/davinci_asp.h | 15 -------------- sound/soc/ti/davinci-i2s.c | 24 ++++------------------- 2 files changed, 4 insertions(+), 35 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index c8645b2ed3c02..b9c8520b4bd3d 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -25,16 +25,6 @@ struct davinci_mcasp_pdata { unsigned sram_size_capture; struct gen_pool *sram_pool; - /* - * If McBSP peripheral gets the clock from an external pin, - * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR - * and MCBSP_CLKS. - * Depending on different hardware connections it is possible - * to use this setting to change the behaviour of McBSP - * driver. - */ - int clk_input_pin; - /* * This flag works when both clock and FS are outputs for the cpu * and makes clock more accurate (FS is not symmetrical and the @@ -91,11 +81,6 @@ enum { MCASP_VERSION_OMAP, /* OMAP4/5 */ }; -enum mcbsp_clk_input_pin { - MCBSP_CLKR = 0, /* as in DM365 */ - MCBSP_CLKS, -}; - #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 07c8b2259208d..5c906641640eb 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include @@ -159,7 +158,6 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; - int clk_input_pin; bool i2s_accurate_sck; }; @@ -239,26 +237,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_BC_FP: - pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; /* - * Selection of the clock input pin that is the - * input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate - * Generator. + * McBSP CLKR pin is the input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - switch (dev->clk_input_pin) { - case MCBSP_CLKS: - pcr |= DAVINCI_MCBSP_PCR_CLKXM | - DAVINCI_MCBSP_PCR_CLKRM; - break; - case MCBSP_CLKR: - pcr |= DAVINCI_MCBSP_PCR_SCLKME; - break; - default: - dev_err(dev->dev, "bad clk_input_pin\n"); - return -EINVAL; - } - + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; case SND_SOC_DAIFMT_BC_FC: /* codec is master */ From 6b1517b30d6dc9442d92f0273726f1e7390eff2c Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:04 +0200 Subject: [PATCH 249/603] ASoC: ti: davinci-i2s: Replace dev_err with dev_err_probe In probe(), the dev_err() is used for every returned error. Replace dev_err() with dev_err_probe() where -EPROBE_DEFER can be returned. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-5-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 5c906641640eb..cd64f1384e18c 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -708,7 +708,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = edma_pcm_platform_register(&pdev->dev); if (ret) { - dev_err(&pdev->dev, "register PCM failed: %d\n", ret); + dev_err_probe(&pdev->dev, ret, "register PCM failed\n"); goto err_unregister_component; } From 714ffb8d36f94bdc6d576417b451e9c568c83894 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:05 +0200 Subject: [PATCH 250/603] ASoC: ti: davinci-i2s: Use external clock to drive sample rate generator McBSP's internal sample rate generator can be programed to be driven by its internal clock or by an external clock source located on CLKS pin. The external clock source case is not handled by the driver. Handle an optional clock related to this external clock source. If present, the driver uses the clock located on CLKS pin as input for the sample rate generator. Thus, the external clock rate is used to compute divisors. If this optional clock is not present, the sample rate generator is driven by the McBSP's functional clock. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-6-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 65 ++++++++++++++++++++++++++++---------- 1 file changed, 49 insertions(+), 16 deletions(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index cd64f1384e18c..578b4ae28b710 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -134,6 +134,7 @@ struct davinci_mcbsp_dev { int mode; u32 pcr; struct clk *clk; + struct clk *ext_clk; /* * Combining both channels into 1 element will at least double the * amount of time between servicing the dma channel, increase @@ -364,7 +365,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); struct snd_interval *i = NULL; int mcbsp_word_length, master; - unsigned int rcr, xcr, srgr, clk_div, freq, framesize; + unsigned int rcr, xcr, clk_div, freq, framesize; + unsigned int srgr = 0; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; @@ -385,9 +387,13 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, switch (master) { case SND_SOC_DAIFMT_BP_FP: - freq = clk_get_rate(dev->clk); - srgr = DAVINCI_MCBSP_SRGR_FSGM | - DAVINCI_MCBSP_SRGR_CLKSM; + if (dev->ext_clk) { + freq = clk_get_rate(dev->ext_clk); + } else { + freq = clk_get_rate(dev->clk); + srgr = DAVINCI_MCBSP_SRGR_CLKSM; + } + srgr |= DAVINCI_MCBSP_SRGR_FSGM; srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); if (dev->i2s_accurate_sck) { @@ -691,12 +697,36 @@ static int davinci_i2s_probe(struct platform_device *pdev) return -ENODEV; } - dev->clk = clk_get(&pdev->dev, NULL); + /* + * The optional is there for backward compatibility. + * If 'fck' is not present, the clk_get(dev, NULL) that follows may find something + */ + dev->clk = devm_clk_get_optional(&pdev->dev, "fck"); if (IS_ERR(dev->clk)) - return -ENODEV; - ret = clk_enable(dev->clk); + return dev_err_probe(&pdev->dev, PTR_ERR(dev->clk), "Invalid functional clock\n"); + if (!dev->clk) { + dev->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(dev->clk), + "Missing functional clock\n"); + } + + dev->ext_clk = devm_clk_get_optional(&pdev->dev, "clks"); + if (IS_ERR(dev->ext_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(dev->ext_clk), "Invalid external clock\n"); + + ret = clk_prepare_enable(dev->clk); if (ret) - goto err_put_clk; + return ret; + + if (dev->ext_clk) { + dev_dbg(&pdev->dev, "External clock used for sample rate generator\n"); + ret = clk_prepare_enable(dev->ext_clk); + if (ret) { + dev_err_probe(&pdev->dev, ret, "Failed to enable external clock\n"); + goto err_disable_clk; + } + } dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); @@ -704,7 +734,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component, &davinci_i2s_dai, 1); if (ret != 0) - goto err_release_clk; + goto err_disable_ext_clk; ret = edma_pcm_platform_register(&pdev->dev); if (ret) { @@ -716,10 +746,12 @@ static int davinci_i2s_probe(struct platform_device *pdev) err_unregister_component: snd_soc_unregister_component(&pdev->dev); -err_release_clk: - clk_disable(dev->clk); -err_put_clk: - clk_put(dev->clk); +err_disable_ext_clk: + if (dev->ext_clk) + clk_disable_unprepare(dev->ext_clk); +err_disable_clk: + clk_disable_unprepare(dev->clk); + return ret; } @@ -729,9 +761,10 @@ static void davinci_i2s_remove(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); - clk_disable(dev->clk); - clk_put(dev->clk); - dev->clk = NULL; + clk_disable_unprepare(dev->clk); + + if (dev->ext_clk) + clk_disable_unprepare(dev->ext_clk); } static const struct of_device_id davinci_i2s_match[] __maybe_unused = { From 7dd7a6d2648b0b253cb8be3cdf8e895a995548fe Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:06 +0200 Subject: [PATCH 251/603] ASoC: ti: davinci-i2s: Delete unnecessary assignment In davinci_i2s_hw_params(), mcbsp_word_length is set twice to asp_word_length[fmt]. Remove second unnecessary assignment. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-7-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 578b4ae28b710..a2c7c812ea508 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -479,7 +479,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } - mcbsp_word_length = asp_word_length[fmt]; switch (master) { case SND_SOC_DAIFMT_BP_FP: From 37e313cda35aa16623ccae630530651c786a1392 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:07 +0200 Subject: [PATCH 252/603] ASoC: ti: davinci-i2s: Add TDM support TDM is not supported by the McBSP driver. The McBSP datasheet does not name explicitly TDM as a supported format but it is possible to configure the McBSP to do TDM if all slots are used by McBSP. Add TDM support. It uses single-phase frame. Slot width is used to compute the McBSP's word length. Implement the set_tdm_slot() hook of snd_soc_dai_ops struct. It only supports TDM if all slots are used by McBSP. The snd_soc_dai_driver's channels_max is updated from 2 to 128. This was tested with BP_FC format on a platform designed off of DAVINCI/OMAP_L138. A check is done in davinci_i2s_set_dai_fmt() to prevent TDM to be used with BC_FC and BC_FP formats. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-8-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 92 +++++++++++++++++++++++++++++++++++--- 1 file changed, 87 insertions(+), 5 deletions(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index a2c7c812ea508..f9a67c2bc2f55 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -160,6 +160,9 @@ struct davinci_mcbsp_dev { unsigned int fmt; int clk_div; bool i2s_accurate_sck; + + int tdm_slots; + int slot_width; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -213,6 +216,63 @@ static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback) toggle_clock(dev, playback); } +static int davinci_i2s_tdm_word_length(int tdm_slot_width) +{ + switch (tdm_slot_width) { + case 8: + return DAVINCI_MCBSP_WORD_8; + case 12: + return DAVINCI_MCBSP_WORD_12; + case 16: + return DAVINCI_MCBSP_WORD_16; + case 20: + return DAVINCI_MCBSP_WORD_20; + case 24: + return DAVINCI_MCBSP_WORD_24; + case 32: + return DAVINCI_MCBSP_WORD_32; + default: + return -EINVAL; + } +} + +static int davinci_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + + dev_dbg(dev->dev, "slots %d, slot_width %d\n", slots, slot_width); + + if (slots > 128 || !slots) { + dev_err(dev->dev, "Invalid number of slots\n"); + return -EINVAL; + } + + if (rx_mask != (1 << slots) - 1) { + dev_err(dev->dev, "Invalid RX mask (0x%08x) : all slots must be used by McBSP\n", + rx_mask); + return -EINVAL; + } + + if (tx_mask != (1 << slots) - 1) { + dev_err(dev->dev, "Invalid TX mask (0x%08x) : all slots must be used by McBSP\n", + tx_mask); + return -EINVAL; + } + + if (davinci_i2s_tdm_word_length(slot_width) < 0) { + dev_err(dev->dev, "%s: Unsupported slot_width %d\n", __func__, slot_width); + return -EINVAL; + } + + dev->tdm_slots = slots; + dev->slot_width = slot_width; + + return 0; +} + #define DEFAULT_BITPERSAMPLE 16 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -238,6 +298,11 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_BC_FP: + if (dev->tdm_slots || dev->slot_width) { + dev_err(dev->dev, "TDM is not supported for BC_FP format\n"); + return -EINVAL; + } + /* * McBSP CLKR pin is the input for the Sample Rate Generator. * McBSP FSR and FSX are driven by the Sample Rate Generator. @@ -246,6 +311,11 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; case SND_SOC_DAIFMT_BC_FC: + if (dev->tdm_slots || dev->slot_width) { + dev_err(dev->dev, "TDM is not supported for BC_FC format\n"); + return -EINVAL; + } + /* codec is master */ pcr = 0; break; @@ -383,7 +453,13 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, master = dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; fmt = params_format(params); - mcbsp_word_length = asp_word_length[fmt]; + if (dev->slot_width) + mcbsp_word_length = davinci_i2s_tdm_word_length(dev->slot_width); + else + mcbsp_word_length = asp_word_length[fmt]; + + if (mcbsp_word_length < 0) + return mcbsp_word_length; switch (master) { case SND_SOC_DAIFMT_BP_FP: @@ -483,8 +559,13 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, switch (master) { case SND_SOC_DAIFMT_BP_FP: case SND_SOC_DAIFMT_BP_FC: - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + if (dev->tdm_slots > 0) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(dev->tdm_slots - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(dev->tdm_slots - 1); + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } break; case SND_SOC_DAIFMT_BC_FC: case SND_SOC_DAIFMT_BC_FP: @@ -609,19 +690,20 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, .set_clkdiv = davinci_i2s_dai_set_clkdiv, + .set_tdm_slot = davinci_i2s_set_tdm_slot, }; static struct snd_soc_dai_driver davinci_i2s_dai = { .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 128, .rates = DAVINCI_I2S_RATES, .formats = DAVINCI_I2S_FORMATS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 128, .rates = DAVINCI_I2S_RATES, .formats = DAVINCI_I2S_FORMATS, }, From eff21f5f8ea01834835ebe35995dba40f8435795 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:08 +0200 Subject: [PATCH 253/603] ASoC: ti: davinci-i2s: Add handling of BP_FC format McBSP is able to drive bit clock and consume frame clock but BP_FC format is not handled by McBSP driver. Add BP_FC format support. When BP_FC is selected: - CLKX and CLKR are configured as outputs - The sample rate generator is configured to be able to provide bit clock. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-9-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index f9a67c2bc2f55..e51f05cda9410 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -310,6 +310,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; + case SND_SOC_DAIFMT_BP_FC: + /* cpu is bitclock provider */ + pcr = DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + break; + case SND_SOC_DAIFMT_BC_FC: if (dev->tdm_slots || dev->slot_width) { dev_err(dev->dev, "TDM is not supported for BC_FC format\n"); @@ -500,6 +506,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, clk_div &= 0xFF; srgr |= clk_div; break; + case SND_SOC_DAIFMT_BP_FC: + if (dev->ext_clk) { + freq = clk_get_rate(dev->ext_clk); + } else { + freq = clk_get_rate(dev->clk); + srgr = DAVINCI_MCBSP_SRGR_CLKSM; + } + if (dev->tdm_slots && dev->slot_width) { + clk_div = freq / (params->rate_num * params->rate_den) + / (dev->tdm_slots * dev->slot_width) - 1; + } else { + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + } + clk_div &= 0xFF; + srgr |= clk_div; + break; case SND_SOC_DAIFMT_BC_FC: /* Clock and frame sync given from external sources */ i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); From 94d57c541dbdd350a91baeee94d3f5148e1d4dd7 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:09 +0200 Subject: [PATCH 254/603] ASoC: ti: davinci-i2s: Enable unexpected frame pulses detection McBSP can generate a SYNCERR when unexpected frame pulses are detected. The driver always disables this feature and ignore the unexpected frame pulses. Enable the generation of SYNCERR by the McBSP. Unexpected frame pulses are not ignored anymore. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-10-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index e51f05cda9410..82a0dfa07cec4 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -441,8 +441,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); struct snd_interval *i = NULL; int mcbsp_word_length, master; - unsigned int rcr, xcr, clk_div, freq, framesize; + unsigned int clk_div, freq, framesize; unsigned int srgr = 0; + unsigned int rcr = 0; + unsigned int xcr = 0; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; @@ -539,8 +541,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); - rcr = DAVINCI_MCBSP_RCR_RFIG; - xcr = DAVINCI_MCBSP_XCR_XFIG; if (dev->mode == MOD_DSP_B) { rcr |= DAVINCI_MCBSP_RCR_RDATDLY(0); xcr |= DAVINCI_MCBSP_XCR_XDATDLY(0); From 091b440ffd7cb542fd45c39dddd56bd870f9e180 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:10 +0200 Subject: [PATCH 255/603] ASoC: ti: davinci-i2s: Link free-run mode to SND_SOC_DAIFMT_[GATED/CONT] McBSP has free-running mode where serial clocks continue to run during emulation halts. This mode is always enabled by the driver. Set free-running mode when SND_SOC_DAIFMT_CONT is selected by DAI format, unset it when SND_SOC_DAIFMT_GATED is selected. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-11-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 82a0dfa07cec4..7fb41987175b5 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -280,6 +280,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); unsigned int pcr; + unsigned int spcr; unsigned int srgr; bool inv_fs = false; /* Attention srgr is updated by hw_params! */ @@ -288,6 +289,23 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); dev->fmt = fmt; + + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + spcr |= DAVINCI_MCBSP_SPCR_FREE; + dev_dbg(dev->dev, "Free-running mode ON\n"); + break; + case SND_SOC_DAIFMT_GATED: + spcr &= ~DAVINCI_MCBSP_SPCR_FREE; + dev_dbg(dev->dev, "Free-running mode OFF\n"); + break; + default: + dev_err(dev->dev, "Invalid clock gating\n"); + return -EINVAL; + } + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { case SND_SOC_DAIFMT_BP_FP: @@ -452,10 +470,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - spcr |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + spcr |= DAVINCI_MCBSP_SPCR_RINTM(3); davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } else { - spcr |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + spcr |= DAVINCI_MCBSP_SPCR_XINTM(3); davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } From 92e7bb2b6aa374c130dcf052f2c52f63c5b75d38 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:11 +0200 Subject: [PATCH 256/603] ASoC: ti: davinci-i2s: Add S24_LE to supported formats S24_LE is supported by McBSP but not by the driver. Add S24_LE to driver's supported formats. Using it enables the sign extension in DRR (Data Receive Register). The other formats are kept with the zero extension in DRR. Remove data_type table as it is no longer used. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-12-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 34 +++++++++++++++++++++------------- 1 file changed, 21 insertions(+), 13 deletions(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 7fb41987175b5..fb1e09c78bdfd 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -61,6 +61,9 @@ #define DAVINCI_MCBSP_SPCR_RRST (1 << 0) #define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4) +#define DAVINCI_MCBSP_SPCR_RJUST(v) ((v) << 13) +#define DAVINCI_MCBSP_SPCR_RJUST_Z_LE DAVINCI_MCBSP_SPCR_RJUST(0) +#define DAVINCI_MCBSP_SPCR_RJUST_S_LE DAVINCI_MCBSP_SPCR_RJUST(1) #define DAVINCI_MCBSP_SPCR_XRST (1 << 16) #define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20) #define DAVINCI_MCBSP_SPCR_GRST (1 << 22) @@ -107,15 +110,10 @@ enum { DAVINCI_MCBSP_WORD_32, }; -static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { - [SNDRV_PCM_FORMAT_S8] = 1, - [SNDRV_PCM_FORMAT_S16_LE] = 2, - [SNDRV_PCM_FORMAT_S32_LE] = 4, -}; - static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S24_LE] = DAVINCI_MCBSP_WORD_24, [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, }; @@ -467,8 +465,23 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, snd_pcm_format_t fmt; unsigned element_cnt = 1; - /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + + /* Determine xfer data type */ + fmt = params_format(params); + switch (fmt) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S32_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + spcr |= DAVINCI_MCBSP_SPCR_RJUST_S_LE; + break; + default: + dev_warn(dev->dev, "davinci-i2s: unsupported PCM format\n"); + return -EINVAL; + } + + /* general line settings */ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { spcr |= DAVINCI_MCBSP_SPCR_RINTM(3); davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); @@ -566,12 +579,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1); xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } - /* Determine xfer data type */ - fmt = params_format(params); - if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { - printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); - return -EINVAL; - } if (params_channels(params) == 2) { element_cnt = 2; @@ -710,6 +717,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 #define DAVINCI_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) static int davinci_i2s_dai_probe(struct snd_soc_dai *dai) From 609302ca04a3177463b0fbf4d5fc55a3ab4f900d Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:12 +0200 Subject: [PATCH 257/603] ASoC: dt-bindings: davinci-mcbsp: Add the 'ti,T1-framing-{rx/tx}' flags McBSP's data delay can be configured from 0 to 2 bit clock periods. 0 is used for DSP_B format, 1 for DSP_A format. A data delay of 2 bit clock periods can be used to interface to 'T1 framing' devices where data stream is preceded by a 'framing bit'. This 2 bit clock data delay is not described in the bindings. Add two flags 'ti,T1-framing-[rx/tx]' to enable a data delay of 2 bit clock periods in reception or transmission. Signed-off-by: Bastien Curutchet Reviewed-by: Rob Herring Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-13-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/davinci-mcbsp.yaml | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml index 0c2b1936c6a12..4fa677023827f 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml +++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml @@ -67,6 +67,20 @@ properties: "#sound-dai-cells": const: 0 + ti,T1-framing-tx: + $ref: /schemas/types.yaml#/definitions/flag + description: + If the property is present, tx data delay is set to 2 bit clock periods. + McBSP will insert a blank period (high-impedance period) before the first + data bit. This can be used to interface to T1-framing devices. + + ti,T1-framing-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: + If the property is present, rx data delay is set to 2 bit clock periods. + McBSP will discard the bit preceding the data stream (called framing bit). + This can be used to interface to T1-framing devices. + required: - "#sound-dai-cells" - compatible From 08e02fa48429c34db231cc3b58b940de2f7caf35 Mon Sep 17 00:00:00 2001 From: Bastien Curutchet Date: Tue, 2 Apr 2024 09:12:13 +0200 Subject: [PATCH 258/603] ASoC: ti: davinci-i2s: Add T1 framing support McBSP's data delay can be configured from 0 to 2 bit clock periods. 0 is used for DSP_B format, 1 is used for DSP_A format, 2 is unused. A data delay of 2 bit clock periods can be used to interface to 'T1 framing' devices where data stream is preceded by a 'framing bit'. On transmission, McBSP inserts a blank period (high-impedance period) before the first data bit to leave an opportunity for other devices to set this 'framing bit'. On reception, McBSP discards the 'framing bit' that precedes the data stream. Add support for the 'framing bit' according to the 'ti,T1-framing-[tx/rx]' device-tree properties. If a flag is present, the data delay is set to 2 bit clock periods regardless of the selected DAI format. Signed-off-by: Bastien Curutchet Acked-by: Peter Ujfalusi Link: https://msgid.link/r/20240402071213.11671-14-bastien.curutchet@bootlin.com Signed-off-by: Mark Brown --- sound/soc/ti/davinci-i2s.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index fb1e09c78bdfd..0f15a743c7982 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -161,6 +161,9 @@ struct davinci_mcbsp_dev { int tdm_slots; int slot_width; + + bool tx_framing_bit; + bool rx_framing_bit; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -580,6 +583,15 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } + if (dev->tx_framing_bit) { + xcr &= ~DAVINCI_MCBSP_XCR_XDATDLY(1); + xcr |= DAVINCI_MCBSP_XCR_XDATDLY(2); + } + if (dev->rx_framing_bit) { + rcr &= ~DAVINCI_MCBSP_RCR_RDATDLY(1); + rcr |= DAVINCI_MCBSP_RCR_RDATDLY(2); + } + if (params_channels(params) == 2) { element_cnt = 2; if (double_fmt[fmt] && dev->enable_channel_combine) { @@ -796,6 +808,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->base = io_base; + dev->tx_framing_bit = of_property_read_bool(pdev->dev.of_node, "ti,T1-framing-tx"); + dev->rx_framing_bit = of_property_read_bool(pdev->dev.of_node, "ti,T1-framing-rx"); + /* setup DMA, first TX, then RX */ dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); From 56437a561fefab2be708dfebf80e31c4715eb52e Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 8 Apr 2024 23:32:26 +0530 Subject: [PATCH 259/603] ASoC: amd: fix for soundwire build dependencies for legacy stack The SND_SOC_AMD_SOUNDWIRE Kconfig symbol has build dependency on SOUNDWIRE_AMD. It gets it wrong for a configuration involving SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y,SND_SOC_AMD_PS=y and SOUNDWIRE_AMD=m, which results in a link failure: ld: vmlinux.o: in function `amd_sdw_probe': >> sound/soc/amd/ps/pci-ps.c:271:(.text+0x1d51eff): undefined reference to `sdw_amd_probe' ld: vmlinux.o: in function `acp63_sdw_machine_select': >> sound/soc/amd/ps/pci-ps.c:294:(.text+0x1d525d5): undefined reference to `sdw_amd_get_slave_info' ld: vmlinux.o: in function `amd_sdw_exit': >> sound/soc/amd/ps/pci-ps.c:280:(.text+0x1d538ce): undefined reference to `sdw_amd_exit' Add a top level check config that forbids any of the AMD ACP drivers with version >= 6.3 from being built-in with CONFIG_SOUNDWIRE_AMD=m. Move SND_AMD_ACP_CONFIG common dependency config to SND_SOC_AMD_ACP63_TOPLEVEL config. Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202404061257.khJml82D-lkp@intel.com/ Fixes: a806793f231e ("ASoC: amd: simplify soundwire dependencies for legacy stack") Signed-off-by: Vijendar Mukunda Link: https://msgid.link/r/20240408180229.3287220-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 3508f5a96b751..6dec44f516c13 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -132,6 +132,19 @@ config SND_SOC_AMD_RPL_ACP6x Say m if you have such a device. If unsure select "N". +config SND_SOC_AMD_ACP63_TOPLEVEL + tristate "support for AMD platforms with ACP version >= 6.3" + default SND_AMD_ACP_CONFIG + depends on SND_AMD_ACP_CONFIG + depends on SOUNDWIRE_AMD || !SOUNDWIRE_AMD + depends on X86 || COMPILE_TEST + help + This adds support for AMD platforms with ACP version >= 6.3. + Say Y if you have such a device. + If unsure select "N". + +if SND_SOC_AMD_ACP63_TOPLEVEL + config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE tristate select SND_AMD_SOUNDWIRE_ACPI if ACPI @@ -149,7 +162,6 @@ config SND_SOC_AMD_SOUNDWIRE config SND_SOC_AMD_PS tristate "AMD Audio Coprocessor-v6.3 Pink Sardine support" - select SND_AMD_ACP_CONFIG select SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on X86 && PCI && ACPI help @@ -170,3 +182,5 @@ config SND_SOC_AMD_PS_MACH DMIC can be connected directly to ACP IP. Say m if you have such a device. If unsure select "N". + +endif From f78bf2c933c9cb3b61215378664f83c5abd25374 Mon Sep 17 00:00:00 2001 From: Dragan Simic Date: Tue, 9 Apr 2024 19:56:36 +0200 Subject: [PATCH 260/603] ASoC: pcm: perform power-down delay checks a bit faster When checking whether the power-down delay should be ignored for a specific PCM runtime, there's no need to keep going through all DAI link components after any of them is found to be configured to use the power-down delay. While there, fix a small typo in one of the comment blocks. Signed-off-by: Dragan Simic Link: https://msgid.link/r/90ae761a5b99640ece48363a7099ac2cf402bd37.1712684592.git.dsimic@manjaro.org Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 77ee103b7cd1a..b0e1bd7f588ba 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -315,23 +315,24 @@ EXPORT_SYMBOL_GPL(snd_soc_runtime_action); * @rtd: The ASoC PCM runtime that should be checked. * * This function checks whether the power down delay should be ignored for a - * specific PCM runtime. Returns true if the delay is 0, if it the DAI link has + * specific PCM runtime. Returns true if the delay is 0, if the DAI link has * been configured to ignore the delay, or if none of the components benefits * from having the delay. */ bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component; - bool ignore = true; int i; if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) return true; for_each_rtd_components(rtd, i, component) - ignore &= !component->driver->use_pmdown_time; + if (component->driver->use_pmdown_time) + /* No need to go through all components */ + return false; - return ignore; + return true; } /** From 4693b120299a5893034d1882ddbbe08af1d2ff07 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 10 Apr 2024 17:13:12 +0100 Subject: [PATCH 261/603] ASoC: wm_adsp: Include array_size.h Explicitly #include array_size.h for the ARRAY_SIZE() macro. Signed-off-by: Richard Fitzgerald Link: https://msgid.link/r/20240410161312.22313-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6e348d49a89ce..c9d9a7b28efb0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -7,6 +7,7 @@ * Author: Mark Brown */ +#include #include #include #include From 28d31ffac3d3ef6d60c4eb392a47d5e19dbb15e8 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 10 Apr 2024 17:08:33 +0100 Subject: [PATCH 262/603] ASoC: cs35l56: Include array_size.h Explicitly #include array_size.h for the source files that use ARRAY_SIZE(). Signed-off-by: Richard Fitzgerald Link: https://msgid.link/r/20240410160833.20837-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-shared.c | 1 + sound/soc/codecs/cs35l56.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 08cac58e3ab22..ac2a760aeae14 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -5,6 +5,7 @@ // Copyright (C) 2023 Cirrus Logic, Inc. and // Cirrus Logic International Semiconductor Ltd. +#include #include #include #include diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 8d2f021fb3628..5a0b55e31e3c8 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -6,6 +6,7 @@ // Cirrus Logic International Semiconductor Ltd. #include +#include #include #include #include From 1d165c5a1c85747bf48f37a80636a4a73e1576a1 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 11 Apr 2024 09:33:32 +0100 Subject: [PATCH 263/603] ASoC: codecs: tas2780: remove redundant assignments to variable ret Variable ret is being assigned a value that is never read in a couple of places. The variable is being re-assigned later on. The assignments are redundant and can be removed. Cleans up clang scan build warning: sound/soc/codecs/tas2780.c:84:2: warning: Value stored to 'ret' is never read [deadcode.DeadStores] Signed-off-by: Colin Ian King Link: https://msgid.link/r/20240411083332.304887-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2780.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c index 41076be238542..972e8ea5ebde7 100644 --- a/sound/soc/codecs/tas2780.c +++ b/sound/soc/codecs/tas2780.c @@ -71,7 +71,7 @@ static int tas2780_codec_resume(struct snd_soc_component *component) { struct tas2780_priv *tas2780 = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret; ret = snd_soc_component_update_bits(component, TAS2780_PWR_CTRL, TAS2780_PWR_CTRL_MASK, TAS2780_PWR_CTRL_ACTIVE); @@ -81,7 +81,6 @@ static int tas2780_codec_resume(struct snd_soc_component *component) __func__, ret); goto err; } - ret = 0; regcache_cache_only(tas2780->regmap, false); ret = regcache_sync(tas2780->regmap); err: From 62bef5df35a87d0f8464b1a8837540988cfaf5db Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 11 Apr 2024 18:11:44 +0800 Subject: [PATCH 264/603] ASoC: dt-bindings: imx-audio-spdif: convert to YAML Convert the imx-audio-spdif binding to YAML. When testing dtbs_check, found below compatible strings are not listed in document: fsl,imx-sabreauto-spdif fsl,imx6sx-sdb-spdif So add them in yaml file to pass the test. Signed-off-by: Shengjiu Wang Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/1712830305-31350-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- .../bindings/sound/fsl,imx-audio-spdif.yaml | 66 +++++++++++++++++++ .../bindings/sound/imx-audio-spdif.txt | 36 ---------- 2 files changed, 66 insertions(+), 36 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml new file mode 100644 index 0000000000000..5fc543d02ecb4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale i.MX audio complex with S/PDIF transceiver + +maintainers: + - Shengjiu Wang + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx-sabreauto-spdif + - fsl,imx6sx-sdb-spdif + - const: fsl,imx-audio-spdif + - enum: + - fsl,imx-audio-spdif + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + spdif-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the i.MX S/PDIF controller + + spdif-out: + type: boolean + description: + If present, the transmitting function of S/PDIF will be enabled, + indicating there's a physical S/PDIF out connector or jack on the + board or it's connecting to some other IP block, such as an HDMI + encoder or display-controller. + + spdif-in: + type: boolean + description: + If present, the receiving function of S/PDIF will be enabled, + indicating there is a physical S/PDIF in connector/jack on the board. + +required: + - compatible + - model + - spdif-controller + +anyOf: + - required: + - spdif-in + - required: + - spdif-out + +additionalProperties: false + +examples: + - | + sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; + }; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt deleted file mode 100644 index da84a442ccea5..0000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt +++ /dev/null @@ -1,36 +0,0 @@ -Freescale i.MX audio complex with S/PDIF transceiver - -Required properties: - - - compatible : "fsl,imx-audio-spdif" - - - model : The user-visible name of this sound complex - - - spdif-controller : The phandle of the i.MX S/PDIF controller - - -Optional properties: - - - spdif-out : This is a boolean property. If present, the - transmitting function of S/PDIF will be enabled, - indicating there's a physical S/PDIF out connector - or jack on the board or it's connecting to some - other IP block, such as an HDMI encoder or - display-controller. - - - spdif-in : This is a boolean property. If present, the receiving - function of S/PDIF will be enabled, indicating there - is a physical S/PDIF in connector/jack on the board. - -* Note: At least one of these two properties should be set in the DT binding. - - -Example: - -sound-spdif { - compatible = "fsl,imx-audio-spdif"; - model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; -}; From 7fda0efaa5fe6e93bb79a3a540a4b881ef788f66 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 11 Apr 2024 17:03:36 -0500 Subject: [PATCH 265/603] ASoC: Intel: sof_sdw: add missing sof_sdw_rt_amp_init for Realtek multi-function codecs We do need "info->amp_num++;" for the speaker dai. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b94835448b1b4..8a2cd19d8be6a 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -737,6 +737,8 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-aif2", .dai_type = SOF_SDW_DAI_TYPE_AMP, .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .init = sof_sdw_rt_amp_init, + .exit = sof_sdw_rt_amp_exit, .rtd_init = rt712_spk_rtd_init, }, }, @@ -912,6 +914,8 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_AMP, /* No feedback capability is provided by rt722-sdca codec driver*/ .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .init = sof_sdw_rt_amp_init, + .exit = sof_sdw_rt_amp_exit, .rtd_init = rt722_spk_rtd_init, }, { From 0f8edb15fb6e436f0da7ab25ffcbcaab3def7e8c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 11 Apr 2024 17:03:37 -0500 Subject: [PATCH 266/603] ASoC: Intel: soc-acpi: add support for HP Omen14 SoundWire configuration This platform has an RT711-sdca on link0 and RT1316 on link3 Closes: https://github.com/thesofproject/linux/issues/4880 Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 29 +++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 75935b454e5df..264f5c19af56f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -296,6 +296,15 @@ static const struct snd_soc_acpi_adr_device rt1316_2_group2_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1316_3_single_adr[] = { + { + .adr = 0x000330025D131601ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt1316-1" + } +}; + static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = { { .adr = 0x000130025D131801ull, @@ -562,6 +571,20 @@ static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1316_l12[] = { {} }; +static const struct snd_soc_acpi_link_adr mtl_rt711_l0_rt1316_l3[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt1316_3_single_adr), + .adr_d = rt1316_3_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_adr_device mx8363_2_adr[] = { { .adr = 0x000230019F836300ull, @@ -703,6 +726,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt711-l0-rt1316-l23-rt714-l1.tplg", }, + { + .link_mask = 0x9, /* 2 active links required */ + .links = mtl_rt711_l0_rt1316_l3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt711-l0-rt1316-l3.tplg", + }, { .link_mask = BIT(0), .links = mtl_rt722_only, From 4fee07fbf47d2a5f1065d985459e5ce7bf7969f0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 11 Apr 2024 17:03:38 -0500 Subject: [PATCH 267/603] ASoC: Intel: sof_sdw: add JD2 quirk for HP Omen 14 The default JD1 does not seem to work, use JD2 instead. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 8a2cd19d8be6a..f191dc72bc3d1 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -498,6 +498,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(1) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "OMEN Transcend Gaming Laptop"), + }, + .driver_data = (void *)(RT711_JD2), + }, + /* LunarLake devices */ { .callback = sof_sdw_quirk_cb, From b10cb955c6c0b8dbd9a768166d71cc12680b7fdf Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 11 Apr 2024 17:03:39 -0500 Subject: [PATCH 268/603] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0C0F The JD1 jack detection doesn't seem to work, use JD2. Also use the 4 speaker configuration. Link: https://github.com/thesofproject/linux/issues/4900 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20240411220347.131267-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f191dc72bc3d1..6253fc4641981 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -432,6 +432,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C0F") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { From 744866d28fe6b1a651e63d03a57e06d66e3d460a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 11 Apr 2024 17:03:40 -0500 Subject: [PATCH 269/603] ASoC: Intel: sof_sdw: remove FOUR_SPEAKER quirks This patch removes a confusion between speakers and amplifiers. The existing code keeps track of the number of amplifiers and reports it in the 'cfg-amp' component string. The number of speakers is defined with quirks, but those quirks are not consistently added: in the MeteorLake case, none of the supported platforms used such a quirk, and UCM does not use the values reported anyways. The notion of 'FOUR_SPEAKERS' is also obsolete now with some platforms having more than four speakers. Let's just remove all this and only report the number of amplifiers. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20240411220347.131267-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 68 ++++++++----------------- sound/soc/intel/boards/sof_sdw_common.h | 1 + 2 files changed, 23 insertions(+), 46 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6253fc4641981..d65c5da49000a 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -27,7 +27,7 @@ static void log_quirks(struct device *dev) dev_dbg(dev, "quirk realtek,jack-detect-source %ld\n", SOF_JACK_JDSRC(sof_sdw_quirk)); if (sof_sdw_quirk & SOF_SDW_FOUR_SPK) - dev_dbg(dev, "quirk SOF_SDW_FOUR_SPK enabled\n"); + dev_err(dev, "quirk SOF_SDW_FOUR_SPK enabled but no longer supported\n"); if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) dev_dbg(dev, "quirk SOF_SDW_TGL_HDMI enabled\n"); if (sof_sdw_quirk & SOF_SDW_PCH_DMIC) @@ -80,8 +80,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "098F"), }, - .driver_data = (void *)(RT711_JD2 | - SOF_SDW_FOUR_SPK), + .driver_data = (void *)(RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -89,8 +88,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0990"), }, - .driver_data = (void *)(RT711_JD2 | - SOF_SDW_FOUR_SPK), + .driver_data = (void *)(RT711_JD2), }, /* IceLake devices */ { @@ -141,8 +139,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5D") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -151,8 +148,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A5E") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -162,7 +158,6 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | - SOF_SDW_FOUR_SPK | SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, @@ -173,8 +168,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Ripto"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | - SOF_SDW_FOUR_SPK), + SOF_SDW_PCH_DMIC), }, { /* @@ -258,8 +252,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A32") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -290,7 +283,6 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | SOF_SDW_PCH_DMIC | - SOF_SDW_FOUR_SPK | SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, @@ -301,8 +293,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AF0") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -311,8 +302,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AF3"), }, /* No Jack */ - .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_FOUR_SPK), + .driver_data = (void *)(SOF_SDW_TGL_HDMI), }, { .callback = sof_sdw_quirk_cb, @@ -321,8 +311,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AFE") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -331,8 +320,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0AFF") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -341,8 +329,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B00") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -351,8 +338,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B01") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -361,8 +347,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B11") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -371,8 +356,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B12") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -400,8 +384,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B29"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -429,8 +412,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0BDA") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -449,8 +431,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C10"), }, /* No Jack */ - .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_FOUR_SPK), + .driver_data = (void *)(SOF_SDW_TGL_HDMI), }, { .callback = sof_sdw_quirk_cb, @@ -459,8 +440,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C11") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -469,8 +449,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C40") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, @@ -479,8 +458,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C4F") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, /* MeteorLake devices */ { @@ -2010,9 +1988,7 @@ static int mc_probe(struct platform_device *pdev) amp_num += codec_info_list[i].amp_num; card->components = devm_kasprintf(card->dev, GFP_KERNEL, - "cfg-spk:%d cfg-amp:%d", - (sof_sdw_quirk & SOF_SDW_FOUR_SPK) - ? 4 : 2, amp_num); + "cfg-amp:%d", amp_num); if (!card->components) return -ENOMEM; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 8468487a6bd6d..89253938ebaaa 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -45,6 +45,7 @@ enum { }; #define SOF_JACK_JDSRC(quirk) ((quirk) & GENMASK(3, 0)) +/* Deprecated and no longer supported by the code */ #define SOF_SDW_FOUR_SPK BIT(4) #define SOF_SDW_TGL_HDMI BIT(5) #define SOF_SDW_PCH_DMIC BIT(6) From 36a621070b4854670c43138bb8b19bdf8df77bbd Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 11 Apr 2024 17:03:41 -0500 Subject: [PATCH 270/603] ASoC: Intel: sof_da7219: mach cleanup for adl boards Add a common entry in enumeration table for all da7219 boards with/without speaker amplifier. All other adl_da7219_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-adl-match.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 1ea2d9c582ee1..4167b2e9bc6a7 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -453,11 +453,6 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; -static const struct snd_soc_acpi_codecs adl_max98360a_amp = { - .num_codecs = 1, - .codecs = {"MX98360A"} -}; - static const struct snd_soc_acpi_codecs adl_rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID}, @@ -510,13 +505,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, - { - .id = "DLGS7219", - .drv_name = "adl_da7219_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &adl_max98360a_amp, - .sof_tplg_filename = "sof-adl-max98360a-da7219.tplg", - }, /* place boards for each headphone codec: sof driver will complete the * tplg name and machine driver will detect the amp type */ @@ -527,6 +515,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, }, + { + .id = DA7219_ACPI_HID, + .drv_name = "adl_da7219_def", + .sof_tplg_filename = "sof-adl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, { .id = NAU8825_ACPI_HID, .drv_name = "adl_nau8825_def", From c3d1818f5ef5a9a930e468a799ccf1cade91e350 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 11 Apr 2024 17:03:42 -0500 Subject: [PATCH 271/603] ASoC: Intel: sof_da7219: mach cleanup for rpl boards Add a common entry in enumeration table for all da7219 boards with/without speaker amplifier. All other rpl_da7219_def entries become redundant so get removed. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-rpl-match.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 34588db6138af..b0a49e28ab092 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -361,11 +361,6 @@ static const struct snd_soc_acpi_codecs rpl_max98357a_amp = { .codecs = {"MX98357A"} }; -static const struct snd_soc_acpi_codecs rpl_max98360a_amp = { - .num_codecs = 1, - .codecs = {"MX98360A"}, -}; - static const struct snd_soc_acpi_codecs rpl_lt6911_hdmi = { .num_codecs = 1, .codecs = {"INTC10B0"} @@ -411,6 +406,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, }, + { + .id = DA7219_ACPI_HID, + .drv_name = "rpl_da7219_def", + .sof_tplg_filename = "sof-rpl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, { .id = NAU8825_ACPI_HID, .drv_name = "rpl_nau8825_def", @@ -438,13 +440,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { .drv_name = "rpl_lt6911_hdmi_ssp", .sof_tplg_filename = "sof-rpl-nocodec-hdmi-ssp02.tplg" }, - { - .id = "DLGS7219", - .drv_name = "rpl_da7219_def", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &rpl_max98360a_amp, - .sof_tplg_filename = "sof-rpl-max98360a-da7219.tplg", - }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines); From fe18a4be97d4064c9f3113f819780162cc586f16 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 11 Apr 2024 17:03:43 -0500 Subject: [PATCH 272/603] ASoC: Intel: sof_da7219: add mtl_da7219_def for mtl boards Add the board config mtl_da7219_def to da7219 machine driver for all mtl boards using default SSP port allocation (headphone codec on SSP2, speaker amplifiers on SSP0, and BT offload on SSP1). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 8 ++++++++ sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 7 +++++++ 2 files changed, 15 insertions(+) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index fd3a7be993c19..cbbce07ec5f72 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -398,6 +398,14 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_PORT_BT_OFFLOAD(2) | SOF_BT_OFFLOAD_PRESENT), }, + { + .name = "mtl_da7219_def", + .driver_data = (kernel_ulong_t)(SOF_DA7219_MCLK_EN | + SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(0) | + SOF_SSP_PORT_BT_OFFLOAD(1) | + SOF_BT_OFFLOAD_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 264f5c19af56f..f95490a16b551 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -76,6 +76,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, }, + { + .id = DA7219_ACPI_HID, + .drv_name = "mtl_da7219_def", + .sof_tplg_filename = "sof-mtl", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_AMP_NAME | + SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME, + }, { .id = NAU8825_ACPI_HID, .drv_name = "mtl_nau8825_def", From 551fb5593c2bd34e8711efe01dddf22d9c6200b2 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 11 Apr 2024 17:03:44 -0500 Subject: [PATCH 273/603] ASoC: Intel: sof_realtek_common: support 4xALC1011 amplifier Add support for boards with four ALC1011 amplifiers. Configuration is copied from cml_rt1011_rt5682 machine driver for backward compatibility with existing cml devices. Reviewed-by: Chao Song Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_realtek_common.c | 178 +++++++++++++++++--- sound/soc/intel/boards/sof_realtek_common.h | 4 +- sound/soc/intel/boards/sof_rt5682.c | 4 +- 3 files changed, 162 insertions(+), 24 deletions(-) diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index 80c8687cd1da5..20d094c52d283 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -15,10 +15,40 @@ #include "../../codecs/rt1011.h" #include "../../codecs/rt1015.h" #include "../../codecs/rt1308.h" +#include "../common/soc-intel-quirks.h" #include "sof_realtek_common.h" /* - * Current only 2-amp configuration is supported for rt1011 + * Common structures and functions + */ +static const struct snd_kcontrol_new realtek_4spk_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("WL Ext Spk"), + SOC_DAPM_PIN_SWITCH("WR Ext Spk"), + SOC_DAPM_PIN_SWITCH("TL Ext Spk"), + SOC_DAPM_PIN_SWITCH("TR Ext Spk"), +}; + +static const struct snd_soc_dapm_widget realtek_4spk_widgets[] = { + SND_SOC_DAPM_SPK("WL Ext Spk", NULL), + SND_SOC_DAPM_SPK("WR Ext Spk", NULL), + SND_SOC_DAPM_SPK("TL Ext Spk", NULL), + SND_SOC_DAPM_SPK("TR Ext Spk", NULL), +}; + +/* helper function to get the number of specific codec */ +static unsigned int get_num_codecs(const char *hid) +{ + struct acpi_device *adev; + unsigned int dev_num = 0; + + for_each_acpi_dev_match(adev, hid, NULL, -1) + dev_num++; + + return dev_num; +} + +/* + * Realtek ALC1011 */ static const struct snd_soc_dapm_route speaker_map_lr[] = { /* speaker */ @@ -26,16 +56,14 @@ static const struct snd_soc_dapm_route speaker_map_lr[] = { { "Right Spk", NULL, "Right SPO" }, }; -/* - * Make sure device's Unique ID follows this configuration: - * - * Two speakers: - * 0: left, 1: right - * Four speakers: - * 0: Woofer left, 1: Woofer right - * 2: Tweeter left, 3: Tweeter right - */ -static struct snd_soc_codec_conf rt1011_codec_confs[] = { +static const struct snd_soc_dapm_route rt1011_4spk_routes[] = { + {"WL Ext Spk", NULL, "WL SPO" }, + {"WR Ext Spk", NULL, "WR SPO" }, + {"TL Ext Spk", NULL, "TL SPO" }, + {"TR Ext Spk", NULL, "TR SPO" }, +}; + +static struct snd_soc_codec_conf rt1011_2spk_codec_confs[] = { { .dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME), .name_prefix = "Left", @@ -46,6 +74,25 @@ static struct snd_soc_codec_conf rt1011_codec_confs[] = { }, }; +static struct snd_soc_codec_conf rt1011_4spk_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(RT1011_DEV0_NAME), + .name_prefix = "WL", + }, + { + .dlc = COMP_CODEC_CONF(RT1011_DEV1_NAME), + .name_prefix = "WR", + }, + { + .dlc = COMP_CODEC_CONF(RT1011_DEV2_NAME), + .name_prefix = "TL", + }, + { + .dlc = COMP_CODEC_CONF(RT1011_DEV3_NAME), + .name_prefix = "TR", + }, +}; + static struct snd_soc_dai_link_component rt1011_dai_link_components[] = { { .name = RT1011_DEV0_NAME, @@ -55,6 +102,14 @@ static struct snd_soc_dai_link_component rt1011_dai_link_components[] = { .name = RT1011_DEV1_NAME, .dai_name = RT1011_CODEC_DAI, }, + { + .name = RT1011_DEV2_NAME, + .dai_name = RT1011_CODEC_DAI, + }, + { + .name = RT1011_DEV3_NAME, + .dai_name = RT1011_CODEC_DAI, + }, }; static const struct { @@ -63,6 +118,8 @@ static const struct { } rt1011_tdm_mask[] = { {.tx = 0x4, .rx = 0x1}, {.tx = 0x8, .rx = 0x2}, + {.tx = 0x1, .rx = 0x1}, + {.tx = 0x2, .rx = 0x2}, }; static int rt1011_hw_params(struct snd_pcm_substream *substream, @@ -118,28 +175,109 @@ static const struct snd_soc_ops rt1011_ops = { static int rt1011_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; + unsigned int num_codecs = get_num_codecs(RT1011_ACPI_HID); int ret; - ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr, - ARRAY_SIZE(speaker_map_lr)); - if (ret) - dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + switch (num_codecs) { + case 2: + if (!soc_intel_is_cml()) { + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1011 routes, ret %d\n", + ret); + return ret; + } + + break; + } + + /* + * register speaker widgets "WL Ext Spk" and "WR Ext Spk" to + * keep backward compatible with cml devices + */ + fallthrough; + case 4: + ret = snd_soc_dapm_new_controls(&card->dapm, realtek_4spk_widgets, + num_codecs); + if (ret) { + dev_err(rtd->dev, "fail to add rt1011 widgets, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, realtek_4spk_kcontrols, + num_codecs); + if (ret) { + dev_err(rtd->dev, "fail to add rt1011 controls, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt1011_4spk_routes, + num_codecs); + if (ret) { + dev_err(rtd->dev, "fail to add rt1011 routes, ret %d\n", + ret); + return ret; + } + break; + default: + dev_err(rtd->dev, "rt1011: invalid num_codecs %d\n", num_codecs); + return -EINVAL; + } + return ret; } -void sof_rt1011_dai_link(struct snd_soc_dai_link *link) +void sof_rt1011_dai_link(struct device *dev, struct snd_soc_dai_link *link) { + unsigned int num_codecs = get_num_codecs(RT1011_ACPI_HID); + link->codecs = rt1011_dai_link_components; - link->num_codecs = ARRAY_SIZE(rt1011_dai_link_components); + + switch (num_codecs) { + case 2: + case 4: + link->num_codecs = num_codecs; + break; + default: + dev_err(dev, "rt1011: invalid num_codecs %d\n", num_codecs); + break; + } + link->init = rt1011_init; link->ops = &rt1011_ops; } EXPORT_SYMBOL_NS(sof_rt1011_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON); -void sof_rt1011_codec_conf(struct snd_soc_card *card) +void sof_rt1011_codec_conf(struct device *dev, struct snd_soc_card *card) { - card->codec_conf = rt1011_codec_confs; - card->num_configs = ARRAY_SIZE(rt1011_codec_confs); + unsigned int num_codecs = get_num_codecs(RT1011_ACPI_HID); + + switch (num_codecs) { + case 2: + if (soc_intel_is_cml()) { + /* + * use name prefix 'WL' and 'WR' for speaker widgets to + * keep backward compatible with cml devices + */ + card->codec_conf = rt1011_4spk_codec_confs; + } else { + card->codec_conf = rt1011_2spk_codec_confs; + } + + card->num_configs = num_codecs; + break; + case 4: + card->codec_conf = rt1011_4spk_codec_confs; + card->num_configs = ARRAY_SIZE(rt1011_4spk_codec_confs); + break; + default: + dev_err(dev, "rt1011: invalid num_codecs %d\n", num_codecs); + break; + } + } EXPORT_SYMBOL_NS(sof_rt1011_codec_conf, SND_SOC_INTEL_SOF_REALTEK_COMMON); diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index 289e85b616602..d0e331530b398 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -23,8 +23,8 @@ #define RT1011_DEV2_NAME "i2c-" RT1011_ACPI_HID ":02" #define RT1011_DEV3_NAME "i2c-" RT1011_ACPI_HID ":03" -void sof_rt1011_dai_link(struct snd_soc_dai_link *link); -void sof_rt1011_codec_conf(struct snd_soc_card *card); +void sof_rt1011_dai_link(struct device *dev, struct snd_soc_dai_link *link); +void sof_rt1011_codec_conf(struct device *dev, struct snd_soc_card *card); /* * Realtek ALC1015 (AUTO) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 966bcd3e8bdc5..326d23dd79de6 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -576,7 +576,7 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, max_98390_dai_link(dev, ctx->amp_link); break; case CODEC_RT1011: - sof_rt1011_dai_link(ctx->amp_link); + sof_rt1011_dai_link(dev, ctx->amp_link); break; case CODEC_RT1015: sof_rt1015_dai_link(ctx->amp_link); @@ -683,7 +683,7 @@ static int sof_audio_probe(struct platform_device *pdev) max_98390_set_codec_conf(&pdev->dev, &sof_audio_card_rt5682); break; case CODEC_RT1011: - sof_rt1011_codec_conf(&sof_audio_card_rt5682); + sof_rt1011_codec_conf(&pdev->dev, &sof_audio_card_rt5682); break; case CODEC_RT1015: sof_rt1015_codec_conf(&sof_audio_card_rt5682); From 6136d879f3c2240db636d614a1c39f17c6ceaf36 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 11 Apr 2024 17:03:45 -0500 Subject: [PATCH 274/603] ASoC: Intel: sof_rt5682: support ALC1011 on cml boards For cml boards, ALC1011 speaker amplifier is supported by machine driver cml_rt1011_rt5682. Use same driver name for backward compatibility with existing devices on market. Reviewed-by: Chao Song Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 326d23dd79de6..735e07ea62bdb 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -606,6 +606,7 @@ static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct sof_card_private *ctx; + char *card_name; bool is_legacy_cpu = false; int ret; @@ -631,14 +632,27 @@ static int sof_audio_probe(struct platform_device *pdev) return -ENOMEM; if (ctx->codec_type == CODEC_RT5650) { - sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", - GFP_KERNEL); + card_name = devm_kstrdup(&pdev->dev, "rt5650", GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + sof_audio_card_rt5682.name = card_name; /* create speaker dai link also */ if (ctx->amp_type == CODEC_NONE) ctx->amp_type = CODEC_RT5650; } + if (ctx->amp_type == CODEC_RT1011 && soc_intel_is_cml()) { + /* backward-compatible with existing cml devices */ + card_name = devm_kstrdup(&pdev->dev, "cml_rt1011_rt5682", + GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + sof_audio_card_rt5682.name = card_name; + } + if (is_legacy_cpu) { ctx->rt5682.is_legacy_cpu = true; ctx->dmic_be_num = 0; From ca571e5a2e45b1a4113af2370fd1cf895f0b46d4 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 11 Apr 2024 17:03:46 -0500 Subject: [PATCH 275/603] ASoC: Intel: cml_rt1011_rt5682: delete driver Delete this driver and use sof_rt5682 machine driver instead. Reviewed-by: Chao Song Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 12 +- sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/cml_rt1011_rt5682.c | 609 ------------------ .../intel/common/soc-acpi-intel-cml-match.c | 2 +- 4 files changed, 4 insertions(+), 621 deletions(-) delete mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 9f4a85513702a..0ad7b0a1e237a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -595,17 +595,11 @@ config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH tristate "CML with RT1011 and RT5682 in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_SOC_RT1011 - select SND_SOC_RT5682_I2C - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - select SND_SOC_INTEL_HDA_DSP_COMMON + imply SND_SOC_INTEL_SOF_RT5682_MACH help This adds support for ASoC machine driver for SOF platform with - RT1011 + RT5682 I2S codec. + RT1011 + RT5682 I2S codec. This option is deprecated and please used + SND_SOC_INTEL_SOF_RT5682_MACH instead. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index a93b658ed6728..119413c262de7 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -24,7 +24,6 @@ snd-soc-sof_cs42l42-objs := sof_cs42l42.o snd-soc-sof_es8336-objs := sof_es8336.o snd-soc-sof_nau8825-objs := sof_nau8825.o snd-soc-sof_da7219-objs := sof_da7219.o -snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o @@ -71,7 +70,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_CX2072X_MACH) += snd-soc-sst-byt-cht-cx2072x. obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o -obj-$(CONFIG_SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH) += snd-soc-cml_rt1011_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c deleted file mode 100644 index 679a09b63ea59..0000000000000 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ /dev/null @@ -1,609 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright(c) 2019 Intel Corporation. - -/* - * Intel Cometlake I2S Machine driver for RT1011 + RT5682 codec - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/rt1011.h" -#include "../../codecs/rt5682.h" -#include "../../codecs/hdac_hdmi.h" -#include "hda_dsp_common.h" - -/* The platform clock outputs 24Mhz clock to codec as I2S MCLK */ -#define CML_PLAT_CLK 24000000 -#define CML_RT1011_CODEC_DAI "rt1011-aif" -#define CML_RT5682_CODEC_DAI "rt5682-aif1" -#define NAME_SIZE 32 - -#define SOF_RT1011_SPEAKER_WL BIT(0) -#define SOF_RT1011_SPEAKER_WR BIT(1) -#define SOF_RT1011_SPEAKER_TL BIT(2) -#define SOF_RT1011_SPEAKER_TR BIT(3) - -/* Default: Woofer speakers */ -static unsigned long sof_rt1011_quirk = SOF_RT1011_SPEAKER_WL | - SOF_RT1011_SPEAKER_WR; - -static int sof_rt1011_quirk_cb(const struct dmi_system_id *id) -{ - sof_rt1011_quirk = (unsigned long)id->driver_data; - return 1; -} - -static const struct dmi_system_id sof_rt1011_quirk_table[] = { - { - .callback = sof_rt1011_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Google"), - DMI_MATCH(DMI_PRODUCT_NAME, "Helios"), - }, - .driver_data = (void *)(SOF_RT1011_SPEAKER_WL | SOF_RT1011_SPEAKER_WR | - SOF_RT1011_SPEAKER_TL | SOF_RT1011_SPEAKER_TR), - }, - { - } -}; - -static struct snd_soc_jack hdmi_jack[3]; - -struct hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct card_private { - char codec_name[SND_ACPI_I2C_ID_LEN]; - struct snd_soc_jack headset; - struct list_head hdmi_pcm_list; - bool common_hdmi_codec_drv; -}; - -static const struct snd_kcontrol_new cml_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("WL Ext Spk"), - SOC_DAPM_PIN_SWITCH("WR Ext Spk"), -}; - -static const struct snd_kcontrol_new cml_rt1011_tt_controls[] = { - SOC_DAPM_PIN_SWITCH("TL Ext Spk"), - SOC_DAPM_PIN_SWITCH("TR Ext Spk"), -}; - -static const struct snd_soc_dapm_widget cml_rt1011_rt5682_widgets[] = { - SND_SOC_DAPM_SPK("WL Ext Spk", NULL), - SND_SOC_DAPM_SPK("WR Ext Spk", NULL), - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), -}; - -static const struct snd_soc_dapm_widget cml_rt1011_tt_widgets[] = { - SND_SOC_DAPM_SPK("TL Ext Spk", NULL), - SND_SOC_DAPM_SPK("TR Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route cml_rt1011_rt5682_map[] = { - /*WL/WR speaker*/ - {"WL Ext Spk", NULL, "WL SPO"}, - {"WR Ext Spk", NULL, "WR SPO"}, - - /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, - - /* other jacks */ - { "IN1P", NULL, "Headset Mic" }, - - /* DMIC */ - {"DMic", NULL, "SoC DMIC"}, -}; - -static const struct snd_soc_dapm_route cml_rt1011_tt_map[] = { - /*TL/TR speaker*/ - {"TL Ext Spk", NULL, "TL SPO" }, - {"TR Ext Spk", NULL, "TR SPO" }, -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static int cml_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_jack *jack; - int ret; - - /* need to enable ASRC function for 24MHz mclk rate */ - rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | - RT5682_AD_STEREO1_FILTER, - RT5682_CLK_SEL_I2S1_ASRC); - - /* - * Headset buttons map to the google Reference headset. - * These can be configured by userspace. - */ - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - &ctx->headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); - return ret; - } - - jack = &ctx->headset; - - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); - if (ret) - dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); - - return ret; -}; - -static void cml_rt5682_codec_exit(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - - snd_soc_component_set_jack(component, NULL, NULL); -} - -static int cml_rt1011_spk_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret = 0; - struct snd_soc_card *card = rtd->card; - - if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | - SOF_RT1011_SPEAKER_TR)) { - - ret = snd_soc_add_card_controls(card, cml_rt1011_tt_controls, - ARRAY_SIZE(cml_rt1011_tt_controls)); - if (ret) - return ret; - - ret = snd_soc_dapm_new_controls(&card->dapm, - cml_rt1011_tt_widgets, - ARRAY_SIZE(cml_rt1011_tt_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(&card->dapm, cml_rt1011_tt_map, - ARRAY_SIZE(cml_rt1011_tt_map)); - - if (ret) - return ret; - } - - return ret; -} - -static int cml_rt5682_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int clk_id, clk_freq, pll_out, ret; - - clk_id = RT5682_PLL1_S_MCLK; - clk_freq = CML_PLAT_CLK; - - pll_out = params_rate(params) * 512; - - ret = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out); - if (ret < 0) - dev_warn(rtd->dev, "snd_soc_dai_set_pll err = %d\n", ret); - - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, - pll_out, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_warn(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - /* - * slot_width should be equal or large than data length, set them - * be the same - */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, - params_width(params)); - if (ret < 0) - dev_warn(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; -} - -static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai; - struct snd_soc_card *card = rtd->card; - int srate, i, ret = 0; - - srate = params_rate(params); - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - - /* 100 Fs to drive 24 bit data */ - ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, - 100 * srate, 256 * srate); - if (ret < 0) { - dev_err(card->dev, "codec_dai clock not set\n"); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT1011_FS_SYS_PRE_S_PLL1, - 256 * srate, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "codec_dai clock not set\n"); - return ret; - } - - /* - * Codec TDM is configured as 24 bit capture/ playback. - * 2 CH PB is done over 4 codecs - 2 Woofers and 2 Tweeters. - * The Left woofer and tweeter plays the Left playback data - * and similar by the Right. - * Hence 2 codecs (1 T and 1 W pair) share same Rx slot. - * The feedback is captured for each codec individually. - * Hence all 4 codecs use 1 Tx slot each for feedback. - */ - if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_WL | - SOF_RT1011_SPEAKER_WR)) { - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:00")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x4, 0x1, 4, 24); - if (ret < 0) - break; - } - - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:01")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x8, 0x2, 4, 24); - if (ret < 0) - break; - } - } - - if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | - SOF_RT1011_SPEAKER_TR)) { - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:02")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x1, 0x1, 4, 24); - if (ret < 0) - break; - } - - if (!strcmp(codec_dai->component->name, "i2c-10EC1011:03")) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x2, 0x2, 4, 24); - if (ret < 0) - break; - } - } - } - if (ret < 0) - dev_err(rtd->dev, - "set codec TDM slot for %s failed with error %d\n", - codec_dai->component->name, ret); - return ret; -} - -static struct snd_soc_ops cml_rt5682_ops = { - .hw_params = cml_rt5682_hw_params, -}; - -static const struct snd_soc_ops cml_rt1011_ops = { - .hw_params = cml_rt1011_hw_params, -}; - -static int sof_card_late_probe(struct snd_soc_card *card) -{ - struct card_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_component *component = NULL; - char jack_name[NAME_SIZE]; - struct hdmi_pcm *pcm; - int ret, i = 0; - - if (list_empty(&ctx->hdmi_pcm_list)) - return -EINVAL; - - if (ctx->common_hdmi_codec_drv) { - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm, - head); - component = pcm->codec_dai->component; - return hda_dsp_hdmi_build_controls(card, component); - } - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - ret = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &hdmi_jack[i]); - if (ret) - return ret; - - ret = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &hdmi_jack[i]); - if (ret < 0) - return ret; - - i++; - } - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -static int hdmi_init(struct snd_soc_pcm_runtime *rtd) -{ - struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = dai->id; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -/* Cometlake digital audio interface glue - connects codec <--> CPU */ - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", - CML_RT5682_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec_2spk, - DAILINK_COMP_ARRAY( - /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), - /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI))); -SND_SOC_DAILINK_DEF(ssp1_codec_4spk, - DAILINK_COMP_ARRAY( - /* WL */ COMP_CODEC("i2c-10EC1011:00", CML_RT1011_CODEC_DAI), - /* WR */ COMP_CODEC("i2c-10EC1011:01", CML_RT1011_CODEC_DAI), - /* TL */ COMP_CODEC("i2c-10EC1011:02", CML_RT1011_CODEC_DAI), - /* TR */ COMP_CODEC("i2c-10EC1011:03", CML_RT1011_CODEC_DAI))); - - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); - -SND_SOC_DAILINK_DEF(dmic16k_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin"))); - -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = { - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .init = cml_rt5682_codec_init, - .exit = cml_rt5682_codec_exit, - .ignore_pmdown_time = 1, - .ops = &cml_rt5682_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - .name = "dmic01", - .id = 1, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "dmic16k", - .id = 2, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, - { - /* - * SSP1 - Codec : added to end of list ensuring - * reuse of common topologies for other end points - * and changing only SSP1's codec - */ - .name = "SSP1-Codec", - .id = 6, - .dpcm_playback = 1, - .dpcm_capture = 1, /* Capture stream provides Feedback */ - .no_pcm = 1, - .init = cml_rt1011_spk_init, - .ops = &cml_rt1011_ops, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec_2spk, platform), - }, -}; - -static struct snd_soc_codec_conf rt1011_conf[] = { - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:00"), - .name_prefix = "WL", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:01"), - .name_prefix = "WR", - }, - /* single configuration structure for 2 and 4 channels */ - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:02"), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF("i2c-10EC1011:03"), - .name_prefix = "TR", - }, -}; - -/* Cometlake audio machine driver for RT1011 and RT5682 */ -static struct snd_soc_card snd_soc_card_cml = { - .name = "cml_rt1011_rt5682", - .owner = THIS_MODULE, - .dai_link = cml_rt1011_rt5682_dailink, - .num_links = ARRAY_SIZE(cml_rt1011_rt5682_dailink), - .codec_conf = rt1011_conf, - .num_configs = ARRAY_SIZE(rt1011_conf), - .dapm_widgets = cml_rt1011_rt5682_widgets, - .num_dapm_widgets = ARRAY_SIZE(cml_rt1011_rt5682_widgets), - .dapm_routes = cml_rt1011_rt5682_map, - .num_dapm_routes = ARRAY_SIZE(cml_rt1011_rt5682_map), - .controls = cml_controls, - .num_controls = ARRAY_SIZE(cml_controls), - .fully_routed = true, - .late_probe = sof_card_late_probe, -}; - -static int snd_cml_rt1011_probe(struct platform_device *pdev) -{ - struct snd_soc_dai_link *dai_link; - struct card_private *ctx; - struct snd_soc_acpi_mach *mach; - const char *platform_name; - int ret, i; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - mach = pdev->dev.platform_data; - snd_soc_card_cml.dev = &pdev->dev; - platform_name = mach->mach_params.platform; - - dmi_check_system(sof_rt1011_quirk_table); - - dev_dbg(&pdev->dev, "sof_rt1011_quirk = %lx\n", sof_rt1011_quirk); - - /* when 4 speaker is available, update codec config */ - if (sof_rt1011_quirk & (SOF_RT1011_SPEAKER_TL | - SOF_RT1011_SPEAKER_TR)) { - for_each_card_prelinks(&snd_soc_card_cml, i, dai_link) { - if (!strcmp(dai_link->codecs[0].dai_name, - CML_RT1011_CODEC_DAI)) { - dai_link->codecs = ssp1_codec_4spk; - dai_link->num_codecs = ARRAY_SIZE(ssp1_codec_4spk); - } - } - } - - /* set platform name for each dailink */ - ret = snd_soc_fixup_dai_links_platform_name(&snd_soc_card_cml, - platform_name); - if (ret) - return ret; - - ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; - - snd_soc_card_set_drvdata(&snd_soc_card_cml, ctx); - - return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cml); -} - -static struct platform_driver snd_cml_rt1011_rt5682_driver = { - .probe = snd_cml_rt1011_probe, - .driver = { - .name = "cml_rt1011_rt5682", - .pm = &snd_soc_pm_ops, - }, -}; -module_platform_driver(snd_cml_rt1011_rt5682_driver); - -/* Module information */ -MODULE_DESCRIPTION("Cometlake Audio Machine driver - RT1011 and RT5682 in I2S mode"); -MODULE_AUTHOR("Naveen Manohar "); -MODULE_AUTHOR("Sathya Prakash M R "); -MODULE_AUTHOR("Shuming Fan "); -MODULE_AUTHOR("Mac Chiang "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:cml_rt1011_rt5682"); -MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index d47a548959eaf..160bfa0e5ef5f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -42,7 +42,7 @@ static const struct snd_soc_acpi_codecs max98390_spk_codecs = { struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { { .id = "10EC5682", - .drv_name = "cml_rt1011_rt5682", + .drv_name = "cml_rt5682_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rt1011_spk_codecs, .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", From dcc2cd8000d11a046680a7476b0d96b0b956454a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 11 Apr 2024 17:03:47 -0500 Subject: [PATCH 276/603] ASoC: Intel: sof_rt5682: use RT5682S_PLL1 if needed When 96KHz sample rate is used, and MCLK is 24.576MHz, we will need pll_in = 24576000 and pll_out = 49152000 which is not supported by RT5682S_PLL2. Use RT5682S_PLL1 in this case. We don't test sample rate because RT5682S_PLL2 doesn't support 24.576MHz input and in the MCLK = 24.576MHz, sample rate = 48KHz case, i.e. pll_in == pll_out, PLL will not be used at all. Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240411220347.131267-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 735e07ea62bdb..062012de3de22 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -340,8 +340,19 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, clk_id = RT5682_SCLK_S_PLL1; break; case CODEC_RT5682S: - pll_id = RT5682S_PLL2; - clk_id = RT5682S_SCLK_S_PLL2; + /* + * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test + * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz + * input, so we have no choice but to use PLL1. Besides, we will not use PLL at + * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz + */ + if (pll_in == 24576000) { + pll_id = RT5682S_PLL1; + clk_id = RT5682S_SCLK_S_PLL1; + } else { + pll_id = RT5682S_PLL2; + clk_id = RT5682S_SCLK_S_PLL2; + } break; default: dev_err(rtd->dev, "invalid codec type %d\n", ctx->codec_type); From 4be7bc275253c0f35c8f6362af1ee5d91d5d83e9 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 12 Apr 2024 09:14:10 -0300 Subject: [PATCH 277/603] ASoC: dt-bindings: fsl-asoc-card: Document fsl,imx25-pdk-sgtl5000 Document fsl,imx25-pdk-sgtl5000 to fix the following dt-schema warning: imx25-pdk.dtb: sound: compatible: 'oneOf' conditional failed, one must be fixed: ['fsl,imx25-pdk-sgtl5000', 'fsl,imx-audio-sgtl5000'] is too long Fixes: 4189b54220e5 ("ASoC: dt-bindings: fsl-asoc-card: convert to YAML") Signed-off-by: Fabio Estevam Acked-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240412121410.2948048-1-festevam@gmail.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 42ca39eebd49a..bbf6fa1a20127 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -33,6 +33,7 @@ properties: - items: - enum: - fsl,imx-sgtl5000 + - fsl,imx25-pdk-sgtl5000 - fsl,imx53-cpuvo-sgtl5000 - fsl,imx51-babbage-sgtl5000 - fsl,imx53-m53evk-sgtl5000 From a8cad4a4e431e250edc05242a8ca1be6e4b33749 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 12 Apr 2024 15:07:01 +0300 Subject: [PATCH 278/603] ASoC: soc-card: soc-card-test: Fix some error handling in init() There are two issues here: 1) The get_device() needs a matching put_device() on error paths. 2) The "if (!ret)" was supposed to be "if (ret)". I re-arranged the code a bit to do the allocation before the get_device(). Fixes: ef7784e41db7 ("ASoC: soc-card: Add KUnit test case for snd_soc_card_get_kcontrol") Signed-off-by: Dan Carpenter Reviewed-by: Richard Fitzgerald Link: https://lore.kernel.org/r/450dd21a-b24b-48ba-9aa4-c02e4617852f@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/soc-card-test.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-card-test.c b/sound/soc/soc-card-test.c index 075c52fe82e51..e4a4b101d7438 100644 --- a/sound/soc/soc-card-test.c +++ b/sound/soc/soc-card-test.c @@ -134,22 +134,24 @@ static int soc_card_test_case_init(struct kunit *test) test->priv = priv; + priv->card = kunit_kzalloc(test, sizeof(*priv->card), GFP_KERNEL); + if (!priv->card) + return -ENOMEM; + priv->card_dev = kunit_device_register(test, "sound-soc-card-test"); priv->card_dev = get_device(priv->card_dev); if (!priv->card_dev) return -ENODEV; - priv->card = kunit_kzalloc(test, sizeof(*priv->card), GFP_KERNEL); - if (!priv->card) - return -ENOMEM; - priv->card->name = "soc-card-test"; priv->card->dev = priv->card_dev; priv->card->owner = THIS_MODULE; ret = snd_soc_register_card(priv->card); - if (!ret) + if (ret) { + put_device(priv->card_dev); return ret; + } return 0; } From d6e792ed7dd022a16a637ade224da070b60f4f3b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 12 Apr 2024 05:46:03 +0000 Subject: [PATCH 279/603] ASoC: dt-bindings: renesas: add R8A779H0 V4M Add document for R-Car V4M (R8A779H0). Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/871q7bcew5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 0d7a6b576d880..8ce78ca838ba1 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -49,6 +49,7 @@ properties: # for Gen4 SoC - items: - const: renesas,rcar_sound-r8a779g0 # R-Car V4H + - const: renesas,rcar_sound-r8a779h0 # R-Car V4M - const: renesas,rcar_sound-gen4 # for Generic - enum: From f284b23809bf54f8189f82f822f099e43d6a0a35 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 12 Apr 2024 05:45:43 +0000 Subject: [PATCH 280/603] ASoC: dt-bindings: renesas,rsnd: add missing renesas,rcar_sound-gen4 It is missing generic compatible for R-Car Gen4 Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/8734rrcewo.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 8ce78ca838ba1..b3da278cf141c 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -56,6 +56,7 @@ properties: - renesas,rcar_sound-gen1 - renesas,rcar_sound-gen2 - renesas,rcar_sound-gen3 + - renesas,rcar_sound-gen4 reg: minItems: 1 From c21614d5c0e76cf1d7a864f15d495fd3e5876acc Mon Sep 17 00:00:00 2001 From: "end.to.start" Date: Fri, 12 Apr 2024 10:55:33 +0300 Subject: [PATCH 281/603] ASoC: amd: Support microphone from Acer Aspire A315-24P Add support microphone from Acer Aspire A315-24P and for some other similar devices with such vendor Signed-off-by: "end.to.start" Link: https://lore.kernel.org/r/20240412075533.10214-1-end.to.start@mail.ru Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 69c68d8e7a6b5..1760b5d42460a 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -430,6 +430,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "MRID6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "MDC"), + DMI_MATCH(DMI_BOARD_NAME, "Herbag_MDU"), + } + }, { .driver_data = &acp6x_card, .matches = { From 615169c727b13ff4ec6c43b62501f73f9f88a11a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sun, 14 Apr 2024 17:48:38 +0200 Subject: [PATCH 282/603] ASoC: wcd934x: Drop unneeded MODULE_ALIAS The ID table already has respective entry and MODULE_DEVICE_TABLE and creates proper alias for platform driver. Having another MODULE_ALIAS causes the alias to be duplicated. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240414154839.126852-1-krzk@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 6813268e6a19f..de870c7819caa 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -5967,7 +5967,6 @@ static struct platform_driver wcd934x_codec_driver = { } }; -MODULE_ALIAS("platform:wcd934x-codec"); module_platform_driver(wcd934x_codec_driver); MODULE_DESCRIPTION("WCD934x codec driver"); MODULE_LICENSE("GPL v2"); From 9d85ec4e91e354ed2d8291e1ea3ba76e660205a0 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sun, 14 Apr 2024 17:48:39 +0200 Subject: [PATCH 283/603] ASoC: samsung: i2s: Drop unneeded MODULE_ALIAS The ID table already has respective entry and MODULE_DEVICE_TABLE and creates proper alias for platform driver. Having another MODULE_ALIAS causes the alias to be duplicated. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240414154839.126852-2-krzk@kernel.org Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 9552748aea2e6..1bcabb114e29f 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1755,5 +1755,4 @@ module_platform_driver(samsung_i2s_driver); /* Module information */ MODULE_AUTHOR("Jaswinder Singh, "); MODULE_DESCRIPTION("Samsung I2S Interface"); -MODULE_ALIAS("platform:samsung-i2s"); MODULE_LICENSE("GPL"); From dd8c3473bbb8b4fdc44ed3a5a864cfefab9038dd Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sun, 14 Apr 2024 18:47:03 +0200 Subject: [PATCH 284/603] ASoC: amd: acp-da7219-max98357a: Constify regulator_ops Neither core nor the driver modifes 'struct regulator_ops', so it can be const for code safety. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240414164703.239851-1-krzk@kernel.org Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 84f3d65ba52e3..77cf72082e73d 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -733,7 +733,7 @@ static struct regulator_config acp_da7219_cfg = { .init_data = &acp_da7219_data, }; -static struct regulator_ops acp_da7219_ops = { +static const struct regulator_ops acp_da7219_ops = { }; static const struct regulator_desc acp_da7219_desc = { From 863f94ac5f4481a7c5665e8152d551701ac71bf3 Mon Sep 17 00:00:00 2001 From: John Watts Date: Wed, 3 Apr 2024 14:47:35 +1100 Subject: [PATCH 285/603] ASoC: sunxi: sun4i-i2s: Support 32-bit audio formats The I2S cores used in the H3 onwards support 32-bit sample rates. Support these by adding a per-variant PCM format list. Signed-off-by: John Watts Acked-by: Jernej Skrabec Link: https://lore.kernel.org/r/20240403-sunxi_s32-v2-1-29ebf6ad590a@jookia.org Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-i2s.c | 33 ++++++++++++++++++++++++++++----- 1 file changed, 28 insertions(+), 5 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a736f632bf0b6..59830f2a0d308 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -156,6 +156,7 @@ struct sun4i_i2s; /** * struct sun4i_i2s_quirks - Differences between SoC variants. * @has_reset: SoC needs reset deasserted. + * @pcm_formats: available PCM formats * @reg_offset_txdata: offset of the tx fifo. * @sun4i_i2s_regmap: regmap config to use. * @field_clkdiv_mclk_en: regmap field to enable mclk output. @@ -175,6 +176,7 @@ struct sun4i_i2s; */ struct sun4i_i2s_quirks { bool has_reset; + snd_pcm_format_t pcm_formats; unsigned int reg_offset_txdata; /* TX FIFO */ const struct regmap_config *sun4i_i2s_regmap; @@ -1092,8 +1094,18 @@ static int sun4i_i2s_dai_probe(struct snd_soc_dai *dai) return 0; } +static int sun4i_i2s_dai_startup(struct snd_pcm_substream *sub, struct snd_soc_dai *dai) +{ + struct sun4i_i2s *i2s = snd_soc_dai_get_drvdata(dai); + struct snd_pcm_runtime *runtime = sub->runtime; + + return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, + i2s->variant->pcm_formats); +} + static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .probe = sun4i_i2s_dai_probe, + .startup = sun4i_i2s_dai_startup, .hw_params = sun4i_i2s_hw_params, .set_fmt = sun4i_i2s_set_fmt, .set_sysclk = sun4i_i2s_set_sysclk, @@ -1101,9 +1113,10 @@ static const struct snd_soc_dai_ops sun4i_i2s_dai_ops = { .trigger = sun4i_i2s_trigger, }; -#define SUN4I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S20_LE | \ - SNDRV_PCM_FMTBIT_S24_LE) +#define SUN4I_FORMATS_ALL (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver sun4i_i2s_dai = { .capture = { @@ -1111,14 +1124,14 @@ static struct snd_soc_dai_driver sun4i_i2s_dai = { .channels_min = 1, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SUN4I_FORMATS, + .formats = SUN4I_FORMATS_ALL, }, .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SUN4I_FORMATS, + .formats = SUN4I_FORMATS_ALL, }, .ops = &sun4i_i2s_dai_ops, .symmetric_rate = 1, @@ -1340,8 +1353,12 @@ static int sun4i_i2s_runtime_suspend(struct device *dev) return 0; } +#define SUN4I_FORMATS_A10 (SUN4I_FORMATS_ALL & ~SNDRV_PCM_FMTBIT_S32_LE) +#define SUN4I_FORMATS_H3 SUN4I_FORMATS_ALL + static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = { .has_reset = false, + .pcm_formats = SUN4I_FORMATS_A10, .reg_offset_txdata = SUN4I_I2S_FIFO_TX_REG, .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7), @@ -1360,6 +1377,7 @@ static const struct sun4i_i2s_quirks sun4i_a10_i2s_quirks = { static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { .has_reset = true, + .pcm_formats = SUN4I_FORMATS_A10, .reg_offset_txdata = SUN4I_I2S_FIFO_TX_REG, .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7), @@ -1383,6 +1401,7 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { */ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, + .pcm_formats = SUN4I_FORMATS_A10, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7), @@ -1401,6 +1420,7 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { .has_reset = true, + .pcm_formats = SUN4I_FORMATS_H3, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), @@ -1419,6 +1439,7 @@ static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { .has_reset = true, + .pcm_formats = SUN4I_FORMATS_H3, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, .sun4i_i2s_regmap = &sun4i_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 7, 7), @@ -1437,6 +1458,7 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { static const struct sun4i_i2s_quirks sun50i_h6_i2s_quirks = { .has_reset = true, + .pcm_formats = SUN4I_FORMATS_H3, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, .sun4i_i2s_regmap = &sun50i_h6_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), @@ -1455,6 +1477,7 @@ static const struct sun4i_i2s_quirks sun50i_h6_i2s_quirks = { static const struct sun4i_i2s_quirks sun50i_r329_i2s_quirks = { .has_reset = true, + .pcm_formats = SUN4I_FORMATS_H3, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, .sun4i_i2s_regmap = &sun50i_h6_i2s_regmap_config, .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), From 51f67862ea6ea83f9fa4cda2e59d7bfd4387f63b Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 15 Apr 2024 15:17:35 +0800 Subject: [PATCH 286/603] ASoC: fsl-asoc-card: add wm8904 codec support wm8904 codec is used on i.MX95 Toradex board Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1713165456-3494-2-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index bc07f26ba303f..f6d2564864c60 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -28,6 +28,7 @@ #include "../codecs/wm8994.h" #include "../codecs/tlv320aic31xx.h" #include "../codecs/nau8822.h" +#include "../codecs/wm8904.h" #define DRIVER_NAME "fsl-asoc-card" @@ -709,6 +710,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; if (codec_dev) priv->codec_priv.mclk = devm_clk_get(codec_dev, NULL); + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) { + codec_dai_name = "wm8904-hifi"; + priv->codec_priv.mclk_id = WM8904_FLL_MCLK; + priv->codec_priv.fll_id = WM8904_CLK_FLL; + priv->codec_priv.pll_id = WM8904_FLL_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -935,6 +942,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-si476x", }, { .compatible = "fsl,imx-audio-wm8958", }, { .compatible = "fsl,imx-audio-nau8822", }, + { .compatible = "fsl,imx-audio-wm8904", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); From 62c48dd33b4f2e037554d1322ae4f9f60e9461ef Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Mon, 15 Apr 2024 15:17:36 +0800 Subject: [PATCH 287/603] ASoC: dt-bindings: fsl-asoc-card: Add compatbile string for wm8904 codec The wm8904 codec is used on an i.MX95 Toradex board. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1713165456-3494-3-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index bbf6fa1a20127..9922664d5ccca 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -77,6 +77,7 @@ properties: - fsl,imx-audio-tlv320aic31xx - fsl,imx-audio-tlv320aic32x4 - fsl,imx-audio-wm8524 + - fsl,imx-audio-wm8904 - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 From bce93a162505f0bc0499888ace2f66bd56fb94b3 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:13:53 +0200 Subject: [PATCH 288/603] ASoC: mediatek: Add common machine soundcard driver probe mechanism Add a common machine soundcard driver probe function that supports both DSP and AFE-direct usecases and also provides a hook for legacy machine soundcard driver probe mechanisms. Note that the hook is there because, even for legacy probe, a lot of the actual code can still be commonized, hence still reducing duplication for the legacy devicetree retrocompatibility cases. This common probe function deprecates all of the inconsistent previous probe mechanisms and aims to settle all of the MediaTek card drivers on consistent and common devicetree properties describing wanted DAIs, device specific DAI configuration and DAI links to codecs found on each device/board. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-2-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 148 ++++++++++++++++++ .../mediatek/common/mtk-soundcard-driver.h | 18 +++ 3 files changed, 172 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-soc-card.h b/sound/soc/mediatek/common/mtk-soc-card.h index eeda793700498..d92dc36d45774 100644 --- a/sound/soc/mediatek/common/mtk-soc-card.h +++ b/sound/soc/mediatek/common/mtk-soc-card.h @@ -9,9 +9,14 @@ #ifndef _MTK_SOC_CARD_H_ #define _MTK_SOC_CARD_H_ +struct mtk_platform_card_data; +struct mtk_sof_priv; + struct mtk_soc_card_data { + struct mtk_sof_priv *sof_priv; + struct list_head sof_dai_link_list; + struct mtk_platform_card_data *card_data; void *mach_priv; - void *sof_priv; }; #endif diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.c b/sound/soc/mediatek/common/mtk-soundcard-driver.c index 000a086a8cf44..b1db17e392d52 100644 --- a/sound/soc/mediatek/common/mtk-soundcard-driver.c +++ b/sound/soc/mediatek/common/mtk-soundcard-driver.c @@ -10,6 +10,8 @@ #include #include +#include "mtk-dsp-sof-common.h" +#include "mtk-soc-card.h" #include "mtk-soundcard-driver.h" static int set_card_codec_info(struct snd_soc_card *card, @@ -136,3 +138,149 @@ void clean_card_reference(struct snd_soc_card *card) snd_soc_of_put_dai_link_codecs(dai_link); } EXPORT_SYMBOL_GPL(clean_card_reference); + +int mtk_soundcard_common_probe(struct platform_device *pdev) +{ + struct device_node *platform_node, *adsp_node; + const struct mtk_soundcard_pdata *pdata; + struct mtk_soc_card_data *soc_card_data; + struct snd_soc_dai_link *orig_dai_link, *dai_link; + struct snd_soc_jack *jacks; + struct snd_soc_card *card; + int i, orig_num_links, ret; + bool needs_legacy_probe; + + pdata = device_get_match_data(&pdev->dev); + if (!pdata) + return -EINVAL; + + card = pdata->card_data->card; + card->dev = &pdev->dev; + orig_dai_link = card->dai_link; + orig_num_links = card->num_links; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) + return ret; + + if (!card->name) { + if (!pdata->card_name) + return -EINVAL; + + card->name = pdata->card_name; + } + + needs_legacy_probe = !of_property_read_bool(pdev->dev.of_node, "audio-routing"); + if (needs_legacy_probe) { + /* + * If we have no .soc_probe() callback there's no way of using + * any legacy probe mechanism, as that cannot not be generic. + */ + if (!pdata->soc_probe) + return -EINVAL; + + dev_info_once(&pdev->dev, "audio-routing not found: using legacy probe\n"); + } else { + ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); + if (ret) + return ret; + } + + soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*soc_card_data), GFP_KERNEL); + if (!soc_card_data) + return -ENOMEM; + + soc_card_data->card_data = pdata->card_data; + + jacks = devm_kcalloc(card->dev, soc_card_data->card_data->num_jacks, + sizeof(*jacks), GFP_KERNEL); + if (!jacks) + return -ENOMEM; + + soc_card_data->card_data->jacks = jacks; + + platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0); + if (!platform_node) + return dev_err_probe(&pdev->dev, -EINVAL, + "Property mediatek,platform missing or invalid\n"); + + /* Check if this SoC has an Audio DSP */ + if (pdata->sof_priv) + adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); + else + adsp_node = NULL; + + if (adsp_node) { + if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { + ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, + "mediatek,dai-link", + card->dai_link, card->num_links); + if (ret) { + of_node_put(adsp_node); + of_node_put(platform_node); + return dev_err_probe(&pdev->dev, ret, + "Cannot parse mediatek,dai-link\n"); + } + } + + soc_card_data->sof_priv = pdata->sof_priv; + card->probe = mtk_sof_card_probe; + card->late_probe = mtk_sof_card_late_probe; + if (!card->topology_shortname_created) { + snprintf(card->topology_shortname, 32, "sof-%s", card->name); + card->topology_shortname_created = true; + } + card->name = card->topology_shortname; + } + + /* + * Regardless of whether the ADSP is wanted and/or present in a machine + * specific device tree or not and regardless of whether any AFE_SOF + * link is present, we have to make sure that the platforms->of_node + * is not NULL, and set to either ADSP (adsp_node) or AFE (platform_node). + */ + for_each_card_prelinks(card, i, dai_link) { + if (adsp_node && !strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF"))) + dai_link->platforms->of_node = adsp_node; + else if (!dai_link->platforms->name && !dai_link->platforms->of_node) + dai_link->platforms->of_node = platform_node; + } + + if (!needs_legacy_probe) { + ret = parse_dai_link_info(card); + if (ret) + goto err_restore_dais; + } else { + if (adsp_node) + of_node_put(adsp_node); + of_node_put(platform_node); + } + + if (pdata->soc_probe) { + ret = pdata->soc_probe(soc_card_data, needs_legacy_probe); + if (ret) { + if (!needs_legacy_probe) + clean_card_reference(card); + goto err_restore_dais; + } + } + snd_soc_card_set_drvdata(card, soc_card_data); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + + if (!needs_legacy_probe) + clean_card_reference(card); + + if (ret) { + dev_err_probe(&pdev->dev, ret, "Cannot register card\n"); + goto err_restore_dais; + } + + return 0; + +err_restore_dais: + card->dai_link = orig_dai_link; + card->num_links = orig_num_links; + return ret; +} +EXPORT_SYMBOL_GPL(mtk_soundcard_common_probe); diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.h b/sound/soc/mediatek/common/mtk-soundcard-driver.h index d92cac1d7b72a..44320efff5f81 100644 --- a/sound/soc/mediatek/common/mtk-soundcard-driver.h +++ b/sound/soc/mediatek/common/mtk-soundcard-driver.h @@ -9,6 +9,24 @@ #ifndef _MTK_SOUNDCARD_DRIVER_H_ #define _MTK_SOUNDCARD_DRIVER_H_ +struct mtk_sof_priv; +struct mtk_soc_card_data; + +struct mtk_platform_card_data { + struct snd_soc_card *card; + struct snd_soc_jack *jacks; + u8 num_jacks; + u8 flags; +}; + +struct mtk_soundcard_pdata { + const char *card_name; + struct mtk_platform_card_data *card_data; + struct mtk_sof_priv *sof_priv; + int (*soc_probe)(struct mtk_soc_card_data *card_data, bool legacy); +}; + int parse_dai_link_info(struct snd_soc_card *card); void clean_card_reference(struct snd_soc_card *card); +int mtk_soundcard_common_probe(struct platform_device *pdev); #endif From f8782f776db33066a9b1cd14d319142c4175561f Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:13:54 +0200 Subject: [PATCH 289/603] ASoC: mediatek: common: Constify struct mtk_sof_priv Apart from a dai_link_list variable, the mtk_sof_priv currently holds data that never gets modified during runtime. Constify the mtk_sof_priv structure and move the SOF dai_link_list as sof_dai_link_list in struct mtk_soc_card_data, which is a structure that already holds the card's machine specific, runtime modified data. This allows to safely pass the mtk_sof_priv structure as platform data for the commonized card probe mechanism. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-3-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-dsp-sof-common.c | 15 +++++++-------- sound/soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 2 +- sound/soc/mediatek/common/mtk-soundcard-driver.h | 2 +- 4 files changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c index 7ec8965a70c06..bca758dca2c9a 100644 --- a/sound/soc/mediatek/common/mtk-dsp-sof-common.c +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c @@ -15,7 +15,7 @@ int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_card *card = rtd->card; struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); - struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; + const struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; int i, j, ret = 0; for (i = 0; i < sof_priv->num_streams; i++) { @@ -55,7 +55,6 @@ int mtk_sof_card_probe(struct snd_soc_card *card) int i; struct snd_soc_dai_link *dai_link; struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); - struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; /* Set stream_name to help sof bind widgets */ for_each_card_prelinks(card, i, dai_link) { @@ -63,7 +62,7 @@ int mtk_sof_card_probe(struct snd_soc_card *card) dai_link->stream_name = dai_link->name; } - INIT_LIST_HEAD(&sof_priv->dai_link_list); + INIT_LIST_HEAD(&soc_card_data->sof_dai_link_list); return 0; } @@ -73,7 +72,7 @@ static struct snd_soc_pcm_runtime *mtk_sof_find_tplg_be(struct snd_soc_pcm_runti { struct snd_soc_card *card = rtd->card; struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); - struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; + const struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; struct snd_soc_pcm_runtime *fe; struct snd_soc_pcm_runtime *be; struct snd_soc_dpcm *dpcm; @@ -113,7 +112,7 @@ static int mtk_sof_check_tplg_be_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_card *card = rtd->card; struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); - struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; + const struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; struct snd_soc_pcm_runtime *sof_be; struct mtk_dai_link *dai_link; int ret = 0; @@ -125,7 +124,7 @@ static int mtk_sof_check_tplg_be_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, else if (sof_be->dai_link->be_hw_params_fixup) ret = sof_be->dai_link->be_hw_params_fixup(sof_be, params); } else { - list_for_each_entry(dai_link, &sof_priv->dai_link_list, list) { + list_for_each_entry(dai_link, &soc_card_data->sof_dai_link_list, list) { if (strcmp(dai_link->name, rtd->dai_link->name) == 0) { if (dai_link->be_hw_params_fixup) ret = dai_link->be_hw_params_fixup(rtd, params); @@ -144,7 +143,7 @@ int mtk_sof_card_late_probe(struct snd_soc_card *card) struct snd_soc_component *sof_comp = NULL; struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); - struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; + const struct mtk_sof_priv *sof_priv = soc_card_data->sof_priv; struct snd_soc_dai_link *dai_link; struct mtk_dai_link *mtk_dai_link; int i; @@ -173,7 +172,7 @@ int mtk_sof_card_late_probe(struct snd_soc_card *card) mtk_dai_link->be_hw_params_fixup = dai_link->be_hw_params_fixup; mtk_dai_link->name = dai_link->name; - list_add(&mtk_dai_link->list, &sof_priv->dai_link_list); + list_add(&mtk_dai_link->list, &soc_card_data->sof_dai_link_list); } if (dai_link->no_pcm) diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.h b/sound/soc/mediatek/common/mtk-dsp-sof-common.h index 4bc5e1c0c8ed9..8784ee471132a 100644 --- a/sound/soc/mediatek/common/mtk-dsp-sof-common.h +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.h @@ -30,7 +30,6 @@ struct mtk_sof_priv { int num_streams; int (*sof_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); - struct list_head dai_link_list; }; int mtk_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/mediatek/common/mtk-soc-card.h b/sound/soc/mediatek/common/mtk-soc-card.h index d92dc36d45774..3f6e24dd22df1 100644 --- a/sound/soc/mediatek/common/mtk-soc-card.h +++ b/sound/soc/mediatek/common/mtk-soc-card.h @@ -13,7 +13,7 @@ struct mtk_platform_card_data; struct mtk_sof_priv; struct mtk_soc_card_data { - struct mtk_sof_priv *sof_priv; + const struct mtk_sof_priv *sof_priv; struct list_head sof_dai_link_list; struct mtk_platform_card_data *card_data; void *mach_priv; diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.h b/sound/soc/mediatek/common/mtk-soundcard-driver.h index 44320efff5f81..4fd2ffb7e486f 100644 --- a/sound/soc/mediatek/common/mtk-soundcard-driver.h +++ b/sound/soc/mediatek/common/mtk-soundcard-driver.h @@ -22,7 +22,7 @@ struct mtk_platform_card_data { struct mtk_soundcard_pdata { const char *card_name; struct mtk_platform_card_data *card_data; - struct mtk_sof_priv *sof_priv; + const struct mtk_sof_priv *sof_priv; int (*soc_probe)(struct mtk_soc_card_data *card_data, bool legacy); }; From aad45b8aa973a863dee2f256cea8c527acaaf56e Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:13:55 +0200 Subject: [PATCH 290/603] ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe Add mtk_soundcard_pdata platform data for the MediaTek common sound card probe mechanism, including a driver/soc-specific probe extension (used for bits that cannot be commonized hence specific to this driver), and change the probe function to mtk_soundcard_common_probe. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-4-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +++++++--------------- 1 file changed, 64 insertions(+), 139 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index a391066ab2045..f629fc6bbb531 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -236,11 +236,11 @@ static const struct sof_conn_stream g_sof_conn_streams[] = { }, }; -struct mt8188_mt6359_priv { - struct snd_soc_jack dp_jack; - struct snd_soc_jack hdmi_jack; - struct snd_soc_jack headset_jack; - void *private_data; +enum mt8188_jacks { + MT8188_JACK_HEADSET, + MT8188_JACK_DP, + MT8188_JACK_HDMI, + MT8188_JACK_MAX, }; static struct snd_soc_jack_pin mt8188_hdmi_jack_pins[] = { @@ -268,11 +268,6 @@ static struct snd_soc_jack_pin nau8825_jack_pins[] = { }, }; -struct mt8188_card_data { - const char *name; - unsigned long quirk; -}; - static const struct snd_kcontrol_new mt8188_dumb_spk_controls[] = { SOC_DAPM_PIN_SWITCH("Ext Spk"), }; @@ -590,12 +585,12 @@ static int mt8188_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8188_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) { struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); - struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv; + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_HDMI]; struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; int ret = 0; ret = snd_soc_card_jack_new_pins(rtd->card, "HDMI Jack", - SND_JACK_LINEOUT, &priv->hdmi_jack, + SND_JACK_LINEOUT, jack, mt8188_hdmi_jack_pins, ARRAY_SIZE(mt8188_hdmi_jack_pins)); if (ret) { @@ -603,7 +598,7 @@ static int mt8188_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_component_set_jack(component, &priv->hdmi_jack, NULL); + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "%s, set jack failed on %s (ret=%d)\n", __func__, component->name, ret); @@ -616,19 +611,19 @@ static int mt8188_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) static int mt8188_dptx_codec_init(struct snd_soc_pcm_runtime *rtd) { struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); - struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv; + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_DP]; struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; int ret = 0; ret = snd_soc_card_jack_new_pins(rtd->card, "DP Jack", SND_JACK_LINEOUT, - &priv->dp_jack, mt8188_dp_jack_pins, + jack, mt8188_dp_jack_pins, ARRAY_SIZE(mt8188_dp_jack_pins)); if (ret) { dev_err(rtd->dev, "%s, new jack failed: %d\n", __func__, ret); return ret; } - ret = snd_soc_component_set_jack(component, &priv->dp_jack, NULL); + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "%s, set jack failed on %s (ret=%d)\n", __func__, component->name, ret); @@ -736,10 +731,9 @@ static int mt8188_max98390_codec_init(struct snd_soc_pcm_runtime *rtd) static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); - struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv; + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_HEADSET]; struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_jack *jack = &priv->headset_jack; int ret; ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_nau8825_widgets, @@ -1224,11 +1218,10 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { static void mt8188_fixup_controls(struct snd_soc_card *card) { struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); - struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv; - struct mt8188_card_data *card_data = (struct mt8188_card_data *)priv->private_data; + struct mtk_platform_card_data *card_data = soc_card_data->card_data; struct snd_kcontrol *kctl; - if (card_data->quirk & (NAU8825_HS_PRESENT | RT5682S_HS_PRESENT | ES8326_HS_PRESENT)) { + if (card_data->flags & (NAU8825_HS_PRESENT | RT5682S_HS_PRESENT | ES8326_HS_PRESENT)) { struct snd_soc_dapm_widget *w, *next_w; for_each_card_widgets_safe(card, w, next_w) { @@ -1259,14 +1252,10 @@ static struct snd_soc_card mt8188_mt6359_soc_card = { .fixup_controls = mt8188_fixup_controls, }; -static int mt8188_mt6359_dev_probe(struct platform_device *pdev) +static int mt8188_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy) { - struct snd_soc_card *card = &mt8188_mt6359_soc_card; - struct device_node *platform_node; - struct device_node *adsp_node; - struct mtk_soc_card_data *soc_card_data; - struct mt8188_mt6359_priv *priv; - struct mt8188_card_data *card_data; + struct mtk_platform_card_data *card_data = soc_card_data->card_data; + struct snd_soc_card *card = soc_card_data->card_data->card; struct snd_soc_dai_link *dai_link; bool init_mt6359 = false; bool init_es8326 = false; @@ -1274,91 +1263,12 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev) bool init_rt5682s = false; bool init_max98390 = false; bool init_dumb = false; - int ret, i; - - card_data = (struct mt8188_card_data *)of_device_get_match_data(&pdev->dev); - card->dev = &pdev->dev; - - ret = snd_soc_of_parse_card_name(card, "model"); - if (ret) - return dev_err_probe(&pdev->dev, ret, "%s new card name parsing error\n", - __func__); - - if (!card->name) - card->name = card_data->name; - - if (of_property_read_bool(pdev->dev.of_node, "audio-routing")) { - ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); - if (ret) - return ret; - } - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL); - if (!soc_card_data) - return -ENOMEM; - - soc_card_data->mach_priv = priv; - - adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); - if (adsp_node) { - struct mtk_sof_priv *sof_priv; - - sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL); - if (!sof_priv) { - ret = -ENOMEM; - goto err_adsp_node; - } - sof_priv->conn_streams = g_sof_conn_streams; - sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams); - soc_card_data->sof_priv = sof_priv; - card->probe = mtk_sof_card_probe; - card->late_probe = mtk_sof_card_late_probe; - if (!card->topology_shortname_created) { - snprintf(card->topology_shortname, 32, "sof-%s", card->name); - card->topology_shortname_created = true; - } - card->name = card->topology_shortname; - } - - if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { - ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, - "mediatek,dai-link", - mt8188_mt6359_dai_links, - ARRAY_SIZE(mt8188_mt6359_dai_links)); - if (ret) { - dev_err_probe(&pdev->dev, ret, "Parse dai-link fail\n"); - goto err_adsp_node; - } - } else { - if (!adsp_node) - card->num_links = DAI_LINK_REGULAR_NUM; - } - - platform_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,platform", 0); - if (!platform_node) { - ret = dev_err_probe(&pdev->dev, -EINVAL, - "Property 'platform' missing or invalid\n"); - goto err_adsp_node; - - } + int i; - ret = parse_dai_link_info(card); - if (ret) - goto err; + if (legacy) + return -EINVAL; for_each_card_prelinks(card, i, dai_link) { - if (!dai_link->platforms->name) { - if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && adsp_node) - dai_link->platforms->of_node = adsp_node; - else - dai_link->platforms->of_node = platform_node; - } - if (strcmp(dai_link->name, "DPTX_BE") == 0) { if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8188_dptx_codec_init; @@ -1381,7 +1291,7 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev) * mt8188_max98390_ops. Two amps is I2S mode, * SOC and codec don't require TDM settings. */ - if (!(card_data->quirk & MAX98390_TWO_AMP)) { + if (!(card_data->flags & MAX98390_TWO_AMP)) { dai_link->ops = &mt8188_max98390_ops; } if (!init_max98390) { @@ -1420,40 +1330,55 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev) } } - priv->private_data = card_data; - snd_soc_card_set_drvdata(card, soc_card_data); - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", - __func__); -err: - of_node_put(platform_node); - clean_card_reference(card); - -err_adsp_node: - of_node_put(adsp_node); - - return ret; + return 0; } -static struct mt8188_card_data mt8188_evb_card = { - .name = "mt8188_mt6359", +static const struct mtk_sof_priv mt8188_sof_priv = { + .conn_streams = g_sof_conn_streams, + .num_streams = ARRAY_SIZE(g_sof_conn_streams), +}; + +static const struct mtk_soundcard_pdata mt8188_evb_card = { + .card_name = "mt8188_mt6359", + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8188_mt6359_soc_card, + .num_jacks = MT8188_JACK_MAX, + }, + .sof_priv = &mt8188_sof_priv, + .soc_probe = mt8188_mt6359_soc_card_probe, }; -static struct mt8188_card_data mt8188_nau8825_card = { - .name = "mt8188_nau8825", - .quirk = NAU8825_HS_PRESENT, +static const struct mtk_soundcard_pdata mt8188_nau8825_card = { + .card_name = "mt8188_nau8825", + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8188_mt6359_soc_card, + .num_jacks = MT8188_JACK_MAX, + .flags = NAU8825_HS_PRESENT + }, + .sof_priv = &mt8188_sof_priv, + .soc_probe = mt8188_mt6359_soc_card_probe, }; -static struct mt8188_card_data mt8188_rt5682s_card = { - .name = "mt8188_rt5682s", - .quirk = RT5682S_HS_PRESENT | MAX98390_TWO_AMP, +static const struct mtk_soundcard_pdata mt8188_rt5682s_card = { + .card_name = "mt8188_rt5682s", + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8188_mt6359_soc_card, + .num_jacks = MT8188_JACK_MAX, + .flags = RT5682S_HS_PRESENT | MAX98390_TWO_AMP + }, + .sof_priv = &mt8188_sof_priv, + .soc_probe = mt8188_mt6359_soc_card_probe, }; -static struct mt8188_card_data mt8188_es8326_card = { - .name = "mt8188_es8326", - .quirk = ES8326_HS_PRESENT | MAX98390_TWO_AMP, +static const struct mtk_soundcard_pdata mt8188_es8326_card = { + .card_name = "mt8188_es8326", + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8188_mt6359_soc_card, + .num_jacks = MT8188_JACK_MAX, + .flags = ES8326_HS_PRESENT | MAX98390_TWO_AMP + }, + .sof_priv = &mt8188_sof_priv, + .soc_probe = mt8188_mt6359_soc_card_probe, }; static const struct of_device_id mt8188_mt6359_dt_match[] = { @@ -1471,7 +1396,7 @@ static struct platform_driver mt8188_mt6359_driver = { .of_match_table = mt8188_mt6359_dt_match, .pm = &snd_soc_pm_ops, }, - .probe = mt8188_mt6359_dev_probe, + .probe = mtk_soundcard_common_probe, }; module_platform_driver(mt8188_mt6359_driver); From 6718e1ed93cc28f0faefe21b7d21679ff0b04020 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:13:56 +0200 Subject: [PATCH 291/603] ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe Add mtk_soundcard_pdata platform data for the MediaTek common sound card probe mechanism, including a driver/soc-specific probe extension (used for bits that cannot be commonized hence specific to this driver), and change the probe function to mtk_soundcard_common_probe. This is also adding the possibility of specifying the links and routing with the audio-routing property and (x)-dai-link nodes in device trees to stop hardcoding machine specific links in the card driver assupported by the common probe function, but support for legacy device trees is retained with a legacy_probe function, which is used only in case the new properties are not found. Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-5-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 292 ++++++++++++---------- 1 file changed, 159 insertions(+), 133 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 53fd8a897b9d2..f694618e76354 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -22,6 +22,7 @@ #include "../common/mtk-afe-platform-driver.h" #include "../common/mtk-dsp-sof-common.h" #include "../common/mtk-soc-card.h" +#include "../common/mtk-soundcard-driver.h" #include "mt8195-afe-clk.h" #include "mt8195-afe-common.h" @@ -29,6 +30,13 @@ #define RT1019_SPEAKER_AMP_PRESENT BIT(1) #define MAX98390_SPEAKER_AMP_PRESENT BIT(2) +#define DUMB_CODEC_INIT BIT(0) +#define MT6359_CODEC_INIT BIT(1) +#define RT1011_CODEC_INIT BIT(2) +#define RT1019_CODEC_INIT BIT(3) +#define MAX98390_CODEC_INIT BIT(4) +#define RT5682_CODEC_INIT BIT(5) + #define RT1011_CODEC_DAI "rt1011-aif" #define RT1011_DEV0_NAME "rt1011.2-0038" #define RT1011_DEV1_NAME "rt1011.2-0039" @@ -51,18 +59,17 @@ #define SOF_DMA_UL4 "SOF_DMA_UL4" #define SOF_DMA_UL5 "SOF_DMA_UL5" -struct mt8195_card_data { - const char *name; - unsigned long quirk; -}; - struct mt8195_mt6359_priv { - struct snd_soc_jack headset_jack; - struct snd_soc_jack dp_jack; - struct snd_soc_jack hdmi_jack; struct clk *i2so1_mclk; }; +enum mt8195_jacks { + MT8195_JACK_HEADSET, + MT8195_JACK_DP, + MT8195_JACK_HDMI, + MT8195_JACK_MAX, +}; + /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin mt8195_jack_pins[] = { { @@ -382,33 +389,31 @@ static const struct snd_soc_ops mt8195_dptx_ops = { static int mt8195_dptx_codec_init(struct snd_soc_pcm_runtime *rtd) { struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); - struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8195_JACK_DP]; struct snd_soc_component *cmpnt_codec = snd_soc_rtd_to_codec(rtd, 0)->component; int ret; - ret = snd_soc_card_jack_new(rtd->card, "DP Jack", SND_JACK_LINEOUT, - &priv->dp_jack); + ret = snd_soc_card_jack_new(rtd->card, "DP Jack", SND_JACK_LINEOUT, jack); if (ret) return ret; - return snd_soc_component_set_jack(cmpnt_codec, &priv->dp_jack, NULL); + return snd_soc_component_set_jack(cmpnt_codec, jack, NULL); } static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd) { struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); - struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8195_JACK_HDMI]; struct snd_soc_component *cmpnt_codec = snd_soc_rtd_to_codec(rtd, 0)->component; int ret; - ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, - &priv->hdmi_jack); + ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, jack); if (ret) return ret; - return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL); + return snd_soc_component_set_jack(cmpnt_codec, jack, NULL); } static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -566,7 +571,7 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) snd_soc_rtd_to_codec(rtd, 0)->component; struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); struct mt8195_mt6359_priv *priv = soc_card_data->mach_priv; - struct snd_soc_jack *jack = &priv->headset_jack; + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8195_JACK_HEADSET]; struct snd_soc_component *cmpnt_afe = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe); @@ -687,7 +692,7 @@ static int mt8195_rt1011_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static int mt8195_rt1019_init(struct snd_soc_pcm_runtime *rtd) +static int mt8195_dumb_amp_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int ret; @@ -707,6 +712,18 @@ static int mt8195_rt1019_init(struct snd_soc_pcm_runtime *rtd) return ret; } + return 0; +} + +static int mt8195_rt1019_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = mt8195_dumb_amp_init(rtd); + if (ret) + return ret; + ret = snd_soc_dapm_add_routes(&card->dapm, mt8195_rt1019_routes, ARRAY_SIZE(mt8195_rt1019_routes)); if (ret) @@ -1371,108 +1388,31 @@ static int mt8195_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } -static int mt8195_mt6359_dev_probe(struct platform_device *pdev) +static int mt8195_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data) { - struct snd_soc_card *card = &mt8195_mt6359_soc_card; + struct mtk_platform_card_data *card_data = soc_card_data->card_data; + struct snd_soc_card *card = card_data->card; + struct device_node *codec_node, *dp_node, *hdmi_node; struct snd_soc_dai_link *dai_link; - struct mtk_soc_card_data *soc_card_data; - struct mt8195_mt6359_priv *mach_priv; - struct device_node *platform_node, *adsp_node, *codec_node, *dp_node, *hdmi_node; - struct mt8195_card_data *card_data; - int is5682s = 0; - int init6359 = 0; - int sof_on = 0; - int ret, i; - - card_data = (struct mt8195_card_data *)of_device_get_match_data(&pdev->dev); - card->dev = &pdev->dev; - - ret = snd_soc_of_parse_card_name(card, "model"); - if (ret) { - dev_err(&pdev->dev, "%s new card name parsing error %d\n", - __func__, ret); - return ret; - } - - if (!card->name) - card->name = card_data->name; + struct device *dev = card->dev; + bool is5682s, init6359 = false; + int i; if (strstr(card->name, "_5682s")) { codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682s"); - is5682s = 1; - } else - codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682i"); - - soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*card_data), GFP_KERNEL); - if (!soc_card_data) - return -ENOMEM; - - mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL); - if (!mach_priv) - return -ENOMEM; - - soc_card_data->mach_priv = mach_priv; - - adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); - if (adsp_node) { - struct mtk_sof_priv *sof_priv; - - sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL); - if (!sof_priv) { - ret = -ENOMEM; - goto err_kzalloc; - } - sof_priv->conn_streams = g_sof_conn_streams; - sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams); - sof_priv->sof_dai_link_fixup = mt8195_dai_link_fixup; - soc_card_data->sof_priv = sof_priv; - card->probe = mtk_sof_card_probe; - card->late_probe = mtk_sof_card_late_probe; - if (!card->topology_shortname_created) { - snprintf(card->topology_shortname, 32, "sof-%s", card->name); - card->topology_shortname_created = true; - } - card->name = card->topology_shortname; - sof_on = 1; - } - - if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { - ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, - "mediatek,dai-link", - mt8195_mt6359_dai_links, - ARRAY_SIZE(mt8195_mt6359_dai_links)); - if (ret) { - dev_dbg(&pdev->dev, "Parse dai-link fail\n"); - goto err_parse_of; - } + is5682s = true; } else { - if (!sof_on) - card->num_links = DAI_LINK_REGULAR_NUM; - } - - platform_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,platform", 0); - if (!platform_node) { - dev_dbg(&pdev->dev, "Property 'platform' missing or invalid\n"); - ret = -EINVAL; - goto err_platform_node; + codec_node = of_find_compatible_node(NULL, NULL, "realtek,rt5682i"); + is5682s = false; } - dp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,dptx-codec", 0); - hdmi_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,hdmi-codec", 0); + dp_node = of_parse_phandle(dev->of_node, "mediatek,dptx-codec", 0); + hdmi_node = of_parse_phandle(dev->of_node, "mediatek,hdmi-codec", 0); for_each_card_prelinks(card, i, dai_link) { - if (!dai_link->platforms->name) { - if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && sof_on) - dai_link->platforms->of_node = adsp_node; - else - dai_link->platforms->of_node = platform_node; - } - if (strcmp(dai_link->name, "DPTX_BE") == 0) { if (!dp_node) { - dev_dbg(&pdev->dev, "No property 'dptx-codec'\n"); + dev_dbg(dev, "No property 'dptx-codec'\n"); } else { dai_link->codecs->of_node = dp_node; dai_link->codecs->name = NULL; @@ -1481,7 +1421,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) } } else if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { if (!hdmi_node) { - dev_dbg(&pdev->dev, "No property 'hdmi-codec'\n"); + dev_dbg(dev, "No property 'hdmi-codec'\n"); } else { dai_link->codecs->of_node = hdmi_node; dai_link->codecs->name = NULL; @@ -1490,7 +1430,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) } } else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0) { if (!codec_node) { - dev_err(&pdev->dev, "Codec not found!\n"); + dev_err(dev, "Codec not found!\n"); } else { dai_link->codecs->of_node = codec_node; dai_link->codecs->name = NULL; @@ -1501,7 +1441,7 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) } } else if (strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { if (!codec_node) { - dev_err(&pdev->dev, "Codec not found!\n"); + dev_err(dev, "Codec not found!\n"); } else { dai_link->codecs->of_node = codec_node; dai_link->codecs->name = NULL; @@ -1514,10 +1454,10 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) strcmp(dai_link->name, "UL_SRC2_BE") == 0) { if (!init6359) { dai_link->init = mt8195_mt6359_init; - init6359 = 1; + init6359 = true; } } else if (strcmp(dai_link->name, "ETDM2_OUT_BE") == 0) { - switch (card_data->quirk) { + switch (card_data->flags) { case RT1011_SPEAKER_AMP_PRESENT: dai_link->codecs = rt1011_comps; dai_link->num_codecs = ARRAY_SIZE(rt1011_comps); @@ -1545,33 +1485,119 @@ static int mt8195_mt6359_dev_probe(struct platform_device *pdev) } } - snd_soc_card_set_drvdata(card, soc_card_data); + return 0; +} - ret = devm_snd_soc_register_card(&pdev->dev, card); +static int mt8195_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy) +{ + struct mtk_platform_card_data *card_data = soc_card_data->card_data; + struct snd_soc_card *card = card_data->card; + struct mt8195_mt6359_priv *mach_priv; + struct snd_soc_dai_link *dai_link; + u8 codec_init = 0; + int i; - of_node_put(platform_node); - of_node_put(dp_node); - of_node_put(hdmi_node); -err_kzalloc: -err_parse_of: -err_platform_node: - of_node_put(adsp_node); - return ret; + mach_priv = devm_kzalloc(card->dev, sizeof(*mach_priv), GFP_KERNEL); + if (!mach_priv) + return -ENOMEM; + + soc_card_data->mach_priv = mach_priv; + + if (legacy) + return mt8195_mt6359_legacy_probe(soc_card_data); + + for_each_card_prelinks(card, i, dai_link) { + if (strcmp(dai_link->name, "DPTX_BE") == 0) { + if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + dai_link->init = mt8195_dptx_codec_init; + } else if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { + if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + dai_link->init = mt8195_hdmi_codec_init; + } else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 || + strcmp(dai_link->name, "UL_SRC1_BE") == 0 || + strcmp(dai_link->name, "UL_SRC2_BE") == 0) { + if (!(codec_init & MT6359_CODEC_INIT)) { + dai_link->init = mt8195_mt6359_init; + codec_init |= MT6359_CODEC_INIT; + } + } else if (strcmp(dai_link->name, "ETDM1_OUT_BE") == 0 || + strcmp(dai_link->name, "ETDM2_OUT_BE") == 0 || + strcmp(dai_link->name, "ETDM1_IN_BE") == 0 || + strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { + if (!strcmp(dai_link->codecs->dai_name, MAX98390_CODEC_DAI)) { + if (!(codec_init & MAX98390_CODEC_INIT)) { + dai_link->init = mt8195_max98390_init; + codec_init |= MAX98390_CODEC_INIT; + } + } else if (!strcmp(dai_link->codecs->dai_name, RT1011_CODEC_DAI)) { + dai_link->ops = &mt8195_rt1011_etdm_ops; + if (!(codec_init & RT1011_CODEC_INIT)) { + dai_link->init = mt8195_rt1011_init; + codec_init |= RT1011_CODEC_INIT; + } + } else if (!strcmp(dai_link->codecs->dai_name, RT1019_CODEC_DAI)) { + if (!(codec_init & RT1019_CODEC_INIT)) { + dai_link->init = mt8195_rt1019_init; + codec_init |= RT1019_CODEC_INIT; + } + } else if (!strcmp(dai_link->codecs->dai_name, RT5682_CODEC_DAI) || + !strcmp(dai_link->codecs->dai_name, RT5682S_CODEC_DAI)) { + dai_link->ops = &mt8195_rt5682_etdm_ops; + if (!(codec_init & RT5682_CODEC_INIT)) { + dai_link->init = mt8195_rt5682_init; + codec_init |= RT5682_CODEC_INIT; + } + } else { + if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) { + if (!(codec_init & DUMB_CODEC_INIT)) { + dai_link->init = mt8195_dumb_amp_init; + codec_init |= DUMB_CODEC_INIT; + } + } + } + } + } + + return 0; } -static struct mt8195_card_data mt8195_mt6359_rt1019_rt5682_card = { - .name = "mt8195_r1019_5682", - .quirk = RT1019_SPEAKER_AMP_PRESENT, +static const struct mtk_sof_priv mt8195_sof_priv = { + .conn_streams = g_sof_conn_streams, + .num_streams = ARRAY_SIZE(g_sof_conn_streams), + .sof_dai_link_fixup = mt8195_dai_link_fixup }; -static struct mt8195_card_data mt8195_mt6359_rt1011_rt5682_card = { - .name = "mt8195_r1011_5682", - .quirk = RT1011_SPEAKER_AMP_PRESENT, +static const struct mtk_soundcard_pdata mt8195_mt6359_rt1019_rt5682_card = { + .card_name = "mt8195_r1019_5682", + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8195_mt6359_soc_card, + .num_jacks = MT8195_JACK_MAX, + .flags = RT1019_SPEAKER_AMP_PRESENT + }, + .sof_priv = &mt8195_sof_priv, + .soc_probe = mt8195_mt6359_soc_card_probe }; -static struct mt8195_card_data mt8195_mt6359_max98390_rt5682_card = { - .name = "mt8195_m98390_r5682", - .quirk = MAX98390_SPEAKER_AMP_PRESENT, +static const struct mtk_soundcard_pdata mt8195_mt6359_rt1011_rt5682_card = { + .card_name = "mt8195_r1011_5682", + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8195_mt6359_soc_card, + .num_jacks = MT8195_JACK_MAX, + .flags = RT1011_SPEAKER_AMP_PRESENT + }, + .sof_priv = &mt8195_sof_priv, + .soc_probe = mt8195_mt6359_soc_card_probe +}; + +static const struct mtk_soundcard_pdata mt8195_mt6359_max98390_rt5682_card = { + .card_name = "mt8195_m98390_r5682", + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8195_mt6359_soc_card, + .num_jacks = MT8195_JACK_MAX, + .flags = MAX98390_SPEAKER_AMP_PRESENT + }, + .sof_priv = &mt8195_sof_priv, + .soc_probe = mt8195_mt6359_soc_card_probe }; static const struct of_device_id mt8195_mt6359_dt_match[] = { @@ -1597,7 +1623,7 @@ static struct platform_driver mt8195_mt6359_driver = { .of_match_table = mt8195_mt6359_dt_match, .pm = &snd_soc_pm_ops, }, - .probe = mt8195_mt6359_dev_probe, + .probe = mtk_soundcard_common_probe, }; module_platform_driver(mt8195_mt6359_driver); From 2d72cbb56327205ae04a9376c5b78a35b6347294 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:13:57 +0200 Subject: [PATCH 292/603] ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe Add mtk_soundcard_pdata platform data for the MediaTek common sound card probe mechanism, including a driver/soc-specific probe extension (used for bits that cannot be commonized hence specific to this driver), and change the probe function to mtk_soundcard_common_probe. This is also adding the possibility of specifying the links and routing with the audio-routing property and (x)-dai-link nodes in device trees to stop hardcoding machine specific links in the card driver assupported by the common probe function, but support for legacy device trees is retained with a legacy_probe function, which is used only in case the new properties are not found. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-6-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 164 ++++++++++-------- 1 file changed, 91 insertions(+), 73 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index bfcb2c486c39d..645bc1aa67c7f 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -20,6 +20,8 @@ #include "../../codecs/rt1015.h" #include "../../codecs/rt5682.h" #include "../common/mtk-afe-platform-driver.h" +#include "../common/mtk-soc-card.h" +#include "../common/mtk-soundcard-driver.h" #include "mt8192-afe-common.h" #include "mt8192-afe-clk.h" #include "mt8192-afe-gpio.h" @@ -38,9 +40,10 @@ #define RT1015P_RT5682_OF_NAME "mediatek,mt8192_mt6359_rt1015p_rt5682" #define RT1015P_RT5682S_OF_NAME "mediatek,mt8192_mt6359_rt1015p_rt5682s" -struct mt8192_mt6359_priv { - struct snd_soc_jack headset_jack; - struct snd_soc_jack hdmi_jack; +enum mt8192_jacks { + MT8192_JACK_HEADSET, + MT8192_JACK_HDMI, + MT8192_JACK_MAX, }; /* Headset jack detection DAPM pins */ @@ -323,13 +326,13 @@ static int mt8192_mt6359_init(struct snd_soc_pcm_runtime *rtd) static int mt8192_rt5682_init(struct snd_soc_pcm_runtime *rtd) { + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8192_JACK_HEADSET]; struct snd_soc_component *cmpnt_afe = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe); struct snd_soc_component *cmpnt_codec = snd_soc_rtd_to_codec(rtd, 0)->component; - struct mt8192_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_jack *jack = &priv->headset_jack; int ret; ret = mt8192_dai_i2s_set_share(afe, "I2S8", "I2S9"); @@ -359,19 +362,19 @@ static int mt8192_rt5682_init(struct snd_soc_pcm_runtime *rtd) static int mt8192_mt6359_hdmi_init(struct snd_soc_pcm_runtime *rtd) { + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8192_JACK_HDMI]; struct snd_soc_component *cmpnt_codec = snd_soc_rtd_to_codec(rtd, 0)->component; - struct mt8192_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); int ret; - ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, - &priv->hdmi_jack); + ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, jack); if (ret) { dev_err(rtd->dev, "HDMI Jack creation failed: %d\n", ret); return ret; } - return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL); + return snd_soc_component_set_jack(cmpnt_codec, jack, NULL); } static int mt8192_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -1136,71 +1139,53 @@ static int mt8192_mt6359_card_set_be_link(struct snd_soc_card *card, return 0; } -static int mt8192_mt6359_dev_probe(struct platform_device *pdev) +static int mt8192_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data) { - struct snd_soc_card *card; - struct device_node *platform_node, *hdmi_codec, *headset_codec, *speaker_codec; - int ret, i; + struct mtk_platform_card_data *card_data = soc_card_data->card_data; + struct snd_soc_card *card = card_data->card; + struct device *dev = card->dev; + struct device_node *hdmi_codec, *headset_codec, *speaker_codec; struct snd_soc_dai_link *dai_link; - struct mt8192_mt6359_priv *priv; - - card = (struct snd_soc_card *)of_device_get_match_data(&pdev->dev); - if (!card) - return -EINVAL; - card->dev = &pdev->dev; - - if (of_device_is_compatible(pdev->dev.of_node, RT1015P_RT5682_OF_NAME)) - card->name = RT1015P_RT5682_CARD_NAME; - else if (of_device_is_compatible(pdev->dev.of_node, RT1015P_RT5682S_OF_NAME)) - card->name = RT1015P_RT5682S_CARD_NAME; - else - dev_dbg(&pdev->dev, "No need to set card name\n"); + int ret, i; - hdmi_codec = of_parse_phandle(pdev->dev.of_node, "mediatek,hdmi-codec", 0); + hdmi_codec = of_parse_phandle(dev->of_node, "mediatek,hdmi-codec", 0); if (!hdmi_codec) - dev_dbg(&pdev->dev, "The machine has no hdmi-codec\n"); + dev_dbg(dev, "The machine has no hdmi-codec\n"); - platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0); - if (!platform_node) { - ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'platform' missing or invalid\n"); - goto err_platform_node; - } - - speaker_codec = of_get_child_by_name(pdev->dev.of_node, "speaker-codecs"); + speaker_codec = of_get_child_by_name(dev->of_node, "speaker-codecs"); if (!speaker_codec) { ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'speaker-codecs' missing or invalid\n"); + dev_err_probe(dev, ret, "Property 'speaker-codecs' missing or invalid\n"); goto err_speaker_codec; } - headset_codec = of_get_child_by_name(pdev->dev.of_node, "headset-codec"); + headset_codec = of_get_child_by_name(dev->of_node, "headset-codec"); if (!headset_codec) { ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'headset-codec' missing or invalid\n"); + dev_err_probe(dev, ret, "Property 'headset-codec' missing or invalid\n"); goto err_headset_codec; } for_each_card_prelinks(card, i, dai_link) { ret = mt8192_mt6359_card_set_be_link(card, dai_link, speaker_codec, "I2S3"); if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set speaker_codec fail\n", + dev_err_probe(dev, ret, "%s set speaker_codec fail\n", dai_link->name); - goto err_probe; + break; } ret = mt8192_mt6359_card_set_be_link(card, dai_link, headset_codec, "I2S8"); if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n", + dev_err_probe(dev, ret, "%s set headset_codec fail\n", dai_link->name); - goto err_probe; + break; } ret = mt8192_mt6359_card_set_be_link(card, dai_link, headset_codec, "I2S9"); if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n", + dev_err_probe(dev, ret, "%s set headset_codec fail\n", dai_link->name); - goto err_probe; + break; } if (hdmi_codec && strcmp(dai_link->name, "TDM") == 0) { @@ -1211,52 +1196,85 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev) if (dai_link->num_codecs && dai_link->codecs[0].dai_name && strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0) dai_link->ops = &mt8192_rt1015_i2s_ops; - - if (!dai_link->platforms->name) - dai_link->platforms->of_node = platform_node; - } - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) { - ret = -ENOMEM; - goto err_probe; - } - snd_soc_card_set_drvdata(card, priv); - - ret = mt8192_afe_gpio_init(&pdev->dev); - if (ret) { - dev_err_probe(&pdev->dev, ret, "%s init gpio error\n", __func__); - goto err_probe; } - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", __func__); - -err_probe: of_node_put(headset_codec); err_headset_codec: of_node_put(speaker_codec); err_speaker_codec: - of_node_put(platform_node); -err_platform_node: - of_node_put(hdmi_codec); + if (hdmi_codec) + of_node_put(hdmi_codec); + return ret; } +static int mt8192_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy) +{ + struct mtk_platform_card_data *card_data = soc_card_data->card_data; + struct snd_soc_card *card = card_data->card; + int ret; + + if (legacy) { + ret = mt8192_mt6359_legacy_probe(soc_card_data); + if (ret) + return ret; + } else { + struct snd_soc_dai_link *dai_link; + int i; + + for_each_card_prelinks(card, i, dai_link) + if (dai_link->num_codecs && dai_link->codecs[0].dai_name && + strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0) + dai_link->ops = &mt8192_rt1015_i2s_ops; + } + + ret = mt8192_afe_gpio_init(card->dev); + if (ret) + return dev_err_probe(card->dev, ret, "%s init gpio error\n", __func__); + + return 0; +} + +static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015_rt5682_pdata = { + .card_name = RT1015_RT5682_CARD_NAME, + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8192_mt6359_rt1015_rt5682_card, + .num_jacks = MT8192_JACK_MAX, + }, + .soc_probe = mt8192_mt6359_soc_card_probe +}; + +static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015p_rt5682_pdata = { + .card_name = RT1015P_RT5682_CARD_NAME, + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8192_mt6359_rt1015p_rt5682x_card, + .num_jacks = MT8192_JACK_MAX, + }, + .soc_probe = mt8192_mt6359_soc_card_probe +}; + +static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015p_rt5682s_pdata = { + .card_name = RT1015P_RT5682S_CARD_NAME, + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8192_mt6359_rt1015p_rt5682x_card, + .num_jacks = MT8192_JACK_MAX, + }, + .soc_probe = mt8192_mt6359_soc_card_probe +}; + #ifdef CONFIG_OF static const struct of_device_id mt8192_mt6359_dt_match[] = { { .compatible = RT1015_RT5682_OF_NAME, - .data = &mt8192_mt6359_rt1015_rt5682_card, + .data = &mt8192_mt6359_rt1015_rt5682_pdata, }, { .compatible = RT1015P_RT5682_OF_NAME, - .data = &mt8192_mt6359_rt1015p_rt5682x_card, + .data = &mt8192_mt6359_rt1015p_rt5682_pdata, }, { .compatible = RT1015P_RT5682S_OF_NAME, - .data = &mt8192_mt6359_rt1015p_rt5682x_card, + .data = &mt8192_mt6359_rt1015p_rt5682s_pdata, }, {} }; @@ -1276,7 +1294,7 @@ static struct platform_driver mt8192_mt6359_driver = { #endif .pm = &mt8192_mt6359_pm_ops, }, - .probe = mt8192_mt6359_dev_probe, + .probe = mtk_soundcard_common_probe, }; module_platform_driver(mt8192_mt6359_driver); From 44a53c8ec9cad02ce7dfd55094e8d7aeb48eb73c Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:13:58 +0200 Subject: [PATCH 293/603] ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe Add mtk_soundcard_pdata platform data for the MediaTek common sound card probe mechanism, including a driver/soc-specific probe extension (used for bits that cannot be commonized hence specific to this driver), and change the probe function to mtk_soundcard_common_probe. This is also adding the possibility of specifying the links and routing with the audio-routing property and (x)-dai-link nodes in device trees to stop hardcoding machine specific links in the card driver assupported by the common probe function, but support for legacy device trees is retained with a legacy_probe function, which is used only in case the new properties are not found. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-7-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 220 ++++++++---------- 1 file changed, 96 insertions(+), 124 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c index f78197c8e582b..f8856a100b276 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c @@ -21,6 +21,7 @@ #include "../common/mtk-afe-platform-driver.h" #include "../common/mtk-dsp-sof-common.h" #include "../common/mtk-soc-card.h" +#include "../common/mtk-soundcard-driver.h" #include "mt8186-afe-common.h" #include "mt8186-afe-clk.h" #include "mt8186-afe-gpio.h" @@ -38,11 +39,16 @@ #define SOF_DMA_UL2 "SOF_DMA_UL2" struct mt8186_mt6366_rt1019_rt5682s_priv { - struct snd_soc_jack headset_jack, hdmi_jack; struct gpio_desc *dmic_sel; int dmic_switch; }; +enum mt8186_jacks { + MT8186_JACK_HEADSET, + MT8186_JACK_HDMI, + MT8186_JACK_MAX, +}; + /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin mt8186_jack_pins[] = { { @@ -165,8 +171,7 @@ static int mt8186_rt5682s_init(struct snd_soc_pcm_runtime *rtd) struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe); struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); - struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv; - struct snd_soc_jack *jack = &priv->headset_jack; + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HEADSET]; struct snd_soc_component *cmpnt_codec = snd_soc_rtd_to_codec(rtd, 0)->component; int ret; @@ -257,7 +262,7 @@ static int mt8186_mt6366_rt1019_rt5682s_hdmi_init(struct snd_soc_pcm_runtime *rt snd_soc_rtd_to_codec(rtd, 0)->component; struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); - struct mt8186_mt6366_rt1019_rt5682s_priv *priv = soc_card_data->mach_priv; + struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HDMI]; int ret; ret = mt8186_dai_i2s_set_share(afe, "I2S2", "I2S3"); @@ -266,13 +271,13 @@ static int mt8186_mt6366_rt1019_rt5682s_hdmi_init(struct snd_soc_pcm_runtime *rt return ret; } - ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, &priv->hdmi_jack); + ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, jack); if (ret) { dev_err(rtd->dev, "HDMI Jack creation failed: %d\n", ret); return ret; } - return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL); + return snd_soc_component_set_jack(cmpnt_codec, jack, NULL); } static int mt8186_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -1134,166 +1139,133 @@ static struct snd_soc_card mt8186_mt6366_rt5650_soc_card = { .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf), }; -static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev) +static int mt8186_mt6366_legacy_probe(struct mtk_soc_card_data *soc_card_data) { - struct snd_soc_card *card; + struct mtk_platform_card_data *card_data = soc_card_data->card_data; + struct snd_soc_card *card = card_data->card; + struct device *dev = card->dev; struct snd_soc_dai_link *dai_link; - struct mtk_soc_card_data *soc_card_data; - struct mt8186_mt6366_rt1019_rt5682s_priv *mach_priv; - struct device_node *platform_node, *headset_codec, *playback_codec, *adsp_node; - int sof_on = 0; + struct device_node *headset_codec, *playback_codec; int ret, i; - card = (struct snd_soc_card *)device_get_match_data(&pdev->dev); - if (!card) - return -EINVAL; - card->dev = &pdev->dev; - - soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*soc_card_data), GFP_KERNEL); - if (!soc_card_data) - return -ENOMEM; - mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL); - if (!mach_priv) - return -ENOMEM; - - soc_card_data->mach_priv = mach_priv; - - mach_priv->dmic_sel = devm_gpiod_get_optional(&pdev->dev, - "dmic", GPIOD_OUT_LOW); - if (IS_ERR(mach_priv->dmic_sel)) { - dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", - PTR_ERR(mach_priv->dmic_sel)); - return PTR_ERR(mach_priv->dmic_sel); - } - - adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); - if (adsp_node) { - struct mtk_sof_priv *sof_priv; + playback_codec = of_get_child_by_name(dev->of_node, "playback-codecs"); + if (!playback_codec) + return dev_err_probe(dev, -EINVAL, + "Property 'playback-codecs' missing or invalid\n"); - sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL); - if (!sof_priv) { - ret = -ENOMEM; - goto err_adsp_node; - } - sof_priv->conn_streams = g_sof_conn_streams; - sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams); - sof_priv->sof_dai_link_fixup = mt8186_sof_dai_link_fixup; - soc_card_data->sof_priv = sof_priv; - card->probe = mtk_sof_card_probe; - card->late_probe = mtk_sof_card_late_probe; - if (!card->topology_shortname_created) { - snprintf(card->topology_shortname, 32, "sof-%s", card->name); - card->topology_shortname_created = true; - } - card->name = card->topology_shortname; - sof_on = 1; - } else { - dev_dbg(&pdev->dev, "Probe without adsp\n"); - } - - if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { - ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, - "mediatek,dai-link", - mt8186_mt6366_rt1019_rt5682s_dai_links, - ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links)); - if (ret) { - dev_dbg(&pdev->dev, "Parse dai-link fail\n"); - goto err_adsp_node; - } - } else { - if (!sof_on) - card->num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links) - - ARRAY_SIZE(g_sof_conn_streams); - } - - platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0); - if (!platform_node) { - ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'platform' missing or invalid\n"); - goto err_platform_node; - } - - playback_codec = of_get_child_by_name(pdev->dev.of_node, "playback-codecs"); - if (!playback_codec) { - ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'playback-codecs' missing or invalid\n"); - goto err_playback_codec; - } - - headset_codec = of_get_child_by_name(pdev->dev.of_node, "headset-codec"); + headset_codec = of_get_child_by_name(dev->of_node, "headset-codec"); if (!headset_codec) { - ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'headset-codec' missing or invalid\n"); - goto err_headset_codec; + of_node_put(playback_codec); + return dev_err_probe(dev, -EINVAL, + "Property 'headset-codec' missing or invalid\n"); } for_each_card_prelinks(card, i, dai_link) { ret = mt8186_mt6366_card_set_be_link(card, dai_link, playback_codec, "I2S3"); if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set playback_codec fail\n", + dev_err_probe(dev, ret, "%s set playback_codec fail\n", dai_link->name); - goto err_probe; + break; } ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S0"); if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n", + dev_err_probe(dev, ret, "%s set headset_codec fail\n", dai_link->name); - goto err_probe; + break; } ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S1"); if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n", + dev_err_probe(dev, ret, "%s set headset_codec fail\n", dai_link->name); - goto err_probe; + break; } + } + of_node_put(headset_codec); + of_node_put(playback_codec); - if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && sof_on) - dai_link->platforms->of_node = adsp_node; + return ret; +} - if (!dai_link->platforms->name && !dai_link->platforms->of_node) - dai_link->platforms->of_node = platform_node; - } +static int mt8186_mt6366_soc_card_probe(struct mtk_soc_card_data *soc_card_data, bool legacy) +{ + struct mtk_platform_card_data *card_data = soc_card_data->card_data; + struct snd_soc_card *card = card_data->card; + struct mt8186_mt6366_rt1019_rt5682s_priv *mach_priv; + int ret; - snd_soc_card_set_drvdata(card, soc_card_data); + mach_priv = devm_kzalloc(card->dev, sizeof(*mach_priv), GFP_KERNEL); + if (!mach_priv) + return -ENOMEM; - ret = mt8186_afe_gpio_init(&pdev->dev); - if (ret) { - dev_err_probe(&pdev->dev, ret, "%s init gpio error\n", __func__); - goto err_probe; + soc_card_data->mach_priv = mach_priv; + + mach_priv->dmic_sel = devm_gpiod_get_optional(card->dev, + "dmic", GPIOD_OUT_LOW); + if (IS_ERR(mach_priv->dmic_sel)) + return dev_err_probe(card->dev, PTR_ERR(mach_priv->dmic_sel), + "DMIC gpio failed\n"); + + if (legacy) { + ret = mt8186_mt6366_legacy_probe(soc_card_data); + if (ret) + return ret; } - ret = devm_snd_soc_register_card(&pdev->dev, card); + ret = mt8186_afe_gpio_init(card->dev); if (ret) - dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", __func__); + return dev_err_probe(card->dev, ret, "init AFE gpio error\n"); -err_probe: - of_node_put(headset_codec); -err_headset_codec: - of_node_put(playback_codec); -err_playback_codec: - of_node_put(platform_node); -err_platform_node: -err_adsp_node: - of_node_put(adsp_node); - - return ret; + return 0; } +static const struct mtk_sof_priv mt8186_sof_priv = { + .conn_streams = g_sof_conn_streams, + .num_streams = ARRAY_SIZE(g_sof_conn_streams), + .sof_dai_link_fixup = mt8186_sof_dai_link_fixup +}; + +static const struct mtk_soundcard_pdata mt8186_mt6366_rt1019_rt5682s_pdata = { + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8186_mt6366_rt1019_rt5682s_soc_card, + .num_jacks = MT8186_JACK_MAX, + }, + .sof_priv = &mt8186_sof_priv, + .soc_probe = mt8186_mt6366_soc_card_probe +}; + +static const struct mtk_soundcard_pdata mt8186_mt6366_rt5682s_max98360_pdata = { + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8186_mt6366_rt5682s_max98360_soc_card, + .num_jacks = MT8186_JACK_MAX, + }, + .sof_priv = &mt8186_sof_priv, + .soc_probe = mt8186_mt6366_soc_card_probe +}; + +static const struct mtk_soundcard_pdata mt8186_mt6366_rt5650_pdata = { + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8186_mt6366_rt5650_soc_card, + .num_jacks = MT8186_JACK_MAX, + }, + .sof_priv = &mt8186_sof_priv, + .soc_probe = mt8186_mt6366_soc_card_probe +}; + #if IS_ENABLED(CONFIG_OF) static const struct of_device_id mt8186_mt6366_rt1019_rt5682s_dt_match[] = { { .compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound", - .data = &mt8186_mt6366_rt1019_rt5682s_soc_card, + .data = &mt8186_mt6366_rt1019_rt5682s_pdata, }, { .compatible = "mediatek,mt8186-mt6366-rt5682s-max98360-sound", - .data = &mt8186_mt6366_rt5682s_max98360_soc_card, + .data = &mt8186_mt6366_rt5682s_max98360_pdata, }, { .compatible = "mediatek,mt8186-mt6366-rt5650-sound", - .data = &mt8186_mt6366_rt5650_soc_card, + .data = &mt8186_mt6366_rt5650_pdata, }, {} }; @@ -1308,7 +1280,7 @@ static struct platform_driver mt8186_mt6366_rt1019_rt5682s_driver = { #endif .pm = &snd_soc_pm_ops, }, - .probe = mt8186_mt6366_rt1019_rt5682s_dev_probe, + .probe = mtk_soundcard_common_probe, }; module_platform_driver(mt8186_mt6366_rt1019_rt5682s_driver); From fe071237dee6d617e5cb0eb4856b23576a877479 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:13:59 +0200 Subject: [PATCH 294/603] ASoC: mediatek: Add common snd_soc_ops .startup() callback MediaTek platforms are typically setting PCM rate and channels constraints for playback, capture and HDMI/DisplayPort playback: commonize the startup callback by adding the PCM constraints data to the mtk_platform_card_data structure and by reusing the common mtk_soundcard_startup() function for all of them by getting back the parameters from the aforementioned struct. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-8-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../mediatek/common/mtk-soundcard-driver.c | 51 +++++++++++++++++++ .../mediatek/common/mtk-soundcard-driver.h | 24 +++++++++ 2 files changed, 75 insertions(+) diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.c b/sound/soc/mediatek/common/mtk-soundcard-driver.c index b1db17e392d52..3bbf42c42805f 100644 --- a/sound/soc/mediatek/common/mtk-soundcard-driver.c +++ b/sound/soc/mediatek/common/mtk-soundcard-driver.c @@ -139,6 +139,57 @@ void clean_card_reference(struct snd_soc_card *card) } EXPORT_SYMBOL_GPL(clean_card_reference); +int mtk_soundcard_startup(struct snd_pcm_substream *substream, + enum mtk_pcm_constraint_type ctype) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct mtk_soc_card_data *soc_card = snd_soc_card_get_drvdata(rtd->card); + const struct mtk_pcm_constraints_data *mpc = &soc_card->card_data->pcm_constraints[ctype]; + int ret; + + if (unlikely(!mpc)) + return -EINVAL; + + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + mpc->rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + mpc->channels); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list channel failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(mtk_soundcard_startup); + +static int mtk_soundcard_playback_startup(struct snd_pcm_substream *substream) +{ + return mtk_soundcard_startup(substream, MTK_CONSTRAINT_PLAYBACK); +} + +const struct snd_soc_ops mtk_soundcard_common_playback_ops = { + .startup = mtk_soundcard_playback_startup, +}; +EXPORT_SYMBOL_GPL(mtk_soundcard_common_playback_ops); + +static int mtk_soundcard_capture_startup(struct snd_pcm_substream *substream) +{ + return mtk_soundcard_startup(substream, MTK_CONSTRAINT_CAPTURE); +} + +const struct snd_soc_ops mtk_soundcard_common_capture_ops = { + .startup = mtk_soundcard_capture_startup, +}; +EXPORT_SYMBOL_GPL(mtk_soundcard_common_capture_ops); + int mtk_soundcard_common_probe(struct platform_device *pdev) { struct device_node *platform_node, *adsp_node; diff --git a/sound/soc/mediatek/common/mtk-soundcard-driver.h b/sound/soc/mediatek/common/mtk-soundcard-driver.h index 4fd2ffb7e486f..f6c275b540253 100644 --- a/sound/soc/mediatek/common/mtk-soundcard-driver.h +++ b/sound/soc/mediatek/common/mtk-soundcard-driver.h @@ -11,11 +11,26 @@ struct mtk_sof_priv; struct mtk_soc_card_data; +struct snd_pcm_hw_constraint_list; + +enum mtk_pcm_constraint_type { + MTK_CONSTRAINT_PLAYBACK, + MTK_CONSTRAINT_CAPTURE, + MTK_CONSTRAINT_HDMIDP, + MTK_CONSTRAINT_MAX +}; + +struct mtk_pcm_constraints_data { + const struct snd_pcm_hw_constraint_list *channels; + const struct snd_pcm_hw_constraint_list *rates; +}; struct mtk_platform_card_data { struct snd_soc_card *card; struct snd_soc_jack *jacks; + const struct mtk_pcm_constraints_data *pcm_constraints; u8 num_jacks; + u8 num_pcm_constraints; u8 flags; }; @@ -23,9 +38,18 @@ struct mtk_soundcard_pdata { const char *card_name; struct mtk_platform_card_data *card_data; const struct mtk_sof_priv *sof_priv; + int (*soc_probe)(struct mtk_soc_card_data *card_data, bool legacy); }; +/* Common playback/capture card startup ops */ +extern const struct snd_soc_ops mtk_soundcard_common_playback_ops; +extern const struct snd_soc_ops mtk_soundcard_common_capture_ops; + +/* Exported for custom/extended soundcard startup ops */ +int mtk_soundcard_startup(struct snd_pcm_substream *substream, + enum mtk_pcm_constraint_type ctype); + int parse_dai_link_info(struct snd_soc_card *card); void clean_card_reference(struct snd_soc_card *card); int mtk_soundcard_common_probe(struct platform_device *pdev); From 386fe4ceaf196360ef96cbd68d045ed2730251a7 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:00 +0200 Subject: [PATCH 295/603] ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup Add a const mtk_pcm_constraints_data struct array with all of the (again, constant) constraints for all of the supported usecases, remove the duplicated functions and call mtk_soundcard_startup() instead in all of the .startup() callbacks. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-9-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 195 ++++++---------------- 1 file changed, 53 insertions(+), 142 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index f694618e76354..62e4e34cffbbf 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -328,44 +328,7 @@ static int mt8195_mt6359_init(struct snd_soc_pcm_runtime *rtd) static int mt8195_hdmitx_dptx_startup(struct snd_pcm_substream *substream) { - static const unsigned int rates[] = { - 48000 - }; - static const unsigned int channels[] = { - 2, 4, 6, 8 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channel failed\n"); - return ret; - } - - return 0; + return mtk_soundcard_startup(substream, MTK_CONSTRAINT_HDMIDP); } static const struct snd_soc_ops mt8195_hdmitx_dptx_playback_ops = { @@ -428,98 +391,6 @@ static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int mt8195_playback_startup(struct snd_pcm_substream *substream) -{ - static const unsigned int rates[] = { - 48000 - }; - static const unsigned int channels[] = { - 2 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channel failed\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_ops mt8195_playback_ops = { - .startup = mt8195_playback_startup, -}; - -static int mt8195_capture_startup(struct snd_pcm_substream *substream) -{ - static const unsigned int rates[] = { - 48000 - }; - static const unsigned int channels[] = { - 1, 2 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channel failed\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_ops mt8195_capture_ops = { - .startup = mt8195_capture_startup, -}; - static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -1042,7 +913,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, - .ops = &mt8195_playback_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(DL2_FE), }, [DAI_LINK_DL3_FE] = { @@ -1054,7 +925,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, - .ops = &mt8195_playback_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(DL3_FE), }, [DAI_LINK_DL6_FE] = { @@ -1066,7 +937,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, - .ops = &mt8195_playback_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(DL6_FE), }, [DAI_LINK_DL7_FE] = { @@ -1089,7 +960,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, - .ops = &mt8195_playback_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(DL8_FE), }, [DAI_LINK_DL10_FE] = { @@ -1113,7 +984,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, - .ops = &mt8195_playback_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(DL11_FE), }, [DAI_LINK_UL1_FE] = { @@ -1136,7 +1007,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8195_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(UL2_FE), }, [DAI_LINK_UL3_FE] = { @@ -1148,7 +1019,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8195_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(UL3_FE), }, [DAI_LINK_UL4_FE] = { @@ -1160,7 +1031,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8195_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(UL4_FE), }, [DAI_LINK_UL5_FE] = { @@ -1172,7 +1043,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8195_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(UL5_FE), }, [DAI_LINK_UL6_FE] = { @@ -1195,7 +1066,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8195_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(UL8_FE), }, [DAI_LINK_UL9_FE] = { @@ -1207,7 +1078,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8195_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(UL9_FE), }, [DAI_LINK_UL10_FE] = { @@ -1219,7 +1090,7 @@ static struct snd_soc_dai_link mt8195_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8195_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(UL10_FE), }, /* BE */ @@ -1561,6 +1432,40 @@ static int mt8195_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, return 0; } +static const unsigned int mt8195_pcm_playback_channels[] = { 2 }; +static const unsigned int mt8195_pcm_capture_channels[] = { 1, 2 }; +static const unsigned int mt8195_pcm_hdmidp_channels[] = { 2, 4, 6, 8 }; +static const unsigned int mt8195_pcm_rates[] = { 48000 }; + +static const struct snd_pcm_hw_constraint_list mt8195_rate_constraint = { + .list = mt8195_pcm_rates, + .count = ARRAY_SIZE(mt8195_pcm_rates) +}; + +static const struct mtk_pcm_constraints_data mt8195_pcm_constraints[MTK_CONSTRAINT_HDMIDP + 1] = { + [MTK_CONSTRAINT_PLAYBACK] = { + .channels = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8195_pcm_playback_channels, + .count = ARRAY_SIZE(mt8195_pcm_playback_channels) + }, + .rates = &mt8195_rate_constraint, + }, + [MTK_CONSTRAINT_CAPTURE] = { + .channels = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8195_pcm_capture_channels, + .count = ARRAY_SIZE(mt8195_pcm_capture_channels) + }, + .rates = &mt8195_rate_constraint, + }, + [MTK_CONSTRAINT_HDMIDP] = { + .channels = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8195_pcm_hdmidp_channels, + .count = ARRAY_SIZE(mt8195_pcm_hdmidp_channels) + }, + .rates = &mt8195_rate_constraint, + }, +}; + static const struct mtk_sof_priv mt8195_sof_priv = { .conn_streams = g_sof_conn_streams, .num_streams = ARRAY_SIZE(g_sof_conn_streams), @@ -1572,6 +1477,8 @@ static const struct mtk_soundcard_pdata mt8195_mt6359_rt1019_rt5682_card = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8195_mt6359_soc_card, .num_jacks = MT8195_JACK_MAX, + .pcm_constraints = mt8195_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8195_pcm_constraints), .flags = RT1019_SPEAKER_AMP_PRESENT }, .sof_priv = &mt8195_sof_priv, @@ -1583,6 +1490,8 @@ static const struct mtk_soundcard_pdata mt8195_mt6359_rt1011_rt5682_card = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8195_mt6359_soc_card, .num_jacks = MT8195_JACK_MAX, + .pcm_constraints = mt8195_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8195_pcm_constraints), .flags = RT1011_SPEAKER_AMP_PRESENT }, .sof_priv = &mt8195_sof_priv, @@ -1594,6 +1503,8 @@ static const struct mtk_soundcard_pdata mt8195_mt6359_max98390_rt5682_card = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8195_mt6359_soc_card, .num_jacks = MT8195_JACK_MAX, + .pcm_constraints = mt8195_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8195_pcm_constraints), .flags = MAX98390_SPEAKER_AMP_PRESENT }, .sof_priv = &mt8195_sof_priv, From 94142cc1ffd959aa8e83857515a75779e8ba8064 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:01 +0200 Subject: [PATCH 296/603] ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup Add a const mtk_pcm_constraints_data struct array with all of the (again, constant) constraints for all of the supported usecases, remove the duplicated functions and call mtk_soundcard_startup() instead in all of the .startup() callbacks. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-10-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 139 ++++++------------ 1 file changed, 41 insertions(+), 98 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 645bc1aa67c7f..8b323fb199251 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -389,100 +389,6 @@ static int mt8192_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int -mt8192_mt6359_cap1_startup(struct snd_pcm_substream *substream) -{ - static const unsigned int channels[] = { - 1, 2, 4 - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - static const unsigned int rates[] = { - 8000, 16000, 32000, 48000, 96000, 192000 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channels failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_ops mt8192_mt6359_capture1_ops = { - .startup = mt8192_mt6359_cap1_startup, -}; - -static int -mt8192_mt6359_rt5682_startup(struct snd_pcm_substream *substream) -{ - static const unsigned int channels[] = { - 1, 2 - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - static const unsigned int rates[] = { - 48000 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channels failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_ops mt8192_mt6359_rt5682_ops = { - .startup = mt8192_mt6359_rt5682_startup, -}; - /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), @@ -720,7 +626,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, - .ops = &mt8192_mt6359_rt5682_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(playback3), }, { @@ -784,7 +690,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8192_mt6359_capture1_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(capture1), }, { @@ -794,7 +700,7 @@ static struct snd_soc_dai_link mt8192_mt6359_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, - .ops = &mt8192_mt6359_rt5682_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(capture2), }, { @@ -1146,7 +1052,7 @@ static int mt8192_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data) struct device *dev = card->dev; struct device_node *hdmi_codec, *headset_codec, *speaker_codec; struct snd_soc_dai_link *dai_link; - int ret, i; + int i, ret = 0; hdmi_codec = of_parse_phandle(dev->of_node, "mediatek,hdmi-codec", 0); if (!hdmi_codec) @@ -1235,11 +1141,44 @@ static int mt8192_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, return 0; } +static const unsigned int mt8192_pcm_playback_channels[] = { 1, 2 }; +static const unsigned int mt8192_pcm_playback_rates[] = { 48000 }; + +static const unsigned int mt8192_pcm_capture_channels[] = { 1, 2, 4 }; +static const unsigned int mt8192_pcm_capture_rates[] = { + 8000, 16000, 32000, 48000, 96000, 192000 +}; + +static const struct mtk_pcm_constraints_data mt8192_pcm_constraints[MTK_CONSTRAINT_CAPTURE + 1] = { + [MTK_CONSTRAINT_PLAYBACK] = { + .channels = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8192_pcm_playback_channels, + .count = ARRAY_SIZE(mt8192_pcm_playback_channels) + }, + .rates = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8192_pcm_playback_rates, + .count = ARRAY_SIZE(mt8192_pcm_playback_rates) + } + }, + [MTK_CONSTRAINT_CAPTURE] = { + .channels = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8192_pcm_capture_channels, + .count = ARRAY_SIZE(mt8192_pcm_capture_channels) + }, + .rates = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8192_pcm_capture_rates, + .count = ARRAY_SIZE(mt8192_pcm_capture_rates) + } + } +}; + static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015_rt5682_pdata = { .card_name = RT1015_RT5682_CARD_NAME, .card_data = &(struct mtk_platform_card_data) { .card = &mt8192_mt6359_rt1015_rt5682_card, .num_jacks = MT8192_JACK_MAX, + .pcm_constraints = mt8192_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8192_pcm_constraints), }, .soc_probe = mt8192_mt6359_soc_card_probe }; @@ -1249,6 +1188,8 @@ static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015p_rt5682_pdata = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8192_mt6359_rt1015p_rt5682x_card, .num_jacks = MT8192_JACK_MAX, + .pcm_constraints = mt8192_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8192_pcm_constraints), }, .soc_probe = mt8192_mt6359_soc_card_probe }; @@ -1258,6 +1199,8 @@ static const struct mtk_soundcard_pdata mt8192_mt6359_rt1015p_rt5682s_pdata = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8192_mt6359_rt1015p_rt5682x_card, .num_jacks = MT8192_JACK_MAX, + .pcm_constraints = mt8192_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8192_pcm_constraints), }, .soc_probe = mt8192_mt6359_soc_card_probe }; From f103ce4ff6932b6991e1922d1a39c29971ab2938 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:02 +0200 Subject: [PATCH 297/603] ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup Add a const mtk_pcm_constraints_data struct array with all of the (again, constant) constraints for all of the supported usecases, remove the duplicated functions and call mtk_soundcard_startup() instead in all of the .startup() callbacks. Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-11-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 132 +++++------------- 1 file changed, 36 insertions(+), 96 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c index f8856a100b276..28140ca0ebff8 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c @@ -332,98 +332,6 @@ static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, return ret; } -static int mt8186_mt6366_rt1019_rt5682s_playback_startup(struct snd_pcm_substream *substream) -{ - static const unsigned int rates[] = { - 48000 - }; - static const unsigned int channels[] = { - 2 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channel failed\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_ops mt8186_mt6366_rt1019_rt5682s_playback_ops = { - .startup = mt8186_mt6366_rt1019_rt5682s_playback_startup, -}; - -static int mt8186_mt6366_rt1019_rt5682s_capture_startup(struct snd_pcm_substream *substream) -{ - static const unsigned int rates[] = { - 48000 - }; - static const unsigned int channels[] = { - 1, 2 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channel failed\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_ops mt8186_mt6366_rt1019_rt5682s_capture_ops = { - .startup = mt8186_mt6366_rt1019_rt5682s_capture_startup, -}; - /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), @@ -644,7 +552,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .dpcm_merged_format = 1, .dpcm_merged_chan = 1, .dpcm_merged_rate = 1, - .ops = &mt8186_mt6366_rt1019_rt5682s_playback_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(playback1), }, { @@ -678,7 +586,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .dpcm_merged_format = 1, .dpcm_merged_chan = 1, .dpcm_merged_rate = 1, - .ops = &mt8186_mt6366_rt1019_rt5682s_playback_ops, + .ops = &mtk_soundcard_common_playback_ops, SND_SOC_DAILINK_REG(playback3), }, { @@ -745,7 +653,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .dpcm_merged_format = 1, .dpcm_merged_chan = 1, .dpcm_merged_rate = 1, - .ops = &mt8186_mt6366_rt1019_rt5682s_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(capture2), }, { @@ -767,7 +675,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .dpcm_merged_format = 1, .dpcm_merged_chan = 1, .dpcm_merged_rate = 1, - .ops = &mt8186_mt6366_rt1019_rt5682s_capture_ops, + .ops = &mtk_soundcard_common_capture_ops, SND_SOC_DAILINK_REG(capture4), }, { @@ -1220,6 +1128,32 @@ static int mt8186_mt6366_soc_card_probe(struct mtk_soc_card_data *soc_card_data, return 0; } +static const unsigned int mt8186_pcm_playback_channels[] = { 2 }; +static const unsigned int mt8186_pcm_capture_channels[] = { 1, 2 }; +static const unsigned int mt8186_pcm_rates[] = { 48000 }; + +static const struct snd_pcm_hw_constraint_list mt8186_rate_constraint = { + .list = mt8186_pcm_rates, + .count = ARRAY_SIZE(mt8186_pcm_rates) +}; + +static const struct mtk_pcm_constraints_data mt8186_pcm_constraints[MTK_CONSTRAINT_CAPTURE + 1] = { + [MTK_CONSTRAINT_PLAYBACK] = { + .channels = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8186_pcm_playback_channels, + .count = ARRAY_SIZE(mt8186_pcm_playback_channels) + }, + .rates = &mt8186_rate_constraint, + }, + [MTK_CONSTRAINT_CAPTURE] = { + .channels = &(const struct snd_pcm_hw_constraint_list) { + .list = mt8186_pcm_capture_channels, + .count = ARRAY_SIZE(mt8186_pcm_capture_channels) + }, + .rates = &mt8186_rate_constraint, + } +}; + static const struct mtk_sof_priv mt8186_sof_priv = { .conn_streams = g_sof_conn_streams, .num_streams = ARRAY_SIZE(g_sof_conn_streams), @@ -1230,6 +1164,8 @@ static const struct mtk_soundcard_pdata mt8186_mt6366_rt1019_rt5682s_pdata = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8186_mt6366_rt1019_rt5682s_soc_card, .num_jacks = MT8186_JACK_MAX, + .pcm_constraints = mt8186_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints), }, .sof_priv = &mt8186_sof_priv, .soc_probe = mt8186_mt6366_soc_card_probe @@ -1239,6 +1175,8 @@ static const struct mtk_soundcard_pdata mt8186_mt6366_rt5682s_max98360_pdata = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8186_mt6366_rt5682s_max98360_soc_card, .num_jacks = MT8186_JACK_MAX, + .pcm_constraints = mt8186_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints), }, .sof_priv = &mt8186_sof_priv, .soc_probe = mt8186_mt6366_soc_card_probe @@ -1248,6 +1186,8 @@ static const struct mtk_soundcard_pdata mt8186_mt6366_rt5650_pdata = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8186_mt6366_rt5650_soc_card, .num_jacks = MT8186_JACK_MAX, + .pcm_constraints = mt8186_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints), }, .sof_priv = &mt8186_sof_priv, .soc_probe = mt8186_mt6366_soc_card_probe From 9d9b182876a249b393158e9f43c255aeee4ef898 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:03 +0200 Subject: [PATCH 298/603] ASoC: mediatek: Add common mtk_afe_component_probe callback Multiple MediaTek AFE PCM component drivers are using their own .probe() callback, but most of those are simply duplicated functions as they are doing exactly the same thing over and over. Add a common probe callback for this component to reduce duplication. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-12-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../mediatek/common/mtk-afe-platform-driver.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 32edcb6d52198..9b72b2a7ae917 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -126,10 +126,28 @@ int mtk_afe_pcm_new(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(mtk_afe_pcm_new); +static int mtk_afe_component_probe(struct snd_soc_component *component) +{ + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); + int ret; + + snd_soc_component_init_regmap(component, afe->regmap); + + /* If the list was never initialized there are no sub-DAIs */ + if (afe->sub_dais.next && afe->sub_dais.prev) { + ret = mtk_afe_add_sub_dai_control(component); + if (ret) + return ret; + } + + return 0; +} + const struct snd_soc_component_driver mtk_afe_pcm_platform = { .name = AFE_PCM_NAME, .pointer = mtk_afe_pcm_pointer, .pcm_construct = mtk_afe_pcm_new, + .probe = mtk_afe_component_probe, }; EXPORT_SYMBOL_GPL(mtk_afe_pcm_platform); From 1d2a6b3b5189b6c3e51f29e050aa9b2337b774e8 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:04 +0200 Subject: [PATCH 299/603] ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb Since the mtk-afe-platform-driver generic mtk_afe_pcm_platform now has a common .probe() callback, there is no reason to keep duplicating this function over and over in the SoC specific AFE-PCM drivers: switch over to register with the common bits instead. Note that MT8186 was left out of this because it is registering some extra sinegen controls in the AFE-PCM probe callback and needs extra cleanups to be able to use the common bits. Reviewed-by: Alexandre Mergnat Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-13-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +----------- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +----------- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +----------- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +----------------- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 25 ++-------------------- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +----------------- 6 files changed, 7 insertions(+), 102 deletions(-) diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index da7267c684b17..c1dee119e93a7 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -704,18 +704,6 @@ static int mt6797_afe_runtime_resume(struct device *dev) return 0; } -static int mt6797_afe_component_probe(struct snd_soc_component *component) -{ - return mtk_afe_add_sub_dai_control(component); -} - -static const struct snd_soc_component_driver mt6797_afe_component = { - .name = AFE_PCM_NAME, - .probe = mt6797_afe_component_probe, - .pointer = mtk_afe_pcm_pointer, - .pcm_construct = mtk_afe_pcm_new, -}; - static int mt6797_dai_memif_register(struct mtk_base_afe *afe) { struct mtk_base_afe_dai *dai; @@ -852,7 +840,7 @@ static int mt6797_afe_pcm_dev_probe(struct platform_device *pdev) pm_runtime_get_sync(&pdev->dev); /* register component */ - ret = devm_snd_soc_register_component(dev, &mt6797_afe_component, + ret = devm_snd_soc_register_component(dev, &mtk_afe_pcm_platform, NULL, 0); if (ret) { dev_warn(dev, "err_platform\n"); diff --git a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c index d497e11298899..c1c486e275b9d 100644 --- a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c +++ b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c @@ -429,18 +429,6 @@ static int mt7986_afe_runtime_resume(struct device *dev) return 0; } -static int mt7986_afe_component_probe(struct snd_soc_component *component) -{ - return mtk_afe_add_sub_dai_control(component); -} - -static const struct snd_soc_component_driver mt7986_afe_component = { - .name = AFE_PCM_NAME, - .probe = mt7986_afe_component_probe, - .pointer = mtk_afe_pcm_pointer, - .pcm_construct = mtk_afe_pcm_new, -}; - static int mt7986_dai_memif_register(struct mtk_base_afe *afe) { struct mtk_base_afe_dai *dai; @@ -573,7 +561,7 @@ static int mt7986_afe_pcm_dev_probe(struct platform_device *pdev) /* register component */ ret = devm_snd_soc_register_component(&pdev->dev, - &mt7986_afe_component, + &mtk_afe_pcm_platform, NULL, 0); if (ret) return dev_err_probe(dev, ret, "Cannot register AFE component\n"); diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 9e432ed9124b9..25348fdf75fa1 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1042,18 +1042,6 @@ static int mt8183_afe_runtime_resume(struct device *dev) return 0; } -static int mt8183_afe_component_probe(struct snd_soc_component *component) -{ - return mtk_afe_add_sub_dai_control(component); -} - -static const struct snd_soc_component_driver mt8183_afe_component = { - .name = AFE_PCM_NAME, - .probe = mt8183_afe_component_probe, - .pointer = mtk_afe_pcm_pointer, - .pcm_construct = mtk_afe_pcm_new, -}; - static int mt8183_dai_memif_register(struct mtk_base_afe *afe) { struct mtk_base_afe_dai *dai; @@ -1232,7 +1220,7 @@ static int mt8183_afe_pcm_dev_probe(struct platform_device *pdev) /* register component */ ret = devm_snd_soc_register_component(&pdev->dev, - &mt8183_afe_component, + &mtk_afe_pcm_platform, NULL, 0); if (ret) { dev_warn(dev, "err_platform\n"); diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c index 46d6a55404037..9647fe133dc82 100644 --- a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c +++ b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c @@ -3030,25 +3030,6 @@ static int mt8188_afe_runtime_resume(struct device *dev) return 0; } -static int mt8188_afe_component_probe(struct snd_soc_component *component) -{ - struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int ret; - - snd_soc_component_init_regmap(component, afe->regmap); - - ret = mtk_afe_add_sub_dai_control(component); - - return ret; -} - -static const struct snd_soc_component_driver mt8188_afe_component = { - .name = AFE_PCM_NAME, - .pointer = mtk_afe_pcm_pointer, - .pcm_construct = mtk_afe_pcm_new, - .probe = mt8188_afe_component_probe, -}; - static int init_memif_priv_data(struct mtk_base_afe *afe) { struct mt8188_afe_private *afe_priv = afe->platform_priv; @@ -3350,7 +3331,7 @@ static int mt8188_afe_pcm_dev_probe(struct platform_device *pdev) } /* register component */ - ret = devm_snd_soc_register_component(dev, &mt8188_afe_component, + ret = devm_snd_soc_register_component(dev, &mtk_afe_pcm_platform, afe->dai_drivers, afe->num_dai_drivers); if (ret) { dev_warn(dev, "err_platform\n"); diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index aed22baef9fb3..424c5c68f78af 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -2125,22 +2125,6 @@ static int mt8192_afe_runtime_resume(struct device *dev) return 0; } -static int mt8192_afe_component_probe(struct snd_soc_component *component) -{ - return mtk_afe_add_sub_dai_control(component); -} - -static const struct snd_soc_component_driver mt8192_afe_component = { - .name = AFE_PCM_NAME, - .probe = mt8192_afe_component_probe, - .pointer = mtk_afe_pcm_pointer, - .pcm_construct = mtk_afe_pcm_new, -}; - -static const struct snd_soc_component_driver mt8192_afe_pcm_component = { - .name = "mt8192-afe-pcm-dai", -}; - static int mt8192_dai_memif_register(struct mtk_base_afe *afe) { struct mtk_base_afe_dai *dai; @@ -2302,16 +2286,11 @@ static int mt8192_afe_pcm_dev_probe(struct platform_device *pdev) /* register platform */ ret = devm_snd_soc_register_component(&pdev->dev, - &mt8192_afe_component, NULL, 0); - if (ret) - return dev_err_probe(dev, ret, "Couldn't register AFE component\n"); - - ret = devm_snd_soc_register_component(&pdev->dev, - &mt8192_afe_pcm_component, + &mtk_afe_pcm_platform, afe->dai_drivers, afe->num_dai_drivers); if (ret) - return dev_err_probe(dev, ret, "Couldn't register AFE-PCM component\n"); + return dev_err_probe(dev, ret, "Couldn't register AFE component\n"); return 0; } diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 620d7ade1992e..64af9bf363fda 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -2944,25 +2944,6 @@ static int mt8195_afe_runtime_resume(struct device *dev) return 0; } -static int mt8195_afe_component_probe(struct snd_soc_component *component) -{ - struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); - int ret = 0; - - snd_soc_component_init_regmap(component, afe->regmap); - - ret = mtk_afe_add_sub_dai_control(component); - - return ret; -} - -static const struct snd_soc_component_driver mt8195_afe_component = { - .name = AFE_PCM_NAME, - .pointer = mtk_afe_pcm_pointer, - .pcm_construct = mtk_afe_pcm_new, - .probe = mt8195_afe_component_probe, -}; - static int init_memif_priv_data(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; @@ -3164,7 +3145,7 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) } /* register component */ - ret = devm_snd_soc_register_component(dev, &mt8195_afe_component, + ret = devm_snd_soc_register_component(dev, &mtk_afe_pcm_platform, afe->dai_drivers, afe->num_dai_drivers); if (ret) { dev_warn(dev, "err_platform\n"); From 6ed619b348bf0a490e70dea7a330cf44dc81c16d Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:05 +0200 Subject: [PATCH 300/603] ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers Copy the few differences from mt8186-mt6166-da7219-max98357 in the mt8186-mt6366-rt1019-rt5682s driver to greatly reduce code duplication; since now the driver is meant to support MT8186 with the MT6366 PMIC codec and various combinations of I2S codecs, rename the driver to mt8186-mt6366 for consistency with MT8195 and MT8188, and rename the configuration option to SND_SOC_MT8186_MT6366. Since right now there is no machine using the da7219-max98357 yet, the snd_soc_dapm_route array was omitted as it's now possible to specify the audio routing in device trees instead. While at it, also add the missing sentinel comment to the last entry of the of_device_id array. Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-14-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 24 +- sound/soc/mediatek/mt8186/Makefile | 3 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 228 +++- 4 files changed, 202 insertions(+), 1242 deletions(-) delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (83%) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 296b434caf816..5a8476e1ecca7 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -185,27 +185,11 @@ config SND_SOC_MT8186 Select Y if you have such device. If unsure select "N". -config SND_SOC_MT8186_MT6366_DA7219_MAX98357 - tristate "ASoC Audio driver for MT8186 with DA7219 MAX98357A codec" +config SND_SOC_MT8186_MT6366 + tristate "ASoC Audio driver for MT8186 with MT6366 and I2S codecs" depends on I2C && GPIOLIB depends on SND_SOC_MT8186 && MTK_PMIC_WRAP - select SND_SOC_MT6358 - select SND_SOC_MAX98357A select SND_SOC_DA7219 - select SND_SOC_BT_SCO - select SND_SOC_DMIC - select SND_SOC_HDMI_CODEC - help - This adds ASoC driver for Mediatek MT8186 boards - with the MT6366(MT6358) DA7219 MAX98357A codecs. - Select Y if you have such device. - If unsure select "N". - -config SND_SOC_MT8186_MT6366_RT1019_RT5682S - tristate "ASoC Audio driver for MT8186 with RT1019 RT5682S MAX98357A/MAX98360 codec" - depends on I2C && GPIOLIB - depends on SND_SOC_MT8186 && MTK_PMIC_WRAP - select SND_SOC_MAX98357A select SND_SOC_MT6358 select SND_SOC_MAX98357A select SND_SOC_RT1015P @@ -215,8 +199,8 @@ config SND_SOC_MT8186_MT6366_RT1019_RT5682S select SND_SOC_DMIC select SND_SOC_HDMI_CODEC help - This adds ASoC driver for Mediatek MT8186 boards - with the MT6366(MT6358) RT1019 RT5682S codecs. + This adds the ASoC machine driver for Mediatek MT8186 boards + with the MT6366(MT6358) and other I2S audio codecs. Select Y if you have such device. If unsure select "N". diff --git a/sound/soc/mediatek/mt8186/Makefile b/sound/soc/mediatek/mt8186/Makefile index 49b0026628a06..fa2f0efce8159 100644 --- a/sound/soc/mediatek/mt8186/Makefile +++ b/sound/soc/mediatek/mt8186/Makefile @@ -18,5 +18,4 @@ snd-soc-mt8186-afe-objs := \ mt8186-mt6366-common.o obj-$(CONFIG_SND_SOC_MT8186) += snd-soc-mt8186-afe.o -obj-$(CONFIG_SND_SOC_MT8186_MT6366_DA7219_MAX98357) += mt8186-mt6366-da7219-max98357.o -obj-$(CONFIG_SND_SOC_MT8186_MT6366_RT1019_RT5682S) += mt8186-mt6366-rt1019-rt5682s.o +obj-$(CONFIG_SND_SOC_MT8186_MT6366) += mt8186-mt6366.o diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c deleted file mode 100644 index d86dc45be30cf..0000000000000 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c +++ /dev/null @@ -1,1189 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// mt8186-mt6366-da7219-max98357.c -// -- MT8186-MT6366-DA7219-MAX98357 ALSA SoC machine driver -// -// Copyright (c) 2022 MediaTek Inc. -// Author: Jiaxin Yu -// - -#include -#include -#include -#include -#include -#include - -#include "../../codecs/da7219.h" -#include "../../codecs/mt6358.h" -#include "../common/mtk-afe-platform-driver.h" -#include "../common/mtk-dsp-sof-common.h" -#include "../common/mtk-soc-card.h" -#include "mt8186-afe-common.h" -#include "mt8186-afe-clk.h" -#include "mt8186-afe-gpio.h" -#include "mt8186-mt6366-common.h" - -#define DA7219_CODEC_DAI "da7219-hifi" -#define DA7219_DEV_NAME "da7219.5-001a" - -#define SOF_DMA_DL1 "SOF_DMA_DL1" -#define SOF_DMA_DL2 "SOF_DMA_DL2" -#define SOF_DMA_UL1 "SOF_DMA_UL1" -#define SOF_DMA_UL2 "SOF_DMA_UL2" - -struct mt8186_mt6366_da7219_max98357_priv { - struct snd_soc_jack headset_jack, hdmi_jack; -}; - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin mt8186_jack_pins[] = { - { - .pin = "Headphones", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Line Out", - .mask = SND_JACK_LINEOUT, - }, -}; - -static struct snd_soc_codec_conf mt8186_mt6366_da7219_max98357_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF("mt6358-sound"), - .name_prefix = "Mt6366", - }, - { - .dlc = COMP_CODEC_CONF("bt-sco"), - .name_prefix = "Mt8186 bt", - }, - { - .dlc = COMP_CODEC_CONF("hdmi-audio-codec"), - .name_prefix = "Mt8186 hdmi", - }, -}; - -static int mt8186_da7219_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *cmpnt_afe = - snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); - struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe); - struct mtk_soc_card_data *soc_card_data = - snd_soc_card_get_drvdata(rtd->card); - struct mt8186_mt6366_da7219_max98357_priv *priv = soc_card_data->mach_priv; - struct snd_soc_jack *jack = &priv->headset_jack; - struct snd_soc_component *cmpnt_codec = - snd_soc_rtd_to_codec(rtd, 0)->component; - int ret; - - ret = mt8186_dai_i2s_set_share(afe, "I2S1", "I2S0"); - if (ret) { - dev_err(rtd->dev, "Failed to set up shared clocks\n"); - return ret; - } - - /* Enable Headset and 4 Buttons Jack detection */ - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3 | SND_JACK_LINEOUT, - jack, mt8186_jack_pins, - ARRAY_SIZE(mt8186_jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); - return ret; - } - - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); - - snd_soc_component_set_jack(cmpnt_codec, &priv->headset_jack, NULL); - - return 0; -} - -static int mt8186_da7219_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai; - unsigned int rate = params_rate(params); - unsigned int mclk_fs_ratio = 256; - unsigned int mclk_fs = rate * mclk_fs_ratio; - unsigned int freq; - int ret, j; - - ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_cpu(rtd, 0), 0, - mclk_fs, SND_SOC_CLOCK_OUT); - if (ret < 0) { - dev_err(rtd->dev, "failed to set cpu dai sysclk: %d\n", ret); - return ret; - } - - for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, DA7219_DEV_NAME)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - DA7219_CLKSRC_MCLK, - mclk_fs, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "failed to set sysclk: %d\n", - ret); - return ret; - } - - if ((rate % 8000) == 0) - freq = DA7219_PLL_FREQ_OUT_98304; - else - freq = DA7219_PLL_FREQ_OUT_90316; - - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_PLL_SRM, - 0, freq); - if (ret) { - dev_err(rtd->dev, "failed to start PLL: %d\n", - ret); - return ret; - } - } - } - - return 0; -} - -static int mt8186_da7219_i2s_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai; - int ret = 0, j; - - for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, DA7219_DEV_NAME)) { - ret = snd_soc_dai_set_pll(codec_dai, - 0, DA7219_SYSCLK_MCLK, 0, 0); - if (ret < 0) { - dev_err(rtd->dev, "failed to stop PLL: %d\n", - ret); - return ret; - } - } - } - - return 0; -} - -static const struct snd_soc_ops mt8186_da7219_i2s_ops = { - .hw_params = mt8186_da7219_i2s_hw_params, - .hw_free = mt8186_da7219_i2s_hw_free, -}; - -static int mt8186_mt6366_da7219_max98357_hdmi_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *cmpnt_afe = - snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); - struct mtk_base_afe *afe = snd_soc_component_get_drvdata(cmpnt_afe); - struct snd_soc_component *cmpnt_codec = - snd_soc_rtd_to_codec(rtd, 0)->component; - struct mtk_soc_card_data *soc_card_data = - snd_soc_card_get_drvdata(rtd->card); - struct mt8186_mt6366_da7219_max98357_priv *priv = soc_card_data->mach_priv; - int ret; - - ret = mt8186_dai_i2s_set_share(afe, "I2S2", "I2S3"); - if (ret) { - dev_err(rtd->dev, "Failed to set up shared clocks\n"); - return ret; - } - - ret = snd_soc_card_jack_new(rtd->card, "HDMI Jack", SND_JACK_LINEOUT, &priv->hdmi_jack); - if (ret) { - dev_err(rtd->dev, "HDMI Jack creation failed: %d\n", ret); - return ret; - } - - return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL); -} - -static int mt8186_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params, - snd_pcm_format_t fmt) -{ - struct snd_interval *channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - dev_dbg(rtd->dev, "%s(), fix format to %d\n", __func__, fmt); - - /* fix BE i2s channel to 2 channel */ - channels->min = 2; - channels->max = 2; - - /* clean param mask first */ - snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); - - params_set_format(params, fmt); - - return 0; -} - -static int mt8186_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S32_LE); -} - -static int mt8186_anx7625_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); -} - -/* fixup the BE DAI link to match any values from topology */ -static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - int ret; - - ret = mtk_sof_dai_link_fixup(rtd, params); - - if (!strcmp(rtd->dai_link->name, "I2S0") || - !strcmp(rtd->dai_link->name, "I2S1") || - !strcmp(rtd->dai_link->name, "I2S2")) - mt8186_i2s_hw_params_fixup(rtd, params); - else if (!strcmp(rtd->dai_link->name, "I2S3")) - mt8186_anx7625_i2s_hw_params_fixup(rtd, params); - - return ret; -} - -static int mt8186_mt6366_da7219_max98357_playback_startup(struct snd_pcm_substream *substream) -{ - static const unsigned int rates[] = { - 48000 - }; - static const unsigned int channels[] = { - 2 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channel failed\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_ops mt8186_mt6366_da7219_max98357_playback_ops = { - .startup = mt8186_mt6366_da7219_max98357_playback_startup, -}; - -static int mt8186_mt6366_da7219_max98357_capture_startup(struct snd_pcm_substream *substream) -{ - static const unsigned int rates[] = { - 48000 - }; - static const unsigned int channels[] = { - 1, 2 - }; - static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, - }; - static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, - }; - - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list rate failed\n"); - return ret; - } - - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - if (ret < 0) { - dev_err(rtd->dev, "hw_constraint_list channel failed\n"); - return ret; - } - - return 0; -} - -static const struct snd_soc_ops mt8186_mt6366_da7219_max98357_capture_ops = { - .startup = mt8186_mt6366_da7219_max98357_capture_startup, -}; - -/* FE */ -SND_SOC_DAILINK_DEFS(playback1, - DAILINK_COMP_ARRAY(COMP_CPU("DL1")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(playback12, - DAILINK_COMP_ARRAY(COMP_CPU("DL12")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(playback2, - DAILINK_COMP_ARRAY(COMP_CPU("DL2")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(playback3, - DAILINK_COMP_ARRAY(COMP_CPU("DL3")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(playback4, - DAILINK_COMP_ARRAY(COMP_CPU("DL4")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(playback5, - DAILINK_COMP_ARRAY(COMP_CPU("DL5")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(playback6, - DAILINK_COMP_ARRAY(COMP_CPU("DL6")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(playback7, - DAILINK_COMP_ARRAY(COMP_CPU("DL7")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(playback8, - DAILINK_COMP_ARRAY(COMP_CPU("DL8")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(capture1, - DAILINK_COMP_ARRAY(COMP_CPU("UL1")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(capture2, - DAILINK_COMP_ARRAY(COMP_CPU("UL2")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(capture3, - DAILINK_COMP_ARRAY(COMP_CPU("UL3")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(capture4, - DAILINK_COMP_ARRAY(COMP_CPU("UL4")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(capture5, - DAILINK_COMP_ARRAY(COMP_CPU("UL5")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(capture6, - DAILINK_COMP_ARRAY(COMP_CPU("UL6")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(capture7, - DAILINK_COMP_ARRAY(COMP_CPU("UL7")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -/* hostless */ -SND_SOC_DAILINK_DEFS(hostless_lpbk, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless LPBK DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_fm, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless FM DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_src1, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless_SRC_1_DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_src_bargein, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless_SRC_Bargein_DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -/* BE */ -SND_SOC_DAILINK_DEFS(adda, - DAILINK_COMP_ARRAY(COMP_CPU("ADDA")), - DAILINK_COMP_ARRAY(COMP_CODEC("mt6358-sound", - "mt6358-snd-codec-aif1"), - COMP_CODEC("dmic-codec", - "dmic-hifi")), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(i2s0, - DAILINK_COMP_ARRAY(COMP_CPU("I2S0")), - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(i2s1, - DAILINK_COMP_ARRAY(COMP_CPU("I2S1")), - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(i2s2, - DAILINK_COMP_ARRAY(COMP_CPU("I2S2")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(i2s3, - DAILINK_COMP_ARRAY(COMP_CPU("I2S3")), - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hw_gain1, - DAILINK_COMP_ARRAY(COMP_CPU("HW Gain 1")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hw_gain2, - DAILINK_COMP_ARRAY(COMP_CPU("HW Gain 2")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hw_src1, - DAILINK_COMP_ARRAY(COMP_CPU("HW_SRC_1")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hw_src2, - DAILINK_COMP_ARRAY(COMP_CPU("HW_SRC_2")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(connsys_i2s, - DAILINK_COMP_ARRAY(COMP_CPU("CONNSYS_I2S")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(pcm1, - DAILINK_COMP_ARRAY(COMP_CPU("PCM 1")), - DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm-wb")), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(tdm_in, - DAILINK_COMP_ARRAY(COMP_CPU("TDM IN")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -/* hostless */ -SND_SOC_DAILINK_DEFS(hostless_ul1, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL1 DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_ul2, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL2 DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_ul3, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL3 DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_ul5, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL5 DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_ul6, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless_UL6 DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_hw_gain_aaudio, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless HW Gain AAudio DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(hostless_src_aaudio, - DAILINK_COMP_ARRAY(COMP_CPU("Hostless SRC AAudio DAI")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); -SND_SOC_DAILINK_DEFS(AFE_SOF_DL1, - DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL1")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(AFE_SOF_DL2, - DAILINK_COMP_ARRAY(COMP_CPU("SOF_DL2")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(AFE_SOF_UL1, - DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL1")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(AFE_SOF_UL2, - DAILINK_COMP_ARRAY(COMP_CPU("SOF_UL2")), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -static const struct sof_conn_stream g_sof_conn_streams[] = { - { "I2S1", "AFE_SOF_DL1", SOF_DMA_DL1, SNDRV_PCM_STREAM_PLAYBACK}, - { "I2S3", "AFE_SOF_DL2", SOF_DMA_DL2, SNDRV_PCM_STREAM_PLAYBACK}, - { "Primary Codec", "AFE_SOF_UL1", SOF_DMA_UL1, SNDRV_PCM_STREAM_CAPTURE}, - { "I2S0", "AFE_SOF_UL2", SOF_DMA_UL2, SNDRV_PCM_STREAM_CAPTURE}, -}; - -static struct snd_soc_dai_link mt8186_mt6366_da7219_max98357_dai_links[] = { - /* Front End DAI links */ - { - .name = "Playback_1", - .stream_name = "Playback_1", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_merged_format = 1, - .dpcm_merged_chan = 1, - .dpcm_merged_rate = 1, - .ops = &mt8186_mt6366_da7219_max98357_playback_ops, - SND_SOC_DAILINK_REG(playback1), - }, - { - .name = "Playback_12", - .stream_name = "Playback_12", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(playback12), - }, - { - .name = "Playback_2", - .stream_name = "Playback_2", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_merged_format = 1, - .dpcm_merged_chan = 1, - .dpcm_merged_rate = 1, - SND_SOC_DAILINK_REG(playback2), - }, - { - .name = "Playback_3", - .stream_name = "Playback_3", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_merged_format = 1, - .dpcm_merged_chan = 1, - .dpcm_merged_rate = 1, - .ops = &mt8186_mt6366_da7219_max98357_playback_ops, - SND_SOC_DAILINK_REG(playback3), - }, - { - .name = "Playback_4", - .stream_name = "Playback_4", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(playback4), - }, - { - .name = "Playback_5", - .stream_name = "Playback_5", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(playback5), - }, - { - .name = "Playback_6", - .stream_name = "Playback_6", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(playback6), - }, - { - .name = "Playback_7", - .stream_name = "Playback_7", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(playback7), - }, - { - .name = "Playback_8", - .stream_name = "Playback_8", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(playback8), - }, - { - .name = "Capture_1", - .stream_name = "Capture_1", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(capture1), - }, - { - .name = "Capture_2", - .stream_name = "Capture_2", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_capture = 1, - .dpcm_merged_format = 1, - .dpcm_merged_chan = 1, - .dpcm_merged_rate = 1, - .ops = &mt8186_mt6366_da7219_max98357_capture_ops, - SND_SOC_DAILINK_REG(capture2), - }, - { - .name = "Capture_3", - .stream_name = "Capture_3", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(capture3), - }, - { - .name = "Capture_4", - .stream_name = "Capture_4", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_capture = 1, - .dpcm_merged_format = 1, - .dpcm_merged_chan = 1, - .dpcm_merged_rate = 1, - .ops = &mt8186_mt6366_da7219_max98357_capture_ops, - SND_SOC_DAILINK_REG(capture4), - }, - { - .name = "Capture_5", - .stream_name = "Capture_5", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(capture5), - }, - { - .name = "Capture_6", - .stream_name = "Capture_6", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_capture = 1, - .dpcm_merged_format = 1, - .dpcm_merged_chan = 1, - .dpcm_merged_rate = 1, - SND_SOC_DAILINK_REG(capture6), - }, - { - .name = "Capture_7", - .stream_name = "Capture_7", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(capture7), - }, - { - .name = "Hostless_LPBK", - .stream_name = "Hostless_LPBK", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_lpbk), - }, - { - .name = "Hostless_FM", - .stream_name = "Hostless_FM", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_fm), - }, - { - .name = "Hostless_SRC_1", - .stream_name = "Hostless_SRC_1", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_src1), - }, - { - .name = "Hostless_SRC_Bargein", - .stream_name = "Hostless_SRC_Bargein", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_src_bargein), - }, - { - .name = "Hostless_HW_Gain_AAudio", - .stream_name = "Hostless_HW_Gain_AAudio", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_hw_gain_aaudio), - }, - { - .name = "Hostless_SRC_AAudio", - .stream_name = "Hostless_SRC_AAudio", - .trigger = {SND_SOC_DPCM_TRIGGER_PRE, - SND_SOC_DPCM_TRIGGER_PRE}, - .dynamic = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_src_aaudio), - }, - /* Back End DAI links */ - { - .name = "Primary Codec", - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - .init = mt8186_mt6366_init, - SND_SOC_DAILINK_REG(adda), - }, - { - .name = "I2S3", - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_IB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .dpcm_playback = 1, - .ignore_suspend = 1, - .init = mt8186_mt6366_da7219_max98357_hdmi_init, - .be_hw_params_fixup = mt8186_anx7625_i2s_hw_params_fixup, - SND_SOC_DAILINK_REG(i2s3), - }, - { - .name = "I2S0", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - .be_hw_params_fixup = mt8186_i2s_hw_params_fixup, - .ops = &mt8186_da7219_i2s_ops, - SND_SOC_DAILINK_REG(i2s0), - }, - { - .name = "I2S1", - .no_pcm = 1, - .dpcm_playback = 1, - .ignore_suspend = 1, - .be_hw_params_fixup = mt8186_i2s_hw_params_fixup, - .init = mt8186_da7219_init, - .ops = &mt8186_da7219_i2s_ops, - SND_SOC_DAILINK_REG(i2s1), - }, - { - .name = "I2S2", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - .be_hw_params_fixup = mt8186_i2s_hw_params_fixup, - SND_SOC_DAILINK_REG(i2s2), - }, - { - .name = "HW Gain 1", - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hw_gain1), - }, - { - .name = "HW Gain 2", - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hw_gain2), - }, - { - .name = "HW_SRC_1", - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hw_src1), - }, - { - .name = "HW_SRC_2", - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hw_src2), - }, - { - .name = "CONNSYS_I2S", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(connsys_i2s), - }, - { - .name = "PCM 1", - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF, - .no_pcm = 1, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(pcm1), - }, - { - .name = "TDM IN", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(tdm_in), - }, - /* dummy BE for ul memif to record from dl memif */ - { - .name = "Hostless_UL1", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_ul1), - }, - { - .name = "Hostless_UL2", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_ul2), - }, - { - .name = "Hostless_UL3", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_ul3), - }, - { - .name = "Hostless_UL5", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_ul5), - }, - { - .name = "Hostless_UL6", - .no_pcm = 1, - .dpcm_capture = 1, - .ignore_suspend = 1, - SND_SOC_DAILINK_REG(hostless_ul6), - }, - /* SOF BE */ - { - .name = "AFE_SOF_DL1", - .no_pcm = 1, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(AFE_SOF_DL1), - }, - { - .name = "AFE_SOF_DL2", - .no_pcm = 1, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(AFE_SOF_DL2), - }, - { - .name = "AFE_SOF_UL1", - .no_pcm = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(AFE_SOF_UL1), - }, - { - .name = "AFE_SOF_UL2", - .no_pcm = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(AFE_SOF_UL2), - }, -}; - -static const struct snd_soc_dapm_widget -mt8186_mt6366_da7219_max98357_widgets[] = { - SND_SOC_DAPM_SPK("Speakers", NULL), - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_OUTPUT("HDMI1"), - SND_SOC_DAPM_MIXER(SOF_DMA_DL1, SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER(SOF_DMA_DL2, SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER(SOF_DMA_UL1, SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER(SOF_DMA_UL2, SND_SOC_NOPM, 0, 0, NULL, 0), -}; - -static const struct snd_soc_dapm_route -mt8186_mt6366_da7219_max98357_routes[] = { - /* SPK */ - { "Speakers", NULL, "Speaker"}, - /* Headset */ - { "Headphones", NULL, "HPL" }, - { "Headphones", NULL, "HPR" }, - { "MIC", NULL, "Headset Mic" }, - /* HDMI */ - { "HDMI1", NULL, "TX"}, - /* SOF Uplink */ - {SOF_DMA_UL1, NULL, "UL1_CH1"}, - {SOF_DMA_UL1, NULL, "UL1_CH2"}, - {SOF_DMA_UL2, NULL, "UL2_CH1"}, - {SOF_DMA_UL2, NULL, "UL2_CH2"}, - /* SOF Downlink */ - {"DSP_DL1_VIRT", NULL, SOF_DMA_DL1}, - {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2}, -}; - -static const struct snd_kcontrol_new -mt8186_mt6366_da7219_max98357_controls[] = { - SOC_DAPM_PIN_SWITCH("Speakers"), - SOC_DAPM_PIN_SWITCH("Headphones"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Line Out"), - SOC_DAPM_PIN_SWITCH("HDMI1"), -}; - -static struct snd_soc_card mt8186_mt6366_da7219_max98357_soc_card = { - .name = "mt8186_da7219_max98357", - .owner = THIS_MODULE, - .dai_link = mt8186_mt6366_da7219_max98357_dai_links, - .num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links), - .controls = mt8186_mt6366_da7219_max98357_controls, - .num_controls = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_controls), - .dapm_widgets = mt8186_mt6366_da7219_max98357_widgets, - .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_widgets), - .dapm_routes = mt8186_mt6366_da7219_max98357_routes, - .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_routes), - .codec_conf = mt8186_mt6366_da7219_max98357_codec_conf, - .num_configs = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_codec_conf), -}; - -static int mt8186_mt6366_da7219_max98357_dev_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card; - struct snd_soc_dai_link *dai_link; - struct mtk_soc_card_data *soc_card_data; - struct mt8186_mt6366_da7219_max98357_priv *mach_priv; - struct device_node *platform_node, *headset_codec, *playback_codec, *adsp_node; - int sof_on = 0; - int ret, i; - - card = (struct snd_soc_card *)device_get_match_data(&pdev->dev); - if (!card) - return -EINVAL; - card->dev = &pdev->dev; - - soc_card_data = devm_kzalloc(&pdev->dev, sizeof(*soc_card_data), GFP_KERNEL); - if (!soc_card_data) - return -ENOMEM; - mach_priv = devm_kzalloc(&pdev->dev, sizeof(*mach_priv), GFP_KERNEL); - if (!mach_priv) - return -ENOMEM; - - soc_card_data->mach_priv = mach_priv; - - adsp_node = of_parse_phandle(pdev->dev.of_node, "mediatek,adsp", 0); - if (adsp_node) { - struct mtk_sof_priv *sof_priv; - - sof_priv = devm_kzalloc(&pdev->dev, sizeof(*sof_priv), GFP_KERNEL); - if (!sof_priv) { - ret = -ENOMEM; - goto err_adsp_node; - } - sof_priv->conn_streams = g_sof_conn_streams; - sof_priv->num_streams = ARRAY_SIZE(g_sof_conn_streams); - sof_priv->sof_dai_link_fixup = mt8186_sof_dai_link_fixup; - soc_card_data->sof_priv = sof_priv; - card->probe = mtk_sof_card_probe; - card->late_probe = mtk_sof_card_late_probe; - if (!card->topology_shortname_created) { - snprintf(card->topology_shortname, 32, "sof-%s", card->name); - card->topology_shortname_created = true; - } - card->name = card->topology_shortname; - sof_on = 1; - } else { - dev_dbg(&pdev->dev, "Probe without adsp\n"); - } - - if (of_property_read_bool(pdev->dev.of_node, "mediatek,dai-link")) { - ret = mtk_sof_dailink_parse_of(card, pdev->dev.of_node, - "mediatek,dai-link", - mt8186_mt6366_da7219_max98357_dai_links, - ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links)); - if (ret) { - dev_dbg(&pdev->dev, "Parse dai-link fail\n"); - goto err_adsp_node; - } - } else { - if (!sof_on) - card->num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links) - - ARRAY_SIZE(g_sof_conn_streams); - } - - platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0); - if (!platform_node) { - ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'platform' missing or invalid\n"); - goto err_platform_node; - } - - playback_codec = of_get_child_by_name(pdev->dev.of_node, "playback-codecs"); - if (!playback_codec) { - ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'speaker-codecs' missing or invalid\n"); - goto err_playback_codec; - } - - headset_codec = of_get_child_by_name(pdev->dev.of_node, "headset-codec"); - if (!headset_codec) { - ret = -EINVAL; - dev_err_probe(&pdev->dev, ret, "Property 'headset-codec' missing or invalid\n"); - goto err_headset_codec; - } - - for_each_card_prelinks(card, i, dai_link) { - ret = mt8186_mt6366_card_set_be_link(card, dai_link, playback_codec, "I2S3"); - if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set speaker_codec fail\n", - dai_link->name); - goto err_probe; - } - - ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S0"); - if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n", - dai_link->name); - goto err_probe; - } - - ret = mt8186_mt6366_card_set_be_link(card, dai_link, headset_codec, "I2S1"); - if (ret) { - dev_err_probe(&pdev->dev, ret, "%s set headset_codec fail\n", - dai_link->name); - goto err_probe; - } - - if (!strncmp(dai_link->name, "AFE_SOF", strlen("AFE_SOF")) && sof_on) - dai_link->platforms->of_node = adsp_node; - - if (!dai_link->platforms->name && !dai_link->platforms->of_node) - dai_link->platforms->of_node = platform_node; - } - - snd_soc_card_set_drvdata(card, soc_card_data); - - ret = mt8186_afe_gpio_init(&pdev->dev); - if (ret) { - dev_err_probe(&pdev->dev, ret, "%s init gpio error\n", __func__); - goto err_probe; - } - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", __func__); - -err_probe: - of_node_put(headset_codec); -err_headset_codec: - of_node_put(playback_codec); -err_playback_codec: - of_node_put(platform_node); -err_platform_node: -err_adsp_node: - of_node_put(adsp_node); - - return ret; -} - -#if IS_ENABLED(CONFIG_OF) -static const struct of_device_id mt8186_mt6366_da7219_max98357_dt_match[] = { - { .compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound", - .data = &mt8186_mt6366_da7219_max98357_soc_card, - }, - {} -}; -MODULE_DEVICE_TABLE(of, mt8186_mt6366_da7219_max98357_dt_match); -#endif - -static struct platform_driver mt8186_mt6366_da7219_max98357_driver = { - .driver = { - .name = "mt8186_mt6366_da7219_max98357", -#if IS_ENABLED(CONFIG_OF) - .of_match_table = mt8186_mt6366_da7219_max98357_dt_match, -#endif - .pm = &snd_soc_pm_ops, - }, - .probe = mt8186_mt6366_da7219_max98357_dev_probe, -}; - -module_platform_driver(mt8186_mt6366_da7219_max98357_driver); - -/* Module information */ -MODULE_DESCRIPTION("MT8186-MT6366-DA7219-MAX98357 ALSA SoC machine driver"); -MODULE_AUTHOR("Jiaxin Yu "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("mt8186_mt6366_da7219_max98357 soc card"); diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366.c similarity index 83% rename from sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c rename to sound/soc/mediatek/mt8186/mt8186-mt6366.c index 28140ca0ebff8..8fd31c52b077e 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366.c @@ -1,11 +1,14 @@ // SPDX-License-Identifier: GPL-2.0 // -// mt8186-mt6366-rt1019-rt5682s.c -// -- MT8186-MT6366-RT1019-RT5682S ALSA SoC machine driver +// mt8186-mt6366.c +// -- MT8186-MT6366 ALSA SoC machine driver // // Copyright (c) 2022 MediaTek Inc. // Author: Jiaxin Yu // +// Copyright (c) 2024 Collabora Ltd. +// AngeloGioacchino Del Regno +// #include #include @@ -16,6 +19,7 @@ #include #include +#include "../../codecs/da7219.h" #include "../../codecs/mt6358.h" #include "../../codecs/rt5682.h" #include "../common/mtk-afe-platform-driver.h" @@ -33,11 +37,16 @@ #define RT5682S_CODEC_DAI "rt5682s-aif1" #define RT5682S_DEV0_NAME "rt5682s.5-001a" +#define DA7219_CODEC_DAI "da7219-hifi" +#define DA7219_DEV_NAME "da7219.5-001a" + #define SOF_DMA_DL1 "SOF_DMA_DL1" #define SOF_DMA_DL2 "SOF_DMA_DL2" #define SOF_DMA_UL1 "SOF_DMA_UL1" #define SOF_DMA_UL2 "SOF_DMA_UL2" +#define DA7219_CODEC_PRESENT BIT(0) + struct mt8186_mt6366_rt1019_rt5682s_priv { struct gpio_desc *dmic_sel; int dmic_switch; @@ -164,7 +173,7 @@ static int primary_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static int mt8186_rt5682s_init(struct snd_soc_pcm_runtime *rtd) +static int mt8186_headset_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *cmpnt_afe = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); @@ -174,6 +183,13 @@ static int mt8186_rt5682s_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8186_JACK_HEADSET]; struct snd_soc_component *cmpnt_codec = snd_soc_rtd_to_codec(rtd, 0)->component; + const int hs_keys_rt5682[] = { + KEY_PLAYPAUSE, KEY_VOLUMEUP, KEY_VOLUMEDOWN, KEY_VOICECOMMAND + }; + const int hs_keys_da7219[] = { + KEY_PLAYPAUSE, KEY_VOICECOMMAND, KEY_VOLUMEUP, KEY_VOLUMEDOWN + }; + const int *hs_keys; int ret; int type; @@ -194,15 +210,90 @@ static int mt8186_rt5682s_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + if (soc_card_data->card_data->flags & DA7219_CODEC_PRESENT) + hs_keys = hs_keys_da7219; + else + hs_keys = hs_keys_rt5682; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, hs_keys[0]); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, hs_keys[1]); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, hs_keys[2]); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, hs_keys[3]); type = SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3; return snd_soc_component_set_jack(cmpnt_codec, jack, (void *)&type); } +static int mt8186_da7219_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai; + unsigned int rate = params_rate(params); + unsigned int mclk_fs_ratio = 256; + unsigned int mclk_fs = rate * mclk_fs_ratio; + unsigned int freq; + int ret, j; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_fs, SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->dev, "failed to set cpu dai sysclk: %d\n", ret); + return ret; + } + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (strcmp(codec_dai->component->name, DA7219_DEV_NAME)) + continue; + + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, + mclk_fs, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "failed to set sysclk: %d\n", ret); + return ret; + } + + if ((rate % 8000) == 0) + freq = DA7219_PLL_FREQ_OUT_98304; + else + freq = DA7219_PLL_FREQ_OUT_90316; + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, + 0, freq); + if (ret) { + dev_err(rtd->dev, "failed to start PLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int mt8186_da7219_i2s_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int j, ret; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (strcmp(codec_dai->component->name, DA7219_DEV_NAME)) + continue; + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(rtd->dev, "failed to stop PLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_ops mt8186_da7219_i2s_ops = { + .hw_params = mt8186_da7219_i2s_hw_params, + .hw_free = mt8186_da7219_i2s_hw_free, +}; + static int mt8186_rt5682s_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -302,14 +393,14 @@ static int mt8186_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int mt8186_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static int mt8186_i2s_hw_params_24le_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) { return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S24_LE); } -static int mt8186_it6505_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +static int mt8186_i2s_hw_params_32le_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) { return mt8186_hw_params_fixup(rtd, params, SNDRV_PCM_FORMAT_S32_LE); } @@ -318,16 +409,24 @@ static int mt8186_it6505_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8186_sof_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { + struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); int ret; ret = mtk_sof_dai_link_fixup(rtd, params); if (!strcmp(rtd->dai_link->name, "I2S0") || !strcmp(rtd->dai_link->name, "I2S1") || - !strcmp(rtd->dai_link->name, "I2S2")) - mt8186_i2s_hw_params_fixup(rtd, params); - else if (!strcmp(rtd->dai_link->name, "I2S3")) - mt8186_it6505_i2s_hw_params_fixup(rtd, params); + !strcmp(rtd->dai_link->name, "I2S2")) { + if (soc_card_data->card_data->flags & DA7219_CODEC_PRESENT) + mt8186_i2s_hw_params_32le_fixup(rtd, params); + else + mt8186_i2s_hw_params_24le_fixup(rtd, params); + } else if (!strcmp(rtd->dai_link->name, "I2S3")) { + if (soc_card_data->card_data->flags & DA7219_CODEC_PRESENT) + mt8186_i2s_hw_params_24le_fixup(rtd, params); + else + mt8186_i2s_hw_params_32le_fixup(rtd, params); + } return ret; } @@ -792,7 +891,6 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .dpcm_playback = 1, .ignore_suspend = 1, .init = mt8186_mt6366_rt1019_rt5682s_hdmi_init, - .be_hw_params_fixup = mt8186_it6505_i2s_hw_params_fixup, SND_SOC_DAILINK_REG(i2s3), }, { @@ -800,7 +898,6 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .no_pcm = 1, .dpcm_capture = 1, .ignore_suspend = 1, - .be_hw_params_fixup = mt8186_i2s_hw_params_fixup, .ops = &mt8186_rt5682s_i2s_ops, SND_SOC_DAILINK_REG(i2s0), }, @@ -809,9 +906,7 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, - .be_hw_params_fixup = mt8186_i2s_hw_params_fixup, - .init = mt8186_rt5682s_init, - .ops = &mt8186_rt5682s_i2s_ops, + .init = mt8186_headset_codec_init, SND_SOC_DAILINK_REG(i2s1), }, { @@ -819,7 +914,6 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { .no_pcm = 1, .dpcm_capture = 1, .ignore_suspend = 1, - .be_hw_params_fixup = mt8186_i2s_hw_params_fixup, SND_SOC_DAILINK_REG(i2s2), }, { @@ -941,6 +1035,19 @@ static struct snd_soc_dai_link mt8186_mt6366_rt1019_rt5682s_dai_links[] = { }, }; +static const struct snd_soc_dapm_widget +mt8186_mt6366_da7219_max98357_widgets[] = { + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("HDMI1", NULL), + SND_SOC_DAPM_MIXER(SOF_DMA_DL1, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_DL2, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_UL1, SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER(SOF_DMA_UL2, SND_SOC_NOPM, 0, 0, NULL, 0), +}; + static const struct snd_soc_dapm_widget mt8186_mt6366_rt1019_rt5682s_widgets[] = { SND_SOC_DAPM_SPK("Speakers", NULL), @@ -994,6 +1101,14 @@ static const struct snd_soc_dapm_route mt8186_mt6366_rt5650_routes[] = { {"DSP_DL2_VIRT", NULL, SOF_DMA_DL2}, }; +static const struct snd_kcontrol_new mt8186_mt6366_da7219_max98357_controls[] = { + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Line Out"), + SOC_DAPM_PIN_SWITCH("HDMI1"), +}; + static const struct snd_kcontrol_new mt8186_mt6366_rt1019_rt5682s_controls[] = { SOC_DAPM_PIN_SWITCH("Speakers"), @@ -1002,6 +1117,21 @@ mt8186_mt6366_rt1019_rt5682s_controls[] = { SOC_DAPM_PIN_SWITCH("HDMI1"), }; +static struct snd_soc_card mt8186_mt6366_da7219_max98357_soc_card = { + .name = "mt8186_da7219_max98357", + .owner = THIS_MODULE, + .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links, + .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links), + .controls = mt8186_mt6366_da7219_max98357_controls, + .num_controls = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_controls), + .dapm_widgets = mt8186_mt6366_da7219_max98357_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_widgets), + .dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes, + .num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes), + .codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf, + .num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf), +}; + static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = { .name = "mt8186_rt1019_rt5682s", .owner = THIS_MODULE, @@ -1100,8 +1230,9 @@ static int mt8186_mt6366_soc_card_probe(struct mtk_soc_card_data *soc_card_data, { struct mtk_platform_card_data *card_data = soc_card_data->card_data; struct snd_soc_card *card = card_data->card; + struct snd_soc_dai_link *dai_link; struct mt8186_mt6366_rt1019_rt5682s_priv *mach_priv; - int ret; + int i, ret; mach_priv = devm_kzalloc(card->dev, sizeof(*mach_priv), GFP_KERNEL); if (!mach_priv) @@ -1115,6 +1246,25 @@ static int mt8186_mt6366_soc_card_probe(struct mtk_soc_card_data *soc_card_data, return dev_err_probe(card->dev, PTR_ERR(mach_priv->dmic_sel), "DMIC gpio failed\n"); + for_each_card_prelinks(card, i, dai_link) { + if (strcmp(dai_link->name, "I2S0") == 0 || + strcmp(dai_link->name, "I2S1") == 0 || + strcmp(dai_link->name, "I2S2") == 0) { + if (card_data->flags & DA7219_CODEC_PRESENT) { + dai_link->be_hw_params_fixup = mt8186_i2s_hw_params_32le_fixup; + dai_link->ops = &mt8186_da7219_i2s_ops; + } else { + dai_link->be_hw_params_fixup = mt8186_i2s_hw_params_24le_fixup; + dai_link->ops = &mt8186_rt5682s_i2s_ops; + } + } else if (strcmp(dai_link->name, "I2S3") == 0) { + if (card_data->flags & DA7219_CODEC_PRESENT) + dai_link->be_hw_params_fixup = mt8186_i2s_hw_params_24le_fixup; + else + dai_link->be_hw_params_fixup = mt8186_i2s_hw_params_32le_fixup; + } + } + if (legacy) { ret = mt8186_mt6366_legacy_probe(soc_card_data); if (ret) @@ -1160,6 +1310,18 @@ static const struct mtk_sof_priv mt8186_sof_priv = { .sof_dai_link_fixup = mt8186_sof_dai_link_fixup }; +static const struct mtk_soundcard_pdata mt8186_mt6366_da7219_max98357_pdata = { + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8186_mt6366_da7219_max98357_soc_card, + .num_jacks = MT8186_JACK_MAX, + .pcm_constraints = mt8186_pcm_constraints, + .num_pcm_constraints = ARRAY_SIZE(mt8186_pcm_constraints), + .flags = DA7219_CODEC_PRESENT, + }, + .sof_priv = &mt8186_sof_priv, + .soc_probe = mt8186_mt6366_soc_card_probe +}; + static const struct mtk_soundcard_pdata mt8186_mt6366_rt1019_rt5682s_pdata = { .card_data = &(struct mtk_platform_card_data) { .card = &mt8186_mt6366_rt1019_rt5682s_soc_card, @@ -1194,7 +1356,7 @@ static const struct mtk_soundcard_pdata mt8186_mt6366_rt5650_pdata = { }; #if IS_ENABLED(CONFIG_OF) -static const struct of_device_id mt8186_mt6366_rt1019_rt5682s_dt_match[] = { +static const struct of_device_id mt8186_mt6366_dt_match[] = { { .compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound", .data = &mt8186_mt6366_rt1019_rt5682s_pdata, @@ -1207,26 +1369,30 @@ static const struct of_device_id mt8186_mt6366_rt1019_rt5682s_dt_match[] = { .compatible = "mediatek,mt8186-mt6366-rt5650-sound", .data = &mt8186_mt6366_rt5650_pdata, }, - {} + { + .compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound", + .data = &mt8186_mt6366_da7219_max98357_pdata, + }, + { /* sentinel */ } }; -MODULE_DEVICE_TABLE(of, mt8186_mt6366_rt1019_rt5682s_dt_match); +MODULE_DEVICE_TABLE(of, mt8186_mt6366_dt_match); #endif -static struct platform_driver mt8186_mt6366_rt1019_rt5682s_driver = { +static struct platform_driver mt8186_mt6366_driver = { .driver = { - .name = "mt8186_mt6366_rt1019_rt5682s", + .name = "mt8186_mt6366", #if IS_ENABLED(CONFIG_OF) - .of_match_table = mt8186_mt6366_rt1019_rt5682s_dt_match, + .of_match_table = mt8186_mt6366_dt_match, #endif .pm = &snd_soc_pm_ops, }, .probe = mtk_soundcard_common_probe, }; -module_platform_driver(mt8186_mt6366_rt1019_rt5682s_driver); +module_platform_driver(mt8186_mt6366_driver); /* Module information */ -MODULE_DESCRIPTION("MT8186-MT6366-RT1019-RT5682S ALSA SoC machine driver"); +MODULE_DESCRIPTION("MT8186-MT6366 ALSA SoC machine driver"); MODULE_AUTHOR("Jiaxin Yu "); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("mt8186_mt6366_rt1019_rt5682s soc card"); +MODULE_ALIAS("mt8186_mt6366 soc card"); From dc7a31300a8f28af282974512c2d403ada0658b5 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:06 +0200 Subject: [PATCH 301/603] ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode Document the dai-link subnodes and the audio-routing property, allowing to describe machine specific audio hardware and links in device tree. While at it, also deprecate the old properties which were previously used with driver hardcoded configuration. Also, since describing machine specific audio hardware and links replaces the now deprecated old logic doing the same in a driver hardcoded fashion, it is not allowed to have both the old and new properties together. Reviewed-by: Rob Herring Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-15-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../bindings/sound/mt8195-mt6359.yaml | 134 ++++++++++++++++++ 1 file changed, 134 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml index c1ddbf672ca3f..2af1d8ffbd8b5 100644 --- a/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml +++ b/Documentation/devicetree/bindings/sound/mt8195-mt6359.yaml @@ -12,6 +12,9 @@ maintainers: description: This binding describes the MT8195 sound card. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -23,6 +26,33 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: User specified audio sound card name + audio-routing: + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + Valid names could be the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - Ext Spk + - Headphone + - IN1P + - Left Spk + - Right Spk + + # Sources + - Headset Mic + - HPOL + - HPOR + - Left BE_OUT + - Left SPO + - Right BE_OUT + - Right SPO + - Speaker + mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8195 ASoC platform. @@ -30,10 +60,12 @@ properties: mediatek,dptx-codec: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8195 Display Port Tx codec node. + deprecated: true mediatek,hdmi-codec: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8195 HDMI codec node. + deprecated: true mediatek,adsp: $ref: /schemas/types.yaml#/definitions/phandle @@ -45,20 +77,122 @@ properties: A list of the desired dai-links in the sound card. Each entry is a name defined in the machine driver. +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: + - DPTX_BE + - ETDM1_IN_BE + - ETDM2_IN_BE + - ETDM1_OUT_BE + - ETDM2_OUT_BE + - ETDM3_OUT_BE + - PCM1_BE + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + additionalProperties: false required: - compatible - mediatek,platform +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + mediatek,dptx-codec: false + mediatek,hdmi-codec: false + examples: - | sound: mt8195-sound { compatible = "mediatek,mt8195_mt6359_rt1019_rt5682"; + model = "mt8195_r1019_5682"; mediatek,platform = <&afe>; pinctrl-names = "default"; pinctrl-0 = <&aud_pins_default>; + + audio-routing = + "Headphone", "HPOL", + "Headphone", "HPOR", + "IN1P", "Headset Mic", + "Ext Spk", "Speaker"; + + mm-dai-link { + link-name = "ETDM1_IN_BE"; + mediatek,clk-provider = "cpu"; + }; + + hs-playback-dai-link { + link-name = "ETDM1_OUT_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&headset_codec>; + }; + }; + + hs-capture-dai-link { + link-name = "ETDM2_IN_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&headset_codec>; + }; + }; + + spk-playback-dai-link { + link-name = "ETDM2_OUT_BE"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&spk_amplifier>; + }; + }; + + hdmi-dai-link { + link-name = "ETDM3_OUT_BE"; + codec { + sound-dai = <&hdmi_tx>; + }; + }; + + displayport-dai-link { + link-name = "DPTX_BE"; + codec { + sound-dai = <&dp_tx>; + }; + }; }; ... From 5a50483c47b8cf3066eb151fa21e195f3dfab5a8 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:07 +0200 Subject: [PATCH 302/603] ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode Document the dai-link subnodes and the audio-routing property, allowing to describe machine specific audio hardware and links in device tree. While at it, also deprecate the old properties which were previously used with the driver's partially hardcoded configuration. Also, since describing machine specific audio hardware and links replaces the now deprecated old logic doing the same in a driver hardcoded fashion, it is not allowed to have both the old and new properties together. Reviewed-by: Rob Herring Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-16-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 139 ++++++++++++++++-- 1 file changed, 129 insertions(+), 10 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index 7e50f5d65c8fb..c4e68f31aaabd 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -13,6 +13,9 @@ maintainers: description: This binding describes the MT8192 sound card. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -20,6 +23,31 @@ properties: - mediatek,mt8192_mt6359_rt1015p_rt5682 - mediatek,mt8192_mt6359_rt1015p_rt5682s + audio-routing: + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + Valid names could be the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - Speakers + - Headphone Jack + - IN1P + - Left Spk + - Right Spk + + # Sources + - Headset Mic + - HPOL + - HPOR + - Left SPO + - Right SPO + - Speaker + mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8192 ASoC platform. @@ -27,10 +55,12 @@ properties: mediatek,hdmi-codec: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of HDMI codec. + deprecated: true headset-codec: type: object additionalProperties: false + deprecated: true properties: sound-dai: @@ -41,6 +71,7 @@ properties: speaker-codecs: type: object additionalProperties: false + deprecated: true properties: sound-dai: @@ -51,33 +82,121 @@ properties: required: - sound-dai -additionalProperties: false +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: + - I2S0 + - I2S1 + - I2S2 + - I2S3 + - I2S4 + - I2S5 + - I2S6 + - I2S7 + - I2S8 + - I2S9 + - TDM + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + +unevaluatedProperties: false required: - compatible - mediatek,platform - - headset-codec - - speaker-codecs + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false + mediatek,hdmi-codec: false examples: - | sound: mt8192-sound { compatible = "mediatek,mt8192_mt6359_rt1015_rt5682"; - mediatek,platform = <&afe>; - mediatek,hdmi-codec = <&anx_bridge_dp>; + model = "mt8192_mt6359_rt1015_rt5682"; pinctrl-names = "aud_clk_mosi_off", "aud_clk_mosi_on"; pinctrl-0 = <&aud_clk_mosi_off>; pinctrl-1 = <&aud_clk_mosi_on>; + mediatek,platform = <&afe>; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR", + "IN1P", "Headset Mic", + "Speakers", "Speaker"; + + spk-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt1015p>; + }; + }; + + hs-playback-dai-link { + link-name = "I2S8"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682 0>; + }; + }; - headset-codec { - sound-dai = <&rt5682>; + hs-capture-dai-link { + link-name = "I2S9"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682 0>; + }; }; - speaker-codecs { - sound-dai = <&rt1015_l>, - <&rt1015_r>; + displayport-dai-link { + link-name = "TDM"; + dai-format = "dsp_a"; + codec { + sound-dai = <&anx_bridge_dp>; + }; }; }; From 787f190cd1cd3123555e33a3400f3e919ec94a81 Mon Sep 17 00:00:00 2001 From: AngeloGioacchino Del Regno Date: Tue, 16 Apr 2024 09:14:08 +0200 Subject: [PATCH 303/603] ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode Document the dai-link subnodes and the audio-routing property, allowing to describe machine specific audio hardware and links in device tree. While at it, also deprecate the old properties which were previously used with the driver's partially hardcoded configuration. Also, since describing machine specific audio hardware and links replaces the now deprecated old logic doing the same in a driver hardcoded fashion, it is not allowed to have both the old and new properties together. Reviewed-by: Rob Herring Signed-off-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20240416071410.75620-17-angelogioacchino.delregno@collabora.com Signed-off-by: Mark Brown --- .../sound/mt8186-mt6366-da7219-max98357.yaml | 131 ++++++++++++++++-- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 120 ++++++++++++++-- 2 files changed, 233 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml index 9853c11a1330b..cbc641ecbe94a 100644 --- a/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml @@ -12,17 +12,46 @@ maintainers: description: This binding describes the MT8186 sound card. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: - mediatek,mt8186-mt6366-da7219-max98357-sound + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + Valid names could be the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - HDMI1 + - Headphones + - Line Out + - MIC + - Speakers + + # Sources + - Headset Mic + - HPL + - HPR + - Speaker + - TX + mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8186 ASoC platform. headset-codec: type: object + deprecated: true additionalProperties: false properties: sound-dai: @@ -32,6 +61,7 @@ properties: playback-codecs: type: object + deprecated: true additionalProperties: false properties: sound-dai: @@ -53,32 +83,115 @@ properties: A list of the desired dai-links in the sound card. Each entry is a name defined in the machine driver. -additionalProperties: false +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + items: + enum: + - I2S0 + - I2S1 + - I2S2 + - I2S3 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + items: + enum: + - cpu + - codec + + required: + - link-name + +unevaluatedProperties: false required: - compatible - mediatek,platform - - headset-codec - - playback-codecs + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false examples: - | sound: mt8186-sound { compatible = "mediatek,mt8186-mt6366-da7219-max98357-sound"; - mediatek,platform = <&afe>; + model = "mt8186_da7219_m98357"; pinctrl-names = "aud_clk_mosi_off", "aud_clk_mosi_on"; pinctrl-0 = <&aud_clk_mosi_off>; pinctrl-1 = <&aud_clk_mosi_on>; + mediatek,platform = <&afe>; + + audio-routing = + "Headphones", "HPL", + "Headphones", "HPR", + "MIC", "Headset Mic", + "Speakers", "Speaker", + "HDMI1", "TX"; + + hs-playback-dai-link { + link-name = "I2S0"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&da7219>; + }; + }; - headset-codec { - sound-dai = <&da7219>; + hs-capture-dai-link { + link-name = "I2S1"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&da7219>; + }; }; - playback-codecs { - sound-dai = <&anx_bridge_dp>, - <&max98357a>; + spk-dp-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&anx_bridge_dp>, <&max98357a>; + }; }; }; diff --git a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml index bdf7b09605330..ed93f18ef985c 100644 --- a/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml @@ -12,6 +12,9 @@ maintainers: description: This binding describes the MT8186 sound card. +allOf: + - $ref: sound-card-common.yaml# + properties: compatible: enum: @@ -19,6 +22,34 @@ properties: - mediatek,mt8186-mt6366-rt5682s-max98360-sound - mediatek,mt8186-mt6366-rt5650-sound + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + Valid names could be the input or output widgets of audio components, + power supplies, MicBias of codec and the software switch. + minItems: 2 + items: + enum: + # Sinks + - HDMI1 + - Headphone + - IN1P + - IN1N + - Line Out + - Speakers + + # Sources + - Headset Mic + - HPOL + - HPOR + - Speaker + - SPOL + - SPOR + - TX + mediatek,platform: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of MT8186 ASoC platform. @@ -32,6 +63,7 @@ properties: headset-codec: type: object + deprecated: true additionalProperties: false properties: sound-dai: @@ -41,6 +73,7 @@ properties: playback-codecs: type: object + deprecated: true additionalProperties: false properties: sound-dai: @@ -62,13 +95,56 @@ properties: A list of the desired dai-links in the sound card. Each entry is a name defined in the machine driver. -additionalProperties: false +patternProperties: + ".*-dai-link$": + type: object + additionalProperties: false + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + link-name: + description: Indicates dai-link name and PCM stream name + enum: [ I2S0, I2S1, I2S2, I2S3 ] + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + minItems: 1 + maxItems: 2 + required: + - sound-dai + + dai-format: + description: audio format + enum: [ i2s, right_j, left_j, dsp_a, dsp_b ] + + mediatek,clk-provider: + $ref: /schemas/types.yaml#/definitions/string + description: Indicates dai-link clock master. + enum: [ cpu, codec ] + + required: + - link-name + +unevaluatedProperties: false required: - compatible - mediatek,platform - - headset-codec - - playback-codecs + +# Disallow legacy properties if xxx-dai-link nodes are specified +if: + not: + patternProperties: + ".*-dai-link$": false +then: + properties: + headset-codec: false + speaker-codecs: false examples: - | @@ -76,23 +152,49 @@ examples: sound: mt8186-sound { compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound"; - mediatek,platform = <&afe>; + model = "mt8186_rt1019_rt5682s"; pinctrl-names = "aud_clk_mosi_off", "aud_clk_mosi_on", "aud_gpio_dmic_sec"; pinctrl-0 = <&aud_clk_mosi_off>; pinctrl-1 = <&aud_clk_mosi_on>; pinctrl-2 = <&aud_gpio_dmic_sec>; + mediatek,platform = <&afe>; dmic-gpios = <&pio 23 GPIO_ACTIVE_HIGH>; - headset-codec { - sound-dai = <&rt5682s>; + audio-routing = + "Headphone", "HPOL", + "Headphone", "HPOR", + "IN1P", "Headset Mic", + "Speakers", "Speaker", + "HDMI1", "TX"; + + hs-playback-dai-link { + link-name = "I2S0"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682s 0>; + }; + }; + + hs-capture-dai-link { + link-name = "I2S1"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&rt5682s 0>; + }; }; - playback-codecs { - sound-dai = <&it6505dptx>, - <&rt1019p>; + spk-hdmi-playback-dai-link { + link-name = "I2S3"; + dai-format = "i2s"; + mediatek,clk-provider = "cpu"; + codec { + sound-dai = <&it6505dptx>, <&rt1019p>; + }; }; }; From 58300f8d6a48e58d1843199be743f819e2791ea3 Mon Sep 17 00:00:00 2001 From: Steven Rostedt Date: Tue, 16 Apr 2024 00:03:03 -0400 Subject: [PATCH 304/603] ASoC: tracing: Export SND_SOC_DAPM_DIR_OUT to its value The string SND_SOC_DAPM_DIR_OUT is printed in the snd_soc_dapm_path trace event instead of its value: (((REC->path_dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-") User space cannot parse this, as it has no idea what SND_SOC_DAPM_DIR_OUT is. Use TRACE_DEFINE_ENUM() to convert it to its value: (((REC->path_dir) == 1) ? "->" : "<-") So that user space tools, such as perf and trace-cmd, can parse it correctly. Reported-by: Luca Ceresoli Fixes: 6e588a0d839b5 ("ASoC: dapm: Consolidate path trace events") Signed-off-by: Steven Rostedt (Google) Link: https://lore.kernel.org/r/20240416000303.04670cdf@rorschach.local.home Signed-off-by: Mark Brown --- include/trace/events/asoc.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 4eed9028bb119..517015ef36a84 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -12,6 +12,8 @@ #define DAPM_DIRECT "(direct)" #define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-") +TRACE_DEFINE_ENUM(SND_SOC_DAPM_DIR_OUT); + struct snd_soc_jack; struct snd_soc_card; struct snd_soc_dapm_widget; From 5894ff6c4707af645f2faa36cf07cc2745e1658f Mon Sep 17 00:00:00 2001 From: Frank Li Date: Tue, 16 Apr 2024 10:47:47 -0400 Subject: [PATCH 305/603] ASoC: dt-bindings: fsl-esai: Remove 'fsl,*' from required list fsl,fifo-depth have default value 64 in driver(sound/soc/fsl/fsl_esai.c). fsl,esai-synchronous is flag(bool) type. It doesn't make sense to put flag type into 'required'. Fix warning: arch/arm/boot/dts/nxp/imx/imx6q-sabreauto.dtb: esai@2024000: 'fsl,fifo-depth' is a required property arch/arm/boot/dts/nxp/imx/imx6q-sabreauto.dtb: esai@2024000: 'fsl,esai-synchronous' is a required property Signed-off-by: Frank Li Link: https://lore.kernel.org/r/20240416-esai_arm_dts_warning-v2-1-879e59c0c3b8@nxp.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl,esai.yaml | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml index f167f1634d7e3..2a64e682b5178 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml @@ -67,6 +67,7 @@ properties: fsl,fifo-depth: $ref: /schemas/types.yaml#/definitions/uint32 + default: 64 description: The number of elements in the transmit and receive FIFOs. This number is the maximum allowed value for @@ -95,8 +96,6 @@ required: - clock-names - dmas - dma-names - - fsl,fifo-depth - - fsl,esai-synchronous unevaluatedProperties: false From 9aea6d64bb3acbff6b2f33cb5bbff65e824ab9a7 Mon Sep 17 00:00:00 2001 From: Frank Li Date: Tue, 16 Apr 2024 10:47:48 -0400 Subject: [PATCH 306/603] ASoC: dt-bindings: fsl-esai: Add ref: dai-common.yaml Add ref: dai-common.yaml to fix below warning. arch/arm/boot/dts/nxp/imx/imx6q-sabreauto.dtb: esai@2024000: Unevaluated properties are not allowed ('#sound-dai-cells' was unexpected) Reviewed-by: Rob Herring Signed-off-by: Frank Li Link: https://lore.kernel.org/r/20240416-esai_arm_dts_warning-v2-2-879e59c0c3b8@nxp.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl,esai.yaml | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.yaml b/Documentation/devicetree/bindings/sound/fsl,esai.yaml index 2a64e682b5178..f99ed20fa684a 100644 --- a/Documentation/devicetree/bindings/sound/fsl,esai.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,esai.yaml @@ -99,6 +99,9 @@ required: unevaluatedProperties: false +allOf: + - $ref: dai-common.yaml# + examples: - | esai@2024000 { From 9be51470d514eb20d4ce29684cf933ef1c5fff48 Mon Sep 17 00:00:00 2001 From: John Watts Date: Wed, 17 Apr 2024 15:14:43 +1000 Subject: [PATCH 307/603] ASoC: sunxi: sun4i-i2s: Fix pcm_formats type specification pcm_formats should be a u64 as it is a SNDRV_PCM_FMTBIT_* not a SNDRV_PCM_FORMAT_*. Also fix a small grammar error while we're here. Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202404170103.ySYwieqi-lkp@intel.com/ Signed-off-by: John Watts Link: https://lore.kernel.org/r/20240417-sunxi_s32_fix-v1-1-d82e451565c0@jookia.org Signed-off-by: Mark Brown --- sound/soc/sunxi/sun4i-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 59830f2a0d308..5f8d979585b69 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -156,7 +156,7 @@ struct sun4i_i2s; /** * struct sun4i_i2s_quirks - Differences between SoC variants. * @has_reset: SoC needs reset deasserted. - * @pcm_formats: available PCM formats + * @pcm_formats: available PCM formats. * @reg_offset_txdata: offset of the tx fifo. * @sun4i_i2s_regmap: regmap config to use. * @field_clkdiv_mclk_en: regmap field to enable mclk output. @@ -176,7 +176,7 @@ struct sun4i_i2s; */ struct sun4i_i2s_quirks { bool has_reset; - snd_pcm_format_t pcm_formats; + u64 pcm_formats; unsigned int reg_offset_txdata; /* TX FIFO */ const struct regmap_config *sun4i_i2s_regmap; From 4074f8d23278e9e32aabe9dba60f6dddaf68e6ef Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:03:14 +1030 Subject: [PATCH 308/603] ALSA: scarlett2: Move initialisation code lower in the source So that more forward declarations won't be required when we add handling of the ACK notification, move the initialisation functions to after the notification functions. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: <0922071cb8be99a2394705de27b917d1e4e46f3f.1710264833.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 1187 ++++++++++++++++++----------------- 1 file changed, 595 insertions(+), 592 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index bd114be537d7a..8390b646c0aef 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -6383,798 +6383,801 @@ static int scarlett2_add_power_status_ctl(struct usb_mixer_interface *mixer) &private->power_status_ctl); } -/*** Cleanup/Suspend Callbacks ***/ +/*** Notification Handlers ***/ -static void scarlett2_private_free(struct usb_mixer_interface *mixer) +/* Notify on sync change */ +static void scarlett2_notify_sync(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; - cancel_delayed_work_sync(&private->work); - kfree(private); - mixer->private_data = NULL; + private->sync_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->sync_ctl->id); } -static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) +/* Notify on monitor change (Gen 2/3) */ +static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) { + struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; + int i; - if (cancel_delayed_work_sync(&private->work)) - scarlett2_config_save(private->mixer); -} - -/*** Initialisation ***/ - -static void scarlett2_count_io(struct scarlett2_data *private) -{ - const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int port_type, srcs = 0, dsts = 0; + if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH)) + return; - /* Count the number of mux sources and destinations */ - for (port_type = 0; - port_type < SCARLETT2_PORT_TYPE_COUNT; - port_type++) { - srcs += port_count[port_type][SCARLETT2_PORT_IN]; - dsts += port_count[port_type][SCARLETT2_PORT_OUT]; - } + private->vol_updated = 1; - private->num_mux_srcs = srcs; - private->num_mux_dsts = dsts; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); - /* Mixer inputs are mux outputs and vice versa. - * Scarlett Gen 4 DSP I/O uses SCARLETT2_PORT_TYPE_MIX but - * doesn't have mixer controls. - */ - private->num_mix_in = - port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT] - - info->dsp_count; + for (i = 0; i < private->num_line_out; i++) + if (private->vol_sw_hw_switch[line_out_remap(private, i)]) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->vol_ctls[i]->id); +} - private->num_mix_out = - port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN] - - info->dsp_count; +/* Notify on volume change (Gen 4) */ +static void scarlett2_notify_volume(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; - /* Number of analogue line outputs */ - private->num_line_out = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; + private->vol_updated = 1; - /* Number of monitor mix controls */ - private->num_monitor_mix_ctls = - info->direct_monitor * 2 * private->num_mix_in; + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->headphone_vol_ctl->id); } -/* Look through the interface descriptors for the Focusrite Control - * interface (bInterfaceClass = 255 Vendor Specific Class) and set - * bInterfaceNumber, bEndpointAddress, wMaxPacketSize, and bInterval - * in private - */ -static int scarlett2_find_fc_interface(struct usb_device *dev, - struct scarlett2_data *private) +/* Notify on dim/mute change */ +static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) { - struct usb_host_config *config = dev->actconfig; + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; int i; - for (i = 0; i < config->desc.bNumInterfaces; i++) { - struct usb_interface *intf = config->interface[i]; - struct usb_interface_descriptor *desc = - &intf->altsetting[0].desc; - struct usb_endpoint_descriptor *epd; + if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH)) + return; - if (desc->bInterfaceClass != 255) - continue; + private->dim_mute_updated = 1; - epd = get_endpoint(intf->altsetting, 0); - private->bInterfaceNumber = desc->bInterfaceNumber; - private->bEndpointAddress = epd->bEndpointAddress & - USB_ENDPOINT_NUMBER_MASK; - private->wMaxPacketSize = le16_to_cpu(epd->wMaxPacketSize); - private->bInterval = epd->bInterval; - return 0; - } + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->dim_mute_ctls[i]->id); - return -EINVAL; + for (i = 0; i < private->num_line_out; i++) + if (private->vol_sw_hw_switch[line_out_remap(private, i)]) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->mute_ctls[i]->id); } -/* Initialise private data */ -static int scarlett2_init_private(struct usb_mixer_interface *mixer, - const struct scarlett2_device_entry *entry) +/* Notify on input level switch change */ +static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer) { - struct scarlett2_data *private = - kzalloc(sizeof(struct scarlett2_data), GFP_KERNEL); + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; - if (!private) - return -ENOMEM; + private->input_level_updated = 1; - mutex_init(&private->usb_mutex); - mutex_init(&private->data_mutex); - INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work); + for (i = 0; i < info->level_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->level_ctls[i]->id); +} - mixer->private_data = private; - mixer->private_free = scarlett2_private_free; - mixer->private_suspend = scarlett2_private_suspend; +/* Notify on input pad switch change */ +static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; - private->info = entry->info; - private->config_set = entry->info->config_set; - private->series_name = entry->series_name; - scarlett2_count_io(private); - private->scarlett2_seq = 0; - private->mixer = mixer; + private->input_pad_updated = 1; - return scarlett2_find_fc_interface(mixer->chip->dev, private); + for (i = 0; i < info->pad_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->pad_ctls[i]->id); } -/* Cargo cult proprietary initialisation sequence */ -static int scarlett2_usb_init(struct usb_mixer_interface *mixer) +/* Notify on input air switch change */ +static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer) { - struct usb_device *dev = mixer->chip->dev; + struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; - u8 step0_buf[24]; - u8 step2_buf[84]; - int err; + const struct scarlett2_device_info *info = private->info; + int i; - if (usb_pipe_type_check(dev, usb_sndctrlpipe(dev, 0))) - return -EINVAL; + private->input_air_updated = 1; - /* step 0 */ - err = scarlett2_usb_rx(dev, private->bInterfaceNumber, - SCARLETT2_USB_CMD_INIT, - step0_buf, sizeof(step0_buf)); - if (err < 0) - return err; + for (i = 0; i < info->air_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->air_ctls[i]->id); +} - /* step 1 */ - private->scarlett2_seq = 1; - err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_1, NULL, 0, NULL, 0); - if (err < 0) - return err; +/* Notify on input phantom switch change */ +static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; - /* step 2 */ - private->scarlett2_seq = 1; - err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_2, - NULL, 0, - step2_buf, sizeof(step2_buf)); - if (err < 0) - return err; + private->input_phantom_updated = 1; - /* extract 4-byte firmware version from step2_buf[8] */ - private->firmware_version = le32_to_cpu(*(__le32 *)(step2_buf + 8)); - usb_audio_info(mixer->chip, - "Firmware version %d\n", - private->firmware_version); + for (i = 0; i < info->phantom_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->phantom_ctls[i]->id); - return 0; + scarlett2_phantom_notify_access(mixer); } -/* Get the flash segment numbers for the App_Settings and App_Upgrade - * segments and put them in the private data - */ -static int scarlett2_get_flash_segment_nums(struct usb_mixer_interface *mixer) +/* Notify on "input other" change (level/pad/air/phantom) */ +static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer) +{ + scarlett2_notify_input_level(mixer); + scarlett2_notify_input_pad(mixer); + scarlett2_notify_input_air(mixer); + scarlett2_notify_input_phantom(mixer); +} + +/* Notify on input select change */ +static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer) { + struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; - int err, count, i; + const struct scarlett2_device_info *info = private->info; + int i; - struct { - __le32 size; - __le32 count; - u8 unknown[8]; - } __packed flash_info; + if (!info->gain_input_count) + return; - struct { - __le32 size; - __le32 flags; - char name[16]; - } __packed segment_info; + private->input_select_updated = 1; - err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_FLASH, - NULL, 0, - &flash_info, sizeof(flash_info)); - if (err < 0) - return err; + snd_ctl_notify(card, + SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, + &private->input_select_ctl->id); - count = le32_to_cpu(flash_info.count); + for (i = 0; i < info->gain_input_count / 2; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->input_link_ctls[i]->id); +} - /* sanity check count */ - if (count < SCARLETT2_SEGMENT_NUM_MIN || - count > SCARLETT2_SEGMENT_NUM_MAX + 1) { - usb_audio_err(mixer->chip, - "invalid flash segment count: %d\n", count); - return -EINVAL; - } - - for (i = 0; i < count; i++) { - __le32 segment_num_req = cpu_to_le32(i); - int flash_segment_id; - - err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_SEGMENT, - &segment_num_req, sizeof(segment_num_req), - &segment_info, sizeof(segment_info)); - if (err < 0) { - usb_audio_err(mixer->chip, - "failed to get flash segment info %d: %d\n", - i, err); - return err; - } - - if (!strncmp(segment_info.name, - SCARLETT2_SEGMENT_SETTINGS_NAME, 16)) - flash_segment_id = SCARLETT2_SEGMENT_ID_SETTINGS; - else if (!strncmp(segment_info.name, - SCARLETT2_SEGMENT_FIRMWARE_NAME, 16)) - flash_segment_id = SCARLETT2_SEGMENT_ID_FIRMWARE; - else - continue; +/* Notify on input gain change */ +static void scarlett2_notify_input_gain(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; - private->flash_segment_nums[flash_segment_id] = i; - private->flash_segment_blocks[flash_segment_id] = - le32_to_cpu(segment_info.size) / - SCARLETT2_FLASH_BLOCK_SIZE; - } + if (!info->gain_input_count) + return; - /* segment 0 is App_Gold and we never want to touch that, so - * use 0 as the "not-found" value - */ - if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_SETTINGS]) { - usb_audio_err(mixer->chip, - "failed to find flash segment %s\n", - SCARLETT2_SEGMENT_SETTINGS_NAME); - return -EINVAL; - } - if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_FIRMWARE]) { - usb_audio_err(mixer->chip, - "failed to find flash segment %s\n", - SCARLETT2_SEGMENT_FIRMWARE_NAME); - return -EINVAL; - } + private->input_gain_updated = 1; - return 0; + for (i = 0; i < info->gain_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->input_gain_ctls[i]->id); } -/* Read configuration from the interface on start */ -static int scarlett2_read_configs(struct usb_mixer_interface *mixer) +/* Notify on autogain change */ +static void scarlett2_notify_autogain(struct usb_mixer_interface *mixer) { + struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - int err, i; - - if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) { - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_MSD_SWITCH, - 1, &private->msd_switch); - if (err < 0) - return err; - } - - if (private->firmware_version < info->min_firmware_version) { - usb_audio_err(mixer->chip, - "Focusrite %s firmware version %d is too old; " - "need %d", - private->series_name, - private->firmware_version, - info->min_firmware_version); - return 0; - } - - /* no other controls are created if MSD mode is on */ - if (private->msd_switch) - return 0; - - err = scarlett2_update_input_level(mixer); - if (err < 0) - return err; - - err = scarlett2_update_input_pad(mixer); - if (err < 0) - return err; + int i; - err = scarlett2_update_input_air(mixer); - if (err < 0) - return err; + if (!info->gain_input_count) + return; - err = scarlett2_update_input_phantom(mixer); - if (err < 0) - return err; + private->autogain_updated = 1; - err = scarlett2_update_direct_monitor(mixer); - if (err < 0) - return err; + for (i = 0; i < info->gain_input_count; i++) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->autogain_ctls[i]->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->autogain_status_ctls[i]->id); + } - /* the rest of the configuration is for devices with a mixer */ - if (!scarlett2_has_mixer(private)) - return 0; + scarlett2_autogain_notify_access(mixer); +} - err = scarlett2_update_monitor_mix(mixer); - if (err < 0) - return err; +/* Notify on input safe switch change */ +static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; - err = scarlett2_update_monitor_other(mixer); - if (err < 0) - return err; + if (!info->gain_input_count) + return; - if (scarlett2_has_config_item(private, - SCARLETT2_CONFIG_STANDALONE_SWITCH)) { - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_STANDALONE_SWITCH, - 1, &private->standalone_switch); - if (err < 0) - return err; - } + private->input_safe_updated = 1; - if (scarlett2_has_config_item(private, - SCARLETT2_CONFIG_POWER_EXT)) { - err = scarlett2_update_power_status(mixer); - if (err < 0) - return err; - } + for (i = 0; i < info->gain_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->safe_ctls[i]->id); +} - err = scarlett2_update_sync(mixer); - if (err < 0) - return err; +/* Notify on "monitor other" change (speaker switching, talkback) */ +static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; - if (scarlett2_has_config_item(private, - SCARLETT2_CONFIG_LINE_OUT_VOLUME)) { - s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; + private->monitor_other_updated = 1; - /* read SW line out volume */ - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, - private->num_line_out, &sw_vol); - if (err < 0) - return err; + if (info->has_speaker_switching) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->speaker_switching_ctl->id); - for (i = 0; i < private->num_line_out; i++) - private->vol[i] = clamp( - sw_vol[i] + SCARLETT2_VOLUME_BIAS, - 0, SCARLETT2_VOLUME_BIAS); + if (info->has_talkback) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->talkback_ctl->id); - /* read SW mute */ - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_MUTE_SWITCH, - private->num_line_out, &private->mute_switch); - if (err < 0) - return err; + /* if speaker switching was recently enabled or disabled, + * invalidate the dim/mute and mux enum controls + */ + if (private->speaker_switching_switched) { + int i; - for (i = 0; i < private->num_line_out; i++) - private->mute_switch[i] = - !!private->mute_switch[i]; + scarlett2_notify_dim_mute(mixer); - /* read SW/HW switches */ - if (scarlett2_has_config_item(private, - SCARLETT2_CONFIG_SW_HW_SWITCH)) { - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, - private->num_line_out, - &private->vol_sw_hw_switch); - if (err < 0) - return err; + private->speaker_switching_switched = 0; + private->mux_updated = 1; - for (i = 0; i < private->num_line_out; i++) - private->vol_sw_hw_switch[i] = - !!private->vol_sw_hw_switch[i]; - } + for (i = 0; i < private->num_mux_dsts; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->mux_ctls[i]->id); } +} - err = scarlett2_update_volumes(mixer); - if (err < 0) - return err; - - err = scarlett2_update_dim_mute(mixer); - if (err < 0) - return err; - - err = scarlett2_update_input_select(mixer); - if (err < 0) - return err; - - err = scarlett2_update_input_gain(mixer); - if (err < 0) - return err; +/* Notify on direct monitor switch change */ +static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + int count = private->num_mix_in * private->num_mix_out; + int i; - err = scarlett2_update_autogain(mixer); - if (err < 0) - return err; + private->direct_monitor_updated = 1; - err = scarlett2_update_input_safe(mixer); - if (err < 0) - return err; + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->direct_monitor_ctl->id); - if (scarlett2_has_config_item(private, - SCARLETT2_CONFIG_PCM_INPUT_SWITCH)) { - err = scarlett2_update_pcm_input_switch(mixer); - if (err < 0) - return err; - } + if (!scarlett2_has_mixer(private)) + return; - err = scarlett2_update_mix(mixer); - if (err < 0) - return err; + private->mix_updated = 1; - return scarlett2_usb_get_mux(mixer); + /* Notify of change to the mix controls */ + for (i = 0; i < count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->mix_ctls[i]->id); } -/* Notify on sync change */ -static void scarlett2_notify_sync(struct usb_mixer_interface *mixer) +/* Notify on power change */ +static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer) { + struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; - private->sync_updated = 1; + private->power_status_updated = 1; - snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->sync_ctl->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->power_status_ctl->id); } -/* Notify on monitor change (Gen 2/3) */ -static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) +/* Notify on mux change */ +static void scarlett2_notify_mux(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; int i; - if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH)) - return; - - private->vol_updated = 1; - - snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->master_vol_ctl->id); + private->mux_updated = 1; - for (i = 0; i < private->num_line_out; i++) - if (private->vol_sw_hw_switch[line_out_remap(private, i)]) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->vol_ctls[i]->id); + for (i = 0; i < private->num_mux_dsts; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->mux_ctls[i]->id); } -/* Notify on volume change (Gen 4) */ -static void scarlett2_notify_volume(struct usb_mixer_interface *mixer) +/* Notify on PCM input switch change */ +static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer) { + struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; - private->vol_updated = 1; + private->pcm_input_switch_updated = 1; - snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->master_vol_ctl->id); - snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->headphone_vol_ctl->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->pcm_input_switch_ctl->id); + + scarlett2_notify_mux(mixer); } -/* Notify on dim/mute change */ -static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) +/* Interrupt callback */ +static void scarlett2_notify(struct urb *urb) { - struct snd_card *card = mixer->chip->card; + struct usb_mixer_interface *mixer = urb->context; + int len = urb->actual_length; + int ustatus = urb->status; + u32 data; struct scarlett2_data *private = mixer->private_data; - int i; + const struct scarlett2_notification *notifications = + private->config_set->notifications; - if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH)) - return; + if (ustatus != 0 || len != 8) + goto requeue; - private->dim_mute_updated = 1; + data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); - for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->dim_mute_ctls[i]->id); + while (data && notifications->mask) { + if (data & notifications->mask) { + data &= ~notifications->mask; + if (notifications->func) + notifications->func(mixer); + } + notifications++; + } - for (i = 0; i < private->num_line_out; i++) - if (private->vol_sw_hw_switch[line_out_remap(private, i)]) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->mute_ctls[i]->id); + if (data) + usb_audio_warn(mixer->chip, + "%s: Unhandled notification: 0x%08x\n", + __func__, data); + +requeue: + if (ustatus != -ENOENT && + ustatus != -ECONNRESET && + ustatus != -ESHUTDOWN) { + urb->dev = mixer->chip->dev; + usb_submit_urb(urb, GFP_ATOMIC); + } } -/* Notify on input level switch change */ -static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer) +/*** Cleanup/Suspend Callbacks ***/ + +static void scarlett2_private_free(struct usb_mixer_interface *mixer) { - struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int i; - - private->input_level_updated = 1; - for (i = 0; i < info->level_input_count; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->level_ctls[i]->id); + cancel_delayed_work_sync(&private->work); + kfree(private); + mixer->private_data = NULL; } -/* Notify on input pad switch change */ -static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer) +static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) { - struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; + + if (cancel_delayed_work_sync(&private->work)) + scarlett2_config_save(private->mixer); +} + +/*** Initialisation ***/ + +static void scarlett2_count_io(struct scarlett2_data *private) +{ const struct scarlett2_device_info *info = private->info; - int i; + const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; + int port_type, srcs = 0, dsts = 0; - private->input_pad_updated = 1; + /* Count the number of mux sources and destinations */ + for (port_type = 0; + port_type < SCARLETT2_PORT_TYPE_COUNT; + port_type++) { + srcs += port_count[port_type][SCARLETT2_PORT_IN]; + dsts += port_count[port_type][SCARLETT2_PORT_OUT]; + } - for (i = 0; i < info->pad_input_count; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->pad_ctls[i]->id); + private->num_mux_srcs = srcs; + private->num_mux_dsts = dsts; + + /* Mixer inputs are mux outputs and vice versa. + * Scarlett Gen 4 DSP I/O uses SCARLETT2_PORT_TYPE_MIX but + * doesn't have mixer controls. + */ + private->num_mix_in = + port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT] - + info->dsp_count; + + private->num_mix_out = + port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN] - + info->dsp_count; + + /* Number of analogue line outputs */ + private->num_line_out = + port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; + + /* Number of monitor mix controls */ + private->num_monitor_mix_ctls = + info->direct_monitor * 2 * private->num_mix_in; } -/* Notify on input air switch change */ -static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer) +/* Look through the interface descriptors for the Focusrite Control + * interface (bInterfaceClass = 255 Vendor Specific Class) and set + * bInterfaceNumber, bEndpointAddress, wMaxPacketSize, and bInterval + * in private + */ +static int scarlett2_find_fc_interface(struct usb_device *dev, + struct scarlett2_data *private) { - struct snd_card *card = mixer->chip->card; - struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; + struct usb_host_config *config = dev->actconfig; int i; - private->input_air_updated = 1; + for (i = 0; i < config->desc.bNumInterfaces; i++) { + struct usb_interface *intf = config->interface[i]; + struct usb_interface_descriptor *desc = + &intf->altsetting[0].desc; + struct usb_endpoint_descriptor *epd; - for (i = 0; i < info->air_input_count; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->air_ctls[i]->id); + if (desc->bInterfaceClass != 255) + continue; + + epd = get_endpoint(intf->altsetting, 0); + private->bInterfaceNumber = desc->bInterfaceNumber; + private->bEndpointAddress = epd->bEndpointAddress & + USB_ENDPOINT_NUMBER_MASK; + private->wMaxPacketSize = le16_to_cpu(epd->wMaxPacketSize); + private->bInterval = epd->bInterval; + return 0; + } + + return -EINVAL; } -/* Notify on input phantom switch change */ -static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer) +/* Initialise private data */ +static int scarlett2_init_private(struct usb_mixer_interface *mixer, + const struct scarlett2_device_entry *entry) { - struct snd_card *card = mixer->chip->card; - struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int i; + struct scarlett2_data *private = + kzalloc(sizeof(struct scarlett2_data), GFP_KERNEL); - private->input_phantom_updated = 1; + if (!private) + return -ENOMEM; - for (i = 0; i < info->phantom_count; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->phantom_ctls[i]->id); + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + INIT_DELAYED_WORK(&private->work, scarlett2_config_save_work); - scarlett2_phantom_notify_access(mixer); -} + mixer->private_data = private; + mixer->private_free = scarlett2_private_free; + mixer->private_suspend = scarlett2_private_suspend; -/* Notify on "input other" change (level/pad/air/phantom) */ -static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer) -{ - scarlett2_notify_input_level(mixer); - scarlett2_notify_input_pad(mixer); - scarlett2_notify_input_air(mixer); - scarlett2_notify_input_phantom(mixer); + private->info = entry->info; + private->config_set = entry->info->config_set; + private->series_name = entry->series_name; + scarlett2_count_io(private); + private->scarlett2_seq = 0; + private->mixer = mixer; + + return scarlett2_find_fc_interface(mixer->chip->dev, private); } -/* Notify on input select change */ -static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer) +/* Submit a URB to receive notifications from the device */ +static int scarlett2_init_notify(struct usb_mixer_interface *mixer) { - struct snd_card *card = mixer->chip->card; + struct usb_device *dev = mixer->chip->dev; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int i; + unsigned int pipe = usb_rcvintpipe(dev, private->bEndpointAddress); + void *transfer_buffer; - if (!info->gain_input_count) - return; + if (mixer->urb) { + usb_audio_err(mixer->chip, + "%s: mixer urb already in use!\n", __func__); + return 0; + } - private->input_select_updated = 1; + if (usb_pipe_type_check(dev, pipe)) + return -EINVAL; - snd_ctl_notify(card, - SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, - &private->input_select_ctl->id); + mixer->urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->urb) + return -ENOMEM; - for (i = 0; i < info->gain_input_count / 2; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->input_link_ctls[i]->id); + transfer_buffer = kmalloc(private->wMaxPacketSize, GFP_KERNEL); + if (!transfer_buffer) + return -ENOMEM; + + usb_fill_int_urb(mixer->urb, dev, pipe, + transfer_buffer, private->wMaxPacketSize, + scarlett2_notify, mixer, private->bInterval); + + return usb_submit_urb(mixer->urb, GFP_KERNEL); } -/* Notify on input gain change */ -static void scarlett2_notify_input_gain(struct usb_mixer_interface *mixer) +/* Cargo cult proprietary initialisation sequence */ +static int scarlett2_usb_init(struct usb_mixer_interface *mixer) { - struct snd_card *card = mixer->chip->card; + struct usb_device *dev = mixer->chip->dev; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int i; + u8 step0_buf[24]; + u8 step2_buf[84]; + int err; - if (!info->gain_input_count) - return; + if (usb_pipe_type_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; - private->input_gain_updated = 1; + /* step 0 */ + err = scarlett2_usb_rx(dev, private->bInterfaceNumber, + SCARLETT2_USB_CMD_INIT, + step0_buf, sizeof(step0_buf)); + if (err < 0) + return err; - for (i = 0; i < info->gain_input_count; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->input_gain_ctls[i]->id); + /* step 1 */ + private->scarlett2_seq = 1; + err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_1, NULL, 0, NULL, 0); + if (err < 0) + return err; + + /* step 2 */ + private->scarlett2_seq = 1; + err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_2, + NULL, 0, + step2_buf, sizeof(step2_buf)); + if (err < 0) + return err; + + /* extract 4-byte firmware version from step2_buf[8] */ + private->firmware_version = le32_to_cpu(*(__le32 *)(step2_buf + 8)); + usb_audio_info(mixer->chip, + "Firmware version %d\n", + private->firmware_version); + + return 0; } -/* Notify on autogain change */ -static void scarlett2_notify_autogain(struct usb_mixer_interface *mixer) +/* Get the flash segment numbers for the App_Settings and App_Upgrade + * segments and put them in the private data + */ +static int scarlett2_get_flash_segment_nums(struct usb_mixer_interface *mixer) { - struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int i; + int err, count, i; - if (!info->gain_input_count) - return; + struct { + __le32 size; + __le32 count; + u8 unknown[8]; + } __packed flash_info; - private->autogain_updated = 1; + struct { + __le32 size; + __le32 flags; + char name[16]; + } __packed segment_info; - for (i = 0; i < info->gain_input_count; i++) { - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->autogain_ctls[i]->id); - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->autogain_status_ctls[i]->id); + err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_FLASH, + NULL, 0, + &flash_info, sizeof(flash_info)); + if (err < 0) + return err; + + count = le32_to_cpu(flash_info.count); + + /* sanity check count */ + if (count < SCARLETT2_SEGMENT_NUM_MIN || + count > SCARLETT2_SEGMENT_NUM_MAX + 1) { + usb_audio_err(mixer->chip, + "invalid flash segment count: %d\n", count); + return -EINVAL; } - scarlett2_autogain_notify_access(mixer); -} + for (i = 0; i < count; i++) { + __le32 segment_num_req = cpu_to_le32(i); + int flash_segment_id; -/* Notify on input safe switch change */ -static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer) -{ - struct snd_card *card = mixer->chip->card; - struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int i; + err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_SEGMENT, + &segment_num_req, sizeof(segment_num_req), + &segment_info, sizeof(segment_info)); + if (err < 0) { + usb_audio_err(mixer->chip, + "failed to get flash segment info %d: %d\n", + i, err); + return err; + } - if (!info->gain_input_count) - return; + if (!strncmp(segment_info.name, + SCARLETT2_SEGMENT_SETTINGS_NAME, 16)) + flash_segment_id = SCARLETT2_SEGMENT_ID_SETTINGS; + else if (!strncmp(segment_info.name, + SCARLETT2_SEGMENT_FIRMWARE_NAME, 16)) + flash_segment_id = SCARLETT2_SEGMENT_ID_FIRMWARE; + else + continue; - private->input_safe_updated = 1; + private->flash_segment_nums[flash_segment_id] = i; + private->flash_segment_blocks[flash_segment_id] = + le32_to_cpu(segment_info.size) / + SCARLETT2_FLASH_BLOCK_SIZE; + } - for (i = 0; i < info->gain_input_count; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->safe_ctls[i]->id); + /* segment 0 is App_Gold and we never want to touch that, so + * use 0 as the "not-found" value + */ + if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_SETTINGS]) { + usb_audio_err(mixer->chip, + "failed to find flash segment %s\n", + SCARLETT2_SEGMENT_SETTINGS_NAME); + return -EINVAL; + } + if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_FIRMWARE]) { + usb_audio_err(mixer->chip, + "failed to find flash segment %s\n", + SCARLETT2_SEGMENT_FIRMWARE_NAME); + return -EINVAL; + } + + return 0; } -/* Notify on "monitor other" change (speaker switching, talkback) */ -static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer) +/* Read configuration from the interface on start */ +static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { - struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; + int err, i; - private->monitor_other_updated = 1; - - if (info->has_speaker_switching) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->speaker_switching_ctl->id); - - if (info->has_talkback) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->talkback_ctl->id); - - /* if speaker switching was recently enabled or disabled, - * invalidate the dim/mute and mux enum controls - */ - if (private->speaker_switching_switched) { - int i; - - scarlett2_notify_dim_mute(mixer); - - private->speaker_switching_switched = 0; - private->mux_updated = 1; - - for (i = 0; i < private->num_mux_dsts; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->mux_ctls[i]->id); + if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_MSD_SWITCH, + 1, &private->msd_switch); + if (err < 0) + return err; } -} -/* Notify on direct monitor switch change */ -static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer) -{ - struct snd_card *card = mixer->chip->card; - struct scarlett2_data *private = mixer->private_data; - int count = private->num_mix_in * private->num_mix_out; - int i; + if (private->firmware_version < info->min_firmware_version) { + usb_audio_err(mixer->chip, + "Focusrite %s firmware version %d is too old; " + "need %d", + private->series_name, + private->firmware_version, + info->min_firmware_version); + return 0; + } - private->direct_monitor_updated = 1; + /* no other controls are created if MSD mode is on */ + if (private->msd_switch) + return 0; - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->direct_monitor_ctl->id); + err = scarlett2_update_input_level(mixer); + if (err < 0) + return err; - if (!scarlett2_has_mixer(private)) - return; + err = scarlett2_update_input_pad(mixer); + if (err < 0) + return err; - private->mix_updated = 1; + err = scarlett2_update_input_air(mixer); + if (err < 0) + return err; - /* Notify of change to the mix controls */ - for (i = 0; i < count; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->mix_ctls[i]->id); -} + err = scarlett2_update_input_phantom(mixer); + if (err < 0) + return err; -/* Notify on power change */ -static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer) -{ - struct snd_card *card = mixer->chip->card; - struct scarlett2_data *private = mixer->private_data; + err = scarlett2_update_direct_monitor(mixer); + if (err < 0) + return err; - private->power_status_updated = 1; + /* the rest of the configuration is for devices with a mixer */ + if (!scarlett2_has_mixer(private)) + return 0; - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->power_status_ctl->id); -} + err = scarlett2_update_monitor_mix(mixer); + if (err < 0) + return err; -/* Notify on mux change */ -static void scarlett2_notify_mux(struct usb_mixer_interface *mixer) -{ - struct snd_card *card = mixer->chip->card; - struct scarlett2_data *private = mixer->private_data; - int i; + err = scarlett2_update_monitor_other(mixer); + if (err < 0) + return err; - private->mux_updated = 1; + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_STANDALONE_SWITCH)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_STANDALONE_SWITCH, + 1, &private->standalone_switch); + if (err < 0) + return err; + } - for (i = 0; i < private->num_mux_dsts; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->mux_ctls[i]->id); -} + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_POWER_EXT)) { + err = scarlett2_update_power_status(mixer); + if (err < 0) + return err; + } -/* Notify on PCM input switch change */ -static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer) -{ - struct snd_card *card = mixer->chip->card; - struct scarlett2_data *private = mixer->private_data; + err = scarlett2_update_sync(mixer); + if (err < 0) + return err; - private->pcm_input_switch_updated = 1; + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_LINE_OUT_VOLUME)) { + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->pcm_input_switch_ctl->id); + /* read SW line out volume */ + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + private->num_line_out, &sw_vol); + if (err < 0) + return err; - scarlett2_notify_mux(mixer); -} + for (i = 0; i < private->num_line_out; i++) + private->vol[i] = clamp( + sw_vol[i] + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); -/* Interrupt callback */ -static void scarlett2_notify(struct urb *urb) -{ - struct usb_mixer_interface *mixer = urb->context; - int len = urb->actual_length; - int ustatus = urb->status; - u32 data; - struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_notification *notifications = - private->config_set->notifications; + /* read SW mute */ + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_MUTE_SWITCH, + private->num_line_out, &private->mute_switch); + if (err < 0) + return err; - if (ustatus != 0 || len != 8) - goto requeue; + for (i = 0; i < private->num_line_out; i++) + private->mute_switch[i] = + !!private->mute_switch[i]; - data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); + /* read SW/HW switches */ + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_SW_HW_SWITCH)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + private->num_line_out, + &private->vol_sw_hw_switch); + if (err < 0) + return err; - while (data && notifications->mask) { - if (data & notifications->mask) { - data &= ~notifications->mask; - if (notifications->func) - notifications->func(mixer); + for (i = 0; i < private->num_line_out; i++) + private->vol_sw_hw_switch[i] = + !!private->vol_sw_hw_switch[i]; } - notifications++; } - if (data) - usb_audio_warn(mixer->chip, - "%s: Unhandled notification: 0x%08x\n", - __func__, data); + err = scarlett2_update_volumes(mixer); + if (err < 0) + return err; -requeue: - if (ustatus != -ENOENT && - ustatus != -ECONNRESET && - ustatus != -ESHUTDOWN) { - urb->dev = mixer->chip->dev; - usb_submit_urb(urb, GFP_ATOMIC); - } -} + err = scarlett2_update_dim_mute(mixer); + if (err < 0) + return err; -static int scarlett2_init_notify(struct usb_mixer_interface *mixer) -{ - struct usb_device *dev = mixer->chip->dev; - struct scarlett2_data *private = mixer->private_data; - unsigned int pipe = usb_rcvintpipe(dev, private->bEndpointAddress); - void *transfer_buffer; + err = scarlett2_update_input_select(mixer); + if (err < 0) + return err; - if (mixer->urb) { - usb_audio_err(mixer->chip, - "%s: mixer urb already in use!\n", __func__); - return 0; - } + err = scarlett2_update_input_gain(mixer); + if (err < 0) + return err; - if (usb_pipe_type_check(dev, pipe)) - return -EINVAL; + err = scarlett2_update_autogain(mixer); + if (err < 0) + return err; - mixer->urb = usb_alloc_urb(0, GFP_KERNEL); - if (!mixer->urb) - return -ENOMEM; + err = scarlett2_update_input_safe(mixer); + if (err < 0) + return err; - transfer_buffer = kmalloc(private->wMaxPacketSize, GFP_KERNEL); - if (!transfer_buffer) - return -ENOMEM; + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_PCM_INPUT_SWITCH)) { + err = scarlett2_update_pcm_input_switch(mixer); + if (err < 0) + return err; + } - usb_fill_int_urb(mixer->urb, dev, pipe, - transfer_buffer, private->wMaxPacketSize, - scarlett2_notify, mixer, private->bInterval); + err = scarlett2_update_mix(mixer); + if (err < 0) + return err; - return usb_submit_urb(mixer->urb, GFP_KERNEL); + return scarlett2_usb_get_mux(mixer); } static const struct scarlett2_device_entry *get_scarlett2_device_entry( From 1b65088958cadab04d5d34a8615e2466b1b48ecb Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:04:14 +1030 Subject: [PATCH 309/603] ALSA: scarlett2: Implement handling of the ACK notification After scarlett2_usb() sends a command, it seems that we should wait for an ACK before attempting to read the response. Not doing that didn't seem necessary previously but seems to be causing occasional issues with 4th Gen devices. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: <452d1263c40fa8eba1cfb24e2055e40a84cbc437.1710264833.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 70 ++++++++++++++++++++++++++++++++----- 1 file changed, 61 insertions(+), 9 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 8390b646c0aef..02c488c80b7ec 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -321,6 +321,7 @@ struct scarlett2_notification { void (*func)(struct usb_mixer_interface *mixer); }; +static void scarlett2_notify_ack(struct usb_mixer_interface *mixer); static void scarlett2_notify_sync(struct usb_mixer_interface *mixer); static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer); static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer); @@ -343,7 +344,7 @@ static void scarlett2_notify_pcm_input_switch( /* Arrays of notification callback functions */ static const struct scarlett2_notification scarlett2_notifications[] = { - { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000001, scarlett2_notify_ack }, { 0x00000008, scarlett2_notify_sync }, { 0x00200000, scarlett2_notify_dim_mute }, { 0x00400000, scarlett2_notify_monitor }, @@ -353,14 +354,14 @@ static const struct scarlett2_notification scarlett2_notifications[] = { }; static const struct scarlett2_notification scarlett3a_notifications[] = { - { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000001, scarlett2_notify_ack }, { 0x00800000, scarlett2_notify_input_other }, { 0x01000000, scarlett2_notify_direct_monitor }, { 0, NULL } }; static const struct scarlett2_notification scarlett4_solo_notifications[] = { - { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000001, scarlett2_notify_ack }, { 0x00000008, scarlett2_notify_sync }, { 0x00400000, scarlett2_notify_input_air }, { 0x00800000, scarlett2_notify_direct_monitor }, @@ -371,7 +372,7 @@ static const struct scarlett2_notification scarlett4_solo_notifications[] = { }; static const struct scarlett2_notification scarlett4_2i2_notifications[] = { - { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000001, scarlett2_notify_ack }, { 0x00000008, scarlett2_notify_sync }, { 0x00200000, scarlett2_notify_input_safe }, { 0x00400000, scarlett2_notify_autogain }, @@ -387,7 +388,7 @@ static const struct scarlett2_notification scarlett4_2i2_notifications[] = { }; static const struct scarlett2_notification scarlett4_4i4_notifications[] = { - { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000001, scarlett2_notify_ack }, { 0x00000008, scarlett2_notify_sync }, { 0x00200000, scarlett2_notify_input_safe }, { 0x00400000, scarlett2_notify_autogain }, @@ -942,7 +943,9 @@ struct scarlett2_device_info { struct scarlett2_data { struct usb_mixer_interface *mixer; struct mutex usb_mutex; /* prevent sending concurrent USB requests */ + struct completion cmd_done; struct mutex data_mutex; /* lock access to this data */ + u8 running; u8 hwdep_in_use; u8 selected_flash_segment_id; u8 flash_write_state; @@ -1960,6 +1963,17 @@ static int scarlett2_usb( goto unlock; } + if (!wait_for_completion_timeout(&private->cmd_done, + msecs_to_jiffies(1000))) { + usb_audio_err( + mixer->chip, + "%s USB request timed out, cmd %x\n", + private->series_name, cmd); + + err = -ETIMEDOUT; + goto unlock; + } + /* send a second message to get the response */ err = scarlett2_usb_rx(dev, private->bInterfaceNumber, @@ -6702,6 +6716,18 @@ static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer) scarlett2_notify_mux(mixer); } +/* Handle acknowledgement that a command was received; let + * scarlett2_usb() know that it can proceed + */ +static void scarlett2_notify_ack(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + + /* if running == 0, ignore ACKs */ + if (private->running) + complete(&private->cmd_done); +} + /* Interrupt callback */ static void scarlett2_notify(struct urb *urb) { @@ -6718,6 +6744,12 @@ static void scarlett2_notify(struct urb *urb) data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); + /* Ignore notifications except ACK during initialisation. + * ACK is 0x00000001 on every device. + */ + if (private->running < 2) + data &= 1; + while (data && notifications->mask) { if (data & notifications->mask) { data &= ~notifications->mask; @@ -6738,6 +6770,8 @@ static void scarlett2_notify(struct urb *urb) ustatus != -ESHUTDOWN) { urb->dev = mixer->chip->dev; usb_submit_urb(urb, GFP_ATOMIC); + } else { + complete(&private->cmd_done); } } @@ -6889,6 +6923,8 @@ static int scarlett2_init_notify(struct usb_mixer_interface *mixer) transfer_buffer, private->wMaxPacketSize, scarlett2_notify, mixer, private->bInterval); + init_completion(&private->cmd_done); + return usb_submit_urb(mixer->urb, GFP_KERNEL); } @@ -6911,6 +6947,24 @@ static int scarlett2_usb_init(struct usb_mixer_interface *mixer) if (err < 0) return err; + /* Set up the interrupt polling for notifications. + * When running is: + * 0: all notifications are ignored + * 1: only ACKs are handled + * 2: all notifications are handled + */ + err = scarlett2_init_notify(mixer); + if (err < 0) + return err; + + /* sleep for a moment in case of an outstanding ACK */ + msleep(20); + + /* start handling ACKs, but no other notifications until the + * ALSA controls have been created + */ + private->running = 1; + /* step 1 */ private->scarlett2_seq = 1; err = scarlett2_usb(mixer, SCARLETT2_USB_INIT_1, NULL, 0, NULL, 0); @@ -7308,10 +7362,8 @@ static int snd_scarlett2_controls_create( scarlett2_phantom_update_access(mixer); } - /* Set up the interrupt polling */ - err = scarlett2_init_notify(mixer); - if (err < 0) - return err; + /* Start handling all notifications */ + private->running = 2; return 0; } From 5bfb7c2ae4275be338d2a1a85904b97835a13ec5 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:04:42 +1030 Subject: [PATCH 310/603] ALSA: scarlett2: Add support for reading from flash Add hwdep read op so flash segments can be read. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: <800d20a801e8c59c2905c82ecae5676cd4f31429.1710264833.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 88 +++++++++++++++++++++++++++++++++++-- 1 file changed, 84 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 02c488c80b7ec..981ec48a811a8 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1859,6 +1859,7 @@ static int scarlett2_get_port_start_num( #define SCARLETT2_USB_ERASE_SEGMENT 0x00004002 #define SCARLETT2_USB_GET_ERASE 0x00004003 #define SCARLETT2_USB_WRITE_SEGMENT 0x00004004 +#define SCARLETT2_USB_READ_SEGMENT 0x00004005 #define SCARLETT2_USB_GET_SYNC 0x00006004 #define SCARLETT2_USB_GET_DATA 0x00800000 #define SCARLETT2_USB_SET_DATA 0x00800001 @@ -1869,7 +1870,7 @@ static int scarlett2_get_port_start_num( #define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 #define SCARLETT2_FLASH_BLOCK_SIZE 4096 -#define SCARLETT2_FLASH_WRITE_MAX 1024 +#define SCARLETT2_FLASH_RW_MAX 1024 #define SCARLETT2_SEGMENT_NUM_MIN 1 #define SCARLETT2_SEGMENT_NUM_MAX 4 @@ -7452,7 +7453,7 @@ static int scarlett2_reboot(struct usb_mixer_interface *mixer) return scarlett2_usb(mixer, SCARLETT2_USB_REBOOT, NULL, 0, NULL, 0); } -/* Select a flash segment for erasing (and possibly writing to) */ +/* Select a flash segment for reading/erasing/writing */ static int scarlett2_ioctl_select_flash_segment( struct usb_mixer_interface *mixer, unsigned long arg) @@ -7633,6 +7634,84 @@ static int scarlett2_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, } } +static long scarlett2_hwdep_read(struct snd_hwdep *hw, + char __user *buf, + long count, loff_t *offset) +{ + struct usb_mixer_interface *mixer = hw->private_data; + struct scarlett2_data *private = mixer->private_data; + int segment_id, segment_num, err; + int flash_size; + + /* SCARLETT2_USB_READ_SEGMENT request data */ + struct { + __le32 segment_num; + __le32 offset; + __le32 len; + } __packed req; + + u8 *resp; + + /* Flash segment must first be selected */ + if (private->flash_write_state != SCARLETT2_FLASH_WRITE_STATE_SELECTED) + return -EINVAL; + + /* Get the selected flash segment number */ + segment_id = private->selected_flash_segment_id; + if (segment_id < 0 || segment_id >= SCARLETT2_SEGMENT_ID_COUNT) + return -EINVAL; + + segment_num = private->flash_segment_nums[segment_id]; + if (segment_num < 0 || + segment_num > SCARLETT2_SEGMENT_NUM_MAX) + return -EFAULT; + + /* Validate the offset and count */ + if (count < 0 || *offset < 0) + return -EINVAL; + + /* Reached EOF? */ + flash_size = private->flash_segment_blocks[segment_id] * + SCARLETT2_FLASH_BLOCK_SIZE; + if (!count || *offset >= flash_size) + return 0; + + /* Limit the numbers of bytes read to SCARLETT2_FLASH_RW_MAX */ + if (count > SCARLETT2_FLASH_RW_MAX) + count = SCARLETT2_FLASH_RW_MAX; + + /* Limit read to EOF */ + if (*offset + count >= flash_size) + count = flash_size - *offset; + + /* Create and send the request */ + req.segment_num = cpu_to_le32(segment_num); + req.offset = cpu_to_le32(*offset); + req.len = cpu_to_le32(count); + + resp = kzalloc(count, GFP_KERNEL); + if (!resp) + return -ENOMEM; + + err = scarlett2_usb(mixer, SCARLETT2_USB_READ_SEGMENT, + &req, sizeof(req), resp, count); + if (err < 0) + goto error; + + /* Copy the response to userspace */ + if (copy_to_user(buf, resp, count)) { + err = -EFAULT; + goto error; + } + + *offset += count; + err = count; + +error: + kfree(resp); + return err; +} + static long scarlett2_hwdep_write(struct snd_hwdep *hw, const char __user *buf, long count, loff_t *offset) @@ -7651,7 +7730,7 @@ static long scarlett2_hwdep_write(struct snd_hwdep *hw, } __packed *req; /* Calculate the maximum permitted in data[] */ - const size_t max_data_size = SCARLETT2_FLASH_WRITE_MAX - + const size_t max_data_size = SCARLETT2_FLASH_RW_MAX - offsetof(typeof(*req), data); /* If erasing, wait for it to complete */ @@ -7688,7 +7767,7 @@ static long scarlett2_hwdep_write(struct snd_hwdep *hw, if (!count) return 0; - /* Limit the *req size to SCARLETT2_FLASH_WRITE_MAX */ + /* Limit the *req size to SCARLETT2_FLASH_RW_MAX */ if (count > max_data_size) count = max_data_size; @@ -7749,6 +7828,7 @@ static int scarlett2_hwdep_init(struct usb_mixer_interface *mixer) hw->exclusive = 1; hw->ops.open = scarlett2_hwdep_open; hw->ops.ioctl = scarlett2_hwdep_ioctl; + hw->ops.read = scarlett2_hwdep_read; hw->ops.write = scarlett2_hwdep_write; hw->ops.release = scarlett2_hwdep_release; From 4390095126eecb83fd02877aec651a24fec99a43 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:04:59 +1030 Subject: [PATCH 311/603] ALSA: scarlett2: Rename gen4_write_addr to param_buf_addr The location pointed to by gen4_write_addr and gen4_write_addr + 1 is officially known as the parameter buffer. Update the code to match. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: --- sound/usb/mixer_scarlett2.c | 37 ++++++++++++++++++------------------- 1 file changed, 18 insertions(+), 19 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 981ec48a811a8..6031bfd98137a 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -438,9 +438,9 @@ enum { /* Location, size, and activation command number for the configuration * parameters. Size is in bits and may be 0, 1, 8, or 16. * - * A size of 0 indicates that the parameter is a byte-sized Scarlett - * Gen 4 configuration which is written through the gen4_write_addr - * location (but still read through the given offset location). + * A size of 0 indicates that the parameter is a byte-sized + * configuration which is set through the parameter buffer (but still + * read through the given offset location). * * Some Gen 4 configuration parameters are written with 0x02 for a * desired value of 0x01, and 0x03 for 0x00. These are indicated with @@ -457,7 +457,7 @@ struct scarlett2_config { struct scarlett2_config_set { const struct scarlett2_notification *notifications; - u16 gen4_write_addr; + u16 param_buf_addr; const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; }; @@ -625,7 +625,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3c = { /* Solo Gen 4 */ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = { .notifications = scarlett4_solo_notifications, - .gen4_write_addr = 0xd8, + .param_buf_addr = 0xd8, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x47, .size = 8, .activate = 4 }, @@ -653,7 +653,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = { /* 2i2 Gen 4 */ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = { .notifications = scarlett4_2i2_notifications, - .gen4_write_addr = 0xfc, + .param_buf_addr = 0xfc, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x49, .size = 8, .activate = 4 }, // 0x41 ?? @@ -696,7 +696,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = { /* 4i4 Gen 4 */ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = { .notifications = scarlett4_4i4_notifications, - .gen4_write_addr = 0x130, + .param_buf_addr = 0x130, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x5c, .size = 8, .activate = 4 }, @@ -2080,7 +2080,7 @@ static int scarlett2_usb_get_config( if (!config_item->offset) return -EFAULT; - /* Gen 4 style parameters are always 1 byte */ + /* Writes to the parameter buffer are always 1 byte */ size = config_item->size ? config_item->size : 8; /* For byte-sized parameters, retrieve directly into buf */ @@ -2167,23 +2167,23 @@ static int scarlett2_usb_set_config( if (!config_item->offset) return -EFAULT; - /* Gen 4 style writes are selected with size = 0; + /* Writes via the parameter buffer are selected with size = 0; * these are only byte-sized values written through a shared * location, different to the read address */ if (!config_item->size) { - if (!config_set->gen4_write_addr) + if (!config_set->param_buf_addr) return -EFAULT; - /* Place index in gen4_write_addr + 1 */ + /* Place index in param_buf_addr + 1 */ err = scarlett2_usb_set_data( - mixer, config_set->gen4_write_addr + 1, 1, index); + mixer, config_set->param_buf_addr + 1, 1, index); if (err < 0) return err; - /* Place value in gen4_write_addr */ + /* Place value in param_buf_addr */ err = scarlett2_usb_set_data( - mixer, config_set->gen4_write_addr, 1, value); + mixer, config_set->param_buf_addr, 1, value); if (err < 0) return err; @@ -2192,9 +2192,8 @@ static int scarlett2_usb_set_config( mixer, config_item->activate); } - /* Not-Gen 4 style needs NVRAM save, supports - * bit-modification, and writing is done to the same place - * that the value can be read from + /* Direct writes (not via the parameter buffer) need NVRAM + * save and support bit-modification */ /* Cancel any pending NVRAM save */ @@ -2238,8 +2237,8 @@ static int scarlett2_usb_set_config( if (err < 0) return err; - /* Gen 2 style writes to Gen 4 devices don't need saving */ - if (config_set->gen4_write_addr) + /* Writes via the parameter buffer don't need a separate save step */ + if (config_set->param_buf_addr) return 0; /* Schedule the change to be written to NVRAM */ From 7d20f7b4f337ac075cf79f5047e68ea46189743f Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:05:15 +1030 Subject: [PATCH 312/603] ALSA: scarlett2: Add pbuf field to struct scarlett2_config scarlett2_usb_set_config() was using size = 0 as a signal to use the parameter buffer. Replace that with an explicit indication (pbuf = 1), as the upcoming Vocaster support has a config item written via the parameter buffer with size = 1 rather than the implicit size of 8. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: <50a7d85bb04f9a7f13f667c70a706826c8d3ef93.1710264833.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 85 ++++++++++++++++++++----------------- 1 file changed, 46 insertions(+), 39 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 6031bfd98137a..8d08f71bd4d6c 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -436,11 +436,12 @@ enum { }; /* Location, size, and activation command number for the configuration - * parameters. Size is in bits and may be 0, 1, 8, or 16. + * parameters. Size is in bits and may be 1, 8, or 16. * - * A size of 0 indicates that the parameter is a byte-sized - * configuration which is set through the parameter buffer (but still - * read through the given offset location). + * Vocaster and 4th Gen devices have a parameter buffer to set certain + * configuration parameters. When pbuf is set, rather than writing to + * the given offset, the channel and value are written to the + * parameter buffer and the activate command is sent to the device. * * Some Gen 4 configuration parameters are written with 0x02 for a * desired value of 0x01, and 0x03 for 0x00. These are indicated with @@ -452,6 +453,7 @@ struct scarlett2_config { u16 offset; u8 size; u8 activate; + u8 pbuf; u8 mute; }; @@ -631,19 +633,21 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = { .offset = 0x47, .size = 8, .activate = 4 }, [SCARLETT2_CONFIG_DIRECT_MONITOR] = { - .offset = 0x108, .activate = 12 }, + .offset = 0x108, .size = 8, .activate = 12, .pbuf = 1 }, [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { - .offset = 0x46, .activate = 9, .mute = 1 }, + .offset = 0x46, .size = 8, .activate = 9, .pbuf = 1, + .mute = 1 }, [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x3d, .activate = 10, .mute = 1 }, + .offset = 0x3d, .size = 8, .activate = 10, .pbuf = 1, + .mute = 1 }, [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x3e, .activate = 11 }, + .offset = 0x3e, .size = 8, .activate = 11, .pbuf = 1 }, [SCARLETT2_CONFIG_PCM_INPUT_SWITCH] = { - .offset = 0x206, .activate = 25 }, + .offset = 0x206, .size = 8, .activate = 25, .pbuf = 1 }, [SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN] = { .offset = 0x232, .size = 16, .activate = 26 } @@ -656,37 +660,39 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = { .param_buf_addr = 0xfc, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { - .offset = 0x49, .size = 8, .activate = 4 }, // 0x41 ?? + .offset = 0x49, .size = 8, .activate = 4 }, [SCARLETT2_CONFIG_DIRECT_MONITOR] = { - .offset = 0x14a, .activate = 16 }, + .offset = 0x14a, .size = 8, .activate = 16, .pbuf = 1 }, [SCARLETT2_CONFIG_AUTOGAIN_SWITCH] = { - .offset = 0x135, .activate = 10 }, + .offset = 0x135, .size = 8, .activate = 10, .pbuf = 1 }, [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = { - .offset = 0x137 }, + .offset = 0x137, .size = 8 }, [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { - .offset = 0x48, .activate = 11, .mute = 1 }, + .offset = 0x48, .size = 8, .activate = 11, .pbuf = 1, + .mute = 1 }, [SCARLETT2_CONFIG_INPUT_GAIN] = { - .offset = 0x4b, .activate = 12 }, + .offset = 0x4b, .size = 8, .activate = 12, .pbuf = 1 }, [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x3c, .activate = 13, .mute = 1 }, + .offset = 0x3c, .size = 8, .activate = 13, .pbuf = 1, + .mute = 1 }, [SCARLETT2_CONFIG_SAFE_SWITCH] = { - .offset = 0x147, .activate = 14 }, + .offset = 0x147, .size = 8, .activate = 14, .pbuf = 1 }, [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x3e, .activate = 15 }, + .offset = 0x3e, .size = 8, .activate = 15, .pbuf = 1 }, [SCARLETT2_CONFIG_INPUT_SELECT_SWITCH] = { - .offset = 0x14b, .activate = 17 }, + .offset = 0x14b, .size = 8, .activate = 17, .pbuf = 1 }, [SCARLETT2_CONFIG_INPUT_LINK_SWITCH] = { - .offset = 0x14e, .activate = 18 }, + .offset = 0x14e, .size = 8, .activate = 18, .pbuf = 1 }, [SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN] = { .offset = 0x2a0, .size = 16, .activate = 36 } @@ -702,31 +708,33 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = { .offset = 0x5c, .size = 8, .activate = 4 }, [SCARLETT2_CONFIG_AUTOGAIN_SWITCH] = { - .offset = 0x13e, .activate = 10 }, + .offset = 0x13e, .size = 8, .activate = 10, .pbuf = 1 }, [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = { - .offset = 0x140 }, + .offset = 0x140, .size = 8 }, [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { - .offset = 0x5a, .activate = 11, .mute = 1 }, + .offset = 0x5a, .size = 8, .activate = 11, .pbuf = 1, + .mute = 1 }, [SCARLETT2_CONFIG_INPUT_GAIN] = { - .offset = 0x5e, .activate = 12 }, + .offset = 0x5e, .size = 8, .activate = 12, .pbuf = 1 }, [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x4e, .activate = 13, .mute = 1 }, + .offset = 0x4e, .size = 8, .activate = 13, .pbuf = 1, + .mute = 1 }, [SCARLETT2_CONFIG_SAFE_SWITCH] = { - .offset = 0x150, .activate = 14 }, + .offset = 0x150, .size = 8, .activate = 14, .pbuf = 1 }, [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x50, .activate = 15 }, + .offset = 0x50, .size = 8, .activate = 15, .pbuf = 1 }, [SCARLETT2_CONFIG_INPUT_SELECT_SWITCH] = { - .offset = 0x153, .activate = 16 }, + .offset = 0x153, .size = 8, .activate = 16, .pbuf = 1 }, [SCARLETT2_CONFIG_INPUT_LINK_SWITCH] = { - .offset = 0x156, .activate = 17 }, + .offset = 0x156, .size = 8, .activate = 17, .pbuf = 1 }, [SCARLETT2_CONFIG_MASTER_VOLUME] = { .offset = 0x32, .size = 16 }, @@ -735,10 +743,10 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = { .offset = 0x3a, .size = 16 }, [SCARLETT2_CONFIG_POWER_EXT] = { - .offset = 0x168 }, + .offset = 0x168, .size = 8 }, [SCARLETT2_CONFIG_POWER_LOW] = { - .offset = 0x16d } + .offset = 0x16d, .size = 8 } } }; @@ -2167,11 +2175,8 @@ static int scarlett2_usb_set_config( if (!config_item->offset) return -EFAULT; - /* Writes via the parameter buffer are selected with size = 0; - * these are only byte-sized values written through a shared - * location, different to the read address - */ - if (!config_item->size) { + /* Write via the parameter buffer? */ + if (config_item->pbuf) { if (!config_set->param_buf_addr) return -EFAULT; @@ -2187,7 +2192,7 @@ static int scarlett2_usb_set_config( if (err < 0) return err; - /* Request the interface do the write */ + /* Activate the write through the parameter buffer */ return scarlett2_usb_activate_config( mixer, config_item->activate); } @@ -2227,7 +2232,7 @@ static int scarlett2_usb_set_config( value = tmp; } - /* Send the configuration parameter data */ + /* Write the new value */ err = scarlett2_usb_set_data(mixer, offset, size, value); if (err < 0) return err; @@ -2237,7 +2242,9 @@ static int scarlett2_usb_set_config( if (err < 0) return err; - /* Writes via the parameter buffer don't need a separate save step */ + /* Interfaces with parameter buffer writes don't need a + * separate save step + */ if (config_set->param_buf_addr) return 0; From b1b3b258242c4e7c70b8da09904a8c5e2aadaa3a Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:05:30 +1030 Subject: [PATCH 313/603] ALSA: scarlett2: Add support for config items with size = 32 Update scarlett2_usb_get_config() to support 32-bit values which are needed by the upcoming Vocaster support. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: --- sound/usb/mixer_scarlett2.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 8d08f71bd4d6c..f2bbf90680f8b 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -436,7 +436,7 @@ enum { }; /* Location, size, and activation command number for the configuration - * parameters. Size is in bits and may be 1, 8, or 16. + * parameters. Size is in bits and may be 1, 8, 16, or 32. * * Vocaster and 4th Gen devices have a parameter buffer to set certain * configuration parameters. When pbuf is set, rather than writing to @@ -2102,6 +2102,11 @@ static int scarlett2_usb_get_config( for (i = 0; i < count; i++, buf_16++) *buf_16 = le16_to_cpu(*(__le16 *)buf_16); + } else if (size == 4) { + u32 *buf_32 = buf; + + for (i = 0; i < count; i++, buf_32++) + *buf_32 = le32_to_cpu(*(__le32 *)buf_32); } return 0; } From 1e48ddb7d71f6a48f5d871864046e33fd94deb99 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:05:47 +1030 Subject: [PATCH 314/603] ALSA: scarlett2: Add additional input configuration parameters The 4th Gen Scarlett interfaces added software-controllable input gain along with channel select, channel link, auto-gain, and "safe" mode. Vocaster has software-controllable input gain and auto-gain but not channel select, channel link, or safe mode. Add a device info field safe_input_count to indicate how many channels have a safe mode control, and use the presence of the input select and input link switch configuration parameters to determine if those controls should be created. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: <167f04a37d0fb23f3077705df835adbc4f2b6a8e.1710264833.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 136 +++++++++++++++++++++--------------- 1 file changed, 80 insertions(+), 56 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index f2bbf90680f8b..4289661b453f5 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -916,6 +916,9 @@ struct scarlett2_device_info { /* the number of inputs with software-controllable gain */ u8 gain_input_count; + /* the number of inputs with safe mode */ + u8 safe_input_count; + /* the number of direct monitor options * (0 = none, 1 = mono only, 2 = mono/stereo) */ @@ -1550,6 +1553,7 @@ static const struct scarlett2_device_info s2i2_gen4_info = { .phantom_count = 1, .inputs_per_phantom = 2, .gain_input_count = 2, + .safe_input_count = 2, .direct_monitor = 2, .dsp_count = 2, @@ -1603,6 +1607,7 @@ static const struct scarlett2_device_info s4i4_gen4_info = { .phantom_count = 2, .inputs_per_phantom = 1, .gain_input_count = 2, + .safe_input_count = 2, .dsp_count = 2, .port_count = { @@ -2937,13 +2942,18 @@ static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer) int val = !scarlett2_autogain_is_running(private); int i; - scarlett2_set_ctl_access(private->input_select_ctl, val); - for (i = 0; i < info->gain_input_count / 2; i++) - scarlett2_set_ctl_access(private->input_link_ctls[i], val); - for (i = 0; i < info->gain_input_count; i++) { + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_INPUT_SELECT_SWITCH)) + scarlett2_set_ctl_access(private->input_select_ctl, val); + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_INPUT_LINK_SWITCH)) + for (i = 0; i < info->gain_input_count / 2; i++) + scarlett2_set_ctl_access(private->input_link_ctls[i], + val); + for (i = 0; i < info->gain_input_count; i++) scarlett2_set_ctl_access(private->input_gain_ctls[i], val); + for (i = 0; i < info->safe_input_count; i++) scarlett2_set_ctl_access(private->safe_ctls[i], val); - } for (i = 0; i < info->level_input_count; i++) scarlett2_set_ctl_access(private->level_ctls[i], val); for (i = 0; i < info->air_input_count; i++) @@ -2962,17 +2972,21 @@ static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer) const struct scarlett2_device_info *info = private->info; int i; - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, - &private->input_select_ctl->id); - for (i = 0; i < info->gain_input_count / 2; i++) + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_INPUT_SELECT_SWITCH)) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, - &private->input_link_ctls[i]->id); - for (i = 0; i < info->gain_input_count; i++) { + &private->input_select_ctl->id); + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_INPUT_LINK_SWITCH)) + for (i = 0; i < info->gain_input_count / 2; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->input_link_ctls[i]->id); + for (i = 0; i < info->gain_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &private->input_gain_ctls[i]->id); + for (i = 0; i < info->safe_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &private->safe_ctls[i]->id); - } for (i = 0; i < info->level_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &private->level_ctls[i]->id); @@ -3183,7 +3197,9 @@ static int scarlett2_update_input_select(struct usb_mixer_interface *mixer) private->input_select_updated = 0; - if (!link_count) + if (!scarlett2_has_config_item(private, + SCARLETT2_CONFIG_INPUT_SELECT_SWITCH) || + !link_count) return 0; err = scarlett2_usb_get_config( @@ -3596,12 +3612,12 @@ static int scarlett2_update_input_safe(struct usb_mixer_interface *mixer) private->input_safe_updated = 0; - if (!info->gain_input_count) + if (!info->safe_input_count) return 0; return scarlett2_usb_get_config( mixer, SCARLETT2_CONFIG_SAFE_SWITCH, - info->gain_input_count, private->safe_switch); + info->safe_input_count, private->safe_switch); } static int scarlett2_safe_ctl_get(struct snd_kcontrol *kctl, @@ -5507,58 +5523,65 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) return err; } - /* Add software-controllable input gain controls */ - if (info->gain_input_count) { + /* Add input select/link controls */ + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_INPUT_SELECT_SWITCH)) { err = scarlett2_add_new_ctl( mixer, &scarlett2_input_select_ctl, 0, 1, "Input Select Capture Enum", &private->input_select_ctl); if (err < 0) return err; + } - for (i = 0; i < info->gain_input_count; i++) { - if (i % 2) { - scnprintf(s, sizeof(s), - "Line In %d-%d Link Capture Switch", - i, i + 1); - err = scarlett2_add_new_ctl( - mixer, &scarlett2_input_link_ctl, - i / 2, 1, s, - &private->input_link_ctls[i / 2]); - if (err < 0) - return err; - } - - scnprintf(s, sizeof(s), fmt, i + 1, - "Gain", "Volume"); + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_INPUT_LINK_SWITCH)) { + for (i = 0; i < info->gain_input_count / 2; i++) { + scnprintf(s, sizeof(s), + "Line In %d-%d Link Capture Switch", + (i * 2) + 1, (i * 2) + 2); err = scarlett2_add_new_ctl( - mixer, &scarlett2_input_gain_ctl, - i, 1, s, &private->input_gain_ctls[i]); + mixer, &scarlett2_input_link_ctl, + i, 1, s, &private->input_link_ctls[i]); if (err < 0) return err; + } + } - scnprintf(s, sizeof(s), fmt, i + 1, - "Autogain", "Switch"); - err = scarlett2_add_new_ctl( - mixer, &scarlett2_autogain_switch_ctl, - i, 1, s, &private->autogain_ctls[i]); - if (err < 0) - return err; + /* Add software-controllable input gain controls */ + for (i = 0; i < info->gain_input_count; i++) { + scnprintf(s, sizeof(s), fmt, i + 1, + "Gain", "Volume"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_input_gain_ctl, + i, 1, s, &private->input_gain_ctls[i]); + if (err < 0) + return err; - scnprintf(s, sizeof(s), fmt, i + 1, - "Autogain Status", "Enum"); - err = scarlett2_add_new_ctl( - mixer, &scarlett2_autogain_status_ctl, - i, 1, s, &private->autogain_status_ctls[i]); + scnprintf(s, sizeof(s), fmt, i + 1, + "Autogain", "Switch"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_autogain_switch_ctl, + i, 1, s, &private->autogain_ctls[i]); + if (err < 0) + return err; - scnprintf(s, sizeof(s), fmt, i + 1, - "Safe", "Switch"); - err = scarlett2_add_new_ctl( - mixer, &scarlett2_safe_ctl, - i, 1, s, &private->safe_ctls[i]); - if (err < 0) - return err; - } + scnprintf(s, sizeof(s), fmt, i + 1, + "Autogain Status", "Enum"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_autogain_status_ctl, + i, 1, s, &private->autogain_status_ctls[i]); + } + + /* Add safe-mode input switch controls */ + for (i = 0; i < info->safe_input_count; i++) { + scnprintf(s, sizeof(s), fmt, i + 1, + "Safe", "Switch"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_safe_ctl, + i, 1, s, &private->safe_ctls[i]); + if (err < 0) + return err; } /* Add PCM Input Switch control */ @@ -6557,7 +6580,8 @@ static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer) const struct scarlett2_device_info *info = private->info; int i; - if (!info->gain_input_count) + if (!scarlett2_has_config_item(private, + SCARLETT2_CONFIG_INPUT_SELECT_SWITCH)) return; private->input_select_updated = 1; @@ -6620,12 +6644,12 @@ static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer) const struct scarlett2_device_info *info = private->info; int i; - if (!info->gain_input_count) + if (!info->safe_input_count) return; private->input_safe_updated = 1; - for (i = 0; i < info->gain_input_count; i++) + for (i = 0; i < info->safe_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->safe_ctls[i]->id); } From 87b73d48a58359922a98352091c4ccb6f463df55 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:05:57 +1030 Subject: [PATCH 315/603] ALSA: scarlett2: Define the maximum preamp input gain per-config-set Remove the #define SCARLETT2_MAX_GAIN_DB and replace with a per-config-set TLV as the Vocaster has a maximum gain of 70dB vs the 4th Gen 69dB. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: --- sound/usb/mixer_scarlett2.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 4289661b453f5..a891e92048b25 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -174,11 +174,10 @@ /* some gui mixers can't handle negative ctl values */ #define SCARLETT2_VOLUME_BIAS 127 -/* maximum preamp input gain and value - * values are from 0 to 70, preamp gain is from 0 to 69 dB +/* maximum preamp input gain value + * (the corresponding value in dB is per-device) */ #define SCARLETT2_MAX_GAIN_VALUE 70 -#define SCARLETT2_MAX_GAIN_DB 69 /* mixer range from -80dB to +6dB in 0.5dB steps */ #define SCARLETT2_MIXER_MIN_DB -80 @@ -460,9 +459,15 @@ struct scarlett2_config { struct scarlett2_config_set { const struct scarlett2_notification *notifications; u16 param_buf_addr; + const unsigned int *input_gain_tlv; const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; }; +/* Input gain TLV dB ranges */ +static const DECLARE_TLV_DB_MINMAX( + db_scale_gen4_gain, 0, 69 * 100 +); + /* Gen 2 devices without SW/HW volume switch: 6i6, 18i8 */ static const struct scarlett2_config_set scarlett2_config_set_gen2a = { @@ -658,6 +663,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = { static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = { .notifications = scarlett4_2i2_notifications, .param_buf_addr = 0xfc, + .input_gain_tlv = db_scale_gen4_gain, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x49, .size = 8, .activate = 4 }, @@ -703,6 +709,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = { static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = { .notifications = scarlett4_4i4_notifications, .param_buf_addr = 0x130, + .input_gain_tlv = db_scale_gen4_gain, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x5c, .size = 8, .activate = 4 }, @@ -3587,10 +3594,6 @@ static int scarlett2_input_gain_ctl_put(struct snd_kcontrol *kctl, return err; } -static const DECLARE_TLV_DB_MINMAX( - db_scale_scarlett2_gain, 0, SCARLETT2_MAX_GAIN_DB * 100 -); - static const struct snd_kcontrol_new scarlett2_input_gain_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -3600,7 +3603,6 @@ static const struct snd_kcontrol_new scarlett2_input_gain_ctl = { .get = scarlett2_input_gain_ctl_get, .put = scarlett2_input_gain_ctl_put, .private_value = 0, /* max value */ - .tlv = { .p = db_scale_scarlett2_gain } }; /*** Safe Controls ***/ @@ -5557,6 +5559,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) i, 1, s, &private->input_gain_ctls[i]); if (err < 0) return err; + private->input_gain_ctls[i]->tlv.p = + private->config_set->input_gain_tlv; scnprintf(s, sizeof(s), fmt, i + 1, "Autogain", "Switch"); From 23715a2176bc747cac4d5b5291419d51c6ac1a9e Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:06:04 +1030 Subject: [PATCH 316/603] ALSA: scarlett2: Define autogain status texts per-config-set The autogain status texts are different for Vocaster vs. Scarlett 4th Gen, so make them configurable per-config-set. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: --- sound/usb/mixer_scarlett2.c | 32 +++++++++++++++++++++++++------- 1 file changed, 25 insertions(+), 7 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index a891e92048b25..0962277947bf5 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -294,7 +294,7 @@ static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { * If autogain_switch is set, autogain_status is set to 0 (Running). * The other status values are from the raw_autogain_status value + 1. */ -static const char *const scarlett2_autogain_status_texts[] = { +static const char *const scarlett2_autogain_status_gen4[] = { "Running", "Success", "SuccessDRover", @@ -303,7 +303,8 @@ static const char *const scarlett2_autogain_status_texts[] = { "FailMaxGainLimit", "FailClipped", "Cancelled", - "Invalid" + "Invalid", + NULL }; /* Power Status Values */ @@ -460,6 +461,7 @@ struct scarlett2_config_set { const struct scarlett2_notification *notifications; u16 param_buf_addr; const unsigned int *input_gain_tlv; + const char *const *autogain_status_texts; const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; }; @@ -664,6 +666,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = { .notifications = scarlett4_2i2_notifications, .param_buf_addr = 0xfc, .input_gain_tlv = db_scale_gen4_gain, + .autogain_status_texts = scarlett2_autogain_status_gen4, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x49, .size = 8, .activate = 4 }, @@ -710,6 +713,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = { .notifications = scarlett4_4i4_notifications, .param_buf_addr = 0x130, .input_gain_tlv = db_scale_gen4_gain, + .autogain_status_texts = scarlett2_autogain_status_gen4, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x5c, .size = 8, .activate = 4 }, @@ -981,6 +985,7 @@ struct scarlett2_data { u8 num_mix_out; u8 num_line_out; u8 num_monitor_mix_ctls; + u8 num_autogain_status_texts; u32 firmware_version; u8 flash_segment_nums[SCARLETT2_SEGMENT_ID_COUNT]; u8 flash_segment_blocks[SCARLETT2_SEGMENT_ID_COUNT]; @@ -2931,12 +2936,12 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer) if (private->autogain_switch[i]) private->autogain_status[i] = 0; else if (raw_autogain_status[i] < - ARRAY_SIZE(scarlett2_autogain_status_texts) - 1) + private->num_autogain_status_texts - 1) private->autogain_status[i] = raw_autogain_status[i] + 1; else private->autogain_status[i] = - ARRAY_SIZE(scarlett2_autogain_status_texts) - 1; + private->num_autogain_status_texts - 1; return 0; } @@ -3171,10 +3176,13 @@ static int scarlett2_autogain_switch_ctl_put( static int scarlett2_autogain_status_ctl_info( struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; + return snd_ctl_enum_info( uinfo, 1, - ARRAY_SIZE(scarlett2_autogain_status_texts), - scarlett2_autogain_status_texts); + private->num_autogain_status_texts, + private->config_set->autogain_status_texts); } static const struct snd_kcontrol_new scarlett2_autogain_switch_ctl = { @@ -6839,8 +6847,9 @@ static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) static void scarlett2_count_io(struct scarlett2_data *private) { const struct scarlett2_device_info *info = private->info; + const struct scarlett2_config_set *config_set = info->config_set; const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int port_type, srcs = 0, dsts = 0; + int port_type, srcs = 0, dsts = 0, i; /* Count the number of mux sources and destinations */ for (port_type = 0; @@ -6872,6 +6881,15 @@ static void scarlett2_count_io(struct scarlett2_data *private) /* Number of monitor mix controls */ private->num_monitor_mix_ctls = info->direct_monitor * 2 * private->num_mix_in; + + /* Number of autogain status texts */ + if (config_set->autogain_status_texts) { + const char * const *texts = config_set->autogain_status_texts; + + for (i = 0; texts[i]; i++) + ; + private->num_autogain_status_texts = i; + } } /* Look through the interface descriptors for the Focusrite Control From bff5421a2c3fd12c719d69662bd92af20359e606 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:06:23 +1030 Subject: [PATCH 317/603] ALSA: scarlett2: Add input mute controls Add controls for the input mute switches that the Vocaster interfaces have. Mark scarlett2_notify_input_mute() as __always_unused until it gets used when the Vocaster callback function array is added. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: <3b384b4e759241bd06f0c223e9f4f00467d88318.1710264833.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 142 +++++++++++++++++++++++++++++++++++- 1 file changed, 141 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 0962277947bf5..17fadfddc2403 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -85,7 +85,7 @@ * controls * - disable/enable MSD mode * - disable/enable standalone mode - * - input gain, autogain, safe mode + * - input mute, gain, autogain, safe mode * - direct monitor mixes * * @@ -214,6 +214,7 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { #define SCARLETT2_LEVEL_SWITCH_MAX 2 #define SCARLETT2_PAD_SWITCH_MAX 8 #define SCARLETT2_AIR_SWITCH_MAX 8 +#define SCARLETT2_INPUT_MUTE_SWITCH_MAX 2 #define SCARLETT2_PHANTOM_SWITCH_MAX 2 #define SCARLETT2_INPUT_GAIN_MAX 2 @@ -329,6 +330,7 @@ static void scarlett2_notify_volume(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_mute(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer); @@ -415,6 +417,7 @@ enum { SCARLETT2_CONFIG_PAD_SWITCH, SCARLETT2_CONFIG_MSD_SWITCH, SCARLETT2_CONFIG_AIR_SWITCH, + SCARLETT2_CONFIG_INPUT_MUTE_SWITCH, SCARLETT2_CONFIG_STANDALONE_SWITCH, SCARLETT2_CONFIG_PHANTOM_SWITCH, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, @@ -915,6 +918,11 @@ struct scarlett2_device_info { */ u8 air_option; + /* the number of analogue inputs with a software switchable + * mute control + */ + u8 mute_input_count; + /* the number of phantom (48V) software switchable controls */ u8 phantom_count; @@ -996,6 +1004,7 @@ struct scarlett2_data { u8 input_level_updated; u8 input_pad_updated; u8 input_air_updated; + u8 input_mute_updated; u8 input_phantom_updated; u8 input_select_updated; u8 input_gain_updated; @@ -1018,6 +1027,7 @@ struct scarlett2_data { u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT]; u8 air_switch[SCARLETT2_AIR_SWITCH_MAX]; + u8 input_mute_switch[SCARLETT2_INPUT_MUTE_SWITCH_MAX]; u8 phantom_switch[SCARLETT2_PHANTOM_SWITCH_MAX]; u8 phantom_persistence; u8 input_select_switch; @@ -1045,6 +1055,7 @@ struct scarlett2_data { struct snd_kcontrol *level_ctls[SCARLETT2_LEVEL_SWITCH_MAX]; struct snd_kcontrol *pad_ctls[SCARLETT2_PAD_SWITCH_MAX]; struct snd_kcontrol *air_ctls[SCARLETT2_AIR_SWITCH_MAX]; + struct snd_kcontrol *input_mute_ctls[SCARLETT2_INPUT_MUTE_SWITCH_MAX]; struct snd_kcontrol *phantom_ctls[SCARLETT2_PHANTOM_SWITCH_MAX]; struct snd_kcontrol *input_select_ctl; struct snd_kcontrol *input_link_ctls[SCARLETT2_INPUT_GAIN_MAX / 2]; @@ -2970,6 +2981,8 @@ static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer) scarlett2_set_ctl_access(private->level_ctls[i], val); for (i = 0; i < info->air_input_count; i++) scarlett2_set_ctl_access(private->air_ctls[i], val); + for (i = 0; i < info->mute_input_count; i++) + scarlett2_set_ctl_access(private->input_mute_ctls[i], val); for (i = 0; i < info->phantom_count; i++) scarlett2_set_ctl_access(private->phantom_ctls[i], val); } @@ -3005,6 +3018,9 @@ static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer) for (i = 0; i < info->air_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &private->air_ctls[i]->id); + for (i = 0; i < info->mute_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->input_mute_ctls[i]->id); for (i = 0; i < info->phantom_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &private->phantom_ctls[i]->id); @@ -4619,6 +4635,100 @@ static const struct snd_kcontrol_new scarlett2_air_ctl[2] = { } }; +/*** Input Mute Switch Controls ***/ + +static int scarlett2_update_input_mute(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + private->input_mute_updated = 0; + + if (!info->mute_input_count) + return 0; + + return scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH, + info->mute_input_count, private->input_mute_switch); +} + +static int scarlett2_input_mute_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->input_mute_updated) { + err = scarlett2_update_input_mute(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.integer.value[0] = + private->input_mute_switch[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + + oval = private->input_mute_switch[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->input_mute_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH, + index, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_input_mute_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_autogain_disables_ctl_info, + .get = scarlett2_input_mute_ctl_get, + .put = scarlett2_input_mute_ctl_put, +}; + /*** Phantom Switch Controls ***/ static int scarlett2_update_input_phantom(struct usb_mixer_interface *mixer) @@ -5497,6 +5607,16 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) return err; } + /* Add input mute controls */ + for (i = 0; i < info->mute_input_count; i++) { + scnprintf(s, sizeof(s), fmt, i + 1, "Mute", "Switch"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_input_mute_ctl, + i, 1, s, &private->input_mute_ctls[i]); + if (err < 0) + return err; + } + /* Add input phantom controls */ if (info->inputs_per_phantom == 1) { for (i = 0; i < info->phantom_count; i++) { @@ -6558,6 +6678,22 @@ static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer) &private->air_ctls[i]->id); } +/* Notify on input mute switch change */ +static __always_unused void scarlett2_notify_input_mute( + struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + private->input_mute_updated = 1; + + for (i = 0; i < info->mute_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->input_mute_ctls[i]->id); +} + /* Notify on input phantom switch change */ static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer) { @@ -7171,6 +7307,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; + err = scarlett2_update_input_mute(mixer); + if (err < 0) + return err; + err = scarlett2_update_input_phantom(mixer); if (err < 0) return err; From b64678eb4e703aa7631d75d622bd295a717e74a8 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:07:12 +1030 Subject: [PATCH 318/603] ALSA: scarlett2: Add DSP controls Add filter and compressor DSP controls for the Vocaster interfaces. Mark scarlett2_notify_input_dsp() as __always_unused until it gets used when the Vocaster callback function array is added. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: --- sound/usb/mixer_scarlett2.c | 1047 +++++++++++++++++++++++++++++++---- 1 file changed, 951 insertions(+), 96 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 17fadfddc2403..77bfaadb99b89 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -87,6 +87,7 @@ * - disable/enable standalone mode * - input mute, gain, autogain, safe mode * - direct monitor mixes + * - compressor and EQ * * * /--------------\ 18chn 20chn /--------------\ @@ -214,6 +215,7 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { #define SCARLETT2_LEVEL_SWITCH_MAX 2 #define SCARLETT2_PAD_SWITCH_MAX 8 #define SCARLETT2_AIR_SWITCH_MAX 8 +#define SCARLETT2_DSP_SWITCH_MAX 2 #define SCARLETT2_INPUT_MUTE_SWITCH_MAX 2 #define SCARLETT2_PHANTOM_SWITCH_MAX 2 #define SCARLETT2_INPUT_GAIN_MAX 2 @@ -245,6 +247,59 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { /* Maximum number of meters (sum of output port counts) */ #define SCARLETT2_MAX_METERS 65 +/* Compressor parameter data + * + * The compressor parameters are 32-bit fixed point values with 24 + * bits of fraction. Integer values are sufficient for the parameters + * except for ratio which we can set in 0.5:1 steps. + */ +struct compressor_param { + const char *name; + snd_ctl_elem_type_t type; + s32 min; + s32 max; + int scale_bits; +}; + +/* The available compressor parameters on the Vocaster: + * - Enable: Off, On + * - Threshold: -40dB to 0dB + * - Ratio: 1:1 to 50:1 in 0.5:1 steps + * - Knee Width: 0dB to 10dB + * - Attack: 30ms to 127ms + * - Release: 30ms to 127ms + * - Makeup Gain: 0dB to 24dB + */ +static const struct compressor_param compressor_params[] = { + { "Enable", SNDRV_CTL_ELEM_TYPE_BOOLEAN, 0, 1, 0 }, + { "Threshold", SNDRV_CTL_ELEM_TYPE_INTEGER, -40, 0, 24 }, + { "Ratio", SNDRV_CTL_ELEM_TYPE_INTEGER, 2, 100, 23 }, + { "Knee Width", SNDRV_CTL_ELEM_TYPE_INTEGER, 0, 10, 24 }, + { "Attack", SNDRV_CTL_ELEM_TYPE_INTEGER, 30, 127, 24 }, + { "Release", SNDRV_CTL_ELEM_TYPE_INTEGER, 30, 127, 24 }, + { "Makeup Gain", SNDRV_CTL_ELEM_TYPE_INTEGER, 0, 24, 24 }, +}; + +#define SCARLETT2_COMPRESSOR_PARAM_COUNT ARRAY_SIZE(compressor_params) +#define SCARLETT2_COMPRESSOR_CTLS_MAX \ + (SCARLETT2_COMPRESSOR_PARAM_COUNT * SCARLETT2_DSP_SWITCH_MAX) + +/* Maximum number of filter controls */ +#define SCARLETT2_PRECOMP_FLT_CTLS_MAX (2 * SCARLETT2_DSP_SWITCH_MAX) +#define SCARLETT2_PEQ_FLT_CTLS_MAX (3 * SCARLETT2_DSP_SWITCH_MAX) + +/* Number of biquad filter coefficients */ +#define SCARLETT2_BIQUAD_COEFFS 5 + +/* Maximum number of filter coefficient values */ +#define SCARLETT2_PRECOMP_FLT_VALUES_MAX \ + (SCARLETT2_PRECOMP_FLT_CTLS_MAX * SCARLETT2_BIQUAD_COEFFS) +#define SCARLETT2_PEQ_FLT_VALUES_MAX \ + (SCARLETT2_PEQ_FLT_CTLS_MAX * SCARLETT2_BIQUAD_COEFFS) + +/* Maximum number of PEQ filter slots */ +#define SCARLETT2_PEQ_FLT_SLOTS_MAX 4 + /* Hardware port types: * - None (no input to mux) * - Analogue I/O @@ -330,6 +385,7 @@ static void scarlett2_notify_volume(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_dsp(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_mute(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer); @@ -417,6 +473,12 @@ enum { SCARLETT2_CONFIG_PAD_SWITCH, SCARLETT2_CONFIG_MSD_SWITCH, SCARLETT2_CONFIG_AIR_SWITCH, + SCARLETT2_CONFIG_DSP_SWITCH, + SCARLETT2_CONFIG_COMPRESSOR_PARAMS, + SCARLETT2_CONFIG_PRECOMP_FLT_SWITCH, + SCARLETT2_CONFIG_PRECOMP_FLT_PARAMS, + SCARLETT2_CONFIG_PEQ_FLT_SWITCH, + SCARLETT2_CONFIG_PEQ_FLT_PARAMS, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH, SCARLETT2_CONFIG_STANDALONE_SWITCH, SCARLETT2_CONFIG_PHANTOM_SWITCH, @@ -918,6 +980,18 @@ struct scarlett2_device_info { */ u8 air_option; + /* the number of analogue inputs with DSP control */ + u8 dsp_input_count; + + /* number of pre-compressor filters */ + u8 precomp_flt_count; + + /* number of parametric EQ filters */ + u8 peq_flt_count; + + /* number of PEQ filters plus unused slots */ + u8 peq_flt_total_count; + /* the number of analogue inputs with a software switchable * mute control */ @@ -1004,6 +1078,7 @@ struct scarlett2_data { u8 input_level_updated; u8 input_pad_updated; u8 input_air_updated; + u8 input_dsp_updated; u8 input_mute_updated; u8 input_phantom_updated; u8 input_select_updated; @@ -1027,6 +1102,12 @@ struct scarlett2_data { u8 pad_switch[SCARLETT2_PAD_SWITCH_MAX]; u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT]; u8 air_switch[SCARLETT2_AIR_SWITCH_MAX]; + u8 dsp_switch[SCARLETT2_DSP_SWITCH_MAX]; + s32 compressor_values[SCARLETT2_COMPRESSOR_CTLS_MAX]; + s32 precomp_flt_values[SCARLETT2_PRECOMP_FLT_VALUES_MAX]; + s32 peq_flt_values[SCARLETT2_PEQ_FLT_VALUES_MAX]; + u8 precomp_flt_switch[SCARLETT2_DSP_SWITCH_MAX]; + u8 peq_flt_switch[SCARLETT2_DSP_SWITCH_MAX]; u8 input_mute_switch[SCARLETT2_INPUT_MUTE_SWITCH_MAX]; u8 phantom_switch[SCARLETT2_PHANTOM_SWITCH_MAX]; u8 phantom_persistence; @@ -1055,6 +1136,7 @@ struct scarlett2_data { struct snd_kcontrol *level_ctls[SCARLETT2_LEVEL_SWITCH_MAX]; struct snd_kcontrol *pad_ctls[SCARLETT2_PAD_SWITCH_MAX]; struct snd_kcontrol *air_ctls[SCARLETT2_AIR_SWITCH_MAX]; + struct snd_kcontrol *dsp_ctls[SCARLETT2_DSP_SWITCH_MAX]; struct snd_kcontrol *input_mute_ctls[SCARLETT2_INPUT_MUTE_SWITCH_MAX]; struct snd_kcontrol *phantom_ctls[SCARLETT2_PHANTOM_SWITCH_MAX]; struct snd_kcontrol *input_select_ctl; @@ -1066,6 +1148,11 @@ struct scarlett2_data { struct snd_kcontrol *pcm_input_switch_ctl; struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX]; struct snd_kcontrol *mix_ctls[SCARLETT2_MIX_MAX]; + struct snd_kcontrol *compressor_ctls[SCARLETT2_COMPRESSOR_CTLS_MAX]; + struct snd_kcontrol *precomp_flt_ctls[SCARLETT2_PRECOMP_FLT_CTLS_MAX]; + struct snd_kcontrol *peq_flt_ctls[SCARLETT2_PEQ_FLT_CTLS_MAX]; + struct snd_kcontrol *precomp_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX]; + struct snd_kcontrol *peq_flt_switch_ctls[SCARLETT2_DSP_SWITCH_MAX]; struct snd_kcontrol *direct_monitor_ctl; struct snd_kcontrol *speaker_switching_ctl; struct snd_kcontrol *talkback_ctl; @@ -2174,6 +2261,54 @@ static int scarlett2_usb_set_data( &req, sizeof(u32) * 2 + size, NULL, 0); } +/* Send a SCARLETT2_USB_SET_DATA command with multiple values. + * offset: location in the device's data space + * size: size in bytes of each value (1, 2, 4) + * count: number of values + */ +static int scarlett2_usb_set_data_buf( + struct usb_mixer_interface *mixer, + int offset, int size, int count, void *buf) +{ + struct scarlett2_data *private = mixer->private_data; + int bytes = size * count; + struct { + __le32 offset; + __le32 size; + u8 data[]; + } __packed *req; + int err; + int buf_size = struct_size(req, data, bytes); + + req = kmalloc(buf_size, GFP_KERNEL); + if (!req) + return -ENOMEM; + + req->offset = cpu_to_le32(offset); + req->size = cpu_to_le32(bytes); + if (size == 1) { + memcpy(req->data, buf, count); + } else if (size == 2) { + u16 *buf_16 = buf; + int i; + + for (i = 0; i < count; i++) + ((__le16 *)req->data)[i] = cpu_to_le16(buf_16[i]); + } else { + u32 *buf_32 = buf; + int i; + + for (i = 0; i < count; i++) + ((__le32 *)req->data)[i] = cpu_to_le32(buf_32[i]); + } + + err = scarlett2_usb(private->mixer, SCARLETT2_USB_SET_DATA, + req, buf_size, NULL, 0); + + kfree(req); + return err; +} + /* Send a SCARLETT2_USB_DATA_CMD command. * Configuration changes require activation with this after they have * been uploaded by a previous SCARLETT2_USB_SET_DATA. @@ -2288,6 +2423,47 @@ static int scarlett2_usb_set_config( return 0; } +/* Send USB messages to set a SCARLETT2_CONFIG_* parameter with + * multiple values + */ +static int scarlett2_usb_set_config_buf( + struct usb_mixer_interface *mixer, + int config_item_num, int index, int count, void *buf) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_config_set *config_set = private->config_set; + const struct scarlett2_config *config_item = + &config_set->items[config_item_num]; + int offset, size; + int err; + + /* Check that the configuration item is present in the + * configuration set used by this device + */ + if (!config_item->offset) + return -EFAULT; + + /* Convert config_item->size in bits to size in bytes and + * calculate offset + */ + if (config_item->size >= 8) { + size = config_item->size / 8; + offset = config_item->offset + index * size; + + /* Bit updates not supported */ + } else { + return -EFAULT; + } + + /* Write the new values */ + err = scarlett2_usb_set_data_buf(mixer, offset, size, count, buf); + if (err < 0) + return err; + + /* Activate the change */ + return scarlett2_usb_activate_config(mixer, config_item->activate); +} + /* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ static void scarlett2_config_save(struct usb_mixer_interface *mixer) { @@ -2985,6 +3161,8 @@ static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer) scarlett2_set_ctl_access(private->input_mute_ctls[i], val); for (i = 0; i < info->phantom_count; i++) scarlett2_set_ctl_access(private->phantom_ctls[i], val); + for (i = 0; i < info->dsp_input_count; i++) + scarlett2_set_ctl_access(private->dsp_ctls[i], val); } /* Notify of access mode change for all controls read-only while @@ -3018,6 +3196,9 @@ static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer) for (i = 0; i < info->air_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &private->air_ctls[i]->id); + for (i = 0; i < info->dsp_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->dsp_ctls[i]->id); for (i = 0; i < info->mute_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &private->input_mute_ctls[i]->id); @@ -4635,25 +4816,25 @@ static const struct snd_kcontrol_new scarlett2_air_ctl[2] = { } }; -/*** Input Mute Switch Controls ***/ +/*** DSP Switch Control ***/ -static int scarlett2_update_input_mute(struct usb_mixer_interface *mixer) +static int scarlett2_update_input_dsp(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - private->input_mute_updated = 0; + private->input_dsp_updated = 0; - if (!info->mute_input_count) + if (!info->dsp_input_count) return 0; return scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH, - info->mute_input_count, private->input_mute_switch); + mixer, SCARLETT2_CONFIG_DSP_SWITCH, + info->dsp_input_count, private->dsp_switch); } -static int scarlett2_input_mute_ctl_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int scarlett2_dsp_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; @@ -4667,21 +4848,20 @@ static int scarlett2_input_mute_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - if (private->input_mute_updated) { - err = scarlett2_update_input_mute(mixer); + if (private->input_dsp_updated) { + err = scarlett2_update_input_dsp(mixer); if (err < 0) goto unlock; } - ucontrol->value.integer.value[0] = - private->input_mute_switch[elem->control]; + ucontrol->value.integer.value[0] = private->dsp_switch[elem->control]; unlock: mutex_unlock(&private->data_mutex); return err; } -static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int scarlett2_dsp_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; @@ -4701,18 +4881,17 @@ static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl, if (err < 0) goto unlock; - oval = private->input_mute_switch[index]; + oval = private->dsp_switch[index]; val = ucontrol->value.integer.value[0]; if (oval == val) goto unlock; - private->input_mute_switch[index] = val; + private->dsp_switch[index] = val; /* Send switch change to the device */ - err = scarlett2_usb_set_config( - mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH, - index, val); + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_DSP_SWITCH, + index, val); if (err == 0) err = 1; @@ -4721,113 +4900,173 @@ static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl, return err; } -static const struct snd_kcontrol_new scarlett2_input_mute_ctl = { +static const struct snd_kcontrol_new scarlett2_dsp_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", .info = scarlett2_autogain_disables_ctl_info, - .get = scarlett2_input_mute_ctl_get, - .put = scarlett2_input_mute_ctl_put, + .get = scarlett2_dsp_ctl_get, + .put = scarlett2_dsp_ctl_put, }; -/*** Phantom Switch Controls ***/ +/*** DSP Compressor Parameter Controls ***/ -static int scarlett2_update_input_phantom(struct usb_mixer_interface *mixer) +static int scarlett2_update_compressor_values(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - int err; - - private->input_phantom_updated = 0; + int err, i, j; - if (!info->phantom_count) + if (!info->dsp_input_count) return 0; err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_PHANTOM_SWITCH, - info->phantom_count, private->phantom_switch); + mixer, SCARLETT2_CONFIG_COMPRESSOR_PARAMS, + SCARLETT2_COMPRESSOR_PARAM_COUNT * info->dsp_input_count, + private->compressor_values); + if (err < 0) return err; - if (scarlett2_has_config_item(private, - SCARLETT2_CONFIG_PHANTOM_PERSISTENCE)) { - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, - 1, &private->phantom_persistence); - if (err < 0) - return err; + for (i = 0; i < SCARLETT2_COMPRESSOR_PARAM_COUNT; i++) { + const struct compressor_param *param = &compressor_params[i]; + + for (j = 0; j < info->dsp_input_count; j++) { + int idx = i + j * SCARLETT2_COMPRESSOR_PARAM_COUNT; + int val = private->compressor_values[idx]; + + val >>= param->scale_bits; + val = clamp(val, param->min, param->max); + private->compressor_values[idx] = val; + } } return 0; } -/* Check if phantom power on the given input is currently changing state */ -static int scarlett2_phantom_is_switching( - struct scarlett2_data *private, int line_num) +static int scarlett2_compressor_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - const struct scarlett2_device_info *info = private->info; - int index = line_num / info->inputs_per_phantom; + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; - return !!(private->phantom_switch[index] & 0x02); + ucontrol->value.integer.value[0] = + private->compressor_values[elem->control]; + return 0; } -/* Update autogain controls' access mode when phantom power changes state */ -static void scarlett2_phantom_update_access(struct usb_mixer_interface *mixer) +static int scarlett2_compressor_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int i; - /* Disable autogain controls if phantom power is changing state */ - for (i = 0; i < info->gain_input_count; i++) { - int val = !scarlett2_phantom_is_switching(private, i); + int index = elem->control; + int channel = index / SCARLETT2_COMPRESSOR_PARAM_COUNT; + int param_index = index % SCARLETT2_COMPRESSOR_PARAM_COUNT; + int oval, val, err; + s32 scaled_val; - scarlett2_set_ctl_access(private->autogain_ctls[i], val); + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; } + + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + + oval = private->compressor_values[index]; + val = ucontrol->value.integer.value[0]; + if (oval == val) + goto unlock; + + private->compressor_values[index] = val; + + const struct compressor_param *param = &compressor_params[param_index]; + + scaled_val = val << param->scale_bits; + + /* Send change to the device */ + + /* The channel needs to be put in the parameter buffer index + * field (param_buf_addr + 1); the value field isn't used in + * this case. + */ + err = scarlett2_usb_set_data( + mixer, private->config_set->param_buf_addr + 1, 1, channel); + if (err < 0) + goto unlock; + + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_COMPRESSOR_PARAMS, index, scaled_val); + if (err < 0) + goto unlock; + + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } -/* Notify of access mode change for autogain which can't be enabled - * while phantom power is changing. - */ -static void scarlett2_phantom_notify_access(struct usb_mixer_interface *mixer) +static int scarlett2_compressor_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct snd_card *card = mixer->chip->card; - struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int i; + struct usb_mixer_elem_info *elem = kctl->private_data; + int control = elem->control % SCARLETT2_COMPRESSOR_PARAM_COUNT; - for (i = 0; i < info->gain_input_count; i++) - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, - &private->autogain_ctls[i]->id); + uinfo->type = compressor_params[control].type; + uinfo->count = 1; + uinfo->value.integer.min = compressor_params[control].min; + uinfo->value.integer.max = compressor_params[control].max; + uinfo->value.integer.step = 1; + return 0; } -/* Call scarlett2_update_input_phantom() and - * scarlett2_phantom_update_access() if input_phantom_updated is set. - */ -static int scarlett2_check_input_phantom_updated( - struct usb_mixer_interface *mixer) +static const struct snd_kcontrol_new scarlett2_compressor_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", + .info = scarlett2_compressor_ctl_info, + .get = scarlett2_compressor_ctl_get, + .put = scarlett2_compressor_ctl_put, +}; + +/*** DSP Pre-Compressor and PEQ Filter Controls ***/ + +static int scarlett2_precomp_flt_switch_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct scarlett2_data *private = mixer->private_data; - int err; + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; - if (!private->input_phantom_updated) - return 0; + ucontrol->value.integer.value[0] = private->precomp_flt_switch[elem->control]; - err = scarlett2_update_input_phantom(mixer); - if (err < 0) - return err; + return 0; +} - scarlett2_phantom_update_access(mixer); +static int scarlett2_peq_flt_switch_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = + private->peq_flt_switch[elem->control]; return 0; } -static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int scarlett2_precomp_flt_switch_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - int err; + int oval, val, err = 0; mutex_lock(&private->data_mutex); @@ -4836,28 +5075,33 @@ static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - err = scarlett2_check_input_phantom_updated(mixer); - if (err < 0) + oval = private->precomp_flt_switch[elem->control]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) goto unlock; - ucontrol->value.integer.value[0] = scarlett2_decode_muteable( - private->phantom_switch[elem->control]); + private->precomp_flt_switch[elem->control] = val; + + /* Send change to the device */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_PRECOMP_FLT_SWITCH, + elem->control, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); return err; } -static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int scarlett2_peq_flt_switch_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - - int index = elem->control; - int oval, val, err; + int oval, val, err = 0; mutex_lock(&private->data_mutex); @@ -4866,19 +5110,526 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, goto unlock; } - err = scarlett2_check_put_during_autogain(mixer); - if (err < 0) - goto unlock; - - oval = private->phantom_switch[index]; - val = !!ucontrol->value.integer.value[0]; + oval = private->peq_flt_switch[elem->control]; + val = ucontrol->value.integer.value[0]; if (oval == val) goto unlock; - private->phantom_switch[index] = val; + private->peq_flt_switch[elem->control] = val; - /* To set the Gen 4 muteable controls, bit 1 gets set */ + /* Send change to the device */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_PEQ_FLT_SWITCH, + elem->control, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_precomp_flt_switch_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_precomp_flt_switch_ctl_get, + .put = scarlett2_precomp_flt_switch_ctl_put, +}; + +static const struct snd_kcontrol_new scarlett2_peq_flt_switch_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_peq_flt_switch_ctl_get, + .put = scarlett2_peq_flt_switch_ctl_put, +}; + +static int scarlett2_update_filter_values(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i, j, k, src_idx, dst_idx; + s32 peq_flt_values[SCARLETT2_DSP_SWITCH_MAX * + SCARLETT2_PEQ_FLT_SLOTS_MAX * + SCARLETT2_BIQUAD_COEFFS]; + + if (!info->dsp_input_count) + return 0; + + /* Get filter switch values */ + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PRECOMP_FLT_SWITCH, + info->dsp_input_count, private->precomp_flt_switch); + if (err < 0) + return err; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PEQ_FLT_SWITCH, + info->dsp_input_count * info->peq_flt_count, + private->peq_flt_switch); + if (err < 0) + return err; + + /* Get pre-compressor filter values directly */ + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PRECOMP_FLT_PARAMS, + info->dsp_input_count * + info->precomp_flt_count * + SCARLETT2_BIQUAD_COEFFS, + private->precomp_flt_values); + + if (err < 0) + return err; + + /* PEQ filter values need to be copied via buffer because of + * padding after peq_flt_count up to peq_flt_total_count + */ + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PEQ_FLT_PARAMS, + info->dsp_input_count * + info->peq_flt_total_count * + SCARLETT2_BIQUAD_COEFFS, + peq_flt_values); + + for (i = 0, dst_idx = 0; i < info->dsp_input_count; i++) { + src_idx = i * + info->peq_flt_total_count * + SCARLETT2_BIQUAD_COEFFS; + for (j = 0; j < info->peq_flt_count; j++) + for (k = 0; + k < SCARLETT2_BIQUAD_COEFFS; + k++, src_idx++, dst_idx++) + private->peq_flt_values[dst_idx] = + peq_flt_values[src_idx]; + } + + return 0; +} + +static int scarlett2_precomp_flt_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; + int i, idx; + + for (i = 0, idx = elem->control * SCARLETT2_BIQUAD_COEFFS; + i < SCARLETT2_BIQUAD_COEFFS; + i++, idx++) + ucontrol->value.integer.value[i] = + private->precomp_flt_values[idx]; + + return 0; +} + +static int scarlett2_peq_flt_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; + int i, idx; + + for (i = 0, idx = elem->control * SCARLETT2_BIQUAD_COEFFS; + i < SCARLETT2_BIQUAD_COEFFS; + i++, idx++) + ucontrol->value.integer.value[i] = + private->peq_flt_values[idx]; + + return 0; +} + +static int scarlett2_precomp_flt_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control * SCARLETT2_BIQUAD_COEFFS; + int i, oval, val, err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + + /* Check if any of the values have changed; if not, return */ + for (i = 0; i < SCARLETT2_BIQUAD_COEFFS; i++) { + oval = private->precomp_flt_values[index + i]; + val = ucontrol->value.integer.value[i]; + if (oval != val) + break; + } + + if (i == SCARLETT2_BIQUAD_COEFFS) + goto unlock; + + /* Update the values */ + for (i = 0; i < SCARLETT2_BIQUAD_COEFFS; i++) + private->precomp_flt_values[index + i] = + ucontrol->value.integer.value[i]; + + /* Send change to the device */ + err = scarlett2_usb_set_data( + mixer, private->config_set->param_buf_addr, 1, index); + if (err < 0) + goto unlock; + + err = scarlett2_usb_set_config_buf( + mixer, SCARLETT2_CONFIG_PRECOMP_FLT_PARAMS, + index, SCARLETT2_BIQUAD_COEFFS, + &private->precomp_flt_values[index]); + + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_peq_flt_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + int src_index = elem->control * SCARLETT2_BIQUAD_COEFFS; + int dst_index = ( + elem->control / + info->peq_flt_count * + info->peq_flt_total_count + + elem->control % info->peq_flt_count + ) * SCARLETT2_BIQUAD_COEFFS; + int i, oval, val, err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + + /* Check if any of the values have changed; if not, return */ + for (i = 0; i < SCARLETT2_BIQUAD_COEFFS; i++) { + oval = private->peq_flt_values[src_index + i]; + val = ucontrol->value.integer.value[i]; + if (oval != val) + break; + } + + if (i == SCARLETT2_BIQUAD_COEFFS) + goto unlock; + + /* Update the values */ + for (i = 0; i < SCARLETT2_BIQUAD_COEFFS; i++) + private->peq_flt_values[src_index + i] = + ucontrol->value.integer.value[i]; + + /* Send change to the device */ + err = scarlett2_usb_set_data( + mixer, private->config_set->param_buf_addr, 1, dst_index); + if (err < 0) + goto unlock; + + err = scarlett2_usb_set_config_buf( + mixer, SCARLETT2_CONFIG_PEQ_FLT_PARAMS, + dst_index, SCARLETT2_BIQUAD_COEFFS, + &private->peq_flt_values[src_index]); + + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_flt_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = SCARLETT2_BIQUAD_COEFFS; + uinfo->value.integer.min = INT_MIN; + uinfo->value.integer.max = INT_MAX; + uinfo->value.integer.step = 1; + return 0; +} + +static const struct snd_kcontrol_new scarlett2_precomp_flt_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", + .info = scarlett2_flt_ctl_info, + .get = scarlett2_precomp_flt_ctl_get, + .put = scarlett2_precomp_flt_ctl_put, +}; + +static const struct snd_kcontrol_new scarlett2_peq_flt_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", + .info = scarlett2_flt_ctl_info, + .get = scarlett2_peq_flt_ctl_get, + .put = scarlett2_peq_flt_ctl_put, +}; + +/*** Input Mute Switch Controls ***/ + +static int scarlett2_update_input_mute(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + private->input_mute_updated = 0; + + if (!info->mute_input_count) + return 0; + + return scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH, + info->mute_input_count, private->input_mute_switch); +} + +static int scarlett2_input_mute_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->input_mute_updated) { + err = scarlett2_update_input_mute(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.integer.value[0] = + private->input_mute_switch[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_input_mute_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + + oval = private->input_mute_switch[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->input_mute_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_INPUT_MUTE_SWITCH, + index, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_input_mute_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_autogain_disables_ctl_info, + .get = scarlett2_input_mute_ctl_get, + .put = scarlett2_input_mute_ctl_put, +}; + +/*** Phantom Switch Controls ***/ + +static int scarlett2_update_input_phantom(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err; + + private->input_phantom_updated = 0; + + if (!info->phantom_count) + return 0; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PHANTOM_SWITCH, + info->phantom_count, private->phantom_switch); + if (err < 0) + return err; + + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_PHANTOM_PERSISTENCE)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, + 1, &private->phantom_persistence); + if (err < 0) + return err; + } + + return 0; +} + +/* Check if phantom power on the given input is currently changing state */ +static int scarlett2_phantom_is_switching( + struct scarlett2_data *private, int line_num) +{ + const struct scarlett2_device_info *info = private->info; + int index = line_num / info->inputs_per_phantom; + + return !!(private->phantom_switch[index] & 0x02); +} + +/* Update autogain controls' access mode when phantom power changes state */ +static void scarlett2_phantom_update_access(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + /* Disable autogain controls if phantom power is changing state */ + for (i = 0; i < info->gain_input_count; i++) { + int val = !scarlett2_phantom_is_switching(private, i); + + scarlett2_set_ctl_access(private->autogain_ctls[i], val); + } +} + +/* Notify of access mode change for autogain which can't be enabled + * while phantom power is changing. + */ +static void scarlett2_phantom_notify_access(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + for (i = 0; i < info->gain_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->autogain_ctls[i]->id); +} + +/* Call scarlett2_update_input_phantom() and + * scarlett2_phantom_update_access() if input_phantom_updated is set. + */ +static int scarlett2_check_input_phantom_updated( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err; + + if (!private->input_phantom_updated) + return 0; + + err = scarlett2_update_input_phantom(mixer); + if (err < 0) + return err; + + scarlett2_phantom_update_access(mixer); + + return 0; +} + +static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_input_phantom_updated(mixer); + if (err < 0) + goto unlock; + + ucontrol->value.integer.value[0] = scarlett2_decode_muteable( + private->phantom_switch[elem->control]); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + int index = elem->control; + int oval, val, err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + + oval = private->phantom_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->phantom_switch[index] = val; + + /* To set the Gen 4 muteable controls, bit 1 gets set */ if (private->config_set->items[SCARLETT2_CONFIG_PHANTOM_SWITCH].mute) val = (!val) | 0x02; @@ -5568,6 +6319,69 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) /*** Create the analogue input controls ***/ +static int scarlett2_add_dsp_ctls(struct usb_mixer_interface *mixer, int i) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int j, err; + char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + const char *compr_fmt = "Line In %d Compressor %s"; + const char *flt_switch_fmt = "Line In %d %s Filter Enable"; + const char *flt_fmt = "Line In %d %s Coefficients %d"; + + /* Add compressor controls */ + for (j = 0; j < SCARLETT2_COMPRESSOR_PARAM_COUNT; j++) { + const struct compressor_param *param = &compressor_params[j]; + int idx = i * SCARLETT2_COMPRESSOR_PARAM_COUNT + j; + + scnprintf(s, sizeof(s), compr_fmt, i + 1, param->name); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_compressor_ctl, + i * SCARLETT2_COMPRESSOR_PARAM_COUNT + j, + 1, s, &private->compressor_ctls[idx]); + if (err < 0) + return err; + } + + /* Add filter enable controls */ + scnprintf(s, sizeof(s), flt_switch_fmt, i + 1, "Pre-Comp"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_precomp_flt_switch_ctl, + i, 1, s, &private->precomp_flt_switch_ctls[i]); + if (err < 0) + return err; + + scnprintf(s, sizeof(s), flt_switch_fmt, i + 1, "PEQ"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_peq_flt_switch_ctl, + i, 1, s, &private->peq_flt_switch_ctls[i]); + if (err < 0) + return err; + + /* Add filter coefficient controls */ + for (j = 0; j < info->precomp_flt_count; j++) { + scnprintf(s, sizeof(s), flt_fmt, i + 1, "Pre-Comp", j + 1); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_precomp_flt_ctl, + i * info->precomp_flt_count + j, + 1, s, &private->precomp_flt_switch_ctls[j]); + if (err < 0) + return err; + } + + for (j = 0; j < info->peq_flt_count; j++) { + scnprintf(s, sizeof(s), flt_fmt, i + 1, "PEQ", j + 1); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_peq_flt_ctl, + i * info->peq_flt_count + j, + 1, s, &private->peq_flt_switch_ctls[j]); + if (err < 0) + return err; + } + + return 0; +} + static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; @@ -5607,6 +6421,19 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) return err; } + /* Add input DSP controls */ + for (i = 0; i < info->dsp_input_count; i++) { + scnprintf(s, sizeof(s), fmt, i + 1, "DSP", "Switch"); + err = scarlett2_add_new_ctl(mixer, &scarlett2_dsp_ctl, + i, 1, s, &private->dsp_ctls[i]); + if (err < 0) + return err; + + err = scarlett2_add_dsp_ctls(mixer, i); + if (err < 0) + return err; + } + /* Add input mute controls */ for (i = 0; i < info->mute_input_count; i++) { scnprintf(s, sizeof(s), fmt, i + 1, "Mute", "Switch"); @@ -6678,6 +7505,22 @@ static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer) &private->air_ctls[i]->id); } +/* Notify on input DSP switch change */ +static __always_unused void scarlett2_notify_input_dsp( + struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + private->input_dsp_updated = 1; + + for (i = 0; i < info->dsp_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->dsp_ctls[i]->id); +} + /* Notify on input mute switch change */ static __always_unused void scarlett2_notify_input_mute( struct usb_mixer_interface *mixer) @@ -7307,6 +8150,18 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; + err = scarlett2_update_input_dsp(mixer); + if (err < 0) + return err; + + err = scarlett2_update_compressor_values(mixer); + if (err < 0) + return err; + + err = scarlett2_update_filter_values(mixer); + if (err < 0) + return err; + err = scarlett2_update_input_mute(mixer); if (err < 0) return err; From 16a7b277c40615befb21eb9420f743dc157d67bd Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:07:43 +1030 Subject: [PATCH 319/603] ALSA: scarlett2: Add support for Focusrite Vocaster One and Two Add Focusrite Vocaster One and Two USB IDs, notification arrays, config sets, and device info data. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: <5fb48555a8db7bb322b25784b165829357cd6e42.1710264833.git.g@b4.vu> --- sound/usb/mixer_quirks.c | 2 + sound/usb/mixer_scarlett2.c | 167 +++++++++++++++++++++++++++++++++++- 2 files changed, 165 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 065a4be0d7714..212b5e6443d88 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3447,6 +3447,8 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x8213): /* Focusrite Scarlett 8i6 3rd Gen */ case USB_ID(0x1235, 0x8214): /* Focusrite Scarlett 18i8 3rd Gen */ case USB_ID(0x1235, 0x8215): /* Focusrite Scarlett 18i20 3rd Gen */ + case USB_ID(0x1235, 0x8216): /* Focusrite Vocaster One */ + case USB_ID(0x1235, 0x8217): /* Focusrite Vocaster Two */ case USB_ID(0x1235, 0x8218): /* Focusrite Scarlett Solo 4th Gen */ case USB_ID(0x1235, 0x8219): /* Focusrite Scarlett 2i2 4th Gen */ case USB_ID(0x1235, 0x821a): /* Focusrite Scarlett 4i4 4th Gen */ diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 77bfaadb99b89..43922e217503f 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -363,6 +363,18 @@ static const char *const scarlett2_autogain_status_gen4[] = { NULL }; +static const char *const scarlett2_autogain_status_vocaster[] = { + "Running", + "Success", + "FailPG", + "FailRange", + "WarnMaxCap", + "WarnMinCap", + "Cancelled", + "Invalid", + NULL +}; + /* Power Status Values */ enum { SCARLETT2_POWER_STATUS_EXT, @@ -418,6 +430,17 @@ static const struct scarlett2_notification scarlett3a_notifications[] = { { 0, NULL } }; +static const struct scarlett2_notification vocaster_notifications[] = { + { 0x00000001, scarlett2_notify_ack }, + { 0x00000008, scarlett2_notify_sync }, + { 0x00200000, scarlett2_notify_input_mute }, + { 0x00400000, scarlett2_notify_autogain }, + { 0x04000000, scarlett2_notify_input_dsp }, + { 0x08000000, scarlett2_notify_input_gain }, + { 0x10000000, scarlett2_notify_input_phantom }, + { 0, NULL } +}; + static const struct scarlett2_notification scarlett4_solo_notifications[] = { { 0x00000001, scarlett2_notify_ack }, { 0x00000008, scarlett2_notify_sync }, @@ -531,6 +554,11 @@ struct scarlett2_config_set { }; /* Input gain TLV dB ranges */ + +static const DECLARE_TLV_DB_MINMAX( + db_scale_vocaster_gain, 0, 70 * 100 +); + static const DECLARE_TLV_DB_MINMAX( db_scale_gen4_gain, 0, 69 * 100 ); @@ -696,6 +724,51 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3c = { } }; +/* Vocaster */ +static const struct scarlett2_config_set scarlett2_config_set_vocaster = { + .notifications = vocaster_notifications, + .param_buf_addr = 0x1bc, + .input_gain_tlv = db_scale_vocaster_gain, + .autogain_status_texts = scarlett2_autogain_status_vocaster, + .items = { + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x9d, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_AUTOGAIN_SWITCH] = { + .offset = 0x1c0, .size = 8, .activate = 19, .pbuf = 1 }, + + [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = { + .offset = 0x1c2, .size = 8, }, + + [SCARLETT2_CONFIG_INPUT_GAIN] = { + .offset = 0x9f, .size = 8, .activate = 21, .pbuf = 1 }, + + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x9c, .size = 1, .activate = 20, .pbuf = 1 }, + + [SCARLETT2_CONFIG_DSP_SWITCH] = { + .offset = 0x1c4, .size = 8, .activate = 22, .pbuf = 1 }, + + [SCARLETT2_CONFIG_COMPRESSOR_PARAMS] = { + .offset = 0x1c8, .size = 32, .activate = 23 }, + + [SCARLETT2_CONFIG_PRECOMP_FLT_SWITCH] = { + .offset = 0x7c, .size = 32, .activate = 27 }, + + [SCARLETT2_CONFIG_PRECOMP_FLT_PARAMS] = { + .offset = 0x200, .size = 32, .activate = 27 }, + + [SCARLETT2_CONFIG_PEQ_FLT_SWITCH] = { + .offset = 0x84, .size = 32, .activate = 27 }, + + [SCARLETT2_CONFIG_PEQ_FLT_PARAMS] = { + .offset = 0x250, .size = 32, .activate = 27 }, + + [SCARLETT2_CONFIG_INPUT_MUTE_SWITCH] = { + .offset = 0x1be, .size = 8, .activate = 17, .pbuf = 1 }, + } +}; + /* Solo Gen 4 */ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = { .notifications = scarlett4_solo_notifications, @@ -1599,6 +1672,90 @@ static const struct scarlett2_device_info s18i20_gen3_info = { } }; +static const struct scarlett2_device_info vocaster_one_info = { + .config_set = &scarlett2_config_set_vocaster, + .min_firmware_version = 1769, + + .phantom_count = 1, + .inputs_per_phantom = 1, + .dsp_count = 1, + .dsp_input_count = 1, + .precomp_flt_count = 2, + .peq_flt_count = 3, + .peq_flt_total_count = 4, + .mute_input_count = 1, + .gain_input_count = 1, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 2, 4 }, + [SCARLETT2_PORT_TYPE_MIX] = { 9, 9 }, + [SCARLETT2_PORT_TYPE_PCM] = { 4, 10 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_MIX, 8, 1 }, + { SCARLETT2_PORT_TYPE_PCM, 5, 5 }, + { SCARLETT2_PORT_TYPE_MIX, 6, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 5 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 6 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 4 }, + { 0, 0, 0 }, + } }, + + .meter_map = { + { 12, 1 }, + { 18, 5 }, + { 10, 2 }, + { 13, 5 }, + { 4, 6 }, + { 0, 4 }, + { 0, 0 } + } +}; + +static const struct scarlett2_device_info vocaster_two_info = { + .config_set = &scarlett2_config_set_vocaster, + .min_firmware_version = 1769, + + .phantom_count = 2, + .inputs_per_phantom = 1, + .dsp_count = 2, + .dsp_input_count = 2, + .precomp_flt_count = 2, + .peq_flt_count = 3, + .peq_flt_total_count = 4, + .mute_input_count = 2, + .gain_input_count = 2, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 6, 6 }, + [SCARLETT2_PORT_TYPE_MIX] = { 12, 14 }, + [SCARLETT2_PORT_TYPE_PCM] = { 4, 14 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_MIX, 12, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 6, 8 }, + { SCARLETT2_PORT_TYPE_MIX, 10, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 6 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 10 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { 0, 0, 0 }, + } }, + + .meter_map = { + { 18, 2 }, + { 26, 8 }, + { 16, 2 }, + { 20, 6 }, + { 6, 10 }, + { 0, 6 }, + { 0, 0 } + } +}; + static const struct scarlett2_device_info solo_gen4_info = { .config_set = &scarlett2_config_set_gen4_solo, .min_firmware_version = 2115, @@ -1932,6 +2089,10 @@ static const struct scarlett2_device_entry scarlett2_devices[] = { { USB_ID(0x1235, 0x8214), &s18i8_gen3_info, "Scarlett Gen 3" }, { USB_ID(0x1235, 0x8215), &s18i20_gen3_info, "Scarlett Gen 3" }, + /* Supported Vocaster devices */ + { USB_ID(0x1235, 0x8216), &vocaster_one_info, "Vocaster" }, + { USB_ID(0x1235, 0x8217), &vocaster_two_info, "Vocaster" }, + /* Supported Gen 4 devices */ { USB_ID(0x1235, 0x8218), &solo_gen4_info, "Scarlett Gen 4" }, { USB_ID(0x1235, 0x8219), &s2i2_gen4_info, "Scarlett Gen 4" }, @@ -7506,8 +7667,7 @@ static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer) } /* Notify on input DSP switch change */ -static __always_unused void scarlett2_notify_input_dsp( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_input_dsp(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; @@ -7522,8 +7682,7 @@ static __always_unused void scarlett2_notify_input_dsp( } /* Notify on input mute switch change */ -static __always_unused void scarlett2_notify_input_mute( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_input_mute(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; From e30ea5340c25d41484d551f61d8c93106989019b Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:07:56 +1030 Subject: [PATCH 320/603] ALSA: scarlett2: Add autogain target controls The Scarlett 4th Gen and Vocaster interfaces allow the autogain target dBFS value(s) to be configured. Add Mean and Peak Target controls for 4th Gen, and a Hot Target control for Vocaster. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: <33d7f6dc965ab09522361ec99745a0685e4b8272.1710264833.git.g@b4.vu> --- sound/usb/mixer_scarlett2.c | 207 ++++++++++++++++++++++++++++++++++++ 1 file changed, 207 insertions(+) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 43922e217503f..6040c4df356f7 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -332,6 +332,17 @@ enum { SCARLETT2_DIM_MUTE_COUNT }; +/* Autogain target values */ + +#define SCARLETT2_AG_TARGET_MIN (-30) + +enum { + SCARLETT2_AG_HOT_TARGET, + SCARLETT2_AG_MEAN_TARGET, + SCARLETT2_AG_PEAK_TARGET, + SCARLETT2_AG_TARGET_COUNT +}; + /* Flash Write State */ enum { SCARLETT2_FLASH_WRITE_STATE_IDLE, @@ -512,6 +523,9 @@ enum { SCARLETT2_CONFIG_TALKBACK_MAP, SCARLETT2_CONFIG_AUTOGAIN_SWITCH, SCARLETT2_CONFIG_AUTOGAIN_STATUS, + SCARLETT2_CONFIG_AG_HOT_TARGET, + SCARLETT2_CONFIG_AG_MEAN_TARGET, + SCARLETT2_CONFIG_AG_PEAK_TARGET, SCARLETT2_CONFIG_INPUT_GAIN, SCARLETT2_CONFIG_SAFE_SWITCH, SCARLETT2_CONFIG_INPUT_SELECT_SWITCH, @@ -523,6 +537,18 @@ enum { SCARLETT2_CONFIG_COUNT }; +/* Autogain target configuration parameters and names */ + +static const int scarlett2_ag_target_configs[] = { + [SCARLETT2_AG_HOT_TARGET] = SCARLETT2_CONFIG_AG_HOT_TARGET, + [SCARLETT2_AG_MEAN_TARGET] = SCARLETT2_CONFIG_AG_MEAN_TARGET, + [SCARLETT2_AG_PEAK_TARGET] = SCARLETT2_CONFIG_AG_PEAK_TARGET +}; + +static const char *const scarlett2_ag_target_names[] = { + "Hot", "Mean", "Peak" +}; + /* Location, size, and activation command number for the configuration * parameters. Size is in bits and may be 1, 8, 16, or 32. * @@ -740,6 +766,9 @@ static const struct scarlett2_config_set scarlett2_config_set_vocaster = { [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = { .offset = 0x1c2, .size = 8, }, + [SCARLETT2_CONFIG_AG_HOT_TARGET] = { + .offset = 0xc1, .size = 8, .activate = 29, .pbuf = 1 }, + [SCARLETT2_CONFIG_INPUT_GAIN] = { .offset = 0x9f, .size = 8, .activate = 21, .pbuf = 1 }, @@ -818,6 +847,12 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = { [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = { .offset = 0x137, .size = 8 }, + [SCARLETT2_CONFIG_AG_MEAN_TARGET] = { + .offset = 0x131, .size = 8, .activate = 29, .pbuf = 1 }, + + [SCARLETT2_CONFIG_AG_PEAK_TARGET] = { + .offset = 0x132, .size = 8, .activate = 30, .pbuf = 1 }, + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { .offset = 0x48, .size = 8, .activate = 11, .pbuf = 1, .mute = 1 }, @@ -862,6 +897,12 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = { [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = { .offset = 0x140, .size = 8 }, + [SCARLETT2_CONFIG_AG_MEAN_TARGET] = { + .offset = 0x13a, .size = 8, .activate = 23, .pbuf = 1 }, + + [SCARLETT2_CONFIG_AG_PEAK_TARGET] = { + .offset = 0x13b, .size = 8, .activate = 24, .pbuf = 1 }, + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { .offset = 0x5a, .size = 8, .activate = 11, .pbuf = 1, .mute = 1 }, @@ -1189,6 +1230,7 @@ struct scarlett2_data { u8 gain[SCARLETT2_INPUT_GAIN_MAX]; u8 autogain_switch[SCARLETT2_INPUT_GAIN_MAX]; u8 autogain_status[SCARLETT2_INPUT_GAIN_MAX]; + s8 ag_targets[SCARLETT2_AG_TARGET_COUNT]; u8 safe_switch[SCARLETT2_INPUT_GAIN_MAX]; u8 pcm_input_switch; u8 direct_monitor_switch; @@ -1217,6 +1259,7 @@ struct scarlett2_data { struct snd_kcontrol *input_gain_ctls[SCARLETT2_INPUT_GAIN_MAX]; struct snd_kcontrol *autogain_ctls[SCARLETT2_INPUT_GAIN_MAX]; struct snd_kcontrol *autogain_status_ctls[SCARLETT2_INPUT_GAIN_MAX]; + struct snd_kcontrol *ag_target_ctls[SCARLETT2_AG_TARGET_COUNT]; struct snd_kcontrol *safe_ctls[SCARLETT2_INPUT_GAIN_MAX]; struct snd_kcontrol *pcm_input_switch_ctl; struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX]; @@ -3253,6 +3296,7 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer) const struct scarlett2_device_info *info = private->info; int err, i; u8 raw_autogain_status[SCARLETT2_INPUT_GAIN_MAX]; + s8 ag_target_values[SCARLETT2_AG_TARGET_COUNT]; private->autogain_updated = 0; @@ -3291,6 +3335,21 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer) private->autogain_status[i] = private->num_autogain_status_texts - 1; + + for (int i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++) + if (scarlett2_has_config_item(private, + scarlett2_ag_target_configs[i])) { + err = scarlett2_usb_get_config( + mixer, scarlett2_ag_target_configs[i], + 1, &ag_target_values[i]); + if (err < 0) + return err; + } + + /* convert from negative dBFS as used by the device */ + for (int i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++) + private->ag_targets[i] = -ag_target_values[i]; + return 0; } @@ -3324,6 +3383,12 @@ static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer) scarlett2_set_ctl_access(private->phantom_ctls[i], val); for (i = 0; i < info->dsp_input_count; i++) scarlett2_set_ctl_access(private->dsp_ctls[i], val); + + for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++) + if (scarlett2_has_config_item(private, + scarlett2_ag_target_configs[i])) + scarlett2_set_ctl_access( + private->ag_target_ctls[i], val); } /* Notify of access mode change for all controls read-only while @@ -3366,6 +3431,12 @@ static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer) for (i = 0; i < info->phantom_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, &private->phantom_ctls[i]->id); + + for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++) + if (scarlett2_has_config_item(private, + scarlett2_ag_target_configs[i])) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->ag_target_ctls[i]->id); } /* Call scarlett2_update_autogain() and @@ -3559,6 +3630,122 @@ static const struct snd_kcontrol_new scarlett2_autogain_status_ctl = { .get = scarlett2_autogain_status_ctl_get, }; +/*** Autogain Target Controls ***/ + +static int scarlett2_ag_target_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_autogain_updated(mixer); + if (err < 0) + goto unlock; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SCARLETT2_AG_TARGET_MIN; + uinfo->value.integer.max = 0; + uinfo->value.integer.step = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_ag_target_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->autogain_updated) { + err = scarlett2_update_autogain(mixer); + if (err < 0) + goto unlock; + } + + ucontrol->value.integer.value[0] = private->ag_targets[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_ag_target_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + + oval = private->ag_targets[index]; + val = clamp(ucontrol->value.integer.value[0], + (long)SCARLETT2_AG_TARGET_MIN, 0L); + + if (oval == val) + goto unlock; + + private->ag_targets[index] = val; + + /* Send new value to the device */ + err = scarlett2_usb_set_config( + mixer, scarlett2_ag_target_configs[index], 1, -val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_ag_target, SCARLETT2_AG_TARGET_MIN * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_ag_target_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_ag_target_ctl_info, + .get = scarlett2_ag_target_ctl_get, + .put = scarlett2_ag_target_ctl_put, + .tlv = { .p = db_scale_ag_target } +}; + /*** Input Select Control ***/ static int scarlett2_update_input_select(struct usb_mixer_interface *mixer) @@ -6693,6 +6880,20 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) i, 1, s, &private->autogain_status_ctls[i]); } + /* Add autogain target controls */ + for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++) + if (scarlett2_has_config_item(private, + scarlett2_ag_target_configs[i])) { + + scnprintf(s, sizeof(s), "Autogain %s Target", + scarlett2_ag_target_names[i]); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_ag_target_ctl, + i, 1, s, &private->ag_target_ctls[i]); + if (err < 0) + return err; + } + /* Add safe-mode input switch controls */ for (i = 0; i < info->safe_input_count; i++) { scnprintf(s, sizeof(s), fmt, i + 1, @@ -7783,6 +7984,12 @@ static void scarlett2_notify_autogain(struct usb_mixer_interface *mixer) &private->autogain_status_ctls[i]->id); } + for (i = 0; i < SCARLETT2_AG_TARGET_COUNT; i++) + if (scarlett2_has_config_item(private, + scarlett2_ag_target_configs[i])) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->ag_target_ctls[i]->id); + scarlett2_autogain_notify_access(mixer); } From 5738cf65e892904d27f91ec974c47669c5f5eab4 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 13 Mar 2024 05:08:10 +1030 Subject: [PATCH 321/603] ALSA: scarlett2: Add Bluetooth volume control for Vocaster Two The Vocaster Two has a Bluetooth module with a volume control. Add a corresponding ALSA mixer control. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Message-ID: --- sound/usb/mixer_scarlett2.c | 156 ++++++++++++++++++++++++++++++++++++ 1 file changed, 156 insertions(+) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 6040c4df356f7..a2de31a0371b0 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -88,6 +88,7 @@ * - input mute, gain, autogain, safe mode * - direct monitor mixes * - compressor and EQ + * - Bluetooth volume * * * /--------------\ 18chn 20chn /--------------\ @@ -180,6 +181,9 @@ */ #define SCARLETT2_MAX_GAIN_VALUE 70 +/* maximum Bluetooth volume value */ +#define SCARLETT2_MAX_BLUETOOTH_VOLUME 30 + /* mixer range from -80dB to +6dB in 0.5dB steps */ #define SCARLETT2_MIXER_MIN_DB -80 #define SCARLETT2_MIXER_BIAS (-SCARLETT2_MIXER_MIN_DB * 2) @@ -421,6 +425,7 @@ static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer); static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer); static void scarlett2_notify_pcm_input_switch( struct usb_mixer_interface *mixer); +static void scarlett2_notify_bluetooth(struct usb_mixer_interface *mixer); /* Arrays of notification callback functions */ @@ -449,6 +454,7 @@ static const struct scarlett2_notification vocaster_notifications[] = { { 0x04000000, scarlett2_notify_input_dsp }, { 0x08000000, scarlett2_notify_input_gain }, { 0x10000000, scarlett2_notify_input_phantom }, + { 0x20000000, scarlett2_notify_bluetooth }, { 0, NULL } }; @@ -534,6 +540,7 @@ enum { SCARLETT2_CONFIG_POWER_LOW, SCARLETT2_CONFIG_PCM_INPUT_SWITCH, SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN, + SCARLETT2_CONFIG_BLUETOOTH_VOLUME, SCARLETT2_CONFIG_COUNT }; @@ -795,6 +802,9 @@ static const struct scarlett2_config_set scarlett2_config_set_vocaster = { [SCARLETT2_CONFIG_INPUT_MUTE_SWITCH] = { .offset = 0x1be, .size = 8, .activate = 17, .pbuf = 1 }, + + [SCARLETT2_CONFIG_BLUETOOTH_VOLUME] = { + .offset = 0xbf, .size = 8, .activate = 28 }, } }; @@ -1134,6 +1144,9 @@ struct scarlett2_device_info { /* the number of DSP channels */ u8 dsp_count; + /* has a Bluetooth module with volume control */ + u8 has_bluetooth; + /* remap analogue outputs; 18i8 Gen 3 has "line 3/4" connected * internally to the analogue 7/8 outputs */ @@ -1206,6 +1219,7 @@ struct scarlett2_data { u8 mix_updated; u8 speaker_switching_switched; u8 power_status_updated; + u8 bluetooth_updated; u8 sync; u8 master_vol; u8 headphone_vol; @@ -1240,6 +1254,7 @@ struct scarlett2_data { u8 msd_switch; u8 standalone_switch; u8 power_status; + u8 bluetooth_volume; u8 meter_level_map[SCARLETT2_MAX_METERS]; struct snd_kcontrol *sync_ctl; struct snd_kcontrol *master_vol_ctl; @@ -1273,6 +1288,7 @@ struct scarlett2_data { struct snd_kcontrol *speaker_switching_ctl; struct snd_kcontrol *talkback_ctl; struct snd_kcontrol *power_status_ctl; + struct snd_kcontrol *bluetooth_volume_ctl; u8 mux[SCARLETT2_MUX_MAX]; u8 mix[SCARLETT2_MIX_MAX]; u8 monitor_mix[SCARLETT2_MONITOR_MIX_MAX]; @@ -1770,6 +1786,7 @@ static const struct scarlett2_device_info vocaster_two_info = { .peq_flt_total_count = 4, .mute_input_count = 2, .gain_input_count = 2, + .has_bluetooth = 1, .port_count = { [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, @@ -7753,6 +7770,121 @@ static int scarlett2_add_power_status_ctl(struct usb_mixer_interface *mixer) &private->power_status_ctl); } +/*** Bluetooth Volume ***/ + +static int scarlett2_update_bluetooth_volume(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err; + + private->bluetooth_updated = 0; + + if (!private->info->has_bluetooth) + return 0; + + err = scarlett2_usb_get_config(mixer, + SCARLETT2_CONFIG_BLUETOOTH_VOLUME, + 1, &private->bluetooth_volume); + if (err < 0) + return err; + + return 0; +} + +static int scarlett2_bluetooth_volume_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->bluetooth_updated) { + err = scarlett2_update_bluetooth_volume(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.integer.value[0] = private->bluetooth_volume; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_bluetooth_volume_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->bluetooth_volume; + val = clamp(ucontrol->value.integer.value[0], + 0L, (long)SCARLETT2_MAX_BLUETOOTH_VOLUME); + + if (oval == val) + goto unlock; + + private->bluetooth_volume = val; + err = scarlett2_usb_set_config(mixer, + SCARLETT2_CONFIG_BLUETOOTH_VOLUME, + 0, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_bluetooth_volume_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT2_MAX_BLUETOOTH_VOLUME; + uinfo->value.integer.step = 1; + return 0; +} + +static const struct snd_kcontrol_new scarlett2_bluetooth_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_bluetooth_volume_ctl_info, + .get = scarlett2_bluetooth_volume_ctl_get, + .put = scarlett2_bluetooth_volume_ctl_put, +}; + +static int scarlett2_add_bluetooth_volume_ctl( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + + if (!private->info->has_bluetooth) + return 0; + + /* Add Bluetooth volume control */ + return scarlett2_add_new_ctl(mixer, &scarlett2_bluetooth_volume_ctl, + 0, 1, "Bluetooth Capture Volume", + &private->bluetooth_volume_ctl); +} + /*** Notification Handlers ***/ /* Notify on sync change */ @@ -8109,6 +8241,21 @@ static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer) scarlett2_notify_mux(mixer); } +/* Notify on Bluetooth change */ +static void scarlett2_notify_bluetooth(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + + if (!private->info->has_bluetooth) + return; + + private->bluetooth_updated = 1; + + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->bluetooth_volume_ctl->id); +} + /* Handle acknowledgement that a command was received; let * scarlett2_usb() know that it can proceed */ @@ -8646,6 +8793,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) return err; } + err = scarlett2_update_bluetooth_volume(mixer); + if (err < 0) + return err; + err = scarlett2_update_mix(mixer); if (err < 0) return err; @@ -8773,6 +8924,11 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; + /* Create the Bluetooth volume control */ + err = scarlett2_add_bluetooth_volume_ctl(mixer); + if (err < 0) + return err; + /* Set the access mode of controls disabled during * autogain/phantom power switching. */ From a1de26c0b74f060ee29f35a6d71765c56c925f5e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 11:38:08 +0300 Subject: [PATCH 322/603] ALSA: hda: Introduce flags to force commands via PIO instead of CORB Add AZX_DCAPS_PIO_COMMANDS quirk (bit 31) and use_pio_for_commands flag to be able to select PIO mode as alternative for CORB based command sending while retaining the RIRB functionality to receive unsolicited responses. This mode differs from the azx single_cmd mode when RIRB is disabled. The mixed mode is needed on Lunar Lake family because it is recommended to use Immediate Command Response (PIO mode) instead of CORB for HDA commands. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Signed-off-by: Takashi Iwai Message-ID: <20240409083812.14001-2-peter.ujfalusi@linux.intel.com> --- include/sound/hdaudio.h | 1 + sound/pci/hda/hda_controller.h | 1 + 2 files changed, 2 insertions(+) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index a73d7f34f4e54..85d2c3d00a271 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -351,6 +351,7 @@ struct hdac_bus { bool needs_damn_long_delay:1; bool not_use_interrupts:1; /* prohibiting the RIRB IRQ */ bool access_sdnctl_in_dword:1; /* accessing the sdnctl register by dword */ + bool use_pio_for_commands:1; /* Use PIO instead of CORB for commands */ int poll_count; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 8556031bcd68e..c2d0109866e62 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -45,6 +45,7 @@ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ +#define AZX_DCAPS_PIO_COMMANDS (1 << 31) /* Use PIO instead of CORB for commands */ enum { AZX_SNOOP_TYPE_NONE, From f81eb6e17d85ca7749781697055becfcff36aac7 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 11:38:09 +0300 Subject: [PATCH 323/603] ALSA: hda: hdac_controller: Implement support for use_pio_for_commands mode In case the use_pio_for_commands flag is set we must not enable the CORB DMA to make sure that it is not interfering with the immediate command mode. Convert the snd_hdac_bus_send_cmd/snd_hdac_bus_get_response as wrappers to call either the PIO or CORB based command handling depending on the use_pio_for_commands flag. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Signed-off-by: Takashi Iwai Message-ID: <20240409083812.14001-3-peter.ujfalusi@linux.intel.com> --- sound/hda/hdac_controller.c | 127 ++++++++++++++++++++++++++++++++++-- 1 file changed, 120 insertions(+), 7 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 7f3a000fab0ce..b5c833b9f8b9c 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -62,7 +62,8 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus) azx_clear_corbrp(bus); /* enable corb dma */ - snd_hdac_chip_writeb(bus, CORBCTL, AZX_CORBCTL_RUN); + if (!bus->use_pio_for_commands) + snd_hdac_chip_writeb(bus, CORBCTL, AZX_CORBCTL_RUN); /* RIRB set up */ bus->rirb.addr = bus->rb.addr + 2048; @@ -135,14 +136,94 @@ static unsigned int azx_command_addr(u32 cmd) return addr; } +/* receive an Immediate Response with PIO */ +static int snd_hdac_bus_wait_for_pio_response(struct hdac_bus *bus, + unsigned int addr) +{ + int timeout = 50; + + while (timeout--) { + /* check IRV bit */ + if (snd_hdac_chip_readw(bus, IRS) & AZX_IRS_VALID) { + /* reuse rirb.res as the response return value */ + bus->rirb.res[addr] = snd_hdac_chip_readl(bus, IR); + return 0; + } + udelay(1); + } + + dev_dbg_ratelimited(bus->dev, "get_response_pio timeout: IRS=%#x\n", + snd_hdac_chip_readw(bus, IRS)); + + bus->rirb.res[addr] = -1; + + return -EIO; +} + /** - * snd_hdac_bus_send_cmd - send a command verb via CORB + * snd_hdac_bus_send_cmd_pio - send a command verb via Immediate Command * @bus: HD-audio core bus * @val: encoded verb value to send * * Returns zero for success or a negative error code. */ -int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val) +static int snd_hdac_bus_send_cmd_pio(struct hdac_bus *bus, unsigned int val) +{ + unsigned int addr = azx_command_addr(val); + int timeout = 50; + int ret = -EIO; + + spin_lock_irq(&bus->reg_lock); + + while (timeout--) { + /* check ICB bit */ + if (!((snd_hdac_chip_readw(bus, IRS) & AZX_IRS_BUSY))) { + /* Clear IRV bit */ + snd_hdac_chip_updatew(bus, IRS, AZX_IRS_VALID, AZX_IRS_VALID); + snd_hdac_chip_writel(bus, IC, val); + /* Set ICB bit */ + snd_hdac_chip_updatew(bus, IRS, AZX_IRS_BUSY, AZX_IRS_BUSY); + + ret = snd_hdac_bus_wait_for_pio_response(bus, addr); + goto out; + } + udelay(1); + } + + dev_dbg_ratelimited(bus->dev, "send_cmd_pio timeout: IRS=%#x, val=%#x\n", + snd_hdac_chip_readw(bus, IRS), val); + +out: + spin_unlock_irq(&bus->reg_lock); + + return ret; +} + +/** + * snd_hdac_bus_get_response_pio - receive a response via Immediate Response + * @bus: HD-audio core bus + * @addr: codec address + * @res: pointer to store the value, NULL when not needed + * + * Returns zero if a value is read, or a negative error code. + */ +static int snd_hdac_bus_get_response_pio(struct hdac_bus *bus, + unsigned int addr, unsigned int *res) +{ + if (res) + *res = bus->rirb.res[addr]; + + return 0; +} + +/** + * snd_hdac_bus_send_cmd_corb - send a command verb via CORB + * @bus: HD-audio core bus + * @val: encoded verb value to send + * + * Returns zero for success or a negative error code. + */ +static int snd_hdac_bus_send_cmd_corb(struct hdac_bus *bus, unsigned int val) { unsigned int addr = azx_command_addr(val); unsigned int wp, rp; @@ -176,7 +257,6 @@ int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val) return 0; } -EXPORT_SYMBOL_GPL(snd_hdac_bus_send_cmd); #define AZX_RIRB_EX_UNSOL_EV (1<<4) @@ -234,15 +314,15 @@ void snd_hdac_bus_update_rirb(struct hdac_bus *bus) EXPORT_SYMBOL_GPL(snd_hdac_bus_update_rirb); /** - * snd_hdac_bus_get_response - receive a response via RIRB + * snd_hdac_bus_get_response_rirb - receive a response via RIRB * @bus: HD-audio core bus * @addr: codec address * @res: pointer to store the value, NULL when not needed * * Returns zero if a value is read, or a negative error code. */ -int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr, - unsigned int *res) +static int snd_hdac_bus_get_response_rirb(struct hdac_bus *bus, + unsigned int addr, unsigned int *res) { unsigned long timeout; unsigned long loopcounter; @@ -293,6 +373,39 @@ int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr, return -EIO; } + +/** + * snd_hdac_bus_send_cmd - send a command verb via CORB or PIO + * @bus: HD-audio core bus + * @val: encoded verb value to send + * + * Returns zero for success or a negative error code. + */ +int snd_hdac_bus_send_cmd(struct hdac_bus *bus, unsigned int val) +{ + if (bus->use_pio_for_commands) + return snd_hdac_bus_send_cmd_pio(bus, val); + + return snd_hdac_bus_send_cmd_corb(bus, val); +} +EXPORT_SYMBOL_GPL(snd_hdac_bus_send_cmd); + +/** + * snd_hdac_bus_get_response - receive a response via RIRB or PIO + * @bus: HD-audio core bus + * @addr: codec address + * @res: pointer to store the value, NULL when not needed + * + * Returns zero if a value is read, or a negative error code. + */ +int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr, + unsigned int *res) +{ + if (bus->use_pio_for_commands) + return snd_hdac_bus_get_response_pio(bus, addr, res); + + return snd_hdac_bus_get_response_rirb(bus, addr, res); +} EXPORT_SYMBOL_GPL(snd_hdac_bus_get_response); #define HDAC_MAX_CAPS 10 From b13593e36ee63d381ed614f28e07cb6c12ecb0bd Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 11:38:10 +0300 Subject: [PATCH 324/603] ALSA: pci: hda: hda_controller: Add support for use_pio_for_commands mode Set the use_pio_for_commands flag in case AZX_DCAPS_PIO_COMMANDS quirk is enabled. When the PIO command mode is used we can re-use the existing azx_single_send_cmd() / azx_single_get_response() functions safely as the CORB DMA is not going to be enabled in snd_hdac_bus_init_cmd_io(). Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Signed-off-by: Takashi Iwai Message-ID: <20240409083812.14001-4-peter.ujfalusi@linux.intel.com> --- sound/pci/hda/hda_controller.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 206306a0eb827..8af5ee1b0ea8d 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -914,7 +914,7 @@ static int azx_send_cmd(struct hdac_bus *bus, unsigned int val) if (chip->disabled) return 0; - if (chip->single_cmd) + if (chip->single_cmd || bus->use_pio_for_commands) return azx_single_send_cmd(bus, val); else return snd_hdac_bus_send_cmd(bus, val); @@ -928,7 +928,7 @@ static int azx_get_response(struct hdac_bus *bus, unsigned int addr, if (chip->disabled) return 0; - if (chip->single_cmd) + if (chip->single_cmd || bus->use_pio_for_commands) return azx_single_get_response(bus, addr, res); else return azx_rirb_get_response(bus, addr, res); @@ -1188,6 +1188,9 @@ int azx_bus_init(struct azx *chip, const char *model) if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) bus->core.align_bdle_4k = true; + if (chip->driver_caps & AZX_DCAPS_PIO_COMMANDS) + bus->core.use_pio_for_commands = true; + /* enable sync_write flag for stable communication as default */ bus->core.sync_write = 1; From f20bee3898c52c13847c4a5667388c56ca059d5f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 11:38:11 +0300 Subject: [PATCH 325/603] ALSA: hda: Intel: Select AZX_DCAPS_PIO_COMMANDS for Lunar Lake It is recommended that on Lunar Lake the PIO (immediate command response) is used instead of CORB/RIRB for commands/verbs. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Signed-off-by: Takashi Iwai Message-ID: <20240409083812.14001-5-peter.ujfalusi@linux.intel.com> --- sound/pci/hda/hda_intel.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b550c42db092..6a3022a87245a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -289,6 +289,9 @@ enum { #define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE +#define AZX_DCAPS_INTEL_LNL \ + (AZX_DCAPS_INTEL_SKYLAKE | AZX_DCAPS_PIO_COMMANDS) + /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB |\ @@ -2503,7 +2506,7 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_RPL_PX, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, { PCI_DEVICE_DATA(INTEL, HDA_MTL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, /* Lunarlake-P */ - { PCI_DEVICE_DATA(INTEL, HDA_LNL_P, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, + { PCI_DEVICE_DATA(INTEL, HDA_LNL_P, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_LNL) }, /* Arrow Lake-S */ { PCI_DEVICE_DATA(INTEL, HDA_ARL_S, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, /* Arrow Lake */ From 05cf17f1bf6d3ffda9a5cba5a2f6175dd155014a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Apr 2024 11:38:12 +0300 Subject: [PATCH 326/603] ASoC: SOF: Intel: hda-bus: Use PIO mode for Lunar Lake It is recommended that on Lunar Lake the PIO (immediate command response) is used instead of CORB/RIRB for commands/verbs. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Liam Girdwood Acked-by: Mark Brown Signed-off-by: Takashi Iwai Message-ID: <20240409083812.14001-6-peter.ujfalusi@linux.intel.com> --- sound/soc/sof/intel/hda-bus.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index fc63085d2d743..f78d6b998be0f 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -72,7 +72,12 @@ void sof_hda_bus_init(struct snd_sof_dev *sdev, struct device *dev) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_LINK) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); + snd_hdac_ext_bus_init(bus, dev, &bus_core_ops, sof_hda_ext_ops); + + if (chip && chip->hw_ip_version == SOF_INTEL_ACE_2_0) + bus->use_pio_for_commands = true; #else snd_hdac_ext_bus_init(bus, dev, NULL, NULL); #endif From ef752c60e41e4e2d7030238e8978e057878a97b1 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 11 Apr 2024 12:08:07 +0100 Subject: [PATCH 327/603] ALSA: hda: cs35l41: Set the max PCM Gain using tuning setting Some systems requires different max PCM Gains settings than the default. The current default value, when running firmware is 17.5 dB, which is used for all systems. Some systems require lower values. Value when running without firmware is 4.5 dB and remains unchanged. Since the gain value is dependent on Tuning and Firmware, it can change, so it cannot be saved in _DSD. Instead we can store it inside a configuration binary file alongside the Firmware and Tuning files. The gain value increments in steps of 1 dB, with value 0 representing 0.5 dB. The max value is 20, which corresponds to 20.5 dB. Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Message-ID: <20240411110813.330483-2-sbinding@opensource.cirrus.com> --- include/sound/cs35l41.h | 5 ++ sound/pci/hda/cs35l41_hda.c | 170 +++++++++++++++++++++++++++++++++--- sound/pci/hda/cs35l41_hda.h | 3 + 3 files changed, 167 insertions(+), 11 deletions(-) diff --git a/include/sound/cs35l41.h b/include/sound/cs35l41.h index 68e053fe73403..bb70782d15d06 100644 --- a/include/sound/cs35l41.h +++ b/include/sound/cs35l41.h @@ -554,6 +554,11 @@ #define CS35L41_LRCLK_FRC_SHIFT 1 #define CS35L41_AMP_GAIN_PCM_MASK 0x3E0 +#define CS35L41_AMP_GAIN_PCM_SHIFT 5 +#define CS35L41_AMP_GAIN_PDM_MASK 0x1F +#define CS35L41_AMP_GAIN_PDM_SHIFT 0 +#define CS35L41_AMP_GAIN_PCM_MAX 20 +#define CS35L41_AMP_GAIN_PDM_MAX 20 #define CS35L41_AMP_GAIN_ZC_MASK 0x0400 #define CS35L41_AMP_GAIN_ZC_SHIFT 10 diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index d3fa6e136744d..12609d6d96c22 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -37,6 +37,32 @@ #define CS35L41_UUID "50d90cdc-3de4-4f18-b528-c7fe3b71f40d" #define CS35L41_DSM_GET_MUTE 5 #define CS35L41_NOTIFY_EVENT 0x91 +#define CS35L41_TUNING_SIG 0x109A4A35 + +enum cs35l41_tuning_param_types { + TUNING_PARAM_GAIN, +}; + +struct cs35l41_tuning_param_hdr { + __le32 tuning_index; + __le32 type; + __le32 size; +} __packed; + +struct cs35l41_tuning_param { + struct cs35l41_tuning_param_hdr hdr; + union { + __le32 gain; + }; +} __packed; + +struct cs35l41_tuning_params { + __le32 signature; + __le32 version; + __le32 size; + __le32 num_entries; + u8 data[]; +} __packed; static bool firmware_autostart = 1; module_param(firmware_autostart, bool, 0444); @@ -92,11 +118,6 @@ static const struct reg_sequence cs35l41_hda_unmute[] = { { CS35L41_AMP_GAIN_CTRL, 0x00000084 }, // AMP_GAIN_PCM 4.5 dB }; -static const struct reg_sequence cs35l41_hda_unmute_dsp[] = { - { CS35L41_AMP_DIG_VOL_CTRL, 0x00008000 }, // AMP_HPF_PCM_EN = 1, AMP_VOL_PCM 0.0 dB - { CS35L41_AMP_GAIN_CTRL, 0x00000233 }, // AMP_GAIN_PCM = 17.5dB AMP_GAIN_PDM = 19.5dB -}; - static const struct reg_sequence cs35l41_hda_mute[] = { { CS35L41_AMP_GAIN_CTRL, 0x00000000 }, // AMP_GAIN_PCM 0.5 dB { CS35L41_AMP_DIG_VOL_CTRL, 0x0000A678 }, // AMP_HPF_PCM_EN = 1, AMP_VOL_PCM Mute @@ -117,6 +138,27 @@ static const struct cs_dsp_client_ops client_ops = { .control_remove = hda_cs_dsp_control_remove, }; +static int cs35l41_request_tuning_param_file(struct cs35l41_hda *cs35l41, char *tuning_filename, + const struct firmware **firmware, char **filename, + const char *ssid) +{ + int ret = 0; + + /* Filename is the same as the tuning file with "cfg" suffix */ + *filename = kasprintf(GFP_KERNEL, "%scfg", tuning_filename); + if (*filename == NULL) + return -ENOMEM; + + ret = firmware_request_nowarn(firmware, *filename, cs35l41->dev); + if (ret != 0) { + dev_dbg(cs35l41->dev, "Failed to request '%s'\n", *filename); + kfree(*filename); + *filename = NULL; + } + + return ret; +} + static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41, const struct firmware **firmware, char **filename, const char *dir, const char *ssid, const char *amp_name, @@ -451,6 +493,94 @@ static int cs35l41_save_calibration(struct cs35l41_hda *cs35l41) } #endif +static void cs35l41_set_default_tuning_params(struct cs35l41_hda *cs35l41) +{ + cs35l41->tuning_gain = DEFAULT_AMP_GAIN_PCM; +} + +static int cs35l41_read_tuning_params(struct cs35l41_hda *cs35l41, const struct firmware *firmware) +{ + struct cs35l41_tuning_params *params; + unsigned int offset = 0; + unsigned int end; + int i; + + params = (void *)&firmware->data[0]; + + if (le32_to_cpu(params->size) != firmware->size) { + dev_err(cs35l41->dev, "Wrong Size for Tuning Param file. Expected %d got %zu\n", + le32_to_cpu(params->size), firmware->size); + return -EINVAL; + } + + if (le32_to_cpu(params->version) != 1) { + dev_err(cs35l41->dev, "Unsupported Tuning Param Version: %d\n", + le32_to_cpu(params->version)); + return -EINVAL; + } + + if (le32_to_cpu(params->signature) != CS35L41_TUNING_SIG) { + dev_err(cs35l41->dev, + "Mismatched Signature for Tuning Param file. Expected %#x got %#x\n", + CS35L41_TUNING_SIG, le32_to_cpu(params->signature)); + return -EINVAL; + } + + end = firmware->size - sizeof(struct cs35l41_tuning_params); + + for (i = 0; i < le32_to_cpu(params->num_entries); i++) { + struct cs35l41_tuning_param *param; + + if ((offset >= end) || ((offset + sizeof(struct cs35l41_tuning_param_hdr)) >= end)) + return -EFAULT; + + param = (void *)¶ms->data[offset]; + offset += le32_to_cpu(param->hdr.size); + + if (offset > end) + return -EFAULT; + + switch (le32_to_cpu(param->hdr.type)) { + case TUNING_PARAM_GAIN: + cs35l41->tuning_gain = le32_to_cpu(param->gain); + dev_dbg(cs35l41->dev, "Applying Gain: %d\n", cs35l41->tuning_gain); + break; + default: + break; + } + } + + return 0; +} + +static int cs35l41_load_tuning_params(struct cs35l41_hda *cs35l41, char *tuning_filename) +{ + const struct firmware *tuning_param_file = NULL; + char *tuning_param_filename = NULL; + int ret; + + ret = cs35l41_request_tuning_param_file(cs35l41, tuning_filename, &tuning_param_file, + &tuning_param_filename, cs35l41->acpi_subsystem_id); + if (ret) { + dev_dbg(cs35l41->dev, "Missing Tuning Param for file: %s: %d\n", tuning_filename, + ret); + return 0; + } + + ret = cs35l41_read_tuning_params(cs35l41, tuning_param_file); + if (ret) { + dev_err(cs35l41->dev, "Error reading Tuning Params from file: %s: %d\n", + tuning_param_filename, ret); + /* Reset to default Tuning Parameters */ + cs35l41_set_default_tuning_params(cs35l41); + } + + release_firmware(tuning_param_file); + kfree(tuning_param_filename); + + return ret; +} + static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) { const struct firmware *coeff_firmware = NULL; @@ -470,27 +600,35 @@ static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) cs35l41->halo_initialized = true; } + cs35l41_set_default_tuning_params(cs35l41); + ret = cs35l41_request_firmware_files(cs35l41, &wmfw_firmware, &wmfw_filename, &coeff_firmware, &coeff_filename); if (ret < 0) return ret; dev_dbg(cs35l41->dev, "Loading WMFW Firmware: %s\n", wmfw_filename); - if (coeff_filename) + if (coeff_filename) { dev_dbg(cs35l41->dev, "Loading Coefficient File: %s\n", coeff_filename); - else + ret = cs35l41_load_tuning_params(cs35l41, coeff_filename); + if (ret) + dev_warn(cs35l41->dev, "Unable to load Tuning Parameters: %d\n", ret); + } else { dev_warn(cs35l41->dev, "No Coefficient File available.\n"); + } ret = cs_dsp_power_up(dsp, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename, hda_cs_dsp_fw_ids[cs35l41->firmware_type]); if (ret) - goto err_release; + goto err; cs35l41_add_controls(cs35l41); ret = cs35l41_save_calibration(cs35l41); -err_release: +err: + if (ret) + cs35l41_set_default_tuning_params(cs35l41); release_firmware(wmfw_firmware); release_firmware(coeff_firmware); kfree(wmfw_filename); @@ -503,6 +641,7 @@ static void cs35l41_shutdown_dsp(struct cs35l41_hda *cs35l41) { struct cs_dsp *dsp = &cs35l41->cs_dsp; + cs35l41_set_default_tuning_params(cs35l41); cs_dsp_stop(dsp); cs_dsp_power_down(dsp); dev_dbg(cs35l41->dev, "Unloaded Firmware\n"); @@ -570,6 +709,7 @@ static void cs35l41_mute(struct device *dev, bool mute) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); struct regmap *reg = cs35l41->regmap; + unsigned int amp_gain; dev_dbg(dev, "Mute(%d:%d) Playback Started: %d\n", mute, cs35l41->mute_override, cs35l41->playback_started); @@ -581,8 +721,13 @@ static void cs35l41_mute(struct device *dev, bool mute) } else { dev_dbg(dev, "Unmuting\n"); if (cs35l41->cs_dsp.running) { - regmap_multi_reg_write(reg, cs35l41_hda_unmute_dsp, - ARRAY_SIZE(cs35l41_hda_unmute_dsp)); + dev_dbg(dev, "Using Tuned Gain: %d\n", cs35l41->tuning_gain); + amp_gain = (cs35l41->tuning_gain << CS35L41_AMP_GAIN_PCM_SHIFT) | + (DEFAULT_AMP_GAIN_PDM << CS35L41_AMP_GAIN_PDM_SHIFT); + + /* AMP_HPF_PCM_EN = 1, AMP_VOL_PCM 0.0 dB */ + regmap_write(reg, CS35L41_AMP_DIG_VOL_CTRL, 0x00008000); + regmap_write(reg, CS35L41_AMP_GAIN_CTRL, amp_gain); } else { regmap_multi_reg_write(reg, cs35l41_hda_unmute, ARRAY_SIZE(cs35l41_hda_unmute)); @@ -1056,6 +1201,9 @@ static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) goto clean_dsp; } + dev_info(cs35l41->dev, "Firmware Loaded - Type: %s, Gain: %d\n", + hda_cs_dsp_fw_ids[cs35l41->firmware_type], cs35l41->tuning_gain); + return 0; clean_dsp: diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index 43d55292b327a..d60aa98bfafca 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -21,6 +21,8 @@ #include #define CS35L41_MAX_ACCEPTABLE_SPI_SPEED_HZ 1000000 +#define DEFAULT_AMP_GAIN_PCM 17 /* 17.5dB Gain */ +#define DEFAULT_AMP_GAIN_PDM 19 /* 19.5dB Gain */ struct cs35l41_amp_cal_data { u32 calTarget[2]; @@ -83,6 +85,7 @@ struct cs35l41_hda { bool mute_override; enum control_bus control_bus; bool bypass_fw; + unsigned int tuning_gain; }; From ce35d1bd9140f987d9c842c8ac6d9993c95e812c Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 11 Apr 2024 12:08:08 +0100 Subject: [PATCH 328/603] ALSA: hda: cs35l41: Support HP Omen models without _DSD Add support for 2 new HP Omen models without _DSD into configuration table. These laptops use the PCM Gain setting for the tuning setting file. Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Message-ID: <20240411110813.330483-3-sbinding@opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda_property.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 8fb688e414148..efa62e99d3301 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -70,6 +70,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "103C8C15", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 }, { "103C8C16", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 }, { "103C8C17", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4000, 24 }, + { "103C8C4D", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8C4E", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8C4F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8C50", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8C51", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, @@ -457,6 +459,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "103C8C15", generic_dsd_config }, { "CSC3551", "103C8C16", generic_dsd_config }, { "CSC3551", "103C8C17", generic_dsd_config }, + { "CSC3551", "103C8C4D", generic_dsd_config }, + { "CSC3551", "103C8C4E", generic_dsd_config }, { "CSC3551", "103C8C4F", generic_dsd_config }, { "CSC3551", "103C8C50", generic_dsd_config }, { "CSC3551", "103C8C51", generic_dsd_config }, From 875e0cd59758a3d636ce94936287787514305095 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 11 Apr 2024 12:08:09 +0100 Subject: [PATCH 329/603] ALSA: hda/realtek: Add quirks for HP Omen models using CS35L41 Add 4 laptops using CS35L41 HDA. None of these laptops have _DSD, so require entries in property configuration table for cs35l41_hda driver. Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Message-ID: <20240411110813.330483-4-sbinding@opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d6940bc4ec393..a1c0381583934 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10040,6 +10040,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8bb3, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8bb4, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bdd, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bde, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bdf, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), @@ -10060,6 +10062,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c48, "HP EliteBook 860 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c49, "HP Elite x360 830 2-in-1 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c4d, "HP Omen", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8c4e, "HP Omen", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c4f, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c50, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c51, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), From b627054837174518a9856ccdbda55ce6c7184f2d Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 11 Apr 2024 12:08:10 +0100 Subject: [PATCH 330/603] ALSA: hda: cs35l41: Update DSP1RX5/6 Sources for DSP config Currently, all PC systems are set to use VBSTMON for DSP1RX5_SRC, however, this is required only for external boost systems. Internal boost systems require VPMON instead of VBSTMON to be the input to DSP1RX5_SRC. All systems require DSP1RX6_SRC to be set to VBSTMON. Also fix incorrect comment for DACPCM1_SRC to use DSP1TX1. Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Message-ID: <20240411110813.330483-5-sbinding@opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 12609d6d96c22..17e812c60ac9b 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -101,7 +101,7 @@ static const struct reg_sequence cs35l41_hda_config_dsp[] = { { CS35L41_SP_HIZ_CTRL, 0x00000003 }, // Hi-Z unused/disabled { CS35L41_SP_TX_WL, 0x00000018 }, // 24 cycles/slot { CS35L41_SP_RX_WL, 0x00000018 }, // 24 cycles/slot - { CS35L41_DAC_PCM1_SRC, 0x00000032 }, // DACPCM1_SRC = ERR_VOL + { CS35L41_DAC_PCM1_SRC, 0x00000032 }, // DACPCM1_SRC = DSP1TX1 { CS35L41_ASP_TX1_SRC, 0x00000018 }, // ASPTX1 SRC = VMON { CS35L41_ASP_TX2_SRC, 0x00000019 }, // ASPTX2 SRC = IMON { CS35L41_ASP_TX3_SRC, 0x00000028 }, // ASPTX3 SRC = VPMON @@ -110,7 +110,7 @@ static const struct reg_sequence cs35l41_hda_config_dsp[] = { { CS35L41_DSP1_RX2_SRC, 0x00000008 }, // DSP1RX2 SRC = ASPRX1 { CS35L41_DSP1_RX3_SRC, 0x00000018 }, // DSP1RX3 SRC = VMON { CS35L41_DSP1_RX4_SRC, 0x00000019 }, // DSP1RX4 SRC = IMON - { CS35L41_DSP1_RX5_SRC, 0x00000029 }, // DSP1RX5 SRC = VBSTMON + { CS35L41_DSP1_RX6_SRC, 0x00000029 }, // DSP1RX6 SRC = VBSTMON }; static const struct reg_sequence cs35l41_hda_unmute[] = { @@ -692,6 +692,10 @@ static void cs35l41_hda_play_start(struct device *dev) if (cs35l41->cs_dsp.running) { regmap_multi_reg_write(reg, cs35l41_hda_config_dsp, ARRAY_SIZE(cs35l41_hda_config_dsp)); + if (cs35l41->hw_cfg.bst_type == CS35L41_INT_BOOST) + regmap_write(reg, CS35L41_DSP1_RX5_SRC, CS35L41_INPUT_SRC_VPMON); + else + regmap_write(reg, CS35L41_DSP1_RX5_SRC, CS35L41_INPUT_SRC_VBSTMON); regmap_update_bits(reg, CS35L41_PWR_CTRL2, CS35L41_VMON_EN_MASK | CS35L41_IMON_EN_MASK, 1 << CS35L41_VMON_EN_SHIFT | 1 << CS35L41_IMON_EN_SHIFT); From 068fc7f326c68dd80c428d2fb3eaebf36cdf2115 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 11 Apr 2024 12:08:11 +0100 Subject: [PATCH 331/603] ALSA: hda: cs35l41: Use shared cs-amp-lib to apply calibration The original mechanism for applying calibration assumed that the calibration data would be ordered the same as the amp instances. However, for some 4 amp laptops, this is not the case. To ensure that the correct calibration is applied to the correct amp, the calibration data contains a unique id, which matches a unique id inside the CS35L41. This can be used to match to the correct data entry. This mechanism is available inside the shared module cs-amp-lib. Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Message-ID: <20240411110813.330483-6-sbinding@opensource.cirrus.com> --- sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l41_hda.c | 140 +++++++++++++++++------------------- sound/pci/hda/cs35l41_hda.h | 3 + 3 files changed, 72 insertions(+), 73 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index f806636242ee9..0da625533afc2 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -128,6 +128,7 @@ config SND_HDA_SCODEC_CS35L41_I2C select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 select SND_HDA_CS_DSP_CONTROLS + select SND_SOC_CS_AMP_LIB help Say Y or M here to include CS35L41 I2C HD-audio side codec support in snd-hda-intel driver, such as ALC287. @@ -144,6 +145,7 @@ config SND_HDA_SCODEC_CS35L41_SPI select SND_SOC_CS35L41_LIB select SND_HDA_SCODEC_CS35L41 select SND_HDA_CS_DSP_CONTROLS + select SND_SOC_CS_AMP_LIB help Say Y or M here to include CS35L41 SPI HD-audio side codec support in snd-hda-intel driver, such as ALC287. diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 17e812c60ac9b..5208e9c06af0b 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -64,6 +64,16 @@ struct cs35l41_tuning_params { u8 data[]; } __packed; +/* Firmware calibration controls */ +static const struct cirrus_amp_cal_controls cs35l41_calibration_controls = { + .alg_id = CAL_DSP_CTL_ALG, + .mem_region = CAL_DSP_CTL_TYPE, + .ambient = CAL_AMBIENT_DSP_CTL_NAME, + .calr = CAL_R_DSP_CTL_NAME, + .status = CAL_STATUS_DSP_CTL_NAME, + .checksum = CAL_CHECKSUM_DSP_CTL_NAME, +}; + static bool firmware_autostart = 1; module_param(firmware_autostart, bool, 0444); MODULE_PARM_DESC(firmware_autostart, "Allow automatic firmware download on boot" @@ -403,95 +413,74 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, coeff_firmware, coeff_filename); } -#if IS_ENABLED(CONFIG_EFI) -static int cs35l41_apply_calibration(struct cs35l41_hda *cs35l41, __be32 ambient, __be32 r0, - __be32 status, __be32 checksum) + +static void cs35l41_hda_apply_calibration(struct cs35l41_hda *cs35l41) { int ret; - ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_AMBIENT_DSP_CTL_NAME, CAL_DSP_CTL_TYPE, - CAL_DSP_CTL_ALG, &ambient, 4); - if (ret) { - dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_AMBIENT_DSP_CTL_NAME, - ret); - return ret; - } - ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_R_DSP_CTL_NAME, CAL_DSP_CTL_TYPE, - CAL_DSP_CTL_ALG, &r0, 4); - if (ret) { - dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_R_DSP_CTL_NAME, ret); - return ret; - } - ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_STATUS_DSP_CTL_NAME, CAL_DSP_CTL_TYPE, - CAL_DSP_CTL_ALG, &status, 4); + if (!cs35l41->cal_data_valid) + return; + + ret = cs_amp_write_cal_coeffs(&cs35l41->cs_dsp, &cs35l41_calibration_controls, + &cs35l41->cal_data); + if (ret < 0) + dev_warn(cs35l41->dev, "Failed to apply calibration: %d\n", ret); + else + dev_info(cs35l41->dev, "Calibration applied: R0=%d\n", cs35l41->cal_data.calR); +} + +static int cs35l41_read_silicon_uid(struct cs35l41_hda *cs35l41, u64 *uid) +{ + u32 tmp; + int ret; + + ret = regmap_read(cs35l41->regmap, CS35L41_DIE_STS2, &tmp); if (ret) { - dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_STATUS_DSP_CTL_NAME, - ret); + dev_err(cs35l41->dev, "Cannot obtain CS35L41_DIE_STS2: %d\n", ret); return ret; } - ret = hda_cs_dsp_write_ctl(&cs35l41->cs_dsp, CAL_CHECKSUM_DSP_CTL_NAME, CAL_DSP_CTL_TYPE, - CAL_DSP_CTL_ALG, &checksum, 4); + + *uid = tmp; + *uid <<= 32; + + ret = regmap_read(cs35l41->regmap, CS35L41_DIE_STS1, &tmp); if (ret) { - dev_err(cs35l41->dev, "Cannot Write Control: %s - %d\n", CAL_CHECKSUM_DSP_CTL_NAME, - ret); + dev_err(cs35l41->dev, "Cannot obtain CS35L41_DIE_STS1: %d\n", ret); return ret; } + *uid |= tmp; + + dev_dbg(cs35l41->dev, "UniqueID = %#llx\n", *uid); + return 0; } -static int cs35l41_save_calibration(struct cs35l41_hda *cs35l41) +static int cs35l41_get_calibration(struct cs35l41_hda *cs35l41) { - static efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, 0x93, 0xfe, - 0x5a, 0xa3, 0x5d, 0xb3); - static efi_char16_t efi_name[] = L"CirrusSmartAmpCalibrationData"; - const struct cs35l41_amp_efi_data *efi_data; - const struct cs35l41_amp_cal_data *cl; - unsigned long data_size = 0; - efi_status_t status; - int ret = 0; - u8 *data = NULL; - u32 attr; + u64 silicon_uid; + int ret; + + ret = cs35l41_read_silicon_uid(cs35l41, &silicon_uid); + if (ret < 0) + return ret; + + ret = cs_amp_get_efi_calibration_data(cs35l41->dev, silicon_uid, + cs35l41->index, + &cs35l41->cal_data); + + /* Only return an error status if probe should be aborted */ + if ((ret == -ENOENT) || (ret == -EOVERFLOW)) + return 0; + + if (ret < 0) + return ret; + + cs35l41->cal_data_valid = true; - /* Get real size of UEFI variable */ - status = efi.get_variable(efi_name, &efi_guid, &attr, &data_size, data); - if (status == EFI_BUFFER_TOO_SMALL) { - ret = -ENODEV; - /* Allocate data buffer of data_size bytes */ - data = vmalloc(data_size); - if (!data) - return -ENOMEM; - /* Get variable contents into buffer */ - status = efi.get_variable(efi_name, &efi_guid, &attr, &data_size, data); - if (status == EFI_SUCCESS) { - efi_data = (struct cs35l41_amp_efi_data *)data; - dev_dbg(cs35l41->dev, "Calibration: Size=%d, Amp Count=%d\n", - efi_data->size, efi_data->count); - if (efi_data->count > cs35l41->index) { - cl = &efi_data->data[cs35l41->index]; - dev_dbg(cs35l41->dev, - "Calibration: Ambient=%02x, Status=%02x, R0=%d\n", - cl->calAmbient, cl->calStatus, cl->calR); - - /* Calibration can only be applied whilst the DSP is not running */ - ret = cs35l41_apply_calibration(cs35l41, - cpu_to_be32(cl->calAmbient), - cpu_to_be32(cl->calR), - cpu_to_be32(cl->calStatus), - cpu_to_be32(cl->calR + 1)); - } - } - vfree(data); - } - return ret; -} -#else -static int cs35l41_save_calibration(struct cs35l41_hda *cs35l41) -{ - dev_warn(cs35l41->dev, "Calibration not supported without EFI support.\n"); return 0; } -#endif + static void cs35l41_set_default_tuning_params(struct cs35l41_hda *cs35l41) { @@ -624,7 +613,7 @@ static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) cs35l41_add_controls(cs35l41); - ret = cs35l41_save_calibration(cs35l41); + cs35l41_hda_apply_calibration(cs35l41); err: if (ret) @@ -1960,6 +1949,10 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i if (ret) goto err; + ret = cs35l41_get_calibration(cs35l41); + if (ret && ret != -ENOENT) + goto err; + cs35l41_mute(cs35l41->dev, true); INIT_WORK(&cs35l41->fw_load_work, cs35l41_fw_load_work); @@ -2040,6 +2033,7 @@ EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41); MODULE_DESCRIPTION("CS35L41 HDA Driver"); MODULE_IMPORT_NS(SND_HDA_CS_DSP_CONTROLS); +MODULE_IMPORT_NS(SND_SOC_CS_AMP_LIB); MODULE_AUTHOR("Lucas Tanure, Cirrus Logic Inc, "); MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(FW_CS_DSP); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index d60aa98bfafca..b0bebb7784623 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -16,6 +16,7 @@ #include #include #include +#include #include #include @@ -86,6 +87,8 @@ struct cs35l41_hda { enum control_bus control_bus; bool bypass_fw; unsigned int tuning_gain; + struct cirrus_amp_cal_data cal_data; + bool cal_data_valid; }; From 89015f962ffb5effb5bba0c6f5b2cc75d343ae32 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 11 Apr 2024 12:08:12 +0100 Subject: [PATCH 332/603] ALSA: hda: cs35l41: Remove redundant argument to cs35l41_request_firmware_file() In every case the 'dir' argument to cs35l41_request_firmware_file() is passed the string "cirrus/", so this is a redundant argument and can be removed. Signed-off-by: Richard Fitzgerald Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Message-ID: <20240411110813.330483-7-sbinding@opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 39 ++++++++++++++----------------------- 1 file changed, 15 insertions(+), 24 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 5208e9c06af0b..7fde8a7a923bc 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -22,7 +22,6 @@ #include "hda_cs_dsp_ctl.h" #include "cs35l41_hda_property.h" -#define CS35L41_FIRMWARE_ROOT "cirrus/" #define CS35L41_PART "cs35l41" #define HALO_STATE_DSP_CTL_NAME "HALO_STATE" @@ -171,7 +170,7 @@ static int cs35l41_request_tuning_param_file(struct cs35l41_hda *cs35l41, char * static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41, const struct firmware **firmware, char **filename, - const char *dir, const char *ssid, const char *amp_name, + const char *ssid, const char *amp_name, int spkid, const char *filetype) { const char * const dsp_name = cs35l41->cs_dsp.name; @@ -179,23 +178,23 @@ static int cs35l41_request_firmware_file(struct cs35l41_hda *cs35l41, int ret = 0; if (spkid > -1 && ssid && amp_name) - *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-spkid%d-%s.%s", dir, CS35L41_PART, + *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s-%s-spkid%d-%s.%s", CS35L41_PART, dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], ssid, spkid, amp_name, filetype); else if (spkid > -1 && ssid) - *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-spkid%d.%s", dir, CS35L41_PART, + *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s-%s-spkid%d.%s", CS35L41_PART, dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], ssid, spkid, filetype); else if (ssid && amp_name) - *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-%s.%s", dir, CS35L41_PART, + *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s-%s-%s.%s", CS35L41_PART, dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], ssid, amp_name, filetype); else if (ssid) - *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s.%s", dir, CS35L41_PART, + *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s-%s.%s", CS35L41_PART, dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], ssid, filetype); else - *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, CS35L41_PART, + *filename = kasprintf(GFP_KERNEL, "cirrus/%s-%s-%s.%s", CS35L41_PART, dsp_name, hda_cs_dsp_fw_ids[cs35l41->firmware_type], filetype); @@ -236,13 +235,11 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, cs35l41->amp_name, cs35l41->speaker_id, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, cs35l41->amp_name, cs35l41->speaker_id, "bin"); if (ret) @@ -253,12 +250,11 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, /* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->acpi_subsystem_id, cs35l41->amp_name, -1, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, cs35l41->amp_name, cs35l41->speaker_id, "bin"); if (ret) @@ -269,18 +265,17 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, /* try cirrus/part-dspN-fwtype-sub<-spkidN>.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->acpi_subsystem_id, NULL, cs35l41->speaker_id, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, cs35l41->amp_name, cs35l41->speaker_id, "bin"); if (ret) /* try cirrus/part-dspN-fwtype-sub<-spkidN>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, - coeff_filename, CS35L41_FIRMWARE_ROOT, + coeff_filename, cs35l41->acpi_subsystem_id, NULL, cs35l41->speaker_id, "bin"); if (ret) @@ -291,18 +286,17 @@ static int cs35l41_request_firmware_files_spkid(struct cs35l41_hda *cs35l41, /* try cirrus/part-dspN-fwtype-sub.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->acpi_subsystem_id, NULL, -1, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-spkidN><-ampname>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, cs35l41->amp_name, cs35l41->speaker_id, "bin"); if (ret) /* try cirrus/part-dspN-fwtype-sub<-spkidN>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, - coeff_filename, CS35L41_FIRMWARE_ROOT, + coeff_filename, cs35l41->acpi_subsystem_id, NULL, cs35l41->speaker_id, "bin"); if (ret) @@ -329,13 +323,13 @@ static int cs35l41_fallback_firmware_file(struct cs35l41_hda *cs35l41, /* fallback try cirrus/part-dspN-fwtype.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "wmfw"); + NULL, NULL, -1, "wmfw"); if (ret) goto err; /* fallback try cirrus/part-dspN-fwtype.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, NULL, NULL, -1, "bin"); + NULL, NULL, -1, "bin"); if (ret) { release_firmware(*wmfw_firmware); kfree(*wmfw_filename); @@ -364,12 +358,11 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, /* try cirrus/part-dspN-fwtype-sub<-ampname>.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->acpi_subsystem_id, cs35l41->amp_name, -1, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-ampname>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, cs35l41->amp_name, -1, "bin"); if (ret) @@ -380,18 +373,16 @@ static int cs35l41_request_firmware_files(struct cs35l41_hda *cs35l41, /* try cirrus/part-dspN-fwtype-sub.wmfw */ ret = cs35l41_request_firmware_file(cs35l41, wmfw_firmware, wmfw_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, + cs35l41->acpi_subsystem_id, NULL, -1, "wmfw"); if (!ret) { /* try cirrus/part-dspN-fwtype-sub<-ampname>.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, cs35l41->amp_name, -1, "bin"); if (ret) /* try cirrus/part-dspN-fwtype-sub.bin */ ret = cs35l41_request_firmware_file(cs35l41, coeff_firmware, coeff_filename, - CS35L41_FIRMWARE_ROOT, cs35l41->acpi_subsystem_id, NULL, -1, "bin"); if (ret) From 4a1a8065f5d3565677347d34a908ff2d0803b14f Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 11 Apr 2024 12:08:13 +0100 Subject: [PATCH 333/603] ALSA: hda: cs35l41: Remove Speaker ID for Lenovo Legion slim 7 16ARHA7 These laptops do not have _DSD and must be added by configuration table, however, the initial entries for them are incorrect: Neither laptop contains a Speaker ID GPIO. This issue would not affect audio playback, but may affect which files are loaded when loading firmware. Fixes: b67a7dc418aa ("ALSA: hda/realtek: Add sound quirks for Lenovo Legion slim 7 16ARHA7 models") Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Message-ID: <20240411110813.330483-8-sbinding@opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda_property.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index efa62e99d3301..6f82b28e26dde 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -112,8 +112,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, - { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, - { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA38A9", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38AB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, From 93d7d38fa61d1f22ab5a882d89d94c077ec8bb01 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Tue, 16 Apr 2024 14:17:24 +0200 Subject: [PATCH 334/603] ALSA: aloop: add DSD formats The snd-aloop loopback driver does not modify or access the actual samples in any way, defines no volume or mute controls, it's strictly bitperfect. Therefore DSD formats can be supported without any modification. Add all DSD formats to the list of supported formats. Signed-off-by: Pavel Hofman Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Message-ID: <20240416121726.628679-2-pavel.hofman@ivitera.com> --- sound/drivers/aloop.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 892c4e29c0a34..44ee2a4bc2b04 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -927,7 +927,10 @@ static const struct snd_pcm_hardware loopback_pcm_hardware = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | - SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE), + SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE | + SNDRV_PCM_FMTBIT_DSD_U8 | + SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U16_BE | + SNDRV_PCM_FMTBIT_DSD_U32_LE | SNDRV_PCM_FMTBIT_DSD_U32_BE), .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, .rate_min = 8000, .rate_max = 192000, From ad88ea67b135f74a9f32dc5404a35c773b2a3925 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Tue, 16 Apr 2024 14:17:25 +0200 Subject: [PATCH 335/603] ALSA: pcm: add support for 705.6kHz and 768kHz sample rates Many modern codecs support 705.6kHz and 768kHz sample rates. Current HW params fail to set 705.6kHz and 768kHz sample rates as these are not in the known-rates list. Add these new rates to the known-rates list to allow them. Also add defines in pcm.h so that drivers can use it. Signed-off-by: Pavel Hofman Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Message-ID: <20240416121726.628679-3-pavel.hofman@ivitera.com> --- include/sound/pcm.h | 5 +++++ sound/core/pcm_native.c | 2 +- 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 210096f124eed..61c6054618c8e 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -120,6 +120,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_192000 (1U<<12) /* 192000Hz */ #define SNDRV_PCM_RATE_352800 (1U<<13) /* 352800Hz */ #define SNDRV_PCM_RATE_384000 (1U<<14) /* 384000Hz */ +#define SNDRV_PCM_RATE_705600 (1U<<15) /* 705600Hz */ +#define SNDRV_PCM_RATE_768000 (1U<<16) /* 768000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1U<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1U<<31) /* supports more non-continuos rates */ @@ -135,6 +137,9 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000|\ SNDRV_PCM_RATE_352800|\ SNDRV_PCM_RATE_384000) +#define SNDRV_PCM_RATE_8000_768000 (SNDRV_PCM_RATE_8000_384000|\ + SNDRV_PCM_RATE_705600|\ + SNDRV_PCM_RATE_768000) #define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt) #define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8) #define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0b76e76823d28..521ba56392a07 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2416,7 +2416,7 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, static const unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000 + 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000, 705600, 768000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { From 3e2f2235b526e0238d4c77fe3396bc6308c5426b Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Tue, 16 Apr 2024 14:17:26 +0200 Subject: [PATCH 336/603] ALSA: aloop: add support for up to 768kHz sample rate Many modern codecs support rates up to 768kHz (including DSD1024). Add support for rates up to 768kHz to the loopback driver. Signed-off-by: Pavel Hofman Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Message-ID: <20240416121726.628679-4-pavel.hofman@ivitera.com> --- sound/drivers/aloop.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 44ee2a4bc2b04..d6dd4b8c750ad 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -931,9 +931,9 @@ static const struct snd_pcm_hardware loopback_pcm_hardware = SNDRV_PCM_FMTBIT_DSD_U8 | SNDRV_PCM_FMTBIT_DSD_U16_LE | SNDRV_PCM_FMTBIT_DSD_U16_BE | SNDRV_PCM_FMTBIT_DSD_U32_LE | SNDRV_PCM_FMTBIT_DSD_U32_BE), - .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_768000, .rate_min = 8000, - .rate_max = 192000, + .rate_max = 768000, .channels_min = 1, .channels_max = 32, .buffer_bytes_max = 2 * 1024 * 1024, From ed37d240d03e84d09d4d2a771fda419da4308d17 Mon Sep 17 00:00:00 2001 From: Mohammad Shehar Yaar Tausif Date: Thu, 18 Apr 2024 22:03:24 +0530 Subject: [PATCH 337/603] ASoC: dt-bindings: tegra20-das: Convert to schema Convert NVIDIA Tegra20 DAS (Digital Audio Switch) binding to schema. Signed-off-by: Mohammad Shehar Yaar Tausif Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240418163326.58365-1-sheharyaar48@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/nvidia,tegra20-das.txt | 12 ------- .../bindings/sound/nvidia,tegra20-das.yaml | 36 +++++++++++++++++++ 2 files changed, 36 insertions(+), 12 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt deleted file mode 100644 index 6de3a7ee4efb1..0000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.txt +++ /dev/null @@ -1,12 +0,0 @@ -NVIDIA Tegra 20 DAS (Digital Audio Switch) controller - -Required properties: -- compatible : "nvidia,tegra20-das" -- reg : Should contain DAS registers location and length - -Example: - -das@70000c00 { - compatible = "nvidia,tegra20-das"; - reg = <0x70000c00 0x80>; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml new file mode 100644 index 0000000000000..44c5ce8ee6be6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-das.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-das.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra 20 DAS (Digital Audio Switch) controller + +maintainers: + - Thierry Reding + - Jon Hunter + +properties: + compatible: + const: nvidia,tegra20-das + + reg: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + bus { + #address-cells = <1>; + #size-cells = <1>; + das@70000c00 { + compatible = "nvidia,tegra20-das"; + reg = <0x70000c00 0x80>; + }; + }; +... From 7ce6ceeb77bfd9fb0b22203190bd6f57fe917b51 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 19 Apr 2024 10:48:56 +0200 Subject: [PATCH 338/603] ASoC: Intel: avs: New IRQ handling implementation The existing code can be both improved and simplified. To make this change easier to manage, first add new implementation and then remove deadcode in a separate patch. Simplification achieved with: - reduce the amount of resources requested by the driver i.e.: IPC and CLDMA request_irq() merged into one - reduce the number of DSP ops from 2 to 1: irq_handler/thread() vs dsp_interrupt() - drop ambiguity around CLDMA interrupt, let skl.c handle that explicitly as it is the only user Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240419084857.2719593-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/apl.c | 18 +++++++++ sound/soc/intel/avs/avs.h | 3 ++ sound/soc/intel/avs/cldma.c | 11 +++++ sound/soc/intel/avs/cldma.h | 1 + sound/soc/intel/avs/cnl.c | 63 +++++++++++++++++++++++++++++ sound/soc/intel/avs/core.c | 80 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/avs/icl.c | 1 + sound/soc/intel/avs/skl.c | 62 ++++++++++++++++++++++++++++ sound/soc/intel/avs/tgl.c | 1 + 9 files changed, 240 insertions(+) diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c index c21ecaef9ebaa..a186d88430b95 100644 --- a/sound/soc/intel/avs/apl.c +++ b/sound/soc/intel/avs/apl.c @@ -8,11 +8,28 @@ #include #include +#include #include "avs.h" #include "messages.h" #include "path.h" #include "topology.h" +static irqreturn_t avs_apl_dsp_interrupt(struct avs_dev *adev) +{ + u32 adspis = snd_hdac_adsp_readl(adev, AVS_ADSP_REG_ADSPIS); + irqreturn_t ret = IRQ_NONE; + + if (adspis == UINT_MAX) + return ret; + + if (adspis & AVS_ADSP_ADSPIS_IPC) { + avs_skl_ipc_interrupt(adev); + ret = IRQ_HANDLED; + } + + return ret; +} + #ifdef CONFIG_DEBUG_FS int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) @@ -237,6 +254,7 @@ const struct avs_dsp_ops avs_apl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, + .dsp_interrupt = avs_apl_dsp_interrupt, .irq_handler = avs_irq_handler, .irq_thread = avs_skl_irq_thread, .int_control = avs_dsp_interrupt_control, diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index eb8c03afdd239..75ae8f3addb8a 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -46,6 +46,7 @@ struct avs_dsp_ops { int (* const power)(struct avs_dev *, u32, bool); int (* const reset)(struct avs_dev *, u32, bool); int (* const stall)(struct avs_dev *, u32, bool); + irqreturn_t (* const dsp_interrupt)(struct avs_dev *); irqreturn_t (* const irq_handler)(struct avs_dev *); irqreturn_t (* const irq_thread)(struct avs_dev *); void (* const int_control)(struct avs_dev *, bool); @@ -269,6 +270,8 @@ int avs_dsp_enable_d0ix(struct avs_dev *adev); irqreturn_t avs_skl_irq_thread(struct avs_dev *adev); irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev); +void avs_skl_ipc_interrupt(struct avs_dev *adev); +irqreturn_t avs_cnl_dsp_interrupt(struct avs_dev *adev); int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, u32 fifo_full_period, unsigned long resource_mask, u32 *priorities); int avs_icl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c index 585579840b646..c79b126f376d4 100644 --- a/sound/soc/intel/avs/cldma.c +++ b/sound/soc/intel/avs/cldma.c @@ -270,6 +270,17 @@ static irqreturn_t cldma_irq_handler(int irq, void *dev_id) return IRQ_HANDLED; } +void hda_cldma_interrupt(struct hda_cldma *cl) +{ + /* disable CLDMA interrupt */ + snd_hdac_adsp_updatel(cl, AVS_ADSP_REG_ADSPIC, AVS_ADSP_ADSPIC_CLDMA, 0); + + cl->sd_status = snd_hdac_stream_readb(cl, SD_STS); + dev_dbg(cl->dev, "%s sd_status: 0x%08x\n", __func__, cl->sd_status); + + complete(&cl->completion); +} + int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp_ba, unsigned int buffer_size) { diff --git a/sound/soc/intel/avs/cldma.h b/sound/soc/intel/avs/cldma.h index 223d3431ab811..7d95e2747f527 100644 --- a/sound/soc/intel/avs/cldma.h +++ b/sound/soc/intel/avs/cldma.h @@ -24,6 +24,7 @@ int hda_cldma_reset(struct hda_cldma *cl); void hda_cldma_set_data(struct hda_cldma *cl, void *data, unsigned int size); void hda_cldma_setup(struct hda_cldma *cl); +void hda_cldma_interrupt(struct hda_cldma *cl); int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp_ba, unsigned int buffer_size); void hda_cldma_free(struct hda_cldma *cl); diff --git a/sound/soc/intel/avs/cnl.c b/sound/soc/intel/avs/cnl.c index 5423c29ecc4eb..c021c0f51a53f 100644 --- a/sound/soc/intel/avs/cnl.c +++ b/sound/soc/intel/avs/cnl.c @@ -42,10 +42,73 @@ irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev) return IRQ_HANDLED; } +static void avs_cnl_ipc_interrupt(struct avs_dev *adev) +{ + const struct avs_spec *spec = adev->spec; + u32 hipc_ack, hipc_rsp; + + snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, + AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY, 0); + + hipc_ack = snd_hdac_adsp_readl(adev, spec->hipc->ack_offset); + hipc_rsp = snd_hdac_adsp_readl(adev, spec->hipc->rsp_offset); + + /* DSP acked host's request. */ + if (hipc_ack & spec->hipc->ack_done_mask) { + complete(&adev->ipc->done_completion); + + /* Tell DSP it has our attention. */ + snd_hdac_adsp_updatel(adev, spec->hipc->ack_offset, spec->hipc->ack_done_mask, + spec->hipc->ack_done_mask); + } + + /* DSP sent new response to process. */ + if (hipc_rsp & spec->hipc->rsp_busy_mask) { + union avs_reply_msg msg; + u32 hipctda; + + msg.primary = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDR); + msg.ext.val = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDD); + + avs_dsp_process_response(adev, msg.val); + + /* Tell DSP we accepted its message. */ + snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDR, + CNL_ADSP_HIPCTDR_BUSY, CNL_ADSP_HIPCTDR_BUSY); + /* Ack this response. */ + snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDA, + CNL_ADSP_HIPCTDA_DONE, CNL_ADSP_HIPCTDA_DONE); + /* HW might have been clock gated, give some time for change to propagate. */ + snd_hdac_adsp_readl_poll(adev, CNL_ADSP_REG_HIPCTDA, hipctda, + !(hipctda & CNL_ADSP_HIPCTDA_DONE), 10, 1000); + } + + snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, + AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY, + AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY); +} + +irqreturn_t avs_cnl_dsp_interrupt(struct avs_dev *adev) +{ + u32 adspis = snd_hdac_adsp_readl(adev, AVS_ADSP_REG_ADSPIS); + irqreturn_t ret = IRQ_NONE; + + if (adspis == UINT_MAX) + return ret; + + if (adspis & AVS_ADSP_ADSPIS_IPC) { + avs_cnl_ipc_interrupt(adev); + ret = IRQ_HANDLED; + } + + return ret; +} + const struct avs_dsp_ops avs_cnl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, + .dsp_interrupt = avs_cnl_dsp_interrupt, .irq_handler = avs_irq_handler, .irq_thread = avs_cnl_irq_thread, .int_control = avs_dsp_interrupt_control, diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 76782a0f32bc3..25759f4f0213a 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -336,6 +336,86 @@ static irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id) return avs_dsp_op(adev, irq_thread); } +static irqreturn_t avs_hda_interrupt(struct hdac_bus *bus) +{ + irqreturn_t ret = IRQ_NONE; + u32 status; + + status = snd_hdac_chip_readl(bus, INTSTS); + if (snd_hdac_bus_handle_stream_irq(bus, status, hdac_update_stream)) + ret = IRQ_HANDLED; + + spin_lock_irq(&bus->reg_lock); + /* Clear RIRB interrupt. */ + status = snd_hdac_chip_readb(bus, RIRBSTS); + if (status & RIRB_INT_MASK) { + if (status & RIRB_INT_RESPONSE) + snd_hdac_bus_update_rirb(bus); + snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); + ret = IRQ_HANDLED; + } + + spin_unlock_irq(&bus->reg_lock); + return ret; +} + +__maybe_unused +static irqreturn_t avs_hda_irq_handler(int irq, void *dev_id) +{ + struct hdac_bus *bus = dev_id; + u32 intsts; + + intsts = snd_hdac_chip_readl(bus, INTSTS); + if (intsts == UINT_MAX || !(intsts & AZX_INT_GLOBAL_EN)) + return IRQ_NONE; + + /* Mask GIE, unmasked in irq_thread(). */ + snd_hdac_chip_updatel(bus, INTCTL, AZX_INT_GLOBAL_EN, 0); + + return IRQ_WAKE_THREAD; +} + +__maybe_unused +static irqreturn_t avs_hda_irq_thread(int irq, void *dev_id) +{ + struct hdac_bus *bus = dev_id; + u32 status; + + status = snd_hdac_chip_readl(bus, INTSTS); + if (status & ~AZX_INT_GLOBAL_EN) + avs_hda_interrupt(bus); + + /* Unmask GIE, masked in irq_handler(). */ + snd_hdac_chip_updatel(bus, INTCTL, AZX_INT_GLOBAL_EN, AZX_INT_GLOBAL_EN); + + return IRQ_HANDLED; +} + +__maybe_unused +static irqreturn_t avs_dsp_irq_handler2(int irq, void *dev_id) +{ + struct avs_dev *adev = dev_id; + + return avs_hda_irq_handler(irq, &adev->base.core); +} + +__maybe_unused +static irqreturn_t avs_dsp_irq_thread2(int irq, void *dev_id) +{ + struct avs_dev *adev = dev_id; + struct hdac_bus *bus = &adev->base.core; + u32 status; + + status = readl(bus->ppcap + AZX_REG_PP_PPSTS); + if (status & AZX_PPCTL_PIE) + avs_dsp_op(adev, dsp_interrupt); + + /* Unmask GIE, masked in irq_handler(). */ + snd_hdac_chip_updatel(bus, INTCTL, AZX_INT_GLOBAL_EN, AZX_INT_GLOBAL_EN); + + return IRQ_HANDLED; +} + static int avs_hdac_acquire_irq(struct avs_dev *adev) { struct hdac_bus *bus = &adev->base.core; diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index e8b4983e03e9f..c18ea41d094be 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -188,6 +188,7 @@ const struct avs_dsp_ops avs_icl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, + .dsp_interrupt = avs_cnl_dsp_interrupt, .irq_handler = avs_irq_handler, .irq_thread = avs_cnl_irq_thread, .int_control = avs_dsp_interrupt_control, diff --git a/sound/soc/intel/avs/skl.c b/sound/soc/intel/avs/skl.c index d19f8953993f4..25abfead9f63a 100644 --- a/sound/soc/intel/avs/skl.c +++ b/sound/soc/intel/avs/skl.c @@ -10,6 +10,7 @@ #include #include #include "avs.h" +#include "cldma.h" #include "messages.h" irqreturn_t avs_skl_irq_thread(struct avs_dev *adev) @@ -37,6 +38,66 @@ irqreturn_t avs_skl_irq_thread(struct avs_dev *adev) return IRQ_HANDLED; } +void avs_skl_ipc_interrupt(struct avs_dev *adev) +{ + const struct avs_spec *spec = adev->spec; + u32 hipc_ack, hipc_rsp; + + snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, + AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY, 0); + + hipc_ack = snd_hdac_adsp_readl(adev, spec->hipc->ack_offset); + hipc_rsp = snd_hdac_adsp_readl(adev, spec->hipc->rsp_offset); + + /* DSP acked host's request. */ + if (hipc_ack & spec->hipc->ack_done_mask) { + complete(&adev->ipc->done_completion); + + /* Tell DSP it has our attention. */ + snd_hdac_adsp_updatel(adev, spec->hipc->ack_offset, spec->hipc->ack_done_mask, + spec->hipc->ack_done_mask); + } + + /* DSP sent new response to process */ + if (hipc_rsp & spec->hipc->rsp_busy_mask) { + union avs_reply_msg msg; + + msg.primary = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCT); + msg.ext.val = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCTE); + + avs_dsp_process_response(adev, msg.val); + + /* Tell DSP we accepted its message. */ + snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCT, SKL_ADSP_HIPCT_BUSY, + SKL_ADSP_HIPCT_BUSY); + } + + snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, + AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY, + AVS_ADSP_HIPCCTL_DONE | AVS_ADSP_HIPCCTL_BUSY); +} + +static irqreturn_t avs_skl_dsp_interrupt(struct avs_dev *adev) +{ + u32 adspis = snd_hdac_adsp_readl(adev, AVS_ADSP_REG_ADSPIS); + irqreturn_t ret = IRQ_NONE; + + if (adspis == UINT_MAX) + return ret; + + if (adspis & AVS_ADSP_ADSPIS_CLDMA) { + hda_cldma_interrupt(&code_loader); + ret = IRQ_HANDLED; + } + + if (adspis & AVS_ADSP_ADSPIS_IPC) { + avs_skl_ipc_interrupt(adev); + ret = IRQ_HANDLED; + } + + return ret; +} + static int __maybe_unused avs_skl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, u32 fifo_full_period, unsigned long resource_mask, u32 *priorities) @@ -128,6 +189,7 @@ const struct avs_dsp_ops avs_skl_dsp_ops = { .power = avs_dsp_core_power, .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, + .dsp_interrupt = avs_skl_dsp_interrupt, .irq_handler = avs_irq_handler, .irq_thread = avs_skl_irq_thread, .int_control = avs_dsp_interrupt_control, diff --git a/sound/soc/intel/avs/tgl.c b/sound/soc/intel/avs/tgl.c index 0e052e7f6bac4..e3723ef38a333 100644 --- a/sound/soc/intel/avs/tgl.c +++ b/sound/soc/intel/avs/tgl.c @@ -39,6 +39,7 @@ const struct avs_dsp_ops avs_tgl_dsp_ops = { .power = avs_tgl_dsp_core_power, .reset = avs_tgl_dsp_core_reset, .stall = avs_tgl_dsp_core_stall, + .dsp_interrupt = avs_cnl_dsp_interrupt, .irq_handler = avs_irq_handler, .irq_thread = avs_cnl_irq_thread, .int_control = avs_dsp_interrupt_control, From 84049e2db59ad9b09461b6d7ec56bd3e8fe75eca Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 19 Apr 2024 10:48:57 +0200 Subject: [PATCH 339/603] ASoC: Intel: avs: Remove unused IRQ-related code Most IRQ-related code is duplicated in the driver. Switch to the new implementation and remove unused members. While the change is non-trivial, from functional perspective status quo is achieved. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240419084857.2719593-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/apl.c | 2 - sound/soc/intel/avs/avs.h | 5 -- sound/soc/intel/avs/cldma.c | 45 ++---------------- sound/soc/intel/avs/cnl.c | 34 -------------- sound/soc/intel/avs/core.c | 91 ++----------------------------------- sound/soc/intel/avs/icl.c | 2 - sound/soc/intel/avs/ipc.c | 48 ------------------- sound/soc/intel/avs/skl.c | 27 ----------- sound/soc/intel/avs/tgl.c | 2 - 9 files changed, 7 insertions(+), 249 deletions(-) diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c index a186d88430b95..bf97e4e428a49 100644 --- a/sound/soc/intel/avs/apl.c +++ b/sound/soc/intel/avs/apl.c @@ -255,8 +255,6 @@ const struct avs_dsp_ops avs_apl_dsp_ops = { .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, .dsp_interrupt = avs_apl_dsp_interrupt, - .irq_handler = avs_irq_handler, - .irq_thread = avs_skl_irq_thread, .int_control = avs_dsp_interrupt_control, .load_basefw = avs_hda_load_basefw, .load_lib = avs_hda_load_library, diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index 75ae8f3addb8a..c905ecd0a1089 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -47,8 +47,6 @@ struct avs_dsp_ops { int (* const reset)(struct avs_dev *, u32, bool); int (* const stall)(struct avs_dev *, u32, bool); irqreturn_t (* const dsp_interrupt)(struct avs_dev *); - irqreturn_t (* const irq_handler)(struct avs_dev *); - irqreturn_t (* const irq_thread)(struct avs_dev *); void (* const int_control)(struct avs_dev *, bool); int (* const load_basefw)(struct avs_dev *, struct firmware *); int (* const load_lib)(struct avs_dev *, struct firmware *, u32); @@ -246,7 +244,6 @@ struct avs_ipc { #define AVS_IPC_RET(ret) \ (((ret) <= 0) ? (ret) : -AVS_EIPC) -irqreturn_t avs_irq_handler(struct avs_dev *adev); void avs_dsp_process_response(struct avs_dev *adev, u64 header); int avs_dsp_send_msg_timeout(struct avs_dev *adev, struct avs_ipc_msg *request, struct avs_ipc_msg *reply, int timeout, const char *name); @@ -268,8 +265,6 @@ void avs_ipc_block(struct avs_ipc *ipc); int avs_dsp_disable_d0ix(struct avs_dev *adev); int avs_dsp_enable_d0ix(struct avs_dev *adev); -irqreturn_t avs_skl_irq_thread(struct avs_dev *adev); -irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev); void avs_skl_ipc_interrupt(struct avs_dev *adev); irqreturn_t avs_cnl_dsp_interrupt(struct avs_dev *adev); int avs_apl_enable_logs(struct avs_dev *adev, enum avs_log_enable enable, u32 aging_period, diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c index c79b126f376d4..945ea376d0312 100644 --- a/sound/soc/intel/avs/cldma.c +++ b/sound/soc/intel/avs/cldma.c @@ -248,28 +248,6 @@ void hda_cldma_setup(struct hda_cldma *cl) snd_hdac_stream_writel(cl, CL_SPBFCTL, 1); } -static irqreturn_t cldma_irq_handler(int irq, void *dev_id) -{ - struct hda_cldma *cl = dev_id; - u32 adspis; - - adspis = snd_hdac_adsp_readl(cl, AVS_ADSP_REG_ADSPIS); - if (adspis == UINT_MAX) - return IRQ_NONE; - if (!(adspis & AVS_ADSP_ADSPIS_CLDMA)) - return IRQ_NONE; - - cl->sd_status = snd_hdac_stream_readb(cl, SD_STS); - dev_warn(cl->dev, "%s sd_status: 0x%08x\n", __func__, cl->sd_status); - - /* disable CLDMA interrupt */ - snd_hdac_adsp_updatel(cl, AVS_ADSP_REG_ADSPIC, AVS_ADSP_ADSPIC_CLDMA, 0); - - complete(&cl->completion); - - return IRQ_HANDLED; -} - void hda_cldma_interrupt(struct hda_cldma *cl) { /* disable CLDMA interrupt */ @@ -284,7 +262,6 @@ void hda_cldma_interrupt(struct hda_cldma *cl) int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp_ba, unsigned int buffer_size) { - struct pci_dev *pci = to_pci_dev(bus->dev); int ret; ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, bus->dev, buffer_size, &cl->dmab_data); @@ -292,8 +269,10 @@ int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp return ret; ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, bus->dev, BDL_SIZE, &cl->dmab_bdl); - if (ret < 0) - goto alloc_err; + if (ret < 0) { + snd_dma_free_pages(&cl->dmab_data); + return ret; + } cl->dev = bus->dev; cl->bus = bus; @@ -301,27 +280,11 @@ int hda_cldma_init(struct hda_cldma *cl, struct hdac_bus *bus, void __iomem *dsp cl->buffer_size = buffer_size; cl->sd_addr = dsp_ba + AZX_CL_SD_BASE; - ret = pci_request_irq(pci, 0, cldma_irq_handler, NULL, cl, "CLDMA"); - if (ret < 0) { - dev_err(cl->dev, "Failed to request CLDMA IRQ handler: %d\n", ret); - goto req_err; - } - return 0; - -req_err: - snd_dma_free_pages(&cl->dmab_bdl); -alloc_err: - snd_dma_free_pages(&cl->dmab_data); - - return ret; } void hda_cldma_free(struct hda_cldma *cl) { - struct pci_dev *pci = to_pci_dev(cl->dev); - - pci_free_irq(pci, 0, cl); snd_dma_free_pages(&cl->dmab_data); snd_dma_free_pages(&cl->dmab_bdl); } diff --git a/sound/soc/intel/avs/cnl.c b/sound/soc/intel/avs/cnl.c index c021c0f51a53f..0d03e1e03c112 100644 --- a/sound/soc/intel/avs/cnl.c +++ b/sound/soc/intel/avs/cnl.c @@ -10,38 +10,6 @@ #include "avs.h" #include "messages.h" -irqreturn_t avs_cnl_irq_thread(struct avs_dev *adev) -{ - union avs_reply_msg msg; - u32 hipctdr, hipctdd, hipctda; - - hipctdr = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDR); - hipctdd = snd_hdac_adsp_readl(adev, CNL_ADSP_REG_HIPCTDD); - - /* Ensure DSP sent new response to process. */ - if (!(hipctdr & CNL_ADSP_HIPCTDR_BUSY)) - return IRQ_NONE; - - msg.primary = hipctdr; - msg.ext.val = hipctdd; - avs_dsp_process_response(adev, msg.val); - - /* Tell DSP we accepted its message. */ - snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDR, - CNL_ADSP_HIPCTDR_BUSY, CNL_ADSP_HIPCTDR_BUSY); - /* Ack this response. */ - snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCTDA, - CNL_ADSP_HIPCTDA_DONE, CNL_ADSP_HIPCTDA_DONE); - /* HW might have been clock gated, give some time for change to propagate. */ - snd_hdac_adsp_readl_poll(adev, CNL_ADSP_REG_HIPCTDA, hipctda, - !(hipctda & CNL_ADSP_HIPCTDA_DONE), 10, 1000); - /* Unmask busy interrupt. */ - snd_hdac_adsp_updatel(adev, CNL_ADSP_REG_HIPCCTL, - AVS_ADSP_HIPCCTL_BUSY, AVS_ADSP_HIPCCTL_BUSY); - - return IRQ_HANDLED; -} - static void avs_cnl_ipc_interrupt(struct avs_dev *adev) { const struct avs_spec *spec = adev->spec; @@ -109,8 +77,6 @@ const struct avs_dsp_ops avs_cnl_dsp_ops = { .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, .dsp_interrupt = avs_cnl_dsp_interrupt, - .irq_handler = avs_irq_handler, - .irq_thread = avs_cnl_irq_thread, .int_control = avs_dsp_interrupt_control, .load_basefw = avs_hda_load_basefw, .load_lib = avs_hda_load_library, diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 25759f4f0213a..a059bb6888d80 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -257,85 +257,6 @@ static void hdac_update_stream(struct hdac_bus *bus, struct hdac_stream *stream) } } -static irqreturn_t hdac_bus_irq_handler(int irq, void *context) -{ - struct hdac_bus *bus = context; - u32 mask, int_enable; - u32 status; - int ret = IRQ_NONE; - - if (!pm_runtime_active(bus->dev)) - return ret; - - spin_lock(&bus->reg_lock); - - status = snd_hdac_chip_readl(bus, INTSTS); - if (status == 0 || status == UINT_MAX) { - spin_unlock(&bus->reg_lock); - return ret; - } - - /* clear rirb int */ - status = snd_hdac_chip_readb(bus, RIRBSTS); - if (status & RIRB_INT_MASK) { - if (status & RIRB_INT_RESPONSE) - snd_hdac_bus_update_rirb(bus); - snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); - } - - mask = (0x1 << bus->num_streams) - 1; - - status = snd_hdac_chip_readl(bus, INTSTS); - status &= mask; - if (status) { - /* Disable stream interrupts; Re-enable in bottom half */ - int_enable = snd_hdac_chip_readl(bus, INTCTL); - snd_hdac_chip_writel(bus, INTCTL, (int_enable & (~mask))); - ret = IRQ_WAKE_THREAD; - } else { - ret = IRQ_HANDLED; - } - - spin_unlock(&bus->reg_lock); - return ret; -} - -static irqreturn_t hdac_bus_irq_thread(int irq, void *context) -{ - struct hdac_bus *bus = context; - u32 status; - u32 int_enable; - u32 mask; - unsigned long flags; - - status = snd_hdac_chip_readl(bus, INTSTS); - - snd_hdac_bus_handle_stream_irq(bus, status, hdac_update_stream); - - /* Re-enable stream interrupts */ - mask = (0x1 << bus->num_streams) - 1; - spin_lock_irqsave(&bus->reg_lock, flags); - int_enable = snd_hdac_chip_readl(bus, INTCTL); - snd_hdac_chip_writel(bus, INTCTL, (int_enable | mask)); - spin_unlock_irqrestore(&bus->reg_lock, flags); - - return IRQ_HANDLED; -} - -static irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id) -{ - struct avs_dev *adev = dev_id; - - return avs_dsp_op(adev, irq_handler); -} - -static irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id) -{ - struct avs_dev *adev = dev_id; - - return avs_dsp_op(adev, irq_thread); -} - static irqreturn_t avs_hda_interrupt(struct hdac_bus *bus) { irqreturn_t ret = IRQ_NONE; @@ -359,7 +280,6 @@ static irqreturn_t avs_hda_interrupt(struct hdac_bus *bus) return ret; } -__maybe_unused static irqreturn_t avs_hda_irq_handler(int irq, void *dev_id) { struct hdac_bus *bus = dev_id; @@ -375,7 +295,6 @@ static irqreturn_t avs_hda_irq_handler(int irq, void *dev_id) return IRQ_WAKE_THREAD; } -__maybe_unused static irqreturn_t avs_hda_irq_thread(int irq, void *dev_id) { struct hdac_bus *bus = dev_id; @@ -391,16 +310,14 @@ static irqreturn_t avs_hda_irq_thread(int irq, void *dev_id) return IRQ_HANDLED; } -__maybe_unused -static irqreturn_t avs_dsp_irq_handler2(int irq, void *dev_id) +static irqreturn_t avs_dsp_irq_handler(int irq, void *dev_id) { struct avs_dev *adev = dev_id; return avs_hda_irq_handler(irq, &adev->base.core); } -__maybe_unused -static irqreturn_t avs_dsp_irq_thread2(int irq, void *dev_id) +static irqreturn_t avs_dsp_irq_thread(int irq, void *dev_id) { struct avs_dev *adev = dev_id; struct hdac_bus *bus = &adev->base.core; @@ -429,7 +346,7 @@ static int avs_hdac_acquire_irq(struct avs_dev *adev) return ret; } - ret = pci_request_irq(pci, 0, hdac_bus_irq_handler, hdac_bus_irq_thread, bus, + ret = pci_request_irq(pci, 0, avs_hda_irq_handler, avs_hda_irq_thread, bus, KBUILD_MODNAME); if (ret < 0) { dev_err(adev->dev, "Failed to request stream IRQ handler: %d\n", ret); @@ -610,8 +527,6 @@ static void avs_pci_shutdown(struct pci_dev *pci) snd_hdac_bus_stop_chip(bus); snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); - if (avs_platattr_test(adev, CLDMA)) - pci_free_irq(pci, 0, &code_loader); pci_free_irq(pci, 0, adev); pci_free_irq(pci, 0, bus); pci_free_irq_vectors(pci); diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index c18ea41d094be..e0880acf8050b 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -189,8 +189,6 @@ const struct avs_dsp_ops avs_icl_dsp_ops = { .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, .dsp_interrupt = avs_cnl_dsp_interrupt, - .irq_handler = avs_irq_handler, - .irq_thread = avs_cnl_irq_thread, .int_control = avs_dsp_interrupt_control, .load_basefw = avs_icl_load_basefw, .load_lib = avs_hda_load_library, diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c index ad0e535b3c2ed..f9b302215c378 100644 --- a/sound/soc/intel/avs/ipc.c +++ b/sound/soc/intel/avs/ipc.c @@ -301,54 +301,6 @@ void avs_dsp_process_response(struct avs_dev *adev, u64 header) complete(&ipc->busy_completion); } -irqreturn_t avs_irq_handler(struct avs_dev *adev) -{ - struct avs_ipc *ipc = adev->ipc; - const struct avs_spec *const spec = adev->spec; - u32 adspis, hipc_rsp, hipc_ack; - irqreturn_t ret = IRQ_NONE; - - adspis = snd_hdac_adsp_readl(adev, AVS_ADSP_REG_ADSPIS); - if (adspis == UINT_MAX || !(adspis & AVS_ADSP_ADSPIS_IPC)) - return ret; - - hipc_ack = snd_hdac_adsp_readl(adev, spec->hipc->ack_offset); - hipc_rsp = snd_hdac_adsp_readl(adev, spec->hipc->rsp_offset); - - /* DSP acked host's request */ - if (hipc_ack & spec->hipc->ack_done_mask) { - /* - * As an extra precaution, mask done interrupt. Code executed - * due to complete() found below does not assume any masking. - */ - snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, - AVS_ADSP_HIPCCTL_DONE, 0); - - complete(&ipc->done_completion); - - /* tell DSP it has our attention */ - snd_hdac_adsp_updatel(adev, spec->hipc->ack_offset, - spec->hipc->ack_done_mask, - spec->hipc->ack_done_mask); - /* unmask done interrupt */ - snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, - AVS_ADSP_HIPCCTL_DONE, - AVS_ADSP_HIPCCTL_DONE); - ret = IRQ_HANDLED; - } - - /* DSP sent new response to process */ - if (hipc_rsp & spec->hipc->rsp_busy_mask) { - /* mask busy interrupt */ - snd_hdac_adsp_updatel(adev, spec->hipc->ctl_offset, - AVS_ADSP_HIPCCTL_BUSY, 0); - - ret = IRQ_WAKE_THREAD; - } - - return ret; -} - static bool avs_ipc_is_busy(struct avs_ipc *ipc) { struct avs_dev *adev = to_avs_dev(ipc->dev); diff --git a/sound/soc/intel/avs/skl.c b/sound/soc/intel/avs/skl.c index 25abfead9f63a..fceed353449a3 100644 --- a/sound/soc/intel/avs/skl.c +++ b/sound/soc/intel/avs/skl.c @@ -13,31 +13,6 @@ #include "cldma.h" #include "messages.h" -irqreturn_t avs_skl_irq_thread(struct avs_dev *adev) -{ - union avs_reply_msg msg; - u32 hipct, hipcte; - - hipct = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCT); - hipcte = snd_hdac_adsp_readl(adev, SKL_ADSP_REG_HIPCTE); - - /* Ensure DSP sent new response to process. */ - if (!(hipct & SKL_ADSP_HIPCT_BUSY)) - return IRQ_NONE; - - msg.primary = hipct; - msg.ext.val = hipcte; - avs_dsp_process_response(adev, msg.val); - - /* Tell DSP we accepted its message. */ - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCT, SKL_ADSP_HIPCT_BUSY, SKL_ADSP_HIPCT_BUSY); - /* Unmask busy interrupt. */ - snd_hdac_adsp_updatel(adev, SKL_ADSP_REG_HIPCCTL, AVS_ADSP_HIPCCTL_BUSY, - AVS_ADSP_HIPCCTL_BUSY); - - return IRQ_HANDLED; -} - void avs_skl_ipc_interrupt(struct avs_dev *adev) { const struct avs_spec *spec = adev->spec; @@ -190,8 +165,6 @@ const struct avs_dsp_ops avs_skl_dsp_ops = { .reset = avs_dsp_core_reset, .stall = avs_dsp_core_stall, .dsp_interrupt = avs_skl_dsp_interrupt, - .irq_handler = avs_irq_handler, - .irq_thread = avs_skl_irq_thread, .int_control = avs_dsp_interrupt_control, .load_basefw = avs_cldma_load_basefw, .load_lib = avs_cldma_load_library, diff --git a/sound/soc/intel/avs/tgl.c b/sound/soc/intel/avs/tgl.c index e3723ef38a333..b985a8299b721 100644 --- a/sound/soc/intel/avs/tgl.c +++ b/sound/soc/intel/avs/tgl.c @@ -40,8 +40,6 @@ const struct avs_dsp_ops avs_tgl_dsp_ops = { .reset = avs_tgl_dsp_core_reset, .stall = avs_tgl_dsp_core_stall, .dsp_interrupt = avs_cnl_dsp_interrupt, - .irq_handler = avs_irq_handler, - .irq_thread = avs_cnl_irq_thread, .int_control = avs_dsp_interrupt_control, .load_basefw = avs_icl_load_basefw, .load_lib = avs_hda_load_library, From 32cb23a0f911317cdb5030035e49a204aa86fef5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Apr 2024 12:11:02 +0200 Subject: [PATCH 340/603] ALSA: seq: dummy: Allow UMP conversion Although the purpose of dummy seq client is a direct pass-through, it's sometimes helpful for debugging if it can convert to a certain UMP MIDI version. This patch adds an option to specify the UMP event conversion. As default, it skips the conversion and does passthrough, while user can pass ump=1 or ump=2 to enforce the conversion to UMP MIDI1 or MIDI2 format. Message-ID: <20240419101105.15571-1-tiwai@suse.de> Signed-off-by: Takashi Iwai --- sound/core/seq/seq_dummy.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index 9308194b2d9ad..783fc72c2ef67 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -58,6 +58,12 @@ MODULE_PARM_DESC(ports, "number of ports to be created"); module_param(duplex, bool, 0444); MODULE_PARM_DESC(duplex, "create DUPLEX ports"); +#if IS_ENABLED(CONFIG_SND_SEQ_UMP) +static int ump; +module_param(ump, int, 0444); +MODULE_PARM_DESC(ump, "UMP conversion (0: no convert, 1: MIDI 1.0, 2: MIDI 2.0)"); +#endif + struct snd_seq_dummy_port { int client; int port; @@ -152,7 +158,9 @@ static int __init register_client(void) { struct snd_seq_dummy_port *rec1, *rec2; +#if IS_ENABLED(CONFIG_SND_SEQ_UMP) struct snd_seq_client *client; +#endif int i; if (ports < 1) { @@ -166,12 +174,24 @@ register_client(void) if (my_client < 0) return my_client; - /* don't convert events but just pass-through */ +#if IS_ENABLED(CONFIG_SND_SEQ_UMP) client = snd_seq_kernel_client_get(my_client); if (!client) return -EINVAL; - client->filter = SNDRV_SEQ_FILTER_NO_CONVERT; + switch (ump) { + case 1: + client->midi_version = SNDRV_SEQ_CLIENT_UMP_MIDI_1_0; + break; + case 2: + client->midi_version = SNDRV_SEQ_CLIENT_UMP_MIDI_2_0; + break; + default: + /* don't convert events but just pass-through */ + client->filter = SNDRV_SEQ_FILTER_NO_CONVERT; + break; + } snd_seq_kernel_client_put(client); +#endif /* create ports */ for (i = 0; i < ports; i++) { From 467d2528d64e8fcf784850ac94f665efae2af149 Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Mon, 15 Apr 2024 15:06:47 +0800 Subject: [PATCH 341/603] ASoC: dt-bindings: nau8821: Add delay control for ADC Change the original fixed delay to the assignment from the property. It will make it more flexible to different platforms to avoid pop noise at the beginning of recording. Signed-off-by: Seven Lee Reviewed-by: Rob Herring (Arm) Link: https://lore.kernel.org/r/20240415070649.3496487-2-wtli@nuvoton.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index 054b53954ac3d..9f44168efb3ed 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -103,6 +103,12 @@ properties: just limited to the left adc for design demand. type: boolean + nuvoton,adc-delay-ms: + description: Delay (in ms) to make input path stable and avoid pop noise. + minimum: 125 + maximum: 500 + default: 125 + '#sound-dai-cells': const: 0 @@ -136,6 +142,7 @@ examples: nuvoton,jack-eject-debounce = <0>; nuvoton,dmic-clk-threshold = <3072000>; nuvoton,dmic-slew-rate = <0>; + nuvoton,adc-delay-ms = <125>; #sound-dai-cells = <0>; }; }; From b37fdd42c00904afa11d9a8eb80e098429b894b3 Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Mon, 15 Apr 2024 15:06:48 +0800 Subject: [PATCH 342/603] ASoC: nau8821: Add delay control for ADC Change the original fixed delay to the assignment from the property. It will make it more flexible to different platforms to avoid pop noise at the beginning of recording. Signed-off-by: Seven Lee Link: https://lore.kernel.org/r/20240415070649.3496487-3-wtli@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8821.c | 11 +++++++++-- sound/soc/codecs/nau8821.h | 1 + 2 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 012e347e63913..fec669e00a100 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -511,7 +511,7 @@ static int nau8821_left_adc_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - msleep(125); + msleep(nau8821->adc_delay); regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCL, NAU8821_EN_ADCL); break; @@ -535,7 +535,7 @@ static int nau8821_right_adc_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - msleep(125); + msleep(nau8821->adc_delay); regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCR, NAU8821_EN_ADCR); break; @@ -1697,6 +1697,7 @@ static void nau8821_print_device_properties(struct nau8821 *nau8821) dev_dbg(dev, "dmic-clk-threshold: %d\n", nau8821->dmic_clk_threshold); dev_dbg(dev, "key_enable: %d\n", nau8821->key_enable); + dev_dbg(dev, "adc-delay-ms: %d\n", nau8821->adc_delay); } static int nau8821_read_device_properties(struct device *dev, @@ -1742,6 +1743,12 @@ static int nau8821_read_device_properties(struct device *dev, &nau8821->dmic_slew_rate); if (ret) nau8821->dmic_slew_rate = 0; + ret = device_property_read_u32(dev, "nuvoton,adc-delay-ms", + &nau8821->adc_delay); + if (ret) + nau8821->adc_delay = 125; + if (nau8821->adc_delay < 125 || nau8821->adc_delay > 500) + dev_warn(dev, "Please set the suitable delay time!\n"); return 0; } diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h index 62eaad130b2ec..f0935ffafcbec 100644 --- a/sound/soc/codecs/nau8821.h +++ b/sound/soc/codecs/nau8821.h @@ -577,6 +577,7 @@ struct nau8821 { int dmic_clk_threshold; int dmic_slew_rate; int key_enable; + int adc_delay; }; int nau8821_enable_jack_detect(struct snd_soc_component *component, From ad8ba241bc93536a2c8af150b237c5be8fafe942 Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Mon, 15 Apr 2024 15:06:49 +0800 Subject: [PATCH 343/603] ASoC: nau8821: Remove redundant ADC controls Remove redundant left/right adc channel enable controls. Signed-off-by: Seven Lee Link: https://lore.kernel.org/r/20240415070649.3496487-4-wtli@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8821.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index fec669e00a100..6818bbd1d3c76 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -512,12 +512,8 @@ static int nau8821_left_adc_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: msleep(nau8821->adc_delay); - regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, - NAU8821_EN_ADCL, NAU8821_EN_ADCL); break; case SND_SOC_DAPM_POST_PMD: - regmap_update_bits(nau8821->regmap, - NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCL, 0); break; default: return -EINVAL; @@ -536,12 +532,8 @@ static int nau8821_right_adc_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: msleep(nau8821->adc_delay); - regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, - NAU8821_EN_ADCR, NAU8821_EN_ADCR); break; case SND_SOC_DAPM_POST_PMD: - regmap_update_bits(nau8821->regmap, - NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCR, 0); break; default: return -EINVAL; From cbd9eed87107bb6d6e537e79aeea65399898ca41 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Tue, 16 Apr 2024 08:00:24 +0200 Subject: [PATCH 344/603] ASoC: dapm: debugfs: add component to route lines debugfs entries for DAPM widgets have the following form: Left ADC: Off in 1 out 0 - R2(0x2) mask 0x2 stream Left HiFi Capture inactive out "static" "Capture" in "static" "Left PGA" Lines with the "in" and "out" prefixes describe routes from/to other widgets presenting the path name and the connected widget name. This is ambiguous in case of cards having multiple widgets with the same name in different components. For example the STM32MP157A-DK1 board (arch/arm/boot/dts/st/stm32mp15xx-dkx.dtsi) has a "Capture" widget in both the "cs42l51.0-004a" and the "hdmi-audio-codec.1.auto" components. Avoid the ambiguity by adding the component name to the "in" and "out" lines. Add the new field at the end to minimize backward compatibility issues. The existing vizdapm tool continues working after this change. The output becomes: Left ADC: Off in 1 out 0 - R2(0x2) mask 0x2 stream Left HiFi Capture inactive out "static" "Capture" "cs42l51.0-004a" in "static" "Left PGA" "cs42l51.0-004a" Signed-off-by: Luca Ceresoli Reviewed-by: Alexandre Belloni Link: https://lore.kernel.org/r/20240416-vizdapm-ng-v1-1-5d33c0b57bc5@bootlin.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ad8ba8fbbaeec..eb9535764693b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2104,6 +2104,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, int in, out; ssize_t ret; struct snd_soc_dapm_path *p = NULL; + const char *c_name; buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) @@ -2145,11 +2146,13 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!p->connect) continue; + c_name = p->node[rdir]->dapm->component ? + p->node[rdir]->dapm->component->name : NULL; ret += scnprintf(buf + ret, PAGE_SIZE - ret, - " %s \"%s\" \"%s\"\n", + " %s \"%s\" \"%s\" \"%s\"\n", (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", - p->node[rdir]->name); + p->node[rdir]->name, c_name); } } From 5b1047dcf80b35bafcacbd10b57342d1a96139c0 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Tue, 16 Apr 2024 08:00:25 +0200 Subject: [PATCH 345/603] ASoC: dapm: debugfs: show the widget type To make the widget debugfs files more informative, add a line showing the widget type string. Keeping backward compatibility is nice to have being debugfs, and ease of parsing by both humans and software is also good. To maximize both with a reasonable effort add a new line without thouching the already complex format of the first line. The syntax is meant to be a key/value pair. The existing vizdapm tool continues working after this change, ignoring the new line. The new format is: Left ADC: Off in 1 out 0 - R2(0x2) mask 0x2 stream Left HiFi Capture inactive widget-type adc out "static" "Capture" "cs42l51.0-004a" in "static" "Left PGA" "cs42l51.0-004a" Signed-off-by: Luca Ceresoli Reviewed-by: Alexandre Belloni Link: https://lore.kernel.org/r/20240416-vizdapm-ng-v1-2-5d33c0b57bc5@bootlin.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 47 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index eb9535764693b..16dad4a454434 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2094,6 +2094,48 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) } #ifdef CONFIG_DEBUG_FS + +static const char * const snd_soc_dapm_type_name[] = { + [snd_soc_dapm_input] = "input", + [snd_soc_dapm_output] = "output", + [snd_soc_dapm_mux] = "mux", + [snd_soc_dapm_demux] = "demux", + [snd_soc_dapm_mixer] = "mixer", + [snd_soc_dapm_mixer_named_ctl] = "mixer_named_ctl", + [snd_soc_dapm_pga] = "pga", + [snd_soc_dapm_out_drv] = "out_drv", + [snd_soc_dapm_adc] = "adc", + [snd_soc_dapm_dac] = "dac", + [snd_soc_dapm_micbias] = "micbias", + [snd_soc_dapm_mic] = "mic", + [snd_soc_dapm_hp] = "hp", + [snd_soc_dapm_spk] = "spk", + [snd_soc_dapm_line] = "line", + [snd_soc_dapm_switch] = "switch", + [snd_soc_dapm_vmid] = "vmid", + [snd_soc_dapm_pre] = "pre", + [snd_soc_dapm_post] = "post", + [snd_soc_dapm_supply] = "supply", + [snd_soc_dapm_pinctrl] = "pinctrl", + [snd_soc_dapm_regulator_supply] = "regulator_supply", + [snd_soc_dapm_clock_supply] = "clock_supply", + [snd_soc_dapm_aif_in] = "aif_in", + [snd_soc_dapm_aif_out] = "aif_out", + [snd_soc_dapm_siggen] = "siggen", + [snd_soc_dapm_sink] = "sink", + [snd_soc_dapm_dai_in] = "dai_in", + [snd_soc_dapm_dai_out] = "dai_out", + [snd_soc_dapm_dai_link] = "dai_link", + [snd_soc_dapm_kcontrol] = "kcontrol", + [snd_soc_dapm_buffer] = "buffer", + [snd_soc_dapm_scheduler] = "scheduler", + [snd_soc_dapm_effect] = "effect", + [snd_soc_dapm_src] = "src", + [snd_soc_dapm_asrc] = "asrc", + [snd_soc_dapm_encoder] = "encoder", + [snd_soc_dapm_decoder] = "decoder", +}; + static ssize_t dapm_widget_power_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) @@ -2106,6 +2148,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, struct snd_soc_dapm_path *p = NULL; const char *c_name; + BUILD_BUG_ON(ARRAY_SIZE(snd_soc_dapm_type_name) != SND_SOC_DAPM_TYPE_COUNT); + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) return -ENOMEM; @@ -2137,6 +2181,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, w->sname, w->active ? "active" : "inactive"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " widget-type %s\n", + snd_soc_dapm_type_name[w->id]); + snd_soc_dapm_for_each_direction(dir) { rdir = SND_SOC_DAPM_DIR_REVERSE(dir); snd_soc_dapm_widget_for_each_path(w, dir, p) { From e7bb43898bcf54da7ffb4819a04c8428f7db24db Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Tue, 16 Apr 2024 08:00:26 +0200 Subject: [PATCH 346/603] ASoC: dapm-graph: new tool to visualize DAPM state Add a tool to generate a picture of the current DAPM state for a sound card. dapm-graph is inspired by vizdapm which used to be published on a Wolfson Micro git repository now disappeared, and has a few forks around: https://github.com/mihais/asoc-tools https://github.com/alexandrebelloni/asoc-tools dapm-graph is a full reimplementation with several improvements while still being a self-contained shell script: Improvements to rendered output: - shows the entire card, not one component hierarchy only - each component is rendered in a separate box - shows widget on/off status based on widget information alone (the original vizdapm propagates the "on" green colour to the first input widget) - use bold line and gray background and not only green/red line to show on/off status (for the color blind) Improvements for embedded system developers: - remote mode: get state of remote device (possibly with minimal rootfs) via SSH, but parsing locally for faster operation - compatible with BusyBox shell, not only bash Usability improvements: - flexible command line (uses getopts for parsing) - detailed help text - flag to enable detailed debug logging - graphviz output format detected from file extension, not hard coded - a self-contained shell script Usage is designed to be simple: dapm-grpah -c CARD - get state from debugfs for CARD dapm-grpah -c CARD -r REMOTE_TARGET - same, but remotely via SSH dapm-grpah -d STATE_DIR - from a local copy of the debugfs tree for a card Signed-off-by: Luca Ceresoli Reviewed-by: Alexandre Belloni Link: https://lore.kernel.org/r/20240416-vizdapm-ng-v1-3-5d33c0b57bc5@bootlin.com Signed-off-by: Mark Brown --- MAINTAINERS | 6 + tools/sound/dapm-graph | 303 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 309 insertions(+) create mode 100755 tools/sound/dapm-graph diff --git a/MAINTAINERS b/MAINTAINERS index a7d0dd91ac198..aa3b455588e19 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -20669,6 +20669,12 @@ F: include/trace/events/sof*.h F: include/uapi/sound/asoc.h F: sound/soc/ +SOUND - SOC LAYER / dapm-graph +M: Luca Ceresoli +L: linux-sound@vger.kernel.org +S: Maintained +F: tools/sound/dapm-graph + SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS M: Pierre-Louis Bossart M: Liam Girdwood diff --git a/tools/sound/dapm-graph b/tools/sound/dapm-graph new file mode 100755 index 0000000000000..57d78f6df0411 --- /dev/null +++ b/tools/sound/dapm-graph @@ -0,0 +1,303 @@ +#!/bin/sh +# SPDX-License-Identifier: GPL-2.0 +# +# Generate a graph of the current DAPM state for an audio card +# +# Copyright 2024 Bootlin +# Author: Luca Ceresoli + +set -eu + +STYLE_NODE_ON="shape=box,style=bold,color=green4" +STYLE_NODE_OFF="shape=box,style=filled,color=gray30,fillcolor=gray95" + +# Print usage and exit +# +# $1 = exit return value +# $2 = error string (required if $1 != 0) +usage() +{ + if [ "${1}" -ne 0 ]; then + echo "${2}" >&2 + fi + + echo " +Generate a graph of the current DAPM state for an audio card. + +The DAPM state can be obtained via debugfs for a card on the local host or +a remote target, or from a local copy of the debugfs tree for the card. + +Usage: + $(basename $0) [options] -c CARD - Local sound card + $(basename $0) [options] -c CARD -r REMOTE_TARGET - Card on remote system + $(basename $0) [options] -d STATE_DIR - Local directory + +Options: + -c CARD Sound card to get DAPM state of + -r REMOTE_TARGET Get DAPM state from REMOTE_TARGET via SSH and SCP + instead of using a local sound card + -d STATE_DIR Get DAPM state from a local copy of a debugfs tree + -o OUT_FILE Output file (default: dapm.dot) + -D Show verbose debugging info + -h Print this help and exit + +The output format is implied by the extension of OUT_FILE: + + * Use the .dot extension to generate a text graph representation in + graphviz dot syntax. + * Any other extension is assumed to be a format supported by graphviz for + rendering, e.g. 'png', 'svg', and will produce both the .dot file and a + picture from it. This requires the 'dot' program from the graphviz + package. +" + + exit ${1} +} + +# Connect to a remote target via SSH, collect all DAPM files from debufs +# into a tarball and get the tarball via SCP into $3/dapm.tar +# +# $1 = target as used by ssh and scp, e.g. "root@192.168.1.1" +# $2 = sound card name +# $3 = temp dir path (present on the host, created on the target) +# $4 = local directory to extract the tarball into +# +# Requires an ssh+scp server, find and tar+gz on the target +# +# Note: the tarball is needed because plain 'scp -r' from debugfs would +# copy only empty files +grab_remote_files() +{ + echo "Collecting DAPM state from ${1}" + dbg_echo "Collected DAPM state in ${3}" + + ssh "${1}" " +set -eu && +cd \"/sys/kernel/debug/asoc/${2}\" && +find * -type d -exec mkdir -p ${3}/dapm-tree/{} \; && +find * -type f -exec cp \"{}\" \"${3}/dapm-tree/{}\" \; && +cd ${3}/dapm-tree && +tar cf ${3}/dapm.tar ." + scp -q "${1}:${3}/dapm.tar" "${3}" + + mkdir -p "${4}" + tar xf "${tmp_dir}/dapm.tar" -C "${4}" +} + +# Parse a widget file and generate graph description in graphviz dot format +# +# Skips any file named "bias_level". +# +# $1 = temporary work dir +# $2 = component name +# $3 = widget filename +process_dapm_widget() +{ + local tmp_dir="${1}" + local c_name="${2}" + local w_file="${3}" + local dot_file="${tmp_dir}/main.dot" + local links_file="${tmp_dir}/links.dot" + + local w_name="$(basename "${w_file}")" + local w_tag="${c_name}_${w_name}" + + if [ "${w_name}" = "bias_level" ]; then + return 0 + fi + + dbg_echo " + Widget: ${w_name}" + + cat "${w_file}" | ( + read line + + if echo "${line}" | grep -q ': On ' + then local node_style="${STYLE_NODE_ON}" + else local node_style="${STYLE_NODE_OFF}" + fi + + local w_type="" + while read line; do + # Collect widget type if present + if echo "${line}" | grep -q '^widget-type '; then + local w_type_raw="$(echo "$line" | cut -d ' ' -f 2)" + dbg_echo " - Widget type: ${w_type_raw}" + + # Note: escaping '\n' is tricky to get working with both + # bash and busybox ash, so use a '%' here and replace it + # later + local w_type="%n[${w_type_raw}]" + fi + + # Collect any links. We could use "in" links or "out" links, + # let's use "in" links + if echo "${line}" | grep -q '^in '; then + local w_src=$(echo "$line" | + awk -F\" '{print $6 "_" $4}' | + sed 's/^(null)_/ROOT_/') + dbg_echo " - Input route from: ${w_src}" + echo " \"${w_src}\" -> \"$w_tag\"" >> "${links_file}" + fi + done + + echo " \"${w_tag}\" [label=\"${w_name}${w_type}\",${node_style}]" | + tr '%' '\\' >> "${dot_file}" + ) +} + +# Parse the DAPM tree for a sound card component and generate graph +# description in graphviz dot format +# +# $1 = temporary work dir +# $2 = component directory +# $3 = forced component name (extracted for path if empty) +process_dapm_component() +{ + local tmp_dir="${1}" + local c_dir="${2}" + local c_name="${3}" + local dot_file="${tmp_dir}/main.dot" + local links_file="${tmp_dir}/links.dot" + + if [ -z "${c_name}" ]; then + # Extract directory name into component name: + # "./cs42l51.0-004a/dapm" -> "cs42l51.0-004a" + c_name="$(basename $(dirname "${c_dir}"))" + fi + + dbg_echo " * Component: ${c_name}" + + echo "" >> "${dot_file}" + echo " subgraph \"${c_name}\" {" >> "${dot_file}" + echo " cluster = true" >> "${dot_file}" + echo " label = \"${c_name}\"" >> "${dot_file}" + echo " color=dodgerblue" >> "${dot_file}" + + # Create empty file to ensure it will exist in all cases + >"${links_file}" + + # Iterate over widgets in the component dir + for w_file in ${c_dir}/*; do + process_dapm_widget "${tmp_dir}" "${c_name}" "${w_file}" + done + + echo " }" >> "${dot_file}" + + cat "${links_file}" >> "${dot_file}" +} + +# Parse the DAPM tree for a sound card and generate graph description in +# graphviz dot format +# +# $1 = temporary work dir +# $2 = directory tree with DAPM state (either in debugfs or a mirror) +process_dapm_tree() +{ + local tmp_dir="${1}" + local dapm_dir="${2}" + local dot_file="${tmp_dir}/main.dot" + + echo "digraph G {" > "${dot_file}" + echo " fontname=\"sans-serif\"" >> "${dot_file}" + echo " node [fontname=\"sans-serif\"]" >> "${dot_file}" + + + # Process root directory (no component) + process_dapm_component "${tmp_dir}" "${dapm_dir}/dapm" "ROOT" + + # Iterate over components + for c_dir in "${dapm_dir}"/*/dapm + do + process_dapm_component "${tmp_dir}" "${c_dir}" "" + done + + echo "}" >> "${dot_file}" +} + +main() +{ + # Parse command line + local out_file="dapm.dot" + local card_name="" + local remote_target="" + local dapm_tree="" + local dbg_on="" + while getopts "c:r:d:o:Dh" arg; do + case $arg in + c) card_name="${OPTARG}" ;; + r) remote_target="${OPTARG}" ;; + d) dapm_tree="${OPTARG}" ;; + o) out_file="${OPTARG}" ;; + D) dbg_on="1" ;; + h) usage 0 ;; + *) usage 1 ;; + esac + done + shift $(($OPTIND - 1)) + + if [ -n "${dapm_tree}" ]; then + if [ -n "${card_name}${remote_target}" ]; then + usage 1 "Cannot use -c and -r with -d" + fi + echo "Using local tree: ${dapm_tree}" + elif [ -n "${remote_target}" ]; then + if [ -z "${card_name}" ]; then + usage 1 "-r requires -c" + fi + echo "Using card ${card_name} from remote target ${remote_target}" + elif [ -n "${card_name}" ]; then + echo "Using local card: ${card_name}" + else + usage 1 "Please choose mode using -c, -r or -d" + fi + + # Define logging function + if [ "${dbg_on}" ]; then + dbg_echo() { + echo "$*" >&2 + } + else + dbg_echo() { + : + } + fi + + # Filename must have a dot in order the infer the format from the + # extension + if ! echo "${out_file}" | grep -qE '\.'; then + echo "Missing extension in output filename ${out_file}" >&2 + usage + exit 1 + fi + + local out_fmt="${out_file##*.}" + local dot_file="${out_file%.*}.dot" + + dbg_echo "dot file: $dot_file" + dbg_echo "Output file: $out_file" + dbg_echo "Output format: $out_fmt" + + tmp_dir="$(mktemp -d /tmp/$(basename $0).XXXXXX)" + trap "{ rm -fr ${tmp_dir}; }" INT TERM EXIT + + if [ -z "${dapm_tree}" ] + then + dapm_tree="/sys/kernel/debug/asoc/${card_name}" + fi + if [ -n "${remote_target}" ]; then + dapm_tree="${tmp_dir}/dapm-tree" + grab_remote_files "${remote_target}" "${card_name}" "${tmp_dir}" "${dapm_tree}" + fi + # In all cases now ${dapm_tree} contains the DAPM state + + process_dapm_tree "${tmp_dir}" "${dapm_tree}" + cp "${tmp_dir}/main.dot" "${dot_file}" + + if [ "${out_file}" != "${dot_file}" ]; then + dot -T"${out_fmt}" "${dot_file}" -o "${out_file}" + fi + + echo "Generated file ${out_file}" +} + +main "${@}" From 1324eafd37aa5c5b970c1b48a857acc81f0685c8 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sun, 7 Apr 2024 17:18:42 +0800 Subject: [PATCH 347/603] ASoc: PCM6240: Create PCM6240 Family driver code PCM6240 driver implements a flexible and configurable setting for register and filter coefficients, to one, two or even multiple PCM6240 Family Audio chips. Signed-off-by: Shenghao Ding Link: https://lore.kernel.org/r/20240407091846.1299-2-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/pcm6240.c | 2217 ++++++++++++++++++++++++++++++++++++ 1 file changed, 2217 insertions(+) create mode 100644 sound/soc/codecs/pcm6240.c diff --git a/sound/soc/codecs/pcm6240.c b/sound/soc/codecs/pcm6240.c new file mode 100644 index 0000000000000..93d592dbee668 --- /dev/null +++ b/sound/soc/codecs/pcm6240.c @@ -0,0 +1,2217 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments PCM6240 Family Audio ADC/DAC Device +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The PCM6240 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// PCM6240 Family chips. +// +// Author: Shenghao Ding +// + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "pcm6240.h" + +static const struct i2c_device_id pcmdevice_i2c_id[] = { + { "adc3120", ADC3120 }, + { "adc5120", ADC5120 }, + { "adc6120", ADC6120 }, + { "dix4192", DIX4192 }, + { "pcm1690", PCM1690 }, + { "pcm3120", PCM3120 }, + { "pcm3140", PCM3140 }, + { "pcm5120", PCM5120 }, + { "pcm5140", PCM5140 }, + { "pcm6120", PCM6120 }, + { "pcm6140", PCM6140 }, + { "pcm6240", PCM6240 }, + { "pcm6260", PCM6260 }, + { "pcm9211", PCM9211 }, + { "pcmd3140", PCMD3140 }, + { "pcmd3180", PCMD3180 }, + { "pcmd512x", PCMD512X }, + { "taa5212", TAA5212 }, + { "taa5412", TAA5412 }, + { "tad5212", TAD5212 }, + { "tad5412", TAD5412 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcmdevice_i2c_id); + +static const char *const pcmdev_ctrl_name[] = { + "%s i2c%d Dev%d Ch%d Ana Volume", + "%s i2c%d Dev%d Ch%d Digi Volume", + "%s i2c%d Dev%d Ch%d Fine Volume", +}; + +static const char *const pcmdev_ctrl_name_with_prefix[] = { + "%s Dev%d Ch%d Ana Volume", + "%s Dev%d Ch%d Digi Volume", + "%s Dev%d Ch%d Fine Volume", +}; + +static const struct pcmdevice_mixer_control adc5120_analog_gain_ctl[] = { + { + .shift = 1, + .reg = ADC5120_REG_CH1_ANALOG_GAIN, + .max = 0x54, + .invert = 0, + }, + { + .shift = 1, + .reg = ADC5120_REG_CH2_ANALOG_GAIN, + .max = 0x54, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control adc5120_digi_gain_ctl[] = { + { + .shift = 0, + .reg = ADC5120_REG_CH1_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = ADC5120_REG_CH2_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcm1690_digi_gain_ctl[] = { + { + .shift = 0, + .reg = PCM1690_REG_CH1_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM1690_REG_CH2_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM1690_REG_CH3_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM1690_REG_CH4_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM1690_REG_CH5_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM1690_REG_CH6_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM1690_REG_CH7_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM1690_REG_CH8_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcm6240_analog_gain_ctl[] = { + { + .shift = 2, + .reg = PCM6240_REG_CH1_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + }, + { + .shift = 2, + .reg = PCM6240_REG_CH2_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + }, + { + .shift = 2, + .reg = PCM6240_REG_CH3_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + }, + { + .shift = 2, + .reg = PCM6240_REG_CH4_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcm6240_digi_gain_ctl[] = { + { + .shift = 0, + .reg = PCM6240_REG_CH1_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM6240_REG_CH2_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM6240_REG_CH3_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM6240_REG_CH4_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcm6260_analog_gain_ctl[] = { + { + .shift = 2, + .reg = PCM6260_REG_CH1_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + }, + { + .shift = 2, + .reg = PCM6260_REG_CH2_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + }, + { + .shift = 2, + .reg = PCM6260_REG_CH3_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + }, + { + .shift = 2, + .reg = PCM6260_REG_CH4_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + }, + { + .shift = 2, + .reg = PCM6260_REG_CH5_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + }, + { + .shift = 2, + .reg = PCM6260_REG_CH6_ANALOG_GAIN, + .max = 0x42, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcm6260_digi_gain_ctl[] = { + { + .shift = 0, + .reg = PCM6260_REG_CH1_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM6260_REG_CH2_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM6260_REG_CH3_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM6260_REG_CH4_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM6260_REG_CH5_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM6260_REG_CH6_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcm9211_digi_gain_ctl[] = { + { + .shift = 0, + .reg = PCM9211_REG_CH1_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCM9211_REG_CH2_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcmd3140_digi_gain_ctl[] = { + { + .shift = 0, + .reg = PCMD3140_REG_CH1_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3140_REG_CH2_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3140_REG_CH3_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3140_REG_CH4_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcmd3140_fine_gain_ctl[] = { + { + .shift = 4, + .reg = PCMD3140_REG_CH1_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3140_REG_CH2_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3140_REG_CH3_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3140_REG_CH4_FINE_GAIN, + .max = 0xf, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcmd3180_digi_gain_ctl[] = { + { + .shift = 0, + .reg = PCMD3180_REG_CH1_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3180_REG_CH2_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3180_REG_CH3_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3180_REG_CH4_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3180_REG_CH5_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3180_REG_CH6_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3180_REG_CH7_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = PCMD3180_REG_CH8_DIGITAL_GAIN, + .max = 0xff, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control pcmd3180_fine_gain_ctl[] = { + { + .shift = 4, + .reg = PCMD3180_REG_CH1_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3180_REG_CH2_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3180_REG_CH3_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3180_REG_CH4_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3180_REG_CH5_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3180_REG_CH6_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3180_REG_CH7_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = PCMD3180_REG_CH8_FINE_GAIN, + .max = 0xf, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control taa5412_digi_vol_ctl[] = { + { + .shift = 0, + .reg = TAA5412_REG_CH1_DIGITAL_VOLUME, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = TAA5412_REG_CH2_DIGITAL_VOLUME, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = TAA5412_REG_CH3_DIGITAL_VOLUME, + .max = 0xff, + .invert = 0, + }, + { + .shift = 0, + .reg = TAA5412_REG_CH4_DIGITAL_VOLUME, + .max = 0xff, + .invert = 0, + } +}; + +static const struct pcmdevice_mixer_control taa5412_fine_gain_ctl[] = { + { + .shift = 4, + .reg = TAA5412_REG_CH1_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = TAA5412_REG_CH2_FINE_GAIN, + .max = 0xf, + .invert = 0, + }, + { + .shift = 4, + .reg = TAA5412_REG_CH3_FINE_GAIN, + .max = 0xf, + .invert = 4, + }, + { + .shift = 0, + .reg = TAA5412_REG_CH4_FINE_GAIN, + .max = 0xf, + .invert = 4, + } +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(pcmd3140_dig_gain_tlv, + -10000, 2700); +static const DECLARE_TLV_DB_MINMAX_MUTE(pcm1690_fine_dig_gain_tlv, + -12750, 0); +static const DECLARE_TLV_DB_MINMAX_MUTE(pcm1690_dig_gain_tlv, + -25500, 0); +static const DECLARE_TLV_DB_MINMAX_MUTE(pcm9211_dig_gain_tlv, + -11450, 2000); +static const DECLARE_TLV_DB_MINMAX_MUTE(adc5120_fgain_tlv, + -10050, 2700); +static const DECLARE_TLV_DB_LINEAR(adc5120_chgain_tlv, 0, 4200); +static const DECLARE_TLV_DB_MINMAX_MUTE(pcm6260_fgain_tlv, + -10000, 2700); +static const DECLARE_TLV_DB_LINEAR(pcm6260_chgain_tlv, 0, 4200); +static const DECLARE_TLV_DB_MINMAX_MUTE(taa5412_dig_vol_tlv, + -8050, 4700); +static const DECLARE_TLV_DB_LINEAR(taa5412_fine_gain_tlv, + -80, 70); + +static int pcmdev_change_dev(struct pcmdevice_priv *pcm_priv, + unsigned short dev_no) +{ + struct i2c_client *client = (struct i2c_client *)pcm_priv->client; + struct regmap *map = pcm_priv->regmap; + int ret; + + if (client->addr == pcm_priv->addr[dev_no]) + return 0; + + client->addr = pcm_priv->addr[dev_no]; + /* All pcmdevices share the same regmap, clear the page + * inside regmap once switching to another pcmdevice. + * Register 0 at any pages inside pcmdevice is the same + * one for page-switching. + */ + ret = regmap_write(map, PCMDEVICE_PAGE_SELECT, 0); + if (ret < 0) + dev_err(pcm_priv->dev, "%s: err = %d\n", __func__, ret); + + return ret; +} + +static int pcmdev_dev_read(struct pcmdevice_priv *pcm_dev, + unsigned int dev_no, unsigned int reg, unsigned int *val) +{ + struct regmap *map = pcm_dev->regmap; + int ret; + + if (dev_no >= pcm_dev->ndev) { + dev_err(pcm_dev->dev, "%s: no such channel(%d)\n", __func__, + dev_no); + return -EINVAL; + } + + ret = pcmdev_change_dev(pcm_dev, dev_no); + if (ret < 0) { + dev_err(pcm_dev->dev, "%s: chg dev err = %d\n", __func__, ret); + return ret; + } + + ret = regmap_read(map, reg, val); + if (ret < 0) + dev_err(pcm_dev->dev, "%s: err = %d\n", __func__, ret); + + return ret; +} + +static int pcmdev_dev_update_bits(struct pcmdevice_priv *pcm_dev, + unsigned int dev_no, unsigned int reg, unsigned int mask, + unsigned int value) +{ + struct regmap *map = pcm_dev->regmap; + int ret; + + if (dev_no >= pcm_dev->ndev) { + dev_err(pcm_dev->dev, "%s: no such channel(%d)\n", __func__, + dev_no); + return -EINVAL; + } + + ret = pcmdev_change_dev(pcm_dev, dev_no); + if (ret < 0) { + dev_err(pcm_dev->dev, "%s: chg dev err = %d\n", __func__, ret); + return ret; + } + + ret = regmap_update_bits(map, reg, mask, value); + if (ret < 0) + dev_err(pcm_dev->dev, "%s: update_bits err=%d\n", + __func__, ret); + + return ret; +} + +static int pcmdev_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, int vol_ctrl_type) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct pcmdevice_priv *pcm_dev = + snd_soc_component_get_drvdata(component); + struct pcmdevice_mixer_control *mc = + (struct pcmdevice_mixer_control *)kcontrol->private_value; + int max = mc->max, ret; + unsigned int mask = BIT(fls(max)) - 1; + unsigned int dev_no = mc->dev_no; + unsigned int shift = mc->shift; + unsigned int reg = mc->reg; + unsigned int val; + + mutex_lock(&pcm_dev->codec_lock); + + if (pcm_dev->chip_id == PCM1690) { + ret = pcmdev_dev_read(pcm_dev, dev_no, PCM1690_REG_MODE_CTRL, + &val); + if (ret) { + dev_err(pcm_dev->dev, "%s: read mode err=%d\n", + __func__, ret); + goto out; + } + val &= PCM1690_REG_MODE_CTRL_DAMS_MSK; + /* Set to wide-range mode, before using vol ctrl. */ + if (!val && vol_ctrl_type == PCMDEV_PCM1690_VOL_CTRL) { + ucontrol->value.integer.value[0] = -25500; + goto out; + } + /* Set to fine mode, before using fine vol ctrl. */ + if (val && vol_ctrl_type == PCMDEV_PCM1690_FINE_VOL_CTRL) { + ucontrol->value.integer.value[0] = -12750; + goto out; + } + } + + ret = pcmdev_dev_read(pcm_dev, dev_no, reg, &val); + if (ret) { + dev_err(pcm_dev->dev, "%s: read err=%d\n", + __func__, ret); + goto out; + } + + val = (val >> shift) & mask; + val = (val > max) ? max : val; + val = mc->invert ? max - val : val; + ucontrol->value.integer.value[0] = val; +out: + mutex_unlock(&pcm_dev->codec_lock); + return ret; +} + +static int pcmdevice_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return pcmdev_get_volsw(kcontrol, ucontrol, PCMDEV_GENERIC_VOL_CTRL); +} + +static int pcm1690_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return pcmdev_get_volsw(kcontrol, ucontrol, PCMDEV_PCM1690_VOL_CTRL); +} + +static int pcm1690_get_finevolsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return pcmdev_get_volsw(kcontrol, ucontrol, + PCMDEV_PCM1690_FINE_VOL_CTRL); +} + +static int pcmdev_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, int vol_ctrl_type) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct pcmdevice_priv *pcm_dev = + snd_soc_component_get_drvdata(component); + struct pcmdevice_mixer_control *mc = + (struct pcmdevice_mixer_control *)kcontrol->private_value; + int max = mc->max, rc; + unsigned int mask = BIT(fls(max)) - 1; + unsigned int dev_no = mc->dev_no; + unsigned int shift = mc->shift; + unsigned int val, val_mask; + unsigned int reg = mc->reg; + + mutex_lock(&pcm_dev->codec_lock); + val = ucontrol->value.integer.value[0] & mask; + val = (val > max) ? max : val; + val = mc->invert ? max - val : val; + val_mask = mask << shift; + val = val << shift; + + switch (vol_ctrl_type) { + case PCMDEV_PCM1690_VOL_CTRL: + val_mask |= PCM1690_REG_MODE_CTRL_DAMS_MSK; + val |= PCM1690_REG_MODE_CTRL_DAMS_WIDE_RANGE; + break; + case PCMDEV_PCM1690_FINE_VOL_CTRL: + val_mask |= PCM1690_REG_MODE_CTRL_DAMS_MSK; + val |= PCM1690_REG_MODE_CTRL_DAMS_FINE_STEP; + break; + } + + rc = pcmdev_dev_update_bits(pcm_dev, dev_no, reg, val_mask, val); + if (rc < 0) + dev_err(pcm_dev->dev, "%s: update_bits err = %d\n", + __func__, rc); + else + rc = 1; + mutex_unlock(&pcm_dev->codec_lock); + return rc; +} + +static int pcmdevice_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return pcmdev_put_volsw(kcontrol, ucontrol, PCMDEV_GENERIC_VOL_CTRL); +} + +static int pcm1690_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return pcmdev_put_volsw(kcontrol, ucontrol, PCMDEV_PCM1690_VOL_CTRL); +} + +static int pcm1690_put_finevolsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return pcmdev_put_volsw(kcontrol, ucontrol, + PCMDEV_PCM1690_FINE_VOL_CTRL); +} + +static const struct pcmdev_ctrl_info pcmdev_gain_ctl_info[][2] = { + // ADC3120 + { + { + .gain = adc5120_chgain_tlv, + .pcmdev_ctrl = adc5120_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = adc5120_fgain_tlv, + .pcmdev_ctrl = adc5120_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // ADC5120 + { + { + .gain = adc5120_chgain_tlv, + .pcmdev_ctrl = adc5120_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = adc5120_fgain_tlv, + .pcmdev_ctrl = adc5120_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // ADC6120 + { + { + .gain = adc5120_chgain_tlv, + .pcmdev_ctrl = adc5120_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = adc5120_fgain_tlv, + .pcmdev_ctrl = adc5120_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // DIX4192 + { + { + .ctrl_array_size = 0, + }, + { + .ctrl_array_size = 0, + }, + }, + // PCM1690 + { + { + .gain = pcm1690_fine_dig_gain_tlv, + .pcmdev_ctrl = pcm1690_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm1690_digi_gain_ctl), + .get = pcm1690_get_volsw, + .put = pcm1690_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + { + .gain = pcm1690_dig_gain_tlv, + .pcmdev_ctrl = pcm1690_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm1690_digi_gain_ctl), + .get = pcm1690_get_finevolsw, + .put = pcm1690_put_finevolsw, + .pcmdev_ctrl_name_id = 2, + }, + }, + // PCM3120 + { + { + .gain = adc5120_chgain_tlv, + .pcmdev_ctrl = adc5120_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = adc5120_fgain_tlv, + .pcmdev_ctrl = adc5120_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCM3140 + { + { + .gain = pcm6260_chgain_tlv, + .pcmdev_ctrl = pcm6240_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6240_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = pcm6260_fgain_tlv, + .pcmdev_ctrl = pcm6240_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6240_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCM5120 + { + { + .gain = adc5120_chgain_tlv, + .pcmdev_ctrl = adc5120_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = adc5120_fgain_tlv, + .pcmdev_ctrl = adc5120_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCM5140 + { + { + .gain = pcm6260_chgain_tlv, + .pcmdev_ctrl = pcm6240_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6240_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = pcm6260_fgain_tlv, + .pcmdev_ctrl = pcm6240_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6240_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCM6120 + { + { + .gain = adc5120_chgain_tlv, + .pcmdev_ctrl = adc5120_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = adc5120_fgain_tlv, + .pcmdev_ctrl = adc5120_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(adc5120_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCM6140 + { + { + .gain = pcm6260_chgain_tlv, + .pcmdev_ctrl = pcm6240_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6240_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = pcm6260_fgain_tlv, + .pcmdev_ctrl = pcm6240_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6240_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCM6240 + { + { + .gain = pcm6260_chgain_tlv, + .pcmdev_ctrl = pcm6240_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6240_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = pcm6260_fgain_tlv, + .pcmdev_ctrl = pcm6240_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6240_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCM6260 + { + { + .gain = pcm6260_chgain_tlv, + .pcmdev_ctrl = pcm6260_analog_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6260_analog_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 0, + }, + { + .gain = pcm6260_fgain_tlv, + .pcmdev_ctrl = pcm6260_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm6260_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCM9211 + { + { + .ctrl_array_size = 0, + }, + { + .gain = pcm9211_dig_gain_tlv, + .pcmdev_ctrl = pcm9211_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcm9211_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + + }, + // PCMD3140 + { + { + .gain = taa5412_fine_gain_tlv, + .pcmdev_ctrl = pcmd3140_fine_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcmd3140_fine_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 2, + }, + { + .gain = pcmd3140_dig_gain_tlv, + .pcmdev_ctrl = pcmd3140_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcmd3140_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCMD3180 + { + { + .gain = taa5412_fine_gain_tlv, + .pcmdev_ctrl = pcmd3180_fine_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcmd3180_fine_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 2, + }, + { + .gain = pcmd3140_dig_gain_tlv, + .pcmdev_ctrl = pcmd3180_digi_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(pcmd3180_digi_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // PCMD512X + { + { + .ctrl_array_size = 0, + }, + { + .ctrl_array_size = 0, + }, + }, + // TAA5212 + { + { + .gain = taa5412_fine_gain_tlv, + .pcmdev_ctrl = taa5412_fine_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(taa5412_fine_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 2, + }, + { + .gain = taa5412_dig_vol_tlv, + .pcmdev_ctrl = taa5412_digi_vol_ctl, + .ctrl_array_size = ARRAY_SIZE(taa5412_digi_vol_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // TAA5412 + { + { + .gain = taa5412_fine_gain_tlv, + .pcmdev_ctrl = taa5412_fine_gain_ctl, + .ctrl_array_size = ARRAY_SIZE(taa5412_fine_gain_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 2, + }, + { + .gain = taa5412_dig_vol_tlv, + .pcmdev_ctrl = taa5412_digi_vol_ctl, + .ctrl_array_size = ARRAY_SIZE(taa5412_digi_vol_ctl), + .get = pcmdevice_get_volsw, + .put = pcmdevice_put_volsw, + .pcmdev_ctrl_name_id = 1, + }, + }, + // TAD5212 + { + { + .ctrl_array_size = 0, + }, + { + .ctrl_array_size = 0, + }, + }, + // TAD5412 + { + { + .ctrl_array_size = 0, + }, + { + .ctrl_array_size = 0, + }, + }, +}; + +static int pcmdev_dev_bulk_write(struct pcmdevice_priv *pcm_dev, + unsigned int dev_no, unsigned int reg, unsigned char *data, + unsigned int len) +{ + struct regmap *map = pcm_dev->regmap; + int ret; + + if (dev_no >= pcm_dev->ndev) { + dev_err(pcm_dev->dev, "%s: no such channel(%d)\n", __func__, + dev_no); + return -EINVAL; + } + + ret = pcmdev_change_dev(pcm_dev, dev_no); + if (ret < 0) { + dev_err(pcm_dev->dev, "%s: chg dev err = %d\n", __func__, ret); + return ret; + } + + ret = regmap_bulk_write(map, reg, data, len); + if (ret < 0) + dev_err(pcm_dev->dev, "%s: bulk_write err = %d\n", __func__, + ret); + + return ret; +} + +static int pcmdev_dev_write(struct pcmdevice_priv *pcm_dev, + unsigned int dev_no, unsigned int reg, unsigned int value) +{ + struct regmap *map = pcm_dev->regmap; + int ret; + + if (dev_no >= pcm_dev->ndev) { + dev_err(pcm_dev->dev, "%s: no such channel(%d)\n", __func__, + dev_no); + return -EINVAL; + } + + ret = pcmdev_change_dev(pcm_dev, dev_no); + if (ret < 0) { + dev_err(pcm_dev->dev, "%s: chg dev err = %d\n", __func__, ret); + return ret; + } + + ret = regmap_write(map, reg, value); + if (ret < 0) + dev_err(pcm_dev->dev, "%s: err = %d\n", __func__, ret); + + return ret; +} + +static int pcmdevice_info_profile( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct pcmdevice_priv *pcm_dev = + snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max(0, pcm_dev->regbin.ncfgs - 1); + + return 0; +} + +static int pcmdevice_get_profile_id( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct pcmdevice_priv *pcm_dev = + snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm_dev->cur_conf; + + return 0; +} + +static int pcmdevice_set_profile_id( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct pcmdevice_priv *pcm_dev = + snd_soc_component_get_drvdata(codec); + int nr_profile = ucontrol->value.integer.value[0]; + int max = pcm_dev->regbin.ncfgs - 1; + int ret = 0; + + nr_profile = clamp(nr_profile, 0, max); + + if (pcm_dev->cur_conf != nr_profile) { + pcm_dev->cur_conf = nr_profile; + ret = 1; + } + + return ret; +} + +static int pcmdevice_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct pcmdevice_mixer_control *mc = + (struct pcmdevice_mixer_control *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mc->max; + return 0; +} + +static void pcm9211_sw_rst(struct pcmdevice_priv *pcm_dev) +{ + int ret, i; + + for (i = 0; i < pcm_dev->ndev; i++) { + ret = pcmdev_dev_update_bits(pcm_dev, i, + PCM9211_REG_SW_CTRL, PCM9211_REG_SW_CTRL_MRST_MSK, + PCM9211_REG_SW_CTRL_MRST); + if (ret < 0) + dev_err(pcm_dev->dev, "%s: dev %d swreset fail %d\n", + __func__, i, ret); + } +} + +static void pcmdevice_sw_rst(struct pcmdevice_priv *pcm_dev) +{ + int ret, i; + + for (i = 0; i < pcm_dev->ndev; i++) { + ret = pcmdev_dev_write(pcm_dev, i, PCMDEVICE_REG_SWRESET, + PCMDEVICE_REG_SWRESET_RESET); + if (ret < 0) + dev_err(pcm_dev->dev, "%s: dev %d swreset fail %d\n", + __func__, i, ret); + } +} + +static struct pcmdevice_config_info *pcmdevice_add_config(void *ctxt, + const unsigned char *config_data, unsigned int config_size, + int *status) +{ + struct pcmdevice_priv *pcm_dev = (struct pcmdevice_priv *)ctxt; + struct pcmdevice_config_info *cfg_info; + struct pcmdevice_block_data **bk_da; + unsigned int config_offset = 0, i; + + cfg_info = kzalloc(sizeof(struct pcmdevice_config_info), GFP_KERNEL); + if (!cfg_info) { + *status = -ENOMEM; + goto out; + } + + if (pcm_dev->regbin.fw_hdr.binary_version_num >= 0x105) { + if (config_offset + 64 > (int)config_size) { + *status = -EINVAL; + dev_err(pcm_dev->dev, + "%s: cfg_name out of boundary\n", __func__); + goto out; + } + memcpy(cfg_info->cfg_name, &config_data[config_offset], 64); + config_offset += 64; + } + + if (config_offset + 4 > config_size) { + *status = -EINVAL; + dev_err(pcm_dev->dev, "%s: nblocks out of boundary\n", + __func__); + goto out; + } + cfg_info->nblocks = + get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + bk_da = cfg_info->blk_data = kcalloc(cfg_info->nblocks, + sizeof(struct pcmdevice_block_data *), GFP_KERNEL); + if (!bk_da) { + *status = -ENOMEM; + goto out; + } + cfg_info->real_nblocks = 0; + for (i = 0; i < cfg_info->nblocks; i++) { + if (config_offset + 12 > config_size) { + *status = -EINVAL; + dev_err(pcm_dev->dev, + "%s: out of boundary i = %d nblocks = %u\n", + __func__, i, cfg_info->nblocks); + break; + } + bk_da[i] = kzalloc(sizeof(struct pcmdevice_block_data), + GFP_KERNEL); + if (!bk_da[i]) { + *status = -ENOMEM; + break; + } + bk_da[i]->dev_idx = config_data[config_offset]; + config_offset++; + + bk_da[i]->block_type = config_data[config_offset]; + config_offset++; + + if (bk_da[i]->block_type == PCMDEVICE_BIN_BLK_PRE_POWER_UP) { + if (bk_da[i]->dev_idx == 0) + cfg_info->active_dev = + (1 << pcm_dev->ndev) - 1; + else + cfg_info->active_dev = + 1 << (bk_da[i]->dev_idx - 1); + } + + bk_da[i]->yram_checksum = + get_unaligned_be16(&config_data[config_offset]); + config_offset += 2; + bk_da[i]->block_size = + get_unaligned_be32(&config_data[config_offset]); + config_offset += 4; + + bk_da[i]->n_subblks = + get_unaligned_be32(&config_data[config_offset]); + + config_offset += 4; + + if (config_offset + bk_da[i]->block_size > config_size) { + *status = -EINVAL; + dev_err(pcm_dev->dev, + "%s: out of boundary: i = %d blks = %u\n", + __func__, i, cfg_info->nblocks); + break; + } + + bk_da[i]->regdata = kmemdup(&config_data[config_offset], + bk_da[i]->block_size, GFP_KERNEL); + if (!bk_da[i]->regdata) { + *status = -ENOMEM; + goto out; + } + config_offset += bk_da[i]->block_size; + cfg_info->real_nblocks += 1; + } +out: + return cfg_info; +} + +static int pcmdev_gain_ctrl_add(struct pcmdevice_priv *pcm_dev, + int dev_no, int ctl_id) +{ + struct i2c_adapter *adap = pcm_dev->client->adapter; + struct snd_soc_component *comp = pcm_dev->component; + struct pcmdevice_mixer_control *pcmdev_ctrl; + struct snd_kcontrol_new *pcmdev_controls; + int ret, mix_index = 0, name_id, chn; + unsigned int id = pcm_dev->chip_id; + const int nr_chn = + pcmdev_gain_ctl_info[id][ctl_id].ctrl_array_size; + const char *ctrl_name; + char *name; + + if (!nr_chn) { + dev_dbg(pcm_dev->dev, "%s: no gain ctrl for %s\n", __func__, + pcm_dev->dev_name); + return 0; + } + + pcmdev_controls = devm_kzalloc(pcm_dev->dev, + nr_chn * sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!pcmdev_controls) + return -ENOMEM; + + name_id = pcmdev_gain_ctl_info[id][ctl_id].pcmdev_ctrl_name_id; + + if (comp->name_prefix) + ctrl_name = pcmdev_ctrl_name_with_prefix[name_id]; + else + ctrl_name = pcmdev_ctrl_name[name_id]; + + for (chn = 1; chn <= nr_chn; chn++) { + name = devm_kzalloc(pcm_dev->dev, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto out; + } + if (comp->name_prefix) + scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + ctrl_name, comp->name_prefix, dev_no, chn); + else + scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + ctrl_name, pcm_dev->upper_dev_name, adap->nr, + dev_no, chn); + pcmdev_controls[mix_index].tlv.p = + pcmdev_gain_ctl_info[id][ctl_id].gain; + pcmdev_ctrl = devm_kmemdup(pcm_dev->dev, + &pcmdev_gain_ctl_info[id][ctl_id].pcmdev_ctrl[chn - 1], + sizeof(*pcmdev_ctrl), GFP_KERNEL); + if (!pcmdev_ctrl) { + ret = -ENOMEM; + goto out; + } + pcmdev_ctrl->dev_no = dev_no; + pcmdev_controls[mix_index].private_value = + (unsigned long)pcmdev_ctrl; + pcmdev_controls[mix_index].name = name; + pcmdev_controls[mix_index].access = + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE; + pcmdev_controls[mix_index].iface = + SNDRV_CTL_ELEM_IFACE_MIXER; + pcmdev_controls[mix_index].info = pcmdevice_info_volsw; + pcmdev_controls[mix_index].get = + pcmdev_gain_ctl_info[id][ctl_id].get; + pcmdev_controls[mix_index].put = + pcmdev_gain_ctl_info[id][ctl_id].put; + mix_index++; + } + + ret = snd_soc_add_component_controls(comp, pcmdev_controls, mix_index); + if (ret) + dev_err(pcm_dev->dev, "%s: add_controls err = %d\n", + __func__, ret); +out: + return ret; +} + +static int pcmdev_profile_ctrl_add(struct pcmdevice_priv *pcm_dev) +{ + struct snd_soc_component *comp = pcm_dev->component; + struct i2c_adapter *adap = pcm_dev->client->adapter; + struct snd_kcontrol_new *pcmdev_ctrl; + char *name; + int ret; + + pcmdev_ctrl = devm_kzalloc(pcm_dev->dev, + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!pcmdev_ctrl) + return -ENOMEM; + + /* Create a mixer item for selecting the active profile */ + name = devm_kzalloc(pcm_dev->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + GFP_KERNEL); + if (!name) + return -ENOMEM; + + if (comp->name_prefix) + scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "%s Profile id", comp->name_prefix); + else + scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "%s i2c%d Profile id", pcm_dev->upper_dev_name, + adap->nr); + pcmdev_ctrl->name = name; + pcmdev_ctrl->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + pcmdev_ctrl->info = pcmdevice_info_profile; + pcmdev_ctrl->get = pcmdevice_get_profile_id; + pcmdev_ctrl->put = pcmdevice_set_profile_id; + + ret = snd_soc_add_component_controls(comp, pcmdev_ctrl, 1); + if (ret) + dev_err(pcm_dev->dev, "%s: add_controls err = %d\n", + __func__, ret); + + return ret; +} + +static void pcmdevice_config_info_remove(void *ctxt) +{ + struct pcmdevice_priv *pcm_dev = (struct pcmdevice_priv *) ctxt; + struct pcmdevice_regbin *regbin = &(pcm_dev->regbin); + struct pcmdevice_config_info **cfg_info = regbin->cfg_info; + int i, j; + + if (!cfg_info) + return; + for (i = 0; i < regbin->ncfgs; i++) { + if (!cfg_info[i]) + continue; + if (cfg_info[i]->blk_data) { + for (j = 0; j < (int)cfg_info[i]->real_nblocks; j++) { + if (!cfg_info[i]->blk_data[j]) + continue; + kfree(cfg_info[i]->blk_data[j]->regdata); + kfree(cfg_info[i]->blk_data[j]); + } + kfree(cfg_info[i]->blk_data); + } + kfree(cfg_info[i]); + } + kfree(cfg_info); +} + +static int pcmdev_regbin_ready(const struct firmware *fmw, void *ctxt) +{ + struct pcmdevice_config_info **cfg_info; + struct pcmdevice_priv *pcm_dev = ctxt; + struct pcmdevice_regbin_hdr *fw_hdr; + struct pcmdevice_regbin *regbin; + unsigned int total_config_sz = 0; + int offset = 0, ret = 0, i; + unsigned char *buf; + + regbin = &(pcm_dev->regbin); + fw_hdr = &(regbin->fw_hdr); + if (!fmw || !fmw->data) { + dev_err(pcm_dev->dev, "%s: failed to read %s\n", + __func__, pcm_dev->bin_name); + pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED; + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + fw_hdr->img_sz = get_unaligned_be32(&buf[offset]); + offset += 4; + if (fw_hdr->img_sz != fmw->size) { + dev_err(pcm_dev->dev, "%s: file size(%d) not match %u", + __func__, (int)fmw->size, fw_hdr->img_sz); + pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED; + ret = -EINVAL; + goto out; + } + + fw_hdr->checksum = get_unaligned_be32(&buf[offset]); + offset += 4; + fw_hdr->binary_version_num = get_unaligned_be32(&buf[offset]); + if (fw_hdr->binary_version_num < 0x103) { + dev_err(pcm_dev->dev, "%s: bin version 0x%04x is out of date", + __func__, fw_hdr->binary_version_num); + pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED; + ret = -EINVAL; + goto out; + } + offset += 4; + fw_hdr->drv_fw_version = get_unaligned_be32(&buf[offset]); + offset += 8; + fw_hdr->plat_type = buf[offset]; + offset += 1; + fw_hdr->dev_family = buf[offset]; + offset += 1; + fw_hdr->reserve = buf[offset]; + offset += 1; + fw_hdr->ndev = buf[offset]; + offset += 1; + if (fw_hdr->ndev != pcm_dev->ndev) { + dev_err(pcm_dev->dev, "%s: invalid ndev(%u)\n", __func__, + fw_hdr->ndev); + pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED; + ret = -EINVAL; + goto out; + } + + if (offset + PCMDEVICE_MAX_REGBIN_DEVICES > fw_hdr->img_sz) { + dev_err(pcm_dev->dev, "%s: devs out of boundary!\n", __func__); + pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED; + ret = -EINVAL; + goto out; + } + + for (i = 0; i < PCMDEVICE_MAX_REGBIN_DEVICES; i++, offset++) + fw_hdr->devs[i] = buf[offset]; + + fw_hdr->nconfig = get_unaligned_be32(&buf[offset]); + offset += 4; + + for (i = 0; i < PCMDEVICE_CONFIG_SUM; i++) { + fw_hdr->config_size[i] = get_unaligned_be32(&buf[offset]); + offset += 4; + total_config_sz += fw_hdr->config_size[i]; + } + + if (fw_hdr->img_sz - total_config_sz != (unsigned int)offset) { + dev_err(pcm_dev->dev, "%s: bin file error!\n", __func__); + pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED; + ret = -EINVAL; + goto out; + } + cfg_info = kcalloc(fw_hdr->nconfig, sizeof(*cfg_info), GFP_KERNEL); + if (!cfg_info) { + pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED; + ret = -ENOMEM; + goto out; + } + regbin->cfg_info = cfg_info; + regbin->ncfgs = 0; + for (i = 0; i < (int)fw_hdr->nconfig; i++) { + cfg_info[i] = pcmdevice_add_config(ctxt, &buf[offset], + fw_hdr->config_size[i], &ret); + if (ret) { + /* In case the bin file is partially destroyed. */ + if (regbin->ncfgs == 0) + pcm_dev->fw_state = PCMDEVICE_FW_LOAD_FAILED; + break; + } + offset += (int)fw_hdr->config_size[i]; + regbin->ncfgs += 1; + } + +out: + if (pcm_dev->fw_state == PCMDEVICE_FW_LOAD_FAILED) { + dev_err(pcm_dev->dev, + "%s: remove config due to fw load error!\n", __func__); + pcmdevice_config_info_remove(pcm_dev); + } + + return ret; +} + +static int pcmdevice_comp_probe(struct snd_soc_component *comp) +{ + struct pcmdevice_priv *pcm_dev = snd_soc_component_get_drvdata(comp); + struct i2c_adapter *adap = pcm_dev->client->adapter; + const struct firmware *fw_entry = NULL; + int ret, i, j; + + mutex_lock(&pcm_dev->codec_lock); + + pcm_dev->component = comp; + + for (i = 0; i < pcm_dev->ndev; i++) { + for (j = 0; j < 2; j++) { + ret = pcmdev_gain_ctrl_add(pcm_dev, i, j); + if (ret < 0) + goto out; + } + } + + if (comp->name_prefix) { + /* There's name_prefix defined in DTS. Bin file name will be + * name_prefix.bin stores the firmware including register + * setting and params for different filters inside chips, it + * must be copied into firmware folder. The same types of + * pcmdevices sitting on the same i2c bus will be aggregated as + * one single codec, all of them share the same bin file. + */ + scnprintf(pcm_dev->bin_name, PCMDEVICE_BIN_FILENAME_LEN, + "%s.bin", comp->name_prefix); + } else { + /* There's NO name_prefix defined in DTS. Bin file name will be + * device-name[defined in pcmdevice_i2c_id]-i2c-bus_id + * [0,1,...,N]-sum[1,...,4]dev.bin stores the firmware + * including register setting and params for different filters + * inside chips, it must be copied into firmware folder. The + * same types of pcmdevices sitting on the same i2c bus will be + * aggregated as one single codec, all of them share the same + * bin file. + */ + scnprintf(pcm_dev->bin_name, PCMDEVICE_BIN_FILENAME_LEN, + "%s-i2c-%d-%udev.bin", pcm_dev->dev_name, adap->nr, + pcm_dev->ndev); + } + + ret = request_firmware(&fw_entry, pcm_dev->bin_name, pcm_dev->dev); + if (ret) { + dev_err(pcm_dev->dev, "%s: request %s err = %d\n", __func__, + pcm_dev->bin_name, ret); + goto out; + } + + ret = pcmdev_regbin_ready(fw_entry, pcm_dev); + if (ret) { + dev_err(pcm_dev->dev, "%s: %s parse err = %d\n", __func__, + pcm_dev->bin_name, ret); + goto out; + } + ret = pcmdev_profile_ctrl_add(pcm_dev); +out: + if (fw_entry) + release_firmware(fw_entry); + + mutex_unlock(&pcm_dev->codec_lock); + return ret; +} + + +static void pcmdevice_comp_remove(struct snd_soc_component *codec) +{ + struct pcmdevice_priv *pcm_dev = snd_soc_component_get_drvdata(codec); + + if (!pcm_dev) + return; + mutex_lock(&pcm_dev->codec_lock); + pcmdevice_config_info_remove(pcm_dev); + mutex_unlock(&pcm_dev->codec_lock); +} + +static const struct snd_soc_dapm_widget pcmdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI1 OUT", "ASI1 Capture", + 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("MIC"), +}; + +static const struct snd_soc_dapm_route pcmdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI1 OUT", NULL, "MIC"}, +}; + +static const struct snd_soc_component_driver + soc_codec_driver_pcmdevice = { + .probe = pcmdevice_comp_probe, + .remove = pcmdevice_comp_remove, + .dapm_widgets = pcmdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcmdevice_dapm_widgets), + .dapm_routes = pcmdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(pcmdevice_audio_map), + .suspend_bias_off = 1, + .idle_bias_on = 0, + .use_pmdown_time = 1, + .endianness = 1, +}; + +static int pcmdev_single_byte_wr(struct pcmdevice_priv *pcm_dev, + unsigned char *data, int devn, int sublocksize) +{ + unsigned short len = get_unaligned_be16(&data[2]); + int offset = 2; + int i, ret; + + offset += 2; + if (offset + 4 * len > sublocksize) { + dev_err(pcm_dev->dev, "%s: dev-%d byt wr out of boundary\n", + __func__, devn); + return -EINVAL; + } + + for (i = 0; i < len; i++) { + ret = pcmdev_dev_write(pcm_dev, devn, + PCMDEVICE_REG(data[offset + 1], data[offset + 2]), + data[offset + 3]); + /* skip this error for next operation or next devices */ + if (ret < 0) + dev_err(pcm_dev->dev, "%s: dev-%d single write err\n", + __func__, devn); + + offset += 4; + } + + return offset; +} + +static int pcmdev_burst_wr(struct pcmdevice_priv *pcm_dev, + unsigned char *data, int devn, int sublocksize) +{ + unsigned short len = get_unaligned_be16(&data[2]); + int offset = 2; + int ret; + + offset += 2; + if (offset + 4 + len > sublocksize) { + dev_err(pcm_dev->dev, "%s: dev-%d burst Out of boundary\n", + __func__, devn); + return -EINVAL; + } + if (len % 4) { + dev_err(pcm_dev->dev, "%s: dev-%d bst-len(%u) not div by 4\n", + __func__, devn, len); + return -EINVAL; + } + ret = pcmdev_dev_bulk_write(pcm_dev, devn, + PCMDEVICE_REG(data[offset + 1], data[offset + 2]), + &(data[offset + 4]), len); + /* skip this error for next devices */ + if (ret < 0) + dev_err(pcm_dev->dev, "%s: dev-%d bulk_write err = %d\n", + __func__, devn, ret); + + offset += (len + 4); + + return offset; +} + +static int pcmdev_delay(struct pcmdevice_priv *pcm_dev, + unsigned char *data, int devn, int sublocksize) +{ + unsigned int delay_time = 0; + int offset = 2; + + if (offset + 2 > sublocksize) { + dev_err(pcm_dev->dev, "%s: dev-%d deley out of boundary\n", + __func__, devn); + return -EINVAL; + } + delay_time = get_unaligned_be16(&data[2]) * 1000; + usleep_range(delay_time, delay_time + 50); + offset += 2; + + return offset; +} + +static int pcmdev_bits_wr(struct pcmdevice_priv *pcm_dev, + unsigned char *data, int devn, int sublocksize) +{ + int offset = 2; + int ret; + + if (offset + 6 > sublocksize) { + dev_err(pcm_dev->dev, "%s: dev-%d bit write out of memory\n", + __func__, devn); + return -EINVAL; + } + ret = pcmdev_dev_update_bits(pcm_dev, devn, + PCMDEVICE_REG(data[offset + 3], data[offset + 4]), + data[offset + 1], data[offset + 5]); + /* skip this error for next devices */ + if (ret < 0) + dev_err(pcm_dev->dev, "%s: dev-%d update_bits err = %d\n", + __func__, devn, ret); + + offset += 6; + + return offset; +} + +static int pcmdevice_process_block(void *ctxt, unsigned char *data, + unsigned char dev_idx, int sublocksize) +{ + struct pcmdevice_priv *pcm_dev = (struct pcmdevice_priv *)ctxt; + int devn, dev_end, ret = 0; + unsigned char subblk_typ = data[1]; + + if (dev_idx) { + devn = dev_idx - 1; + dev_end = dev_idx; + } else { + devn = 0; + dev_end = pcm_dev->ndev; + } + + /* loop in case of several devices sharing the same sub-block */ + for (; devn < dev_end; devn++) { + switch (subblk_typ) { + case PCMDEVICE_CMD_SING_W: + ret = pcmdev_single_byte_wr(pcm_dev, data, devn, sublocksize); + break; + case PCMDEVICE_CMD_BURST: + ret = pcmdev_burst_wr(pcm_dev, data, devn, sublocksize); + break; + case PCMDEVICE_CMD_DELAY: + ret = pcmdev_delay(pcm_dev, data, devn, sublocksize); + break; + case PCMDEVICE_CMD_FIELD_W: + ret = pcmdev_bits_wr(pcm_dev, data, devn, sublocksize); + break; + default: + break; + } + /* + * In case of sub-block error, break the loop for the rest of + * devices. + */ + if (ret < 0) + break; + } + + return ret; +} + +static void pcmdevice_select_cfg_blk(void *ctxt, int conf_no, + unsigned char block_type) +{ + struct pcmdevice_priv *pcm_dev = (struct pcmdevice_priv *)ctxt; + struct pcmdevice_regbin *regbin = &(pcm_dev->regbin); + struct pcmdevice_config_info **cfg_info = regbin->cfg_info; + struct pcmdevice_block_data **blk_data; + int j, k; + + if (conf_no >= regbin->ncfgs || conf_no < 0 || NULL == cfg_info) { + dev_err(pcm_dev->dev, "%s: conf_no should be less than %u\n", + __func__, regbin->ncfgs); + goto out; + } + blk_data = cfg_info[conf_no]->blk_data; + + for (j = 0; j < (int)cfg_info[conf_no]->real_nblocks; j++) { + unsigned int length = 0, ret; + + if (block_type > 5 || block_type < 2) { + dev_err(pcm_dev->dev, + "%s: block_type should be out of range\n", + __func__); + goto out; + } + if (block_type != blk_data[j]->block_type) + continue; + + for (k = 0; k < (int)blk_data[j]->n_subblks; k++) { + ret = pcmdevice_process_block(pcm_dev, + blk_data[j]->regdata + length, + blk_data[j]->dev_idx, + blk_data[j]->block_size - length); + length += ret; + if (blk_data[j]->block_size < length) { + dev_err(pcm_dev->dev, + "%s: %u %u out of boundary\n", + __func__, length, + blk_data[j]->block_size); + break; + } + } + if (length != blk_data[j]->block_size) + dev_err(pcm_dev->dev, "%s: %u %u size is not same\n", + __func__, length, blk_data[j]->block_size); + } + +out: + return; +} + +static int pcmdevice_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *codec = dai->component; + struct pcmdevice_priv *pcm_dev = snd_soc_component_get_drvdata(codec); + unsigned char block_type; + + if (pcm_dev->fw_state == PCMDEVICE_FW_LOAD_FAILED) { + dev_err(pcm_dev->dev, "%s: bin file not loaded\n", __func__); + return -EINVAL; + } + + if (mute) + block_type = PCMDEVICE_BIN_BLK_PRE_SHUTDOWN; + else + block_type = PCMDEVICE_BIN_BLK_PRE_POWER_UP; + + mutex_lock(&pcm_dev->codec_lock); + pcmdevice_select_cfg_blk(pcm_dev, pcm_dev->cur_conf, block_type); + mutex_unlock(&pcm_dev->codec_lock); + return 0; +} + +static int pcmdevice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct pcmdevice_priv *pcm_dev = snd_soc_dai_get_drvdata(dai); + unsigned int fsrate; + unsigned int slot_width; + int bclk_rate; + int ret = 0; + + fsrate = params_rate(params); + switch (fsrate) { + case 48000: + break; + case 44100: + break; + default: + dev_err(pcm_dev->dev, "%s: incorrect sample rate = %u\n", + __func__, fsrate); + ret = -EINVAL; + goto out; + } + + slot_width = params_width(params); + switch (slot_width) { + case 16: + break; + case 20: + break; + case 24: + break; + case 32: + break; + default: + dev_err(pcm_dev->dev, "%s: incorrect slot width = %u\n", + __func__, slot_width); + ret = -EINVAL; + goto out; + } + + bclk_rate = snd_soc_params_to_bclk(params); + if (bclk_rate < 0) { + dev_err(pcm_dev->dev, "%s: incorrect bclk rate = %d\n", + __func__, bclk_rate); + ret = bclk_rate; + } + +out: + return ret; +} + +static const struct snd_soc_dai_ops pcmdevice_dai_ops = { + .mute_stream = pcmdevice_mute, + .hw_params = pcmdevice_hw_params, +}; + +static struct snd_soc_dai_driver pcmdevice_dai_driver[] = { + { + .name = "pcmdevice-codec", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = PCMDEVICE_MAX_CHANNELS, + .rates = PCMDEVICE_RATES, + .formats = PCMDEVICE_FORMATS, + }, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = PCMDEVICE_MAX_CHANNELS, + .rates = PCMDEVICE_RATES, + .formats = PCMDEVICE_FORMATS, + }, + .ops = &pcmdevice_dai_ops, + .symmetric_rate = 1, + } +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcmdevice_of_match[] = { + { .compatible = "ti,adc3120" }, + { .compatible = "ti,adc5120" }, + { .compatible = "ti,adc6120" }, + { .compatible = "ti,dix4192" }, + { .compatible = "ti,pcm1690" }, + { .compatible = "ti,pcm3120" }, + { .compatible = "ti,pcm3140" }, + { .compatible = "ti,pcm5120" }, + { .compatible = "ti,pcm5140" }, + { .compatible = "ti,pcm6120" }, + { .compatible = "ti,pcm6140" }, + { .compatible = "ti,pcm6240" }, + { .compatible = "ti,pcm6260" }, + { .compatible = "ti,pcm9211" }, + { .compatible = "ti,pcmd3140" }, + { .compatible = "ti,pcmd3180" }, + { .compatible = "ti,pcmd512x" }, + { .compatible = "ti,taa5212" }, + { .compatible = "ti,taa5412" }, + { .compatible = "ti,tad5212" }, + { .compatible = "ti,tad5412" }, + {}, +}; +MODULE_DEVICE_TABLE(of, pcmdevice_of_match); +#endif + +static const struct regmap_range_cfg pcmdevice_ranges[] = { + { + .range_min = 0, + .range_max = 256 * 128, + .selector_reg = PCMDEVICE_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config pcmdevice_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .cache_type = REGCACHE_MAPLE, + .ranges = pcmdevice_ranges, + .num_ranges = ARRAY_SIZE(pcmdevice_ranges), + .max_register = 256 * 128, +}; + +static void pcmdevice_remove(struct pcmdevice_priv *pcm_dev) +{ + if (gpio_is_valid(pcm_dev->irq_info.gpio)) { + gpio_free(pcm_dev->irq_info.gpio); + free_irq(pcm_dev->irq_info.nmb, pcm_dev); + } + mutex_destroy(&pcm_dev->codec_lock); +} + +static char *str_to_upper(char *str) +{ + char *orig = str; + + if (!str) + return NULL; + + while (*str) { + *str = toupper(*str); + str++; + } + + return orig; +} + +static int pcmdevice_i2c_probe(struct i2c_client *i2c) +{ + const struct i2c_device_id *id = i2c_match_id(pcmdevice_i2c_id, i2c); + struct pcmdevice_priv *pcm_dev; + struct device_node *np; + unsigned int dev_addrs[PCMDEVICE_MAX_I2C_DEVICES]; + int ret = 0, i = 0, ndev = 0; +#ifdef CONFIG_OF + const __be32 *reg, *reg_end; + int len, sw, aw; +#endif + + pcm_dev = devm_kzalloc(&i2c->dev, sizeof(*pcm_dev), GFP_KERNEL); + if (!pcm_dev) { + ret = -ENOMEM; + goto out; + } + + pcm_dev->chip_id = (id != NULL) ? id->driver_data : 0; + + pcm_dev->dev = &i2c->dev; + pcm_dev->client = i2c; + + if (pcm_dev->chip_id >= MAX_DEVICE) + pcm_dev->chip_id = 0; + + strscpy(pcm_dev->dev_name, pcmdevice_i2c_id[pcm_dev->chip_id].name, + sizeof(pcm_dev->dev_name)); + + strscpy(pcm_dev->upper_dev_name, + pcmdevice_i2c_id[pcm_dev->chip_id].name, + sizeof(pcm_dev->upper_dev_name)); + + str_to_upper(pcm_dev->upper_dev_name); + + pcm_dev->regmap = devm_regmap_init_i2c(i2c, &pcmdevice_i2c_regmap); + if (IS_ERR(pcm_dev->regmap)) { + ret = PTR_ERR(pcm_dev->regmap); + dev_err(&i2c->dev, "%s: failed to allocate register map: %d\n", + __func__, ret); + goto out; + } + + i2c_set_clientdata(i2c, pcm_dev); + mutex_init(&pcm_dev->codec_lock); + np = pcm_dev->dev->of_node; +#ifdef CONFIG_OF + aw = of_n_addr_cells(np); + sw = of_n_size_cells(np); + if (sw == 0) { + reg = (const __be32 *)of_get_property(np, + "reg", &len); + reg_end = reg + len/sizeof(*reg); + ndev = 0; + do { + dev_addrs[ndev] = of_read_number(reg, aw); + reg += aw; + ndev++; + } while (reg < reg_end); + } else { + ndev = 1; + dev_addrs[0] = i2c->addr; + } +#else + ndev = 1; + dev_addrs[0] = i2c->addr; +#endif + pcm_dev->irq_info.gpio = of_irq_get(np, 0); + + for (i = 0; i < ndev; i++) + pcm_dev->addr[i] = dev_addrs[i]; + + pcm_dev->ndev = ndev; + + pcm_dev->hw_rst = devm_gpiod_get_optional(&i2c->dev, + "reset-gpios", GPIOD_OUT_HIGH); + /* No reset GPIO, no side-effect */ + if (IS_ERR(pcm_dev->hw_rst)) { + if (pcm_dev->chip_id == PCM9211 || pcm_dev->chip_id == PCM1690) + pcm9211_sw_rst(pcm_dev); + else + pcmdevice_sw_rst(pcm_dev); + } else { + gpiod_set_value_cansleep(pcm_dev->hw_rst, 0); + usleep_range(500, 1000); + gpiod_set_value_cansleep(pcm_dev->hw_rst, 1); + } + + if (pcm_dev->chip_id == PCM1690) + goto skip_interrupt; + if (gpio_is_valid(pcm_dev->irq_info.gpio)) { + dev_dbg(pcm_dev->dev, "irq-gpio = %d", pcm_dev->irq_info.gpio); + + ret = gpio_request(pcm_dev->irq_info.gpio, "PCMDEV-IRQ"); + if (!ret) { + int gpio = pcm_dev->irq_info.gpio; + + gpio_direction_input(gpio); + pcm_dev->irq_info.nmb = gpio_to_irq(gpio); + + } else + dev_err(pcm_dev->dev, "%s: GPIO %d request error\n", + __func__, pcm_dev->irq_info.gpio); + } else + dev_err(pcm_dev->dev, "Looking up irq-gpio failed %d\n", + pcm_dev->irq_info.gpio); + +skip_interrupt: + ret = devm_snd_soc_register_component(&i2c->dev, + &soc_codec_driver_pcmdevice, pcmdevice_dai_driver, + ARRAY_SIZE(pcmdevice_dai_driver)); + if (ret < 0) + dev_err(&i2c->dev, "probe register comp failed %d\n", ret); + +out: + if (ret < 0) + pcmdevice_remove(pcm_dev); + return ret; +} + +static void pcmdevice_i2c_remove(struct i2c_client *i2c) +{ + struct pcmdevice_priv *pcm_dev = i2c_get_clientdata(i2c); + + pcmdevice_remove(pcm_dev); +} + +static struct i2c_driver pcmdevice_i2c_driver = { + .driver = { + .name = "pcmdevice-codec", + .of_match_table = of_match_ptr(pcmdevice_of_match), + }, + .probe = pcmdevice_i2c_probe, + .remove = pcmdevice_i2c_remove, + .id_table = pcmdevice_i2c_id, +}; +module_i2c_driver(pcmdevice_i2c_driver); + +MODULE_AUTHOR("Shenghao Ding "); +MODULE_DESCRIPTION("ASoC PCM6240 Family Audio ADC/DAC Driver"); +MODULE_LICENSE("GPL"); From d537f7b52d71110a98cd7230cce1b56ae7c8ea71 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sun, 7 Apr 2024 17:18:43 +0800 Subject: [PATCH 348/603] ASoc: PCM6240: Create header file for PCM6240 Family driver code PCM6240 driver implements a flexible and configurable setting for register and filter coefficients, to one, two or even multiple PCM6240 Family Audio chips. Signed-off-by: Shenghao Ding Link: https://lore.kernel.org/r/20240407091846.1299-3-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/pcm6240.h | 252 +++++++++++++++++++++++++++++++++++++ 1 file changed, 252 insertions(+) create mode 100644 sound/soc/codecs/pcm6240.h diff --git a/sound/soc/codecs/pcm6240.h b/sound/soc/codecs/pcm6240.h new file mode 100644 index 0000000000000..1e125bb972860 --- /dev/null +++ b/sound/soc/codecs/pcm6240.h @@ -0,0 +1,252 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments PCM6240 Family Audio ADC/DAC/Router +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The PCM6240 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// PCM6240 Family Audio chips. +// +// Author: Shenghao Ding +// + +#ifndef __PCM6240_H__ +#define __PCM6240_H__ + +enum pcm_device { + ADC3120, + ADC5120, + ADC6120, + DIX4192, + PCM1690, + PCM3120, + PCM3140, + PCM5120, + PCM5140, + PCM6120, + PCM6140, + PCM6240, + PCM6260, + PCM9211, + PCMD3140, + PCMD3180, + PCMD512X, + TAA5212, + TAA5412, + TAD5212, + TAD5412, + MAX_DEVICE, +}; + +#define PCMDEV_GENERIC_VOL_CTRL 0x0 +#define PCMDEV_PCM1690_VOL_CTRL 0x1 +#define PCMDEV_PCM1690_FINE_VOL_CTRL 0x2 + +/* Maximum number of I2C addresses */ +#define PCMDEVICE_MAX_I2C_DEVICES 4 +/* Maximum number defined in REGBIN protocol */ +#define PCMDEVICE_MAX_REGBIN_DEVICES 8 +#define PCMDEVICE_CONFIG_SUM 64 +#define PCMDEVICE_BIN_FILENAME_LEN 64 + +#define PCMDEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) +#define PCMDEVICE_MAX_CHANNELS 8 +#define PCMDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* PAGE Control Register (available in page0 of each book) */ +#define PCMDEVICE_PAGE_SELECT 0x00 +#define PCMDEVICE_REG(page, reg) ((page * 128) + reg) +#define PCMDEVICE_REG_SWRESET PCMDEVICE_REG(0X0, 0x01) +#define PCMDEVICE_REG_SWRESET_RESET BIT(0) + +#define ADC5120_REG_CH1_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x3d) +#define ADC5120_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3e) +#define ADC5120_REG_CH2_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x42) +#define ADC5120_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43) + +#define PCM1690_REG_MODE_CTRL PCMDEVICE_REG(0X0, 0x46) +#define PCM1690_REG_MODE_CTRL_DAMS_MSK BIT(7) +#define PCM1690_REG_MODE_CTRL_DAMS_FINE_STEP 0x0 +#define PCM1690_REG_MODE_CTRL_DAMS_WIDE_RANGE 0x80 + +#define PCM1690_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48) +#define PCM1690_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x49) +#define PCM1690_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4a) +#define PCM1690_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4b) +#define PCM1690_REG_CH5_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4c) +#define PCM1690_REG_CH6_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4d) +#define PCM1690_REG_CH7_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4e) +#define PCM1690_REG_CH8_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4f) + +#define PCM6240_REG_CH1_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x3d) +#define PCM6240_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3e) +#define PCM6240_REG_CH2_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x42) +#define PCM6240_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43) +#define PCM6240_REG_CH3_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x47) +#define PCM6240_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48) +#define PCM6240_REG_CH4_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x4c) +#define PCM6240_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4d) + +#define PCM6260_REG_CH1_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x3d) +#define PCM6260_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3e) +#define PCM6260_REG_CH2_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x42) +#define PCM6260_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43) +#define PCM6260_REG_CH3_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x47) +#define PCM6260_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48) +#define PCM6260_REG_CH4_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x4c) +#define PCM6260_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4d) +#define PCM6260_REG_CH5_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x51) +#define PCM6260_REG_CH5_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x52) +#define PCM6260_REG_CH6_ANALOG_GAIN PCMDEVICE_REG(0X0, 0x56) +#define PCM6260_REG_CH6_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x57) + +#define PCM9211_REG_SW_CTRL PCMDEVICE_REG(0X0, 0x40) +#define PCM9211_REG_SW_CTRL_MRST_MSK BIT(7) +#define PCM9211_REG_SW_CTRL_MRST 0x0 + +#define PCM9211_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x46) +#define PCM9211_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x47) + +#define PCMD3140_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3E) +#define PCMD3140_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43) +#define PCMD3140_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48) +#define PCMD3140_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4D) + +#define PCMD3140_REG_CH1_FINE_GAIN PCMDEVICE_REG(0X0, 0x3F) +#define PCMD3140_REG_CH2_FINE_GAIN PCMDEVICE_REG(0X0, 0x44) +#define PCMD3140_REG_CH3_FINE_GAIN PCMDEVICE_REG(0X0, 0x49) +#define PCMD3140_REG_CH4_FINE_GAIN PCMDEVICE_REG(0X0, 0x4E) + +#define PCMD3180_REG_CH1_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x3E) +#define PCMD3180_REG_CH2_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x43) +#define PCMD3180_REG_CH3_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x48) +#define PCMD3180_REG_CH4_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x4D) +#define PCMD3180_REG_CH5_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x52) +#define PCMD3180_REG_CH6_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x57) +#define PCMD3180_REG_CH7_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x5C) +#define PCMD3180_REG_CH8_DIGITAL_GAIN PCMDEVICE_REG(0X0, 0x61) + +#define PCMD3180_REG_CH1_FINE_GAIN PCMDEVICE_REG(0X0, 0x3F) +#define PCMD3180_REG_CH2_FINE_GAIN PCMDEVICE_REG(0X0, 0x44) +#define PCMD3180_REG_CH3_FINE_GAIN PCMDEVICE_REG(0X0, 0x49) +#define PCMD3180_REG_CH4_FINE_GAIN PCMDEVICE_REG(0X0, 0x4E) +#define PCMD3180_REG_CH5_FINE_GAIN PCMDEVICE_REG(0X0, 0x53) +#define PCMD3180_REG_CH6_FINE_GAIN PCMDEVICE_REG(0X0, 0x58) +#define PCMD3180_REG_CH7_FINE_GAIN PCMDEVICE_REG(0X0, 0x5D) +#define PCMD3180_REG_CH8_FINE_GAIN PCMDEVICE_REG(0X0, 0x62) + +#define TAA5412_REG_CH1_DIGITAL_VOLUME PCMDEVICE_REG(0X0, 0x52) +#define TAA5412_REG_CH2_DIGITAL_VOLUME PCMDEVICE_REG(0X0, 0x57) +#define TAA5412_REG_CH3_DIGITAL_VOLUME PCMDEVICE_REG(0X0, 0x5B) +#define TAA5412_REG_CH4_DIGITAL_VOLUME PCMDEVICE_REG(0X0, 0x5F) + +#define TAA5412_REG_CH1_FINE_GAIN PCMDEVICE_REG(0X0, 0x53) +#define TAA5412_REG_CH2_FINE_GAIN PCMDEVICE_REG(0X0, 0x58) +#define TAA5412_REG_CH3_FINE_GAIN PCMDEVICE_REG(0X0, 0x5C) +#define TAA5412_REG_CH4_FINE_GAIN PCMDEVICE_REG(0X0, 0x60) + +#define PCMDEVICE_CMD_SING_W 0x1 +#define PCMDEVICE_CMD_BURST 0x2 +#define PCMDEVICE_CMD_DELAY 0x3 +#define PCMDEVICE_CMD_FIELD_W 0x4 + +enum pcmdevice_bin_blk_type { + PCMDEVICE_BIN_BLK_COEFF = 1, + PCMDEVICE_BIN_BLK_POST_POWER_UP, + PCMDEVICE_BIN_BLK_PRE_SHUTDOWN, + PCMDEVICE_BIN_BLK_PRE_POWER_UP, + PCMDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +enum pcmdevice_fw_state { + PCMDEVICE_FW_LOAD_OK = 0, + PCMDEVICE_FW_LOAD_FAILED +}; + +struct pcmdevice_regbin_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned int timestamp; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[PCMDEVICE_MAX_REGBIN_DEVICES]; + unsigned int nconfig; + unsigned int config_size[PCMDEVICE_CONFIG_SUM]; +}; + +struct pcmdevice_block_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct pcmdevice_config_info { + char cfg_name[64]; + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct pcmdevice_block_data **blk_data; +}; + +struct pcmdevice_regbin { + struct pcmdevice_regbin_hdr fw_hdr; + int ncfgs; + struct pcmdevice_config_info **cfg_info; +}; + +struct pcmdevice_irqinfo { + int gpio; + int nmb; +}; + +struct pcmdevice_priv { + struct snd_soc_component *component; + struct i2c_client *client; + struct device *dev; + struct mutex codec_lock; + struct gpio_desc *hw_rst; + struct regmap *regmap; + struct pcmdevice_regbin regbin; + struct pcmdevice_irqinfo irq_info; + unsigned int addr[PCMDEVICE_MAX_I2C_DEVICES]; + unsigned int chip_id; + int cur_conf; + int fw_state; + int ndev; + unsigned char bin_name[PCMDEVICE_BIN_FILENAME_LEN]; + /* used for kcontrol name */ + unsigned char upper_dev_name[I2C_NAME_SIZE]; + unsigned char dev_name[I2C_NAME_SIZE]; +}; + +/* mixer control */ +struct pcmdevice_mixer_control { + int max; + int reg; + unsigned int dev_no; + unsigned int shift; + unsigned int invert; +}; +struct pcmdev_ctrl_info { + const unsigned int *gain; + const struct pcmdevice_mixer_control *pcmdev_ctrl; + unsigned int ctrl_array_size; + snd_kcontrol_get_t *get; + snd_kcontrol_put_t *put; + int pcmdev_ctrl_name_id; +}; +#endif /* __PCM6240_H__ */ From 6bb1b70690766367e166b8ff59f4b5f9739f7964 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sun, 7 Apr 2024 17:18:44 +0800 Subject: [PATCH 349/603] ASoc: PCM6240: Add compile item for PCM6240 Family driver PCM6240 driver implements a flexible and configurable setting for register and filter coefficients, to one, two or even multiple PCM6240 Family Audio chips. Signed-off-by: Shenghao Ding Link: https://lore.kernel.org/r/20240407091846.1299-4-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 10 ++++++++++ sound/soc/codecs/Makefile | 2 ++ 2 files changed, 12 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 995eab954dd5d..4afc43d3f71fd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -179,6 +179,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM5102A imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI + imply SND_SOC_PCM6240 imply SND_SOC_PEB2466 imply SND_SOC_RK3308 imply SND_SOC_RK3328 @@ -1424,6 +1425,15 @@ config SND_SOC_PCM512x_SPI select SND_SOC_PCM512x select REGMAP_SPI +config SND_SOC_PCM6240 + tristate "Texas Instruments PCM6240 Family Audio chips based on I2C" + depends on I2C + help + Enable support for Texas Instruments PCM6240 Family Audio chips. + Note the PCM6240 driver implements a flexible and configurable + setting for register and filter coefficients, to one, two or + even multiple PCM6240 Family Audio chips. + config SND_SOC_PEB2466 tristate "Infineon PEB2466 quad PCM codec" depends on SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9ba24fb870b11..cddb16cd6a4c2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -204,6 +204,7 @@ snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o +snd-soc-pcm6240-objs := pcm6240.o snd-soc-peb2466-objs := peb2466.o snd-soc-rk3308-objs := rk3308_codec.o snd-soc-rk3328-objs := rk3328_codec.o @@ -595,6 +596,7 @@ obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o +obj-$(CONFIG_SND_SOC_PCM6240) += snd-soc-pcm6240.o obj-$(CONFIG_SND_SOC_PEB2466) += snd-soc-peb2466.o obj-$(CONFIG_SND_SOC_RK3308) += snd-soc-rk3308.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o From cee56a8e9cf89fd690e18eb2e6f09d3f8a6442fb Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sun, 7 Apr 2024 17:18:45 +0800 Subject: [PATCH 350/603] ASoc: dt-bindings: PCM6240: Add initial DT binding PCM6240 family chips are popular among audio customers, in spite of only a portion of the functionality of codec, such as ADC or DAC, and so on, for different Specifications, range from Personal Electric to Automotive Electric, even some professional fields. Yet their audio performance is far superior to the codec's, and cost is lower than codec, and much easier to program than codec. Reviewed-by: Rob Herring Signed-off-by: Shenghao Ding Link: https://lore.kernel.org/r/20240407091846.1299-5-shenghao-ding@ti.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ti,pcm6240.yaml | 177 ++++++++++++++++++ 1 file changed, 177 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm6240.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml b/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml new file mode 100644 index 0000000000000..dd5b08e3d7a15 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm6240.yaml @@ -0,0 +1,177 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2022 - 2024 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm6240.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments PCM6240 Family Audio ADC/DAC + +maintainers: + - Shenghao Ding + +description: | + The PCM6240 Family is a big family of Audio ADC/DAC for + different Specifications, range from Personal Electric + to Automotive Electric, even some professional fields. + + Specifications about the audio chip can be found at: + https://www.ti.com/lit/gpn/tlv320adc3120 + https://www.ti.com/lit/gpn/tlv320adc5120 + https://www.ti.com/lit/gpn/tlv320adc6120 + https://www.ti.com/lit/gpn/dix4192 + https://www.ti.com/lit/gpn/pcm1690 + https://www.ti.com/lit/gpn/pcm3120-q1 + https://www.ti.com/lit/gpn/pcm3140-q1 + https://www.ti.com/lit/gpn/pcm5120-q1 + https://www.ti.com/lit/gpn/pcm6120-q1 + https://www.ti.com/lit/gpn/pcm6260-q1 + https://www.ti.com/lit/gpn/pcm9211 + https://www.ti.com/lit/gpn/pcmd3140 + https://www.ti.com/lit/gpn/pcmd3180 + https://www.ti.com/lit/gpn/taa5212 + https://www.ti.com/lit/gpn/tad5212 + +properties: + compatible: + description: | + ti,adc3120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to- + digital converter (ADC) with 106-dB SNR. + + ti,adc5120: 2-Channel, 768-kHz, Burr-Brown™ Audio ADC with 120-dB SNR. + + ti,adc6120: Stereo-channel, 768-kHz, Burr-Brown™ audio analog-to- + digital converter (ADC) with 123-dB SNR. + + ti,dix4192: 216-kHz digital audio converter with Quad-Channel In + and One-Channel Out. + + ti,pcm1690: Automotive Catalog 113dB SNR 8-Channel Audio DAC with + Differential Outputs. + + ti,pcm3120: Automotive, stereo, 106-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm3140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC + with 106-dB SNR. + + ti,pcm5120: Automotive, stereo, 120-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm5140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC + with 120-dB SNR. + + ti,pcm6120: Automotive, stereo, 123-dB SNR, 768-kHz, low-power + software-controlled audio ADC. + + ti,pcm6140: Automotive, Quad-Channel, 768-kHz, Burr-Brown™ Audio ADC + with 123-dB SNR. + + ti,pcm6240: Automotive 4-ch audio ADC with integrated programmable mic + bias, boost and input diagnostics. + + ti,pcm6260: Automotive 6-ch audio ADC with integrated programmable mic + bias, boost and input diagnostics. + + ti,pcm9211: 216-kHz digital audio converter With Stereo ADC and + Routing. + + ti,pcmd3140: Four-channel PDM-input to TDM or I2S output converter. + + ti,pcmd3180: Eight-channel pulse-density-modulation input to TDM or + I2S output converter. + + ti,taa5212: Low-power high-performance stereo audio ADC with 118-dB + dynamic range. + + ti,tad5212: Low-power stereo audio DAC with 120-dB dynamic range. + oneOf: + - items: + - enum: + - ti,adc3120 + - ti,adc5120 + - ti,pcm3120 + - ti,pcm5120 + - ti,pcm6120 + - const: ti,adc6120 + - items: + - enum: + - ti,pcmd512x + - ti,pcm9211 + - ti,taa5212 + - ti,tad5212 + - const: ti,adc6120 + - items: + - enum: + - ti,pcm3140 + - ti,pcm5140 + - ti,dix4192 + - ti,pcm6140 + - ti,pcm6260 + - const: ti,pcm6240 + - items: + - enum: + - ti,pcmd3140 + - ti,pcmd3180 + - ti,pcm1690 + - ti,taa5412 + - ti,tad5412 + - const: ti,pcm6240 + - enum: + - ti,adc6120 + - ti,pcm6240 + + reg: + description: + I2C address, in multiple pcmdevices case, all the i2c address + aggregate as one Audio Device to support multiple audio slots. + minItems: 1 + maxItems: 4 + + reset-gpios: + maxItems: 1 + + interrupts: + maxItems: 1 + description: + Invalid only for ti,pcm1690 because of no INT pin. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - ti,pcm1690 + then: + properties: + interrupts: false + +additionalProperties: false + +examples: + - | + #include + i2c { + /* example for two devices with interrupt support */ + #address-cells = <1>; + #size-cells = <0>; + pcm6240: audio-codec@48 { + compatible = "ti,pcm6240"; + reg = <0x48>, /* primary-device */ + <0x4b>; /* secondary-device */ + #sound-dai-cells = <0>; + reset-gpios = <&gpio1 10 GPIO_ACTIVE_HIGH>; + interrupt-parent = <&gpio1>; + interrupts = <15>; + }; + }; +... From f5d20b253d1a51aadb8881d899caaaa989217e89 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 19 Apr 2024 10:43:07 +0200 Subject: [PATCH 351/603] ASoC: Intel: avs: Switch to ACPI NHLT MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Now that NHLT support in ACPI framework was introduced, migrate avs driver to new API. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240419084307.2718881-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 1 + sound/soc/intel/avs/avs.h | 1 - sound/soc/intel/avs/board_selection.c | 7 +++--- sound/soc/intel/avs/core.c | 10 ++++----- sound/soc/intel/avs/path.c | 32 +++++++++++++-------------- 5 files changed, 24 insertions(+), 27 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 4b9e498e33037..38b61dfd1487f 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -214,6 +214,7 @@ config SND_SOC_INTEL_AVS depends on X86 || COMPILE_TEST depends on PCI depends on COMMON_CLK + select ACPI_NHLT if ACPI select SND_SOC_ACPI if ACPI select SND_SOC_TOPOLOGY select SND_SOC_HDA diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index f80f79415344a..900786eb81e4d 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -151,7 +151,6 @@ struct avs_dev { struct completion fw_ready; struct work_struct probe_work; - struct nhlt_acpi_table *nhlt; struct list_head comp_list; struct mutex comp_list_mutex; struct list_head path_list; diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 8360ce557401c..290ea314ace82 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -10,10 +10,10 @@ #include #include #include +#include #include #include #include -#include #include #include #include "avs.h" @@ -434,8 +434,7 @@ static int avs_register_dmic_board(struct avs_dev *adev) struct snd_soc_acpi_mach mach = {{0}}; int ret; - if (!adev->nhlt || - !intel_nhlt_has_endpoint_type(adev->nhlt, NHLT_LINK_DMIC)) { + if (!acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_PDM, -1, -1, -1)) { dev_dbg(adev->dev, "no DMIC endpoints present\n"); return 0; } @@ -523,7 +522,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev) struct snd_soc_acpi_mach *mach; int ret; - if (!adev->nhlt || !intel_nhlt_has_endpoint_type(adev->nhlt, NHLT_LINK_SSP)) { + if (!acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_SSP, -1, -1, -1)) { dev_dbg(adev->dev, "no I2S endpoints present\n"); return 0; } diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index d7f8940099cec..a61ce42b426cb 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -14,15 +14,16 @@ // foundation of this driver // +#include #include #include +#include #include #include #include #include #include #include -#include #include "../../codecs/hda.h" #include "avs.h" #include "cldma.h" @@ -214,9 +215,7 @@ static void avs_hda_probe_work(struct work_struct *work) if (ret < 0) return; - adev->nhlt = intel_nhlt_init(adev->dev); - if (!adev->nhlt) - dev_info(bus->dev, "platform has no NHLT\n"); + acpi_nhlt_get_gbl_table(); avs_debugfs_init(adev); avs_register_all_boards(adev); @@ -549,8 +548,7 @@ static void avs_pci_remove(struct pci_dev *pci) avs_unregister_all_boards(adev); avs_debugfs_exit(adev); - if (adev->nhlt) - intel_nhlt_free(adev->nhlt); + acpi_nhlt_put_gbl_table(); if (avs_platattr_test(adev, CLDMA)) hda_cldma_free(&code_loader); diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index e785fc2a7008f..531a086642be1 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -6,7 +6,8 @@ // Amadeusz Slawinski // -#include +#include +#include #include #include #include "avs.h" @@ -143,10 +144,10 @@ static bool avs_dma_type_is_input(u32 dma_type) static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod) { - struct nhlt_acpi_table *nhlt = adev->nhlt; struct avs_tplg_module *t = mod->template; struct avs_copier_cfg *cfg; - struct nhlt_specific_cfg *ep_blob; + struct acpi_nhlt_format_config *ep_blob; + struct acpi_nhlt_endpoint *ep; union avs_connector_node_id node_id = {0}; size_t cfg_size, data_size = 0; void *data = NULL; @@ -174,18 +175,18 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod) else fmt = t->cfg_ext->copier.out_fmt; - ep_blob = intel_nhlt_get_endpoint_blob(adev->dev, - nhlt, t->cfg_ext->copier.vindex.i2s.instance, - NHLT_LINK_SSP, fmt->valid_bit_depth, fmt->bit_depth, - fmt->num_channels, fmt->sampling_freq, direction, - NHLT_DEVICE_I2S); + ep = acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_SSP, + ACPI_NHLT_DEVICETYPE_CODEC, direction, + t->cfg_ext->copier.vindex.i2s.instance); + ep_blob = acpi_nhlt_endpoint_find_fmtcfg(ep, fmt->num_channels, fmt->sampling_freq, + fmt->valid_bit_depth, fmt->bit_depth); if (!ep_blob) { dev_err(adev->dev, "no I2S ep_blob found\n"); return -ENOENT; } - data = ep_blob->caps; - data_size = ep_blob->size; + data = ep_blob->config.capabilities; + data_size = ep_blob->config.capabilities_size; /* I2S gateway's vindex is statically assigned in topology */ node_id.vindex = t->cfg_ext->copier.vindex.val; @@ -199,17 +200,16 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod) else fmt = t->in_fmt; - ep_blob = intel_nhlt_get_endpoint_blob(adev->dev, nhlt, 0, - NHLT_LINK_DMIC, fmt->valid_bit_depth, - fmt->bit_depth, fmt->num_channels, - fmt->sampling_freq, direction, NHLT_DEVICE_DMIC); + ep = acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_PDM, -1, direction, 0); + ep_blob = acpi_nhlt_endpoint_find_fmtcfg(ep, fmt->num_channels, fmt->sampling_freq, + fmt->valid_bit_depth, fmt->bit_depth); if (!ep_blob) { dev_err(adev->dev, "no DMIC ep_blob found\n"); return -ENOENT; } - data = ep_blob->caps; - data_size = ep_blob->size; + data = ep_blob->config.capabilities; + data_size = ep_blob->config.capabilities_size; /* DMIC gateway's vindex is statically assigned in topology */ node_id.vindex = t->cfg_ext->copier.vindex.val; From 3a56855bb549211031184e45fe1a9d24874d7227 Mon Sep 17 00:00:00 2001 From: Nathan Chancellor Date: Fri, 19 Apr 2024 17:25:59 -0700 Subject: [PATCH 352/603] ALSA: scarlett2: Zero initialize ret in scarlett2_ag_target_ctl_get() Clang warns (or errors with CONFIG_WERROR): sound/usb/mixer_scarlett2.c:3697:6: error: variable 'err' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized] 3697 | if (private->autogain_updated) { | ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/usb/mixer_scarlett2.c:3707:9: note: uninitialized use occurs here 3707 | return err; | ^~~ sound/usb/mixer_scarlett2.c:3697:2: note: remove the 'if' if its condition is always true 3697 | if (private->autogain_updated) { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/usb/mixer_scarlett2.c:3688:9: note: initialize the variable 'err' to silence this warning 3688 | int err; | ^ | = 0 1 error generated. Initialize ret to zero to ensure ret is initialized in all paths within scarlett2_ag_target_ctl_get(), which matches the style of other functions in this driver. Fixes: e30ea5340c25 ("ALSA: scarlett2: Add autogain target controls") Signed-off-by: Nathan Chancellor Message-ID: <20240419-alsa-scarlett2-fix-wsometimes-uninitialized-v1-1-e2ace8642e08@kernel.org> Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index a2de31a0371b0..b251d25f2a858 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -3685,7 +3685,7 @@ static int scarlett2_ag_target_ctl_get( struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - int err; + int err = 0; mutex_lock(&private->data_mutex); From 3249c68e3cdacc2da8fe811a54f8db43df124c0e Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 22 Apr 2024 18:15:13 +0300 Subject: [PATCH 353/603] ASoC: soc.h: Don't use "proxy" headers Update header inclusions to follow IWYU (Include What You Use) principle. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20240422151513.2052167-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc.h | 30 ++++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0376f7e4c15d9..2a1b6c198547c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -11,20 +11,30 @@ #define __LINUX_SND_SOC_H #include +#include +#include +#include +#include +#include +#include +#include +#include #include -#include #include -#include #include -#include -#include -#include -#include -#include -#include + +#include #include #include -#include +#include +#include + +struct module; +struct platform_device; + +/* For the current users of sound/soc.h to avoid build issues */ +#include +#include /* * Convenience kcontrol builders @@ -412,7 +422,6 @@ #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts) -struct device_node; struct snd_jack; struct snd_soc_card; struct snd_soc_pcm_stream; @@ -427,6 +436,7 @@ struct soc_enum; struct snd_soc_jack; struct snd_soc_jack_zone; struct snd_soc_jack_pin; + #include #include #include From ab371a026a1a4ad81817d82ba433c19934082894 Mon Sep 17 00:00:00 2001 From: Kartik Agarwala Date: Sun, 21 Apr 2024 09:01:31 +0530 Subject: [PATCH 354/603] ASoC: dt-bindings: mt2701-wm8960: Convert to dtschema Convert mt2701-wm890 bindings from text to dtschema. This is used by MediaTek mt7623a/n SoC. Signed-off-by: Kartik Agarwala Reviewed-by: Rob Herring (Arm) Link: https://lore.kernel.org/r/20240421033129.13076-1-agarwala.kartik@gmail.com Signed-off-by: Mark Brown --- .../sound/mediatek,mt2701-wm8960.yaml | 54 +++++++++++++++++++ .../bindings/sound/mt2701-wm8960.txt | 24 --------- 2 files changed, 54 insertions(+), 24 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mt2701-wm8960.txt diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml new file mode 100644 index 0000000000000..cf985461a995e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt2701-wm8960.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt2701-wm8960.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT2701 with WM8960 CODEC + +maintainers: + - Kartik Agarwala + +properties: + compatible: + const: mediatek,mt2701-wm8960-machine + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT2701 ASoC platform. + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. + + mediatek,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the WM8960 audio codec. + +unevaluatedProperties: false + +required: + - compatible + - mediatek,platform + - audio-routing + - mediatek,audio-codec + - pinctrl-names + - pinctrl-0 + +examples: + - | + sound { + compatible = "mediatek,mt2701-wm8960-machine"; + mediatek,platform = <&afe>; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + mediatek,audio-codec = <&wm8960>; + pinctrl-names = "default"; + pinctrl-0 = <&aud_pins_default>; + }; diff --git a/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt b/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt deleted file mode 100644 index 809b609ea9d0b..0000000000000 --- a/Documentation/devicetree/bindings/sound/mt2701-wm8960.txt +++ /dev/null @@ -1,24 +0,0 @@ -MT2701 with WM8960 CODEC - -Required properties: -- compatible: "mediatek,mt2701-wm8960-machine" -- mediatek,platform: the phandle of MT2701 ASoC platform -- audio-routing: a list of the connections between audio -- mediatek,audio-codec: the phandles of wm8960 codec -- pinctrl-names: Should contain only one value - "default" -- pinctrl-0: Should specify pin control groups used for this controller. - -Example: - - sound:sound { - compatible = "mediatek,mt2701-wm8960-machine"; - mediatek,platform = <&afe>; - audio-routing = - "Headphone", "HP_L", - "Headphone", "HP_R", - "LINPUT1", "AMIC", - "RINPUT1", "AMIC"; - mediatek,audio-codec = <&wm8960>; - pinctrl-names = "default"; - pinctrl-0 = <&aud_pins_default>; - }; From 181d58cfe90f6a2ad53f6811d318fbc7149df8c1 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 17 Apr 2024 12:52:34 +0300 Subject: [PATCH 355/603] ASoC: cs35l41: Delete unnecessary condition in cs35l41_pcm_hw_params() This code returns -EINVAL if "i" is out of bounds a few lines earlier. Delete this unnecessary check and pull the code in a tab. Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/1ee32bfb-6f6c-4b61-887b-6f655abbfc47@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index dfb4ce53491bb..00258ac2bea6d 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -772,10 +772,9 @@ static int cs35l41_pcm_hw_params(struct snd_pcm_substream *substream, asp_wl = params_width(params); - if (i < ARRAY_SIZE(cs35l41_fs_rates)) - regmap_update_bits(cs35l41->regmap, CS35L41_GLOBAL_CLK_CTRL, - CS35L41_GLOBAL_FS_MASK, - cs35l41_fs_rates[i].fs_cfg << CS35L41_GLOBAL_FS_SHIFT); + regmap_update_bits(cs35l41->regmap, CS35L41_GLOBAL_CLK_CTRL, + CS35L41_GLOBAL_FS_MASK, + cs35l41_fs_rates[i].fs_cfg << CS35L41_GLOBAL_FS_SHIFT); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(cs35l41->regmap, CS35L41_SP_FORMAT, From 701a22fd9ffaa409bbd45c2936870341b3ad9fdb Mon Sep 17 00:00:00 2001 From: Mohammad Shehar Yaar Tausif Date: Tue, 23 Apr 2024 17:27:47 +0530 Subject: [PATCH 356/603] ASoC: dt-bindings: tegra20-ac97: convert to dt schema Convert NVIDIA Tegra20 AC97 binding to DT schema. Change -gpio to -gpios in schema as "gpio" suffix is deprecated. Signed-off-by: Mohammad Shehar Yaar Tausif Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240423115749.15786-1-sheharyaar48@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/nvidia,tegra20-ac97.txt | 36 -------- .../bindings/sound/nvidia,tegra20-ac97.yaml | 82 +++++++++++++++++++ 2 files changed, 82 insertions(+), 36 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt deleted file mode 100644 index eaf00102d92c2..0000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt +++ /dev/null @@ -1,36 +0,0 @@ -NVIDIA Tegra 20 AC97 controller - -Required properties: -- compatible : "nvidia,tegra20-ac97" -- reg : Should contain AC97 controller registers location and length -- interrupts : Should contain AC97 interrupt -- resets : Must contain an entry for each entry in reset-names. - See ../reset/reset.txt for details. -- reset-names : Must include the following entries: - - ac97 -- dmas : Must contain an entry for each entry in clock-names. - See ../dma/dma.txt for details. -- dma-names : Must include the following entries: - - rx - - tx -- clocks : Must contain one entry, for the module clock. - See ../clocks/clock-bindings.txt for details. -- nvidia,codec-reset-gpio : The Tegra GPIO controller's phandle and the number - of the GPIO used to reset the external AC97 codec -- nvidia,codec-sync-gpio : The Tegra GPIO controller's phandle and the number - of the GPIO corresponding with the AC97 DAP _FS line - -Example: - -ac97@70002000 { - compatible = "nvidia,tegra20-ac97"; - reg = <0x70002000 0x200>; - interrupts = <0 81 0x04>; - nvidia,codec-reset-gpio = <&gpio 170 0>; - nvidia,codec-sync-gpio = <&gpio 120 0>; - clocks = <&tegra_car 3>; - resets = <&tegra_car 3>; - reset-names = "ac97"; - dmas = <&apbdma 12>, <&apbdma 12>; - dma-names = "rx", "tx"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml new file mode 100644 index 0000000000000..4ea0a303d995d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.yaml @@ -0,0 +1,82 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra20-ac97.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra20 AC97 controller + +maintainers: + - Thierry Reding + - Jon Hunter + +properties: + compatible: + const: nvidia,tegra20-ac97 + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: ac97 + + interrupts: + maxItems: 1 + + clocks: + maxItems: 1 + + dmas: + maxItems: 2 + + dma-names: + items: + - const: rx + - const: tx + + nvidia,codec-reset-gpios: + description: Reset pin of external AC97 codec + maxItems: 1 + + nvidia,codec-sync-gpios: + description: AC97 DAP _FS line + maxItems: 1 + +required: + - compatible + - reg + - resets + - reset-names + - interrupts + - clocks + - dmas + - dma-names + - nvidia,codec-reset-gpios + - nvidia,codec-sync-gpios + +additionalProperties: false + +examples: + - | + #include + #include + #include + #include + #include + + ac97@70002000 { + compatible = "nvidia,tegra20-ac97"; + reg = <0x70002000 0x200>; + resets = <&tegra_car 3>; + reset-names = "ac97"; + interrupts = ; + clocks = <&tegra_car 3>; + dmas = <&apbdma 12>, <&apbdma 12>; + dma-names = "rx", "tx"; + nvidia,codec-reset-gpios = <&gpio TEGRA_GPIO(V, 2) GPIO_ACTIVE_HIGH>; + nvidia,codec-sync-gpios = <&gpio TEGRA_GPIO(P, 0) GPIO_ACTIVE_HIGH>; + }; +... From fe2a84133724ef56d2c99f2be5d0e06c4b8395c1 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 23 Apr 2024 08:44:28 +0100 Subject: [PATCH 357/603] ASoc: PCM6240: Fix spelling mistake: "deley" -> "delay" There is a spelling mistake in a dev_err message. Fix it. Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20240423074428.1313777-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/pcm6240.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm6240.c b/sound/soc/codecs/pcm6240.c index 93d592dbee668..86e126783a1df 100644 --- a/sound/soc/codecs/pcm6240.c +++ b/sound/soc/codecs/pcm6240.c @@ -1772,7 +1772,7 @@ static int pcmdev_delay(struct pcmdevice_priv *pcm_dev, int offset = 2; if (offset + 2 > sublocksize) { - dev_err(pcm_dev->dev, "%s: dev-%d deley out of boundary\n", + dev_err(pcm_dev->dev, "%s: dev-%d delay out of boundary\n", __func__, devn); return -EINVAL; } From 466c8c46b4ba763435b2a787b7824d9f0fd3e76e Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Tue, 23 Apr 2024 14:19:26 +0200 Subject: [PATCH 358/603] ASoC: dt-bindings: renesas: Fix R-Car Gen4 SoC-specific compatibles make dtbs_check: arch/arm64/boot/dts/renesas/r8a779g0-white-hawk.dtb: sound@ec5a0000: compatible: 'oneOf' conditional failed, one must be fixed: ['renesas,rcar_sound-r8a779g0', 'renesas,rcar_sound-gen4'] is too short ... 'renesas,rcar_sound-r8a779h0' was expected The list of SoC-specific compatible values for R-Car Gen4 SoCs is mutually-exclusive, just like for other R-Car SoC generations, so it should use an enum. Fixes: d6e792ed7dd022a1 ("ASoC: dt-bindings: renesas: add R8A779H0 V4M") Signed-off-by: Geert Uytterhoeven Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/b3a6cf12c46b23a501b0d54c6892b969e2a55145.1713874657.git.geert+renesas@glider.be Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index b3da278cf141c..07ec6247d9def 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -48,8 +48,9 @@ properties: - const: renesas,rcar_sound-gen3 # for Gen4 SoC - items: - - const: renesas,rcar_sound-r8a779g0 # R-Car V4H - - const: renesas,rcar_sound-r8a779h0 # R-Car V4M + - enum: + - renesas,rcar_sound-r8a779g0 # R-Car V4H + - renesas,rcar_sound-r8a779h0 # R-Car V4M - const: renesas,rcar_sound-gen4 # for Generic - enum: From 7a62ab91801960b22fbeba7c67b08bd1169b7893 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 23 Apr 2024 17:23:02 +0100 Subject: [PATCH 359/603] ALSA: hda: cs35l41: Support Lenovo 13X laptop without _DSD This laptop does not have the correct _DSD settings, so needs to obtain its configuration from the configuration table. Signed-off-by: Stefan Binding Message-ID: <20240423162303.638211-2-sbinding@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 6f82b28e26dde..be97e973accb6 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -111,6 +111,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "17AA3865", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "17AA3866", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, @@ -504,6 +506,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, { "CSC3551", "10433A60", generic_dsd_config }, + { "CSC3551", "17AA3865", generic_dsd_config }, + { "CSC3551", "17AA3866", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, { "CSC3551", "17AA3877", generic_dsd_config }, { "CSC3551", "17AA3878", generic_dsd_config }, From 25f46354dca912c84f1f79468fd636a94b8d287a Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 23 Apr 2024 17:23:03 +0100 Subject: [PATCH 360/603] ALSA: hda/realtek: Add quirks for Lenovo 13X Add laptop using CS35L41 HDA. This laptop does not have _DSD, so require entries in property configuration table for cs35l41_hda driver. Signed-off-by: Stefan Binding Message-ID: <20240423162303.638211-3-sbinding@opensource.cirrus.com> Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1c0381583934..b91552c9fa309 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10387,6 +10387,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), + SND_PCI_QUIRK(0x17aa, 0x3865, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3866, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x386f, "Legion 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), From 541b8a266e0ebf7b4f91f753783a2d2b972169ea Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 24 Apr 2024 17:49:41 +0300 Subject: [PATCH 361/603] ALSA: control: Use list_for_each_entry_safe() Instead of reiterating the list, use list_for_each_entry_safe() that allows to continue without starting over. Signed-off-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Message-ID: <20240424145020.1057216-1-andriy.shevchenko@linux.intel.com> Signed-off-by: Takashi Iwai --- sound/core/control_led.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 3d37e9fa7b9c2..ac19d14bd5745 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -285,25 +285,22 @@ static void snd_ctl_led_ctl_destroy(struct snd_ctl_led_ctl *lctl) static void snd_ctl_led_clean(struct snd_card *card) { unsigned int group; + struct snd_ctl_led_ctl *lctl, *_lctl; struct snd_ctl_led *led; - struct snd_ctl_led_ctl *lctl; for (group = 0; group < MAX_LED; group++) { led = &snd_ctl_leds[group]; -repeat: - list_for_each_entry(lctl, &led->controls, list) - if (!card || lctl->card == card) { + list_for_each_entry_safe(lctl, _lctl, &led->controls, list) + if (!card || lctl->card == card) snd_ctl_led_ctl_destroy(lctl); - goto repeat; - } } } static int snd_ctl_led_reset(int card_number, unsigned int group) { struct snd_card *card __free(snd_card_unref) = NULL; + struct snd_ctl_led_ctl *lctl, *_lctl; struct snd_ctl_led *led; - struct snd_ctl_led_ctl *lctl; struct snd_kcontrol_volatile *vd; bool change = false; @@ -315,14 +312,12 @@ static int snd_ctl_led_reset(int card_number, unsigned int group) if (!snd_ctl_led_card_valid[card_number]) return -ENXIO; led = &snd_ctl_leds[group]; -repeat: - list_for_each_entry(lctl, &led->controls, list) + list_for_each_entry_safe(lctl, _lctl, &led->controls, list) if (lctl->card == card) { vd = &lctl->kctl->vd[lctl->index_offset]; vd->access &= ~group_to_access(group); snd_ctl_led_ctl_destroy(lctl); change = true; - goto repeat; } } if (change) From 2da01ca3674c6e90dbeeda02168849e2ec877edc Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Thu, 25 Apr 2024 14:35:06 +0800 Subject: [PATCH 362/603] ASoC: dt-bindings: fsl,ssi: Convert to YAML Convert the fsl,ssi binding to YAML. Add below compatible strings which were not listed in document: fsl,imx50-ssi fsl,imx53-ssi fsl,imx25-ssi fsl,imx27-ssi fsl,imx6q-ssi fsl,imx6sl-ssi fsl,imx6sx-ssi Add below fsl,mode strings which were not listed. i2s-slave i2s-master lj-slave lj-master rj-slave rj-master Add 'ac97-gpios' property which were not listed. Then dtbs_check can pass. And remove the 'codec' description which should be in the 'codec' binding doc. Signed-off-by: Shengjiu Wang Reviewed-by: Rob Herring (Arm) Link: https://lore.kernel.org/r/1714026906-16723-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,ssi.txt | 87 -------- .../devicetree/bindings/sound/fsl,ssi.yaml | 194 ++++++++++++++++++ 2 files changed, 194 insertions(+), 87 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,ssi.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,ssi.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt deleted file mode 100644 index 7e15a85cecd28..0000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ /dev/null @@ -1,87 +0,0 @@ -Freescale Synchronous Serial Interface - -The SSI is a serial device that communicates with audio codecs. It can -be programmed in AC97, I2S, left-justified, or right-justified modes. - -Required properties: -- compatible: Compatible list, should contain one of the following - compatibles: - fsl,mpc8610-ssi - fsl,imx51-ssi - fsl,imx35-ssi - fsl,imx21-ssi -- cell-index: The SSI, <0> = SSI1, <1> = SSI2, and so on. -- reg: Offset and length of the register set for the device. -- interrupts: where a is the interrupt number and b is a - field that represents an encoding of the sense and - level information for the interrupt. This should be - encoded based on the information in section 2) - depending on the type of interrupt controller you - have. -- fsl,fifo-depth: The number of elements in the transmit and receive FIFOs. - This number is the maximum allowed value for SFCSR[TFWM0]. - - clocks: "ipg" - Required clock for the SSI unit - "baud" - Required clock for SSI master mode. Otherwise this - clock is not used - -Required are also ac97 link bindings if ac97 is used. See -Documentation/devicetree/bindings/sound/soc-ac97link.txt for the necessary -bindings. - -Optional properties: -- codec-handle: Phandle to a 'codec' node that defines an audio - codec connected to this SSI. This node is typically - a child of an I2C or other control node. -- fsl,fiq-stream-filter: Bool property. Disabled DMA and use FIQ instead to - filter the codec stream. This is necessary for some boards - where an incompatible codec is connected to this SSI, e.g. - on pca100 and pcm043. -- dmas: Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. -- dma-names: Two dmas have to be defined, "tx" and "rx", if fsl,imx-fiq - is not defined. -- fsl,mode: The operating mode for the AC97 interface only. - "ac97-slave" - AC97 mode, SSI is clock slave - "ac97-master" - AC97 mode, SSI is clock master -- fsl,ssi-asynchronous: - If specified, the SSI is to be programmed in asynchronous - mode. In this mode, pins SRCK, STCK, SRFS, and STFS must - all be connected to valid signals. In synchronous mode, - SRCK and SRFS are ignored. Asynchronous mode allows - playback and capture to use different sample sizes and - sample rates. Some drivers may require that SRCK and STCK - be connected together, and SRFS and STFS be connected - together. This would still allow different sample sizes, - but not different sample rates. -- fsl,playback-dma: Phandle to a node for the DMA channel to use for - playback of audio. This is typically dictated by SOC - design. See the notes below. - Only used on Power Architecture. -- fsl,capture-dma: Phandle to a node for the DMA channel to use for - capture (recording) of audio. This is typically dictated - by SOC design. See the notes below. - Only used on Power Architecture. - -Child 'codec' node required properties: -- compatible: Compatible list, contains the name of the codec - -Child 'codec' node optional properties: -- clock-frequency: The frequency of the input clock, which typically comes - from an on-board dedicated oscillator. - -Notes on fsl,playback-dma and fsl,capture-dma: - -On SOCs that have an SSI, specific DMA channels are hard-wired for playback -and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for -playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for -playback and DMA channel 3 for capture. The developer can choose which -DMA controller to use, but the channels themselves are hard-wired. The -purpose of these two properties is to represent this hardware design. - -The device tree nodes for the DMA channels that are referenced by -"fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with -"fsl,ssi-dma-channel". The SOC-specific compatible string (e.g. -"fsl,mpc8610-dma-channel") can remain. If these nodes are left as -"fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA -drivers (fsldma) will attempt to use them, and it will conflict with the -sound drivers. diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.yaml b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml new file mode 100644 index 0000000000000..4ab10cd3b5208 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.yaml @@ -0,0 +1,194 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,ssi.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Serial Interface + +maintainers: + - Shengjiu Wang + +description: + Notes on fsl,playback-dma and fsl,capture-dma + On SOCs that have an SSI, specific DMA channels are hard-wired for playback + and capture. On the MPC8610, for example, SSI1 must use DMA channel 0 for + playback and DMA channel 1 for capture. SSI2 must use DMA channel 2 for + playback and DMA channel 3 for capture. The developer can choose which + DMA controller to use, but the channels themselves are hard-wired. The + purpose of these two properties is to represent this hardware design. + + The device tree nodes for the DMA channels that are referenced by + "fsl,playback-dma" and "fsl,capture-dma" must be marked as compatible with + "fsl,ssi-dma-channel". The SOC-specific compatible string (e.g. + "fsl,mpc8610-dma-channel") can remain. If these nodes are left as + "fsl,elo-dma-channel" or "fsl,eloplus-dma-channel", then the generic Elo DMA + drivers (fsldma) will attempt to use them, and it will conflict with the + sound drivers. + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx50-ssi + - fsl,imx53-ssi + - const: fsl,imx51-ssi + - const: fsl,imx21-ssi + - items: + - enum: + - fsl,imx25-ssi + - fsl,imx27-ssi + - fsl,imx35-ssi + - fsl,imx51-ssi + - const: fsl,imx21-ssi + - items: + - enum: + - fsl,imx6q-ssi + - fsl,imx6sl-ssi + - fsl,imx6sx-ssi + - const: fsl,imx51-ssi + - items: + - const: fsl,imx21-ssi + - items: + - const: fsl,mpc8610-ssi + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: The ipg clock for register access + - description: clock for SSI master mode + minItems: 1 + + clock-names: + items: + - const: ipg + - const: baud + minItems: 1 + + dmas: + oneOf: + - items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + - items: + - description: DMA controller phandle and request line for RX0 + - description: DMA controller phandle and request line for TX0 + - description: DMA controller phandle and request line for RX1 + - description: DMA controller phandle and request line for TX1 + + dma-names: + oneOf: + - items: + - const: rx + - const: tx + - items: + - const: rx0 + - const: tx0 + - const: rx1 + - const: tx1 + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + + cell-index: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2] + description: The SSI index + + ac97-gpios: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: Please refer to soc-ac97link.txt + + codec-handle: + $ref: /schemas/types.yaml#/definitions/phandle + description: + Phandle to a 'codec' node that defines an audio + codec connected to this SSI. This node is typically + a child of an I2C or other control node. + + fsl,fifo-depth: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + The number of elements in the transmit and receive FIFOs. + This number is the maximum allowed value for SFCSR[TFWM0]. + enum: [8, 15] + + fsl,fiq-stream-filter: + type: boolean + description: + Disabled DMA and use FIQ instead to filter the codec stream. + This is necessary for some boards where an incompatible codec + is connected to this SSI, e.g. on pca100 and pcm043. + + fsl,mode: + $ref: /schemas/types.yaml#/definitions/string + enum: [ ac97-slave, ac97-master, i2s-slave, i2s-master, + lj-slave, lj-master, rj-slave, rj-master ] + description: | + "ac97-slave" - AC97 mode, SSI is clock slave + "ac97-master" - AC97 mode, SSI is clock master + "i2s-slave" - I2S mode, SSI is clock slave + "i2s-master" - I2S mode, SSI is clock master + "lj-slave" - Left justified mode, SSI is clock slave + "lj-master" - Left justified mode, SSI is clock master + "rj-slave" - Right justified mode, SSI is clock slave + "rj-master" - Right justified mode, SSI is clock master + + fsl,ssi-asynchronous: + type: boolean + description: If specified, the SSI is to be programmed in asynchronous + mode. In this mode, pins SRCK, STCK, SRFS, and STFS must + all be connected to valid signals. In synchronous mode, + SRCK and SRFS are ignored. Asynchronous mode allows + playback and capture to use different sample sizes and + sample rates. Some drivers may require that SRCK and STCK + be connected together, and SRFS and STFS be connected + together. This would still allow different sample sizes, + but not different sample rates. + + fsl,playback-dma: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to a node for the DMA channel to use for + playback of audio. This is typically dictated by SOC + design. Only used on Power Architecture. + + fsl,capture-dma: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to a node for the DMA channel to use for + capture (recording) of audio. This is typically dictated + by SOC design. Only used on Power Architecture. + +required: + - compatible + - reg + - interrupts + - fsl,fifo-depth + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include + #include + ssi@2028000 { + compatible = "fsl,imx6q-ssi", "fsl,imx51-ssi"; + reg = <0x02028000 0x4000>; + interrupts = ; + clocks = <&clks IMX6QDL_CLK_SSI1_IPG>, + <&clks IMX6QDL_CLK_SSI1>; + clock-names = "ipg", "baud"; + dmas = <&sdma 37 1 0>, <&sdma 38 1 0>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + fsl,fifo-depth = <15>; + }; From 957a36c4fcf47aaa4f0d54bdb8050d86880b7f01 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Thu, 25 Apr 2024 17:07:54 +0100 Subject: [PATCH 363/603] ALSA: kunit: make read-only array buf_samples static const Don't populate the read-only array buf_samples on the stack at run time, instead make it static const. Signed-off-by: Colin Ian King Acked-by: Ivan Orlov Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai Message-ID: <20240425160754.114716-1-colin.i.king@gmail.com> --- sound/core/sound_kunit.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c index eb90f62228c02..e34c4317f5eb3 100644 --- a/sound/core/sound_kunit.c +++ b/sound/core/sound_kunit.c @@ -167,7 +167,7 @@ static void _test_fill_silence(struct kunit *test, struct snd_format_test_data * static void test_format_fill_silence(struct kunit *test) { - u32 buf_samples[] = { 10, 20, 32, 64, 129, SILENCE_BUFFER_MAX_FRAMES }; + static const u32 buf_samples[] = { 10, 20, 32, 64, 129, SILENCE_BUFFER_MAX_FRAMES }; u8 *buffer; u32 i, j; From 078c95fe8c44e04e8baa3ade62efc21dcefeb52a Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Fri, 26 Apr 2024 08:36:53 +0900 Subject: [PATCH 364/603] ALSA: kunit: use const qualifier for immutable data Some data for testing is immutable. In the case, the const qualifier is available for any loader to place it to read-only segment. Fixes: 3e39acf56ede ("ALSA: core: Add sound core KUnit test") Signed-off-by: Takashi Sakamoto Acked-by: Ivan Orlov Signed-off-by: Takashi Iwai Message-ID: <20240425233653.218434-1-o-takashi@sakamocchi.jp> --- sound/core/sound_kunit.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c index e34c4317f5eb3..165bd0f5338af 100644 --- a/sound/core/sound_kunit.c +++ b/sound/core/sound_kunit.c @@ -45,7 +45,7 @@ struct avail_test_data { snd_pcm_uframes_t expected_avail; }; -static struct snd_format_test_data valid_fmt[] = { +static const struct snd_format_test_data valid_fmt[] = { DEFINE_FORMAT(S8, 8, 8, -1, 1, SILENCE()), DEFINE_FORMAT(U8, 8, 8, -1, 0, SILENCE(0x80)), DEFINE_FORMAT(S16_LE, 16, 16, 1, 1, SILENCE()), @@ -154,7 +154,7 @@ static void test_format_endianness(struct kunit *test) KUNIT_EXPECT_EQ(test, snd_pcm_format_big_endian(WRONG_FORMAT_2), -EINVAL); } -static void _test_fill_silence(struct kunit *test, struct snd_format_test_data *data, +static void _test_fill_silence(struct kunit *test, const struct snd_format_test_data *data, u8 *buffer, size_t samples_count) { size_t sample_bytes = data->physical_bits >> 3; @@ -191,7 +191,7 @@ static snd_pcm_uframes_t calculate_boundary(snd_pcm_uframes_t buffer_size) return boundary; } -static struct avail_test_data p_avail_data[] = { +static const struct avail_test_data p_avail_data[] = { /* buf_size + hw_ptr < appl_ptr => avail = buf_size + hw_ptr - appl_ptr + boundary */ { 128, 1000, 1129, 1073741824UL - 1 }, /* @@ -220,7 +220,7 @@ static void test_playback_avail(struct kunit *test) } } -static struct avail_test_data c_avail_data[] = { +static const struct avail_test_data c_avail_data[] = { /* hw_ptr - appl_ptr < 0 => avail = hw_ptr - appl_ptr + boundary */ { 128, 1000, 1001, 1073741824UL - 1 }, /* standard case: avail = hw_ptr - appl_ptr */ From 7868e4c1794d336d40578e861bb3824d1ccdaaa3 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:14 +0200 Subject: [PATCH 365/603] ALSA: emu10k1: simplify E-MU card FPGA reset sequence Firstly, it is pointless to explicitly disable the power to the dock prior to resetting the FPGA, as the latter will do the former anyway. Secondly, it doesn't make much sense to check whether the FPGA is already programmed. It's much simpler to just presume it is, and issue the self-reset command. If it isn't, the effect isn't worse than the checks themselves. As a side effect, we lose the info if the reset fails, but there is no plausible way how that could happen unless the card burns out while operating, and in that case we'll detect a firmware upload failure a bit later anyway. Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093717.3198716-2-oswald.buddenhagen@gmx.de> --- sound/pci/emu10k1/emu10k1_main.c | 25 +++---------------------- 1 file changed, 3 insertions(+), 22 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 8ccc0178360ce..353dd3b61c615 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -864,28 +864,9 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_emu1010_fpga_lock(emu); - /* Disable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); - - /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); - dev_dbg(emu->card->dev, "reg1 = 0x%x\n", reg); - if ((reg & 0x3f) == 0x15) { - /* FPGA netlist already present so clear it */ - /* Return to programming mode */ - - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_HANA); - } - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); - dev_dbg(emu->card->dev, "reg2 = 0x%x\n", reg); - if ((reg & 0x3f) == 0x15) { - /* FPGA failed to return to programming mode */ - dev_info(emu->card->dev, - "emu1010: FPGA failed to return to programming mode\n"); - return -ENODEV; - } - dev_info(emu->card->dev, "emu1010: EMU_HANA_ID = 0x%x\n", reg); - + dev_info(emu->card->dev, "emu1010: Loading Hana Firmware\n"); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, + EMU_HANA_FPGA_CONFIG_HANA); err = snd_emu1010_load_firmware(emu, 0, &emu->firmware); if (err < 0) { dev_info(emu->card->dev, "emu1010: Loading Firmware failed\n"); From b83587eaf2a85640fd9e2633309b76596ece4fd5 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:15 +0200 Subject: [PATCH 366/603] ALSA: emu10k1: make snd_emu1010_load_firmware_entry() void There is only one call site, and there we already know that we actually have a firmware. Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093717.3198716-3-oswald.buddenhagen@gmx.de> --- sound/pci/emu10k1/emu10k1_main.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 353dd3b61c615..ec010971a2207 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -652,7 +652,7 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu) return 0; } -static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, +static void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, const struct firmware *fw_entry) { int n, i; @@ -660,9 +660,6 @@ static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, u8 value; __always_unused u16 write_post; - if (!fw_entry) - return -EIO; - /* The FPGA is a Xilinx Spartan IIE XC2S50E */ /* On E-MU 0404b it is a Xilinx Spartan III XC3S50 */ /* GPIO7 -> FPGA PGMN @@ -694,8 +691,6 @@ static int snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, outw(0x10, emu->port + A_GPIO); write_post = inw(emu->port + A_GPIO); spin_unlock_irq(&emu->emu_lock); - - return 0; } /* firmware file names, per model, init-fw and dock-fw (optional) */ @@ -729,7 +724,8 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, int dock, return err; } - return snd_emu1010_load_firmware_entry(emu, *fw); + snd_emu1010_load_firmware_entry(emu, *fw); + return 0; } static void snd_emu1010_load_dock_firmware(struct snd_emu10k1 *emu) From 4c0c36863cc560d24ad482b3ba3bccd62bb101d5 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:16 +0200 Subject: [PATCH 367/603] ALSA: emu10k1: move snd_emu1010_load_firmware_entry() to io.c It is a low-level I/O access function, so io.c is the natural place for it. While we're moving the code, reduce the scope of some variables, use compound assignment operators, and add/adjust some comments. Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093717.3198716-4-oswald.buddenhagen@gmx.de> --- include/sound/emu10k1.h | 1 + sound/pci/emu10k1/emu10k1_main.c | 41 --------------------------- sound/pci/emu10k1/io.c | 48 ++++++++++++++++++++++++++++++++ 3 files changed, 49 insertions(+), 41 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 234b5baea69c8..2856f4717c93e 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1843,6 +1843,7 @@ void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 s u32 snd_emu1010_fpga_link_dst_src_read(struct snd_emu10k1 *emu, u32 dst); int snd_emu1010_get_raw_rate(struct snd_emu10k1 *emu, u8 src); void snd_emu1010_update_clock(struct snd_emu10k1 *emu); +void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, const struct firmware *fw_entry); unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc); void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb); void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index ec010971a2207..c8aa4143ac4fe 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -652,47 +652,6 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu) return 0; } -static void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, - const struct firmware *fw_entry) -{ - int n, i; - u16 reg; - u8 value; - __always_unused u16 write_post; - - /* The FPGA is a Xilinx Spartan IIE XC2S50E */ - /* On E-MU 0404b it is a Xilinx Spartan III XC3S50 */ - /* GPIO7 -> FPGA PGMN - * GPIO6 -> FPGA CCLK - * GPIO5 -> FPGA DIN - * FPGA CONFIG OFF -> FPGA PGMN - */ - spin_lock_irq(&emu->emu_lock); - outw(0x00, emu->port + A_GPIO); /* Set PGMN low for 100uS. */ - write_post = inw(emu->port + A_GPIO); - udelay(100); - outw(0x80, emu->port + A_GPIO); /* Leave bit 7 set during netlist setup. */ - write_post = inw(emu->port + A_GPIO); - udelay(100); /* Allow FPGA memory to clean */ - for (n = 0; n < fw_entry->size; n++) { - value = fw_entry->data[n]; - for (i = 0; i < 8; i++) { - reg = 0x80; - if (value & 0x1) - reg = reg | 0x20; - value = value >> 1; - outw(reg, emu->port + A_GPIO); - write_post = inw(emu->port + A_GPIO); - outw(reg | 0x40, emu->port + A_GPIO); - write_post = inw(emu->port + A_GPIO); - } - } - /* After programming, set GPIO bit 4 high again. */ - outw(0x10, emu->port + A_GPIO); - write_post = inw(emu->port + A_GPIO); - spin_unlock_irq(&emu->emu_lock); -} - /* firmware file names, per model, init-fw and dock-fw (optional) */ static const char * const firmware_names[5][2] = { [EMU_MODEL_EMU1010] = { diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index f4a1c2d4b0787..fafa299efa5c5 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -422,6 +422,54 @@ void snd_emu1010_update_clock(struct snd_emu10k1 *emu) snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, leds); } +void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, + const struct firmware *fw_entry) +{ + __always_unused u16 write_post; + + // On E-MU 1010 rev1 the FPGA is a Xilinx Spartan IIE XC2S50E. + // On E-MU 0404b it is a Xilinx Spartan III XC3S50. + // The wiring is as follows: + // GPO7 -> FPGA input & 1K resistor -> FPGA /PGMN <- FPGA output + // In normal operation, the active low reset line is held up by + // an FPGA output, while the GPO pin performs its duty as control + // register access strobe signal. Writing the respective bit to + // EMU_HANA_FPGA_CONFIG puts the FPGA output into high-Z mode, at + // which point the GPO pin can control the reset line through the + // resistor. + // GPO6 -> FPGA CCLK & FPGA input + // GPO5 -> FPGA DIN (dual function) + + // Assert reset line for 100uS + outw(0x00, emu->port + A_GPIO); + write_post = inw(emu->port + A_GPIO); + udelay(100); + outw(0x80, emu->port + A_GPIO); + write_post = inw(emu->port + A_GPIO); + udelay(100); // Allow FPGA memory to clean + + // Upload the netlist. Keep reset line high! + for (int n = 0; n < fw_entry->size; n++) { + u8 value = fw_entry->data[n]; + for (int i = 0; i < 8; i++) { + u16 reg = 0x80; + if (value & 1) + reg |= 0x20; + value >>= 1; + outw(reg, emu->port + A_GPIO); + write_post = inw(emu->port + A_GPIO); + outw(reg | 0x40, emu->port + A_GPIO); + write_post = inw(emu->port + A_GPIO); + } + } + + // After programming, set GPIO bit 4 high again. + // This appears to be a config word that the rev1 Hana + // firmware reads; weird things happen without this. + outw(0x10, emu->port + A_GPIO); + write_post = inw(emu->port + A_GPIO); +} + void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb) { unsigned long flags; From 1aa41272efff530ccf6cd8455133ada1d09474a2 Mon Sep 17 00:00:00 2001 From: Oswald Buddenhagen Date: Sun, 28 Apr 2024 11:37:17 +0200 Subject: [PATCH 368/603] ALSA: emu10k1: move code for entering E-MU card FPGA programming mode ... into snd_emu1010_load_firmware_entry(). This makes it clearer that these steps belong together tightly, as implied by prior commits. Signed-off-by: Oswald Buddenhagen Signed-off-by: Takashi Iwai Message-ID: <20240428093717.3198716-5-oswald.buddenhagen@gmx.de> --- include/sound/emu10k1.h | 2 +- sound/pci/emu10k1/emu10k1_main.c | 7 +------ sound/pci/emu10k1/io.c | 7 ++++++- 3 files changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 2856f4717c93e..b83862259eece 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1843,7 +1843,7 @@ void snd_emu1010_fpga_link_dst_src_write(struct snd_emu10k1 *emu, u32 dst, u32 s u32 snd_emu1010_fpga_link_dst_src_read(struct snd_emu10k1 *emu, u32 dst); int snd_emu1010_get_raw_rate(struct snd_emu10k1 *emu, u8 src); void snd_emu1010_update_clock(struct snd_emu10k1 *emu); -void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, const struct firmware *fw_entry); +void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, int dock, const struct firmware *fw_entry); unsigned int snd_emu10k1_efx_read(struct snd_emu10k1 *emu, unsigned int pc); void snd_emu10k1_intr_enable(struct snd_emu10k1 *emu, unsigned int intrenb); void snd_emu10k1_intr_disable(struct snd_emu10k1 *emu, unsigned int intrenb); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index c8aa4143ac4fe..5b8a5ba825bdb 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -683,7 +683,7 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, int dock, return err; } - snd_emu1010_load_firmware_entry(emu, *fw); + snd_emu1010_load_firmware_entry(emu, dock, *fw); return 0; } @@ -699,9 +699,6 @@ static void snd_emu1010_load_dock_firmware(struct snd_emu10k1 *emu) msleep(200); dev_info(emu->card->dev, "emu1010: Loading Audio Dock Firmware\n"); - /* Return to Audio Dock programming mode */ - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, - EMU_HANA_FPGA_CONFIG_AUDIODOCK); err = snd_emu1010_load_firmware(emu, 1, &emu->dock_fw); if (err < 0) return; @@ -820,8 +817,6 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_emu1010_fpga_lock(emu); dev_info(emu->card->dev, "emu1010: Loading Hana Firmware\n"); - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, - EMU_HANA_FPGA_CONFIG_HANA); err = snd_emu1010_load_firmware(emu, 0, &emu->firmware); if (err < 0) { dev_info(emu->card->dev, "emu1010: Loading Firmware failed\n"); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index fafa299efa5c5..b60ab5671e00d 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -422,7 +422,7 @@ void snd_emu1010_update_clock(struct snd_emu10k1 *emu) snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, leds); } -void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, +void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, int dock, const struct firmware *fw_entry) { __always_unused u16 write_post; @@ -440,6 +440,11 @@ void snd_emu1010_load_firmware_entry(struct snd_emu10k1 *emu, // GPO6 -> FPGA CCLK & FPGA input // GPO5 -> FPGA DIN (dual function) + // If the FPGA is already programmed, return it to programming mode + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, + dock ? EMU_HANA_FPGA_CONFIG_AUDIODOCK : + EMU_HANA_FPGA_CONFIG_HANA); + // Assert reset line for 100uS outw(0x00, emu->port + A_GPIO); write_post = inw(emu->port + A_GPIO); From d2d377fc22d23fd38188ea90b051584069a299a2 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:07 -0500 Subject: [PATCH 369/603] ASoC: Intel: nau8825/rt5682: move speaker widget to common modules Two machine drivers sof_rt5682 and sof_nau8825 always register two speaker widgets 'Left Spk' and 'Right Spk' regardless the actual number of speakers. Move the widget registration to speaker common modules to avoid useless speaker widgets for 1 or 4 speaker boards. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_maxim_common.c | 66 ++++++++++++++- sound/soc/intel/boards/sof_nau8825.c | 4 - sound/soc/intel/boards/sof_realtek_common.c | 93 ++++++++++++++++++++- sound/soc/intel/boards/sof_rt5682.c | 32 +++++-- 4 files changed, 180 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index cf29747182716..53c7392bf3ddc 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -12,6 +12,20 @@ #include #include "sof_maxim_common.h" +/* + * Common structures and functions + */ +static const struct snd_kcontrol_new maxim_2spk_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), + +}; + +static const struct snd_soc_dapm_widget maxim_2spk_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + /* helper function to get the number of specific codec */ static unsigned int get_num_codecs(const char *hid) { @@ -135,12 +149,40 @@ EXPORT_SYMBOL_NS(max_98373_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; + unsigned int num_codecs = get_num_codecs(MAX_98373_ACPI_HID); int ret; - ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, - ARRAY_SIZE(max_98373_dapm_routes)); - if (ret) - dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + switch (num_codecs) { + case 2: + ret = snd_soc_dapm_new_controls(&card->dapm, maxim_2spk_widgets, + ARRAY_SIZE(maxim_2spk_widgets)); + if (ret) { + dev_err(rtd->dev, "fail to add max98373 widgets, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, maxim_2spk_kcontrols, + ARRAY_SIZE(maxim_2spk_kcontrols)); + if (ret) { + dev_err(rtd->dev, "fail to add max98373 kcontrols, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, + ARRAY_SIZE(max_98373_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "fail to add max98373 routes, ret %d\n", + ret); + return ret; + } + break; + default: + dev_err(rtd->dev, "max98373: invalid num_codecs %d\n", num_codecs); + return -EINVAL; + } + return ret; } EXPORT_SYMBOL_NS(max_98373_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); @@ -287,6 +329,22 @@ static int max_98390_init(struct snd_soc_pcm_runtime *rtd) fallthrough; case 2: /* add regular speakers dapm route */ + ret = snd_soc_dapm_new_controls(&card->dapm, maxim_2spk_widgets, + ARRAY_SIZE(maxim_2spk_widgets)); + if (ret) { + dev_err(rtd->dev, "fail to add max98390 woofer widgets, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, maxim_2spk_kcontrols, + ARRAY_SIZE(maxim_2spk_kcontrols)); + if (ret) { + dev_err(rtd->dev, "fail to add max98390 woofer kcontrols, ret %d\n", + ret); + return ret; + } + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes, ARRAY_SIZE(max_98390_dapm_routes)); if (ret) { diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index fe5f9e8dd6520..dd4dc4ec42556 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -140,15 +140,11 @@ static int sof_card_late_probe(struct snd_soc_card *card) static const struct snd_kcontrol_new sof_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), }; static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), }; static const struct snd_soc_dapm_route sof_map[] = { diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index 20d094c52d283..cb1c21d822f6c 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -21,6 +21,17 @@ /* * Common structures and functions */ +static const struct snd_kcontrol_new realtek_2spk_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), + +}; + +static const struct snd_soc_dapm_widget realtek_2spk_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_kcontrol_new realtek_4spk_kcontrols[] = { SOC_DAPM_PIN_SWITCH("WL Ext Spk"), SOC_DAPM_PIN_SWITCH("WR Ext Spk"), @@ -181,6 +192,22 @@ static int rt1011_init(struct snd_soc_pcm_runtime *rtd) switch (num_codecs) { case 2: if (!soc_intel_is_cml()) { + ret = snd_soc_dapm_new_controls(&card->dapm, realtek_2spk_widgets, + ARRAY_SIZE(realtek_2spk_widgets)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1011 widgets, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, realtek_2spk_kcontrols, + ARRAY_SIZE(realtek_2spk_kcontrols)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1011 kcontrols, ret %d\n", + ret); + return ret; + } + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map_lr, ARRAY_SIZE(speaker_map_lr)); if (ret) { @@ -357,6 +384,20 @@ static int rt1015p_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; int ret; + ret = snd_soc_dapm_new_controls(&card->dapm, realtek_2spk_widgets, + ARRAY_SIZE(realtek_2spk_widgets)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1015p widgets, ret %d\n", ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, realtek_2spk_kcontrols, + ARRAY_SIZE(realtek_2spk_kcontrols)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1015p kcontrols, ret %d\n", ret); + return ret; + } + if (rt1015p_get_num_codecs() == 1) ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_1dev_dapm_routes, ARRAY_SIZE(rt1015p_1dev_dapm_routes)); @@ -486,8 +527,42 @@ static struct snd_soc_dai_link_component rt1015_components[] = { static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) { - return snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, - ARRAY_SIZE(speaker_map_lr)); + struct snd_soc_card *card = rtd->card; + unsigned int num_codecs = get_num_codecs(RT1015_ACPI_HID); + int ret; + + switch (num_codecs) { + case 2: + ret = snd_soc_dapm_new_controls(&card->dapm, realtek_2spk_widgets, + ARRAY_SIZE(realtek_2spk_widgets)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1015 widgets, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, realtek_2spk_kcontrols, + ARRAY_SIZE(realtek_2spk_kcontrols)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1015 kcontrols, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&rtd->card->dapm, speaker_map_lr, + ARRAY_SIZE(speaker_map_lr)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1015 routes, ret %d\n", + ret); + return ret; + } + break; + default: + dev_err(rtd->dev, "rt1015: invalid num_codecs %d\n", num_codecs); + return -EINVAL; + } + + return ret; } void sof_rt1015_codec_conf(struct snd_soc_card *card) @@ -624,6 +699,20 @@ static int rt1019p_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; int ret; + ret = snd_soc_dapm_new_controls(&card->dapm, realtek_2spk_widgets, + ARRAY_SIZE(realtek_2spk_widgets)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1019p widgets, ret %d\n", ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, realtek_2spk_kcontrols, + ARRAY_SIZE(realtek_2spk_kcontrols)); + if (ret) { + dev_err(rtd->dev, "fail to add rt1019p kcontrols, ret %d\n", ret); + return ret; + } + ret = snd_soc_dapm_add_routes(&card->dapm, rt1019p_dapm_routes, ARRAY_SIZE(rt1019p_dapm_routes)); if (ret) { diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 062012de3de22..1e70ce4696508 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -431,16 +431,11 @@ static int sof_card_late_probe(struct snd_soc_card *card) static const struct snd_kcontrol_new sof_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), - }; static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), }; static const struct snd_soc_dapm_route sof_map[] = { @@ -452,6 +447,17 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_kcontrol_new rt5650_spk_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), + +}; + +static const struct snd_soc_dapm_widget rt5650_spk_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { /* speaker */ { "Left Spk", NULL, "SPOL" }, @@ -463,6 +469,22 @@ static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; int ret; + ret = snd_soc_dapm_new_controls(&card->dapm, rt5650_spk_widgets, + ARRAY_SIZE(rt5650_spk_widgets)); + if (ret) { + dev_err(rtd->dev, "fail to add rt5650 spk widgets, ret %d\n", + ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, rt5650_spk_kcontrols, + ARRAY_SIZE(rt5650_spk_kcontrols)); + if (ret) { + dev_err(rtd->dev, "fail to add rt5650 spk kcontrols, ret %d\n", + ret); + return ret; + } + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, ARRAY_SIZE(rt5650_spk_dapm_routes)); if (ret) From 3b3ed4752600b6462c184edc3284dcc277891aa6 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:08 -0500 Subject: [PATCH 370/603] ASoC: Intel: sof_maxim_common: support MAX98390 on cml boards Add support for 2xMAX98390 speaker amplifier running in I2S mode for existing cml boards. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_maxim_common.c | 42 +++++++++++++++++++---- 1 file changed, 35 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index 53c7392bf3ddc..bb12aab15ad5e 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -10,6 +10,7 @@ #include #include #include +#include "../common/soc-intel-quirks.h" #include "sof_maxim_common.h" /* @@ -219,6 +220,17 @@ static const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { { "TR Spk", NULL, "Tweeter Right BE_OUT" }, }; +static struct snd_soc_codec_conf max_98390_cml_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Right", + }, +}; + static struct snd_soc_codec_conf max_98390_codec_conf[] = { { .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), @@ -271,6 +283,7 @@ static int max_98390_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *codec_dai; int i, ret; @@ -280,13 +293,24 @@ static int max_98390_hw_params(struct snd_pcm_substream *substream, return -ENODEV; } - ret = snd_soc_dai_set_tdm_slot(codec_dai, max_98390_tdm_mask[i].tx, - max_98390_tdm_mask[i].rx, 4, - params_width(params)); - if (ret < 0) { - dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n", - ret); - return ret; + switch (dai_link->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* 4-slot TDM */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, + max_98390_tdm_mask[i].tx, + max_98390_tdm_mask[i].rx, + 4, + params_width(params)); + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n", + ret); + return ret; + } + break; + default: + dev_dbg(codec_dai->dev, "codec is in I2S mode\n"); + break; } } return 0; @@ -395,6 +419,10 @@ void max_98390_set_codec_conf(struct device *dev, struct snd_soc_card *card) switch (num_codecs) { case 2: + if (soc_intel_is_cml()) + card->codec_conf = max_98390_cml_codec_conf; + + fallthrough; case 4: card->num_configs = num_codecs; break; From b7654a7e217704694ed3e484f0a3c415f8678896 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:09 -0500 Subject: [PATCH 371/603] ASoC: Intel: sof_da7219: support MAX98390 Add support to Maxim MAX98390 speaker amplifier. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_da7219.c | 6 ++++++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 0ad7b0a1e237a..e300af4d6b9f8 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -614,6 +614,7 @@ config SND_SOC_INTEL_SOF_DA7219_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_INTEL_SOF_MAXIM_COMMON diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index cbbce07ec5f72..de5b0a8b5b8c0 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -281,6 +281,9 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, return -EINVAL; } break; + case CODEC_MAX98390: + max_98390_dai_link(dev, ctx->amp_link); + break; default: dev_err(dev, "invalid amp type %d\n", ctx->amp_type); return -EINVAL; @@ -352,6 +355,9 @@ static int audio_probe(struct platform_device *pdev) case CODEC_MAX98373: max_98373_set_codec_conf(&card_da7219); break; + case CODEC_MAX98390: + max_98390_set_codec_conf(&pdev->dev, &card_da7219); + break; case CODEC_MAX98360A: case CODEC_NONE: /* no codec conf required */ From a0cf86d813d22d80046d83e9b36c1c2308903956 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:10 -0500 Subject: [PATCH 372/603] ASoC: Intel: sof_da7219: add cml_da7219_def for cml boards Add the board config cml_da7219_def to da7219 machine driver for all cml boards using default SSP port allocation (headphone codec on SSP0, speaker amplifiers on SSP1). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index de5b0a8b5b8c0..69555eee7c7a9 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -19,8 +19,9 @@ #include "sof_maxim_common.h" /* Driver-specific board quirks: from bit 0 to 7 */ -#define SOF_DA7219_JSL_BOARD BIT(0) -#define SOF_DA7219_MCLK_EN BIT(1) +#define SOF_DA7219_CML_BOARD BIT(0) +#define SOF_DA7219_JSL_BOARD BIT(1) +#define SOF_DA7219_MCLK_EN BIT(2) #define DIALOG_CODEC_DAI "da7219-hifi" @@ -292,6 +293,14 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, return 0; } +#define CML_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \ + SOF_LINK_CODEC, \ + SOF_LINK_DMIC01, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_DMIC16K, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE) + #define JSL_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \ SOF_LINK_CODEC, \ SOF_LINK_DMIC01, \ @@ -320,7 +329,10 @@ static int audio_probe(struct platform_device *pdev) if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - if (board_quirk & SOF_DA7219_JSL_BOARD) { + if (board_quirk & SOF_DA7219_CML_BOARD) { + /* overwrite the DAI link order for CML boards */ + ctx->link_order_overwrite = CML_LINK_ORDER; + } else if (board_quirk & SOF_DA7219_JSL_BOARD) { ctx->da7219.is_jsl_board = true; /* overwrite the DAI link order for JSL boards */ @@ -380,6 +392,12 @@ static int audio_probe(struct platform_device *pdev) } static const struct platform_device_id board_ids[] = { + { + .name = "cml_da7219_def", + .driver_data = (kernel_ulong_t)(SOF_DA7219_CML_BOARD | + SOF_SSP_PORT_CODEC(0) | + SOF_SSP_PORT_AMP(1)), + }, { .name = "jsl_da7219_def", .driver_data = (kernel_ulong_t)(SOF_DA7219_JSL_BOARD | From 5a2bc761fab23e47ef20a847476d504548dfb417 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:11 -0500 Subject: [PATCH 373/603] ASoC: Intel: sof_da7219: support MAX98390 on cml boards For cml boards, MAX98390 speaker amplifier is supported by machine driver bxt_da7219_max98357a with sound card name cml_max98390_da7219. Use same name for backward compatibility with existing devices on market. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 33 ++++++++++++++++++++++++----- 1 file changed, 28 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 69555eee7c7a9..0abb71a10a726 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -313,6 +313,7 @@ static int audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct sof_card_private *ctx; + char *card_name; unsigned long board_quirk = 0; int ret; @@ -332,6 +333,21 @@ static int audio_probe(struct platform_device *pdev) if (board_quirk & SOF_DA7219_CML_BOARD) { /* overwrite the DAI link order for CML boards */ ctx->link_order_overwrite = CML_LINK_ORDER; + + /* backward-compatible with existing devices */ + switch (ctx->amp_type) { + case CODEC_MAX98390: + card_name = devm_kstrdup(&pdev->dev, + "cml_max98390_da7219", + GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + card_da7219.name = card_name; + break; + default: + break; + } } else if (board_quirk & SOF_DA7219_JSL_BOARD) { ctx->da7219.is_jsl_board = true; @@ -341,13 +357,20 @@ static int audio_probe(struct platform_device *pdev) /* backward-compatible with existing devices */ switch (ctx->amp_type) { case CODEC_MAX98360A: - card_da7219.name = devm_kstrdup(&pdev->dev, - "da7219max98360a", - GFP_KERNEL); + card_name = devm_kstrdup(&pdev->dev, "da7219max98360a", + GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + card_da7219.name = card_name; break; case CODEC_MAX98373: - card_da7219.name = devm_kstrdup(&pdev->dev, "da7219max", - GFP_KERNEL); + card_name = devm_kstrdup(&pdev->dev, "da7219max", + GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + card_da7219.name = card_name; break; default: break; From 12e5fe68d0bb079cc5228d09280c1d82a61f0d18 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:12 -0500 Subject: [PATCH 374/603] ASoC: Intel: bxt_da7219_max98357a: remove MAX98390 support Remove MAX98390 support and use sof_da7219 machine driver instead for existing cml boards with MAX98390 speaker amplifier. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 6 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 109 +----------------- .../intel/common/soc-acpi-intel-cml-match.c | 2 +- 3 files changed, 6 insertions(+), 111 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index e300af4d6b9f8..44cf1a8df5104 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -301,7 +301,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC tristate select SND_SOC_DA7219 select SND_SOC_MAX98357A - select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON @@ -313,14 +312,14 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH - tristate "Broxton with DA7219 and MAX98357A/MAX98390 in I2S Mode" + tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Broxton-P platforms - with DA7219 + MAX98357A/MAX98390 I2S audio codec. + with DA7219 + MAX98357A I2S audio codec. Say Y or m if you have such a device. This is a recommended option. If unsure select "N". @@ -587,6 +586,7 @@ config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON + imply SND_SOC_INTEL_SOF_DA7219_MACH help This adds support for ASoC machine driver for Cometlake platforms with DA7219 + MAX98357A I2S audio codec. diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3fe3f38c6cb69..9f167c828d1ce 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -24,14 +24,9 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" -#define MAX98390_DEV0_NAME "i2c-MX98390:00" -#define MAX98390_DEV1_NAME "i2c-MX98390:01" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 -#define SPKAMP_MAX98357A 1 -#define SPKAMP_MAX98390 2 - static struct snd_soc_jack broxton_headset; static struct snd_soc_jack broxton_hdmi[3]; @@ -44,7 +39,6 @@ struct bxt_hdmi_pcm { struct bxt_card_private { struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; - int spkamp; }; enum { @@ -91,17 +85,9 @@ static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Line Out"), -}; - -static const struct snd_kcontrol_new max98357a_controls[] = { SOC_DAPM_PIN_SWITCH("Spk"), }; -static const struct snd_kcontrol_new max98390_controls[] = { - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), -}; - static const struct snd_soc_dapm_widget broxton_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -112,17 +98,9 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = { SND_SOC_DAPM_SPK("HDMI3", NULL), SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), -}; - -static const struct snd_soc_dapm_widget max98357a_widgets[] = { SND_SOC_DAPM_SPK("Spk", NULL), }; -static const struct snd_soc_dapm_widget max98390_widgets[] = { - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), -}; - static const struct snd_soc_dapm_route audio_map[] = { /* HP jack connectors - unknown if we have jack detection */ {"Headphone Jack", NULL, "HPL"}, @@ -153,19 +131,11 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "Platform Clock" }, { "Headset Mic", NULL, "Platform Clock" }, { "Line Out", NULL, "Platform Clock" }, -}; -static const struct snd_soc_dapm_route max98357a_routes[] = { /* speaker */ {"Spk", NULL, "Speaker"}, }; -static const struct snd_soc_dapm_route max98390_routes[] = { - /* Speaker */ - {"Left Spk", NULL, "Left BE_OUT"}, - {"Right Spk", NULL, "Right BE_OUT"}, -}; - static const struct snd_soc_dapm_route broxton_map[] = { {"HiFi Playback", NULL, "ssp5 Tx"}, {"ssp5 Tx", NULL, "codec0_out"}, @@ -453,10 +423,6 @@ SND_SOC_DAILINK_DEF(ssp5_pin, SND_SOC_DAILINK_DEF(ssp5_codec, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", BXT_MAXIM_CODEC_DAI))); -SND_SOC_DAILINK_DEF(max98390_codec, - DAILINK_COMP_ARRAY( - /* Left */ COMP_CODEC(MAX98390_DEV0_NAME, "max98390-aif1"), - /* Right */ COMP_CODEC(MAX98390_DEV1_NAME, "max98390-aif1"))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); @@ -654,69 +620,15 @@ static struct snd_soc_dai_link broxton_dais[] = { }, }; -static struct snd_soc_codec_conf max98390_codec_confs[] = { - { - .dlc = COMP_CODEC_CONF(MAX98390_DEV0_NAME), - .name_prefix = "Left", - }, - { - .dlc = COMP_CODEC_CONF(MAX98390_DEV1_NAME), - .name_prefix = "Right", - }, -}; - #define NAME_SIZE 32 static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; - const struct snd_kcontrol_new *controls; - const struct snd_soc_dapm_widget *widgets; - const struct snd_soc_dapm_route *routes; - int num_controls, num_widgets, num_routes, err, i = 0; + int err, i = 0; char jack_name[NAME_SIZE]; - switch (ctx->spkamp) { - case SPKAMP_MAX98357A: - controls = max98357a_controls; - num_controls = ARRAY_SIZE(max98357a_controls); - widgets = max98357a_widgets; - num_widgets = ARRAY_SIZE(max98357a_widgets); - routes = max98357a_routes; - num_routes = ARRAY_SIZE(max98357a_routes); - break; - case SPKAMP_MAX98390: - controls = max98390_controls; - num_controls = ARRAY_SIZE(max98390_controls); - widgets = max98390_widgets; - num_widgets = ARRAY_SIZE(max98390_widgets); - routes = max98390_routes; - num_routes = ARRAY_SIZE(max98390_routes); - break; - default: - dev_err(card->dev, "Invalid speaker amplifier %d\n", ctx->spkamp); - return -EINVAL; - } - - err = snd_soc_dapm_new_controls(&card->dapm, widgets, num_widgets); - if (err) { - dev_err(card->dev, "Fail to new widgets\n"); - return err; - } - - err = snd_soc_add_card_controls(card, controls, num_controls); - if (err) { - dev_err(card->dev, "Fail to add controls\n"); - return err; - } - - err = snd_soc_dapm_add_routes(&card->dapm, routes, num_routes); - if (err) { - dev_err(card->dev, "Fail to add routes\n"); - return err; - } - if (soc_intel_is_glk()) snd_soc_dapm_add_routes(&card->dapm, gemini_map, ARRAY_SIZE(gemini_map)); @@ -785,11 +697,6 @@ static int broxton_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - if (acpi_dev_present("MX98390", NULL, -1)) - ctx->spkamp = SPKAMP_MAX98390; - else - ctx->spkamp = SPKAMP_MAX98357A; - broxton_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&broxton_audio_card, ctx); if (soc_intel_is_glk()) { @@ -817,13 +724,7 @@ static int broxton_audio_probe(struct platform_device *pdev) } else if (soc_intel_is_cml()) { unsigned int i; - if (ctx->spkamp == SPKAMP_MAX98390) { - broxton_audio_card.name = "cml_max98390_da7219"; - - broxton_audio_card.codec_conf = max98390_codec_confs; - broxton_audio_card.num_configs = ARRAY_SIZE(max98390_codec_confs); - } else - broxton_audio_card.name = "cmlda7219max"; + broxton_audio_card.name = "cmlda7219max"; for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { if (!broxton_dais[i].codecs->dai_name) @@ -834,12 +735,6 @@ static int broxton_audio_probe(struct platform_device *pdev) BXT_MAXIM_CODEC_DAI)) { broxton_dais[i].name = "SSP1-Codec"; broxton_dais[i].cpus->dai_name = "SSP1 Pin"; - - if (ctx->spkamp == SPKAMP_MAX98390) { - broxton_dais[i].codecs = max98390_codec; - broxton_dais[i].num_codecs = ARRAY_SIZE(max98390_codec); - broxton_dais[i].dpcm_capture = 1; - } } /* DIALOG_CODEC is connected to SSP0 */ else if (!strcmp(broxton_dais[i].codecs->dai_name, diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 160bfa0e5ef5f..4217dbb716dcf 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -75,7 +75,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }, { .id = "DLGS7219", - .drv_name = "cml_da7219_mx98357a", + .drv_name = "cml_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &max98390_spk_codecs, .sof_tplg_filename = "sof-cml-da7219-max98390.tplg", From 5b093b0b47efefbb928c4d3a5e982bbeaefda44c Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 26 Apr 2024 10:25:13 -0500 Subject: [PATCH 375/603] ASoC: Intel: sof_rt5682: add missing MAX98357A config sof_rt5682 supports MAX98357A/MAX98360A amp. Select SND_SOC_MAX98357A as needed. Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 44cf1a8df5104..a61c3d0435c4c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -486,6 +486,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) + select SND_SOC_MAX98357A select SND_SOC_MAX98373_I2C select SND_SOC_MAX98390 select SND_SOC_RT1011 From 7873252c0e44ef5c0e04cf3c7c25f5e2c5180272 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:14 -0500 Subject: [PATCH 376/603] ASoC: Intel: sof_da7219: support MAX98357A Add support to Maxim MAX98357A speaker amplifier. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 0abb71a10a726..610b5a8e08600 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -267,6 +267,9 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, /* codec-specific fields for speaker amplifier */ switch (ctx->amp_type) { + case CODEC_MAX98357A: + max_98357a_dai_link(ctx->amp_link); + break; case CODEC_MAX98360A: max_98360a_dai_link(ctx->amp_link); break; @@ -393,6 +396,7 @@ static int audio_probe(struct platform_device *pdev) case CODEC_MAX98390: max_98390_set_codec_conf(&pdev->dev, &card_da7219); break; + case CODEC_MAX98357A: case CODEC_MAX98360A: case CODEC_NONE: /* no codec conf required */ From e895d16f4e8f0e5d90d7188e8fd9c507a97fb8b7 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:15 -0500 Subject: [PATCH 377/603] ASoC: Intel: sof_da7219: support MAX98357A on cml boards For cml boards, MAX98357A speaker amplifier is supported by machine driver bxt_da7219_max98357a with sound card name cmlda7219max. Use same name for backward compatibility with existing devices on market. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index 610b5a8e08600..ecb1d4b25ea60 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -339,6 +339,14 @@ static int audio_probe(struct platform_device *pdev) /* backward-compatible with existing devices */ switch (ctx->amp_type) { + case CODEC_MAX98357A: + card_name = devm_kstrdup(&pdev->dev, "cmlda7219max", + GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + card_da7219.name = card_name; + break; case CODEC_MAX98390: card_name = devm_kstrdup(&pdev->dev, "cml_max98390_da7219", From 35ca48662cdf0749a2b9931d625690967fbce032 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:16 -0500 Subject: [PATCH 378/603] ASoC: Intel: bxt_da7219_max98357a: remove cml support Remove cml platform support and use sof_da7219 machine driver instead for existing cml boards with MAX98357A speaker amplifier. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 6 ++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 28 +------------------ .../intel/common/soc-acpi-intel-cml-match.c | 2 +- 3 files changed, 4 insertions(+), 32 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index a61c3d0435c4c..134512a7d3b8e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -584,13 +584,11 @@ if (SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK) config SND_SOC_INTEL_CML_LP_DA7219_MAX98357A_MACH tristate "CML_LP with DA7219 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON imply SND_SOC_INTEL_SOF_DA7219_MACH help This adds support for ASoC machine driver for Cometlake platforms - with DA7219 + MAX98357A I2S audio codec. + with DA7219 + MAX98357A I2S audio codec. This option is deprecated + and please use SND_SOC_INTEL_SOF_DA7219_MACH instead. Say Y or m if you have such a device. This is a recommended option. If unsure select "N". diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 9f167c828d1ce..45fe1d7b88658 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -201,10 +201,7 @@ static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) int clk_freq; /* Configure sysclk for codec */ - if (soc_intel_is_cml()) - clk_freq = 24000000; - else - clk_freq = 19200000; + clk_freq = 19200000; ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq, SND_SOC_CLOCK_IN); @@ -721,28 +718,6 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_dais[i].cpus->dai_name = "SSP2 Pin"; } } - } else if (soc_intel_is_cml()) { - unsigned int i; - - broxton_audio_card.name = "cmlda7219max"; - - for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { - if (!broxton_dais[i].codecs->dai_name) - continue; - - /* MAXIM_CODEC is connected to SSP1. */ - if (!strcmp(broxton_dais[i].codecs->dai_name, - BXT_MAXIM_CODEC_DAI)) { - broxton_dais[i].name = "SSP1-Codec"; - broxton_dais[i].cpus->dai_name = "SSP1 Pin"; - } - /* DIALOG_CODEC is connected to SSP0 */ - else if (!strcmp(broxton_dais[i].codecs->dai_name, - BXT_DIALOG_CODEC_DAI)) { - broxton_dais[i].name = "SSP0-Codec"; - broxton_dais[i].cpus->dai_name = "SSP0 Pin"; - } - } } /* override platform name, if required */ @@ -762,7 +737,6 @@ static int broxton_audio_probe(struct platform_device *pdev) static const struct platform_device_id bxt_board_ids[] = { { .name = "bxt_da7219_mx98357a" }, { .name = "glk_da7219_mx98357a" }, - { .name = "cml_da7219_mx98357a" }, { } }; MODULE_DEVICE_TABLE(platform, bxt_board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 4217dbb716dcf..f79d7558174a1 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -68,7 +68,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }, { .id = "DLGS7219", - .drv_name = "cml_da7219_mx98357a", + .drv_name = "cml_da7219_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &max98357a_spk_codecs, .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", From c7f9523d21d49b56498ac58e4b1afcb930eb551a Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:17 -0500 Subject: [PATCH 379/603] ASoC: Intel: sof_da7219: add glk_da7219_def for glk boards Add the board config glk_da7219_def to da7219 machine driver for all glk boards using default SSP port allocation (headphone codec on SSP2, speaker amplifiers on SSP1). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index ecb1d4b25ea60..cc5e99b214564 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -19,9 +19,10 @@ #include "sof_maxim_common.h" /* Driver-specific board quirks: from bit 0 to 7 */ -#define SOF_DA7219_CML_BOARD BIT(0) -#define SOF_DA7219_JSL_BOARD BIT(1) -#define SOF_DA7219_MCLK_EN BIT(2) +#define SOF_DA7219_GLK_BOARD BIT(0) +#define SOF_DA7219_CML_BOARD BIT(1) +#define SOF_DA7219_JSL_BOARD BIT(2) +#define SOF_DA7219_MCLK_EN BIT(3) #define DIALOG_CODEC_DAI "da7219-hifi" @@ -296,6 +297,14 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, return 0; } +#define GLK_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \ + SOF_LINK_CODEC, \ + SOF_LINK_DMIC01, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE) + #define CML_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \ SOF_LINK_CODEC, \ SOF_LINK_DMIC01, \ @@ -333,7 +342,13 @@ static int audio_probe(struct platform_device *pdev) if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - if (board_quirk & SOF_DA7219_CML_BOARD) { + if (board_quirk & SOF_DA7219_GLK_BOARD) { + /* dmic16k not support */ + ctx->dmic_be_num = 1; + + /* overwrite the DAI link order for GLK boards */ + ctx->link_order_overwrite = GLK_LINK_ORDER; + } else if (board_quirk & SOF_DA7219_CML_BOARD) { /* overwrite the DAI link order for CML boards */ ctx->link_order_overwrite = CML_LINK_ORDER; @@ -427,6 +442,12 @@ static int audio_probe(struct platform_device *pdev) } static const struct platform_device_id board_ids[] = { + { + .name = "glk_da7219_def", + .driver_data = (kernel_ulong_t)(SOF_DA7219_GLK_BOARD | + SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(1)), + }, { .name = "cml_da7219_def", .driver_data = (kernel_ulong_t)(SOF_DA7219_CML_BOARD | From f46b768b7281e4ef784d06788be2f941c13eddde Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:18 -0500 Subject: [PATCH 380/603] ASoC: Intel: sof_da7219: support MAX98357A on glk boards For glk boards, MAX98357A speaker amplifier is supported by machine driver bxt_da7219_max98357a with sound card name glkda7219max. Use same name for backward compatibility with existing devices on market. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index cc5e99b214564..d7b605896f895 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -348,6 +348,20 @@ static int audio_probe(struct platform_device *pdev) /* overwrite the DAI link order for GLK boards */ ctx->link_order_overwrite = GLK_LINK_ORDER; + + /* backward-compatible with existing devices */ + switch (ctx->amp_type) { + case CODEC_MAX98357A: + card_name = devm_kstrdup(&pdev->dev, "glkda7219max", + GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + card_da7219.name = card_name; + break; + default: + break; + } } else if (board_quirk & SOF_DA7219_CML_BOARD) { /* overwrite the DAI link order for CML boards */ ctx->link_order_overwrite = CML_LINK_ORDER; From 8d6114b81d72c522f1ad55cd84ed37699d58c840 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:19 -0500 Subject: [PATCH 381/603] ASoC: Intel: bxt_da7219_max98357a: remove glk support Remove glk platform support and use sof_da7219 machine driver instead for existing glk boards with MAX98357A speaker amplifier. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 14 ++---- sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 44 ------------------- .../intel/common/soc-acpi-intel-glk-match.c | 2 +- 4 files changed, 6 insertions(+), 56 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 134512a7d3b8e..d48b196817b48 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -305,10 +305,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON -config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON - tristate - select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC - if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH @@ -316,7 +312,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI - select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON + select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC help This adds support for ASoC machine driver for Broxton-P platforms with DA7219 + MAX98357A I2S audio codec. @@ -432,13 +428,11 @@ if SND_SOC_SOF_GEMINILAKE config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH tristate "GLK with DA7219 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON + imply SND_SOC_INTEL_SOF_DA7219_MACH help This adds support for ASoC machine driver for Geminilake platforms - with DA7219 + MAX98357A I2S audio codec. + with DA7219 + MAX98357A I2S audio codec. This option is deprecated + and please use SND_SOC_INTEL_SOF_DA7219_MACH instead. Say Y or m if you have such a device. This is a recommended option. If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 119413c262de7..e4bd5689eb1c5 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -51,7 +51,7 @@ obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MACH) += snd-soc-sof_da7219.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-hsw-rt5640.o -obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o +obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o obj-$(CONFIG_SND_SOC_INTEL_SOF_WM8804_MACH) += snd-soc-sst-sof-wm8804.o diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 45fe1d7b88658..57e75f87bea62 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -134,9 +134,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* speaker */ {"Spk", NULL, "Speaker"}, -}; -static const struct snd_soc_dapm_route broxton_map[] = { {"HiFi Playback", NULL, "ssp5 Tx"}, {"ssp5 Tx", NULL, "codec0_out"}, @@ -147,17 +145,6 @@ static const struct snd_soc_dapm_route broxton_map[] = { {"ssp1 Rx", NULL, "Capture"}, }; -static const struct snd_soc_dapm_route gemini_map[] = { - {"HiFi Playback", NULL, "ssp1 Tx"}, - {"ssp1 Tx", NULL, "codec0_out"}, - - {"Playback", NULL, "ssp2 Tx"}, - {"ssp2 Tx", NULL, "codec1_out"}, - - {"codec0_in", NULL, "ssp2 Rx"}, - {"ssp2 Rx", NULL, "Capture"}, -}; - static struct snd_soc_jack_pin jack_pins[] = { { .pin = "Headphone Jack", @@ -626,13 +613,6 @@ static int bxt_card_late_probe(struct snd_soc_card *card) int err, i = 0; char jack_name[NAME_SIZE]; - if (soc_intel_is_glk()) - snd_soc_dapm_add_routes(&card->dapm, gemini_map, - ARRAY_SIZE(gemini_map)); - else - snd_soc_dapm_add_routes(&card->dapm, broxton_map, - ARRAY_SIZE(broxton_map)); - if (list_empty(&ctx->hdmi_pcm_list)) return -EINVAL; @@ -696,29 +676,6 @@ static int broxton_audio_probe(struct platform_device *pdev) broxton_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&broxton_audio_card, ctx); - if (soc_intel_is_glk()) { - unsigned int i; - - broxton_audio_card.name = "glkda7219max"; - /* Fixup the SSP entries for geminilake */ - for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { - if (!broxton_dais[i].codecs->dai_name) - continue; - - /* MAXIM_CODEC is connected to SSP1. */ - if (!strcmp(broxton_dais[i].codecs->dai_name, - BXT_MAXIM_CODEC_DAI)) { - broxton_dais[i].name = "SSP1-Codec"; - broxton_dais[i].cpus->dai_name = "SSP1 Pin"; - } - /* DIALOG_CODE is connected to SSP2 */ - else if (!strcmp(broxton_dais[i].codecs->dai_name, - BXT_DIALOG_CODEC_DAI)) { - broxton_dais[i].name = "SSP2-Codec"; - broxton_dais[i].cpus->dai_name = "SSP2 Pin"; - } - } - } /* override platform name, if required */ mach = pdev->dev.platform_data; @@ -736,7 +693,6 @@ static int broxton_audio_probe(struct platform_device *pdev) static const struct platform_device_id bxt_board_ids[] = { { .name = "bxt_da7219_mx98357a" }, - { .name = "glk_da7219_mx98357a" }, { } }; MODULE_DEVICE_TABLE(platform, bxt_board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 8911c90bbaf68..1d30991afdc84 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -33,7 +33,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { }, { .id = "DLGS7219", - .drv_name = "glk_da7219_mx98357a", + .drv_name = "glk_da7219_def", .fw_filename = "intel/dsp_fw_glk.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &glk_codecs, From 44567d3d62dfe9df514299a98429a59129e0a2d0 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:20 -0500 Subject: [PATCH 382/603] ASoC: Intel: sof_rt5682: add glk_rt5682_def for glk boards Add the board config glk_rt5682_def to rt5682 machine driver for all glk boards using default SSP port allocation (headphone codec on SSP2, speaker amplifiers on SSP1). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-15-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 48 +++++++++++++++++++++-------- 1 file changed, 36 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 1e70ce4696508..80ecf31c936a4 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -635,6 +635,14 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, return 0; } +#define GLK_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_AMP, \ + SOF_LINK_CODEC, \ + SOF_LINK_DMIC01, \ + SOF_LINK_IDISP_HDMI, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE) + static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; @@ -676,24 +684,34 @@ static int sof_audio_probe(struct platform_device *pdev) ctx->amp_type = CODEC_RT5650; } - if (ctx->amp_type == CODEC_RT1011 && soc_intel_is_cml()) { - /* backward-compatible with existing cml devices */ - card_name = devm_kstrdup(&pdev->dev, "cml_rt1011_rt5682", - GFP_KERNEL); - if (!card_name) - return -ENOMEM; - - sof_audio_card_rt5682.name = card_name; - } + if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) + ctx->hdmi.idisp_codec = true; if (is_legacy_cpu) { ctx->rt5682.is_legacy_cpu = true; ctx->dmic_be_num = 0; /* HDMI is not supported by SOF on Baytrail/CherryTrail */ ctx->hdmi_num = 0; - } else { - if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) - ctx->hdmi.idisp_codec = true; + } else if (soc_intel_is_glk()) { + /* dmic16k not support */ + ctx->dmic_be_num = 1; + + /* overwrite the DAI link order for GLK boards */ + ctx->link_order_overwrite = GLK_LINK_ORDER; + } else if (soc_intel_is_cml()) { + /* backward-compatible with existing devices */ + switch (ctx->amp_type) { + case CODEC_RT1011: + card_name = devm_kstrdup(&pdev->dev, "cml_rt1011_rt5682", + GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + sof_audio_card_rt5682.name = card_name; + break; + default: + break; + } } /* need to get main clock from pmc */ @@ -768,6 +786,12 @@ static const struct platform_device_id board_ids[] = { { .name = "sof_rt5682", }, + { + .name = "glk_rt5682_def", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_SSP_PORT_CODEC(2) | + SOF_SSP_PORT_AMP(1)), + }, { .name = "cml_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | From 5498a4edbf314532b7138aabe705faa34fb5df8d Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:21 -0500 Subject: [PATCH 383/603] ASoC: Intel: sof_rt5682: support MAX98357A on glk boards For glk boards, MAX98357A speaker amplifier is supported by machine driver glk_rt5682_mx98357a with sound card name glkrt5682max. Use same name for backward compatibility with existing devices on market. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-16-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 80ecf31c936a4..efca7604040eb 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -698,6 +698,20 @@ static int sof_audio_probe(struct platform_device *pdev) /* overwrite the DAI link order for GLK boards */ ctx->link_order_overwrite = GLK_LINK_ORDER; + + /* backward-compatible with existing devices */ + switch (ctx->amp_type) { + case CODEC_MAX98357A: + card_name = devm_kstrdup(&pdev->dev, "glkrt5682max", + GFP_KERNEL); + if (!card_name) + return -ENOMEM; + + sof_audio_card_rt5682.name = card_name; + break; + default: + break; + } } else if (soc_intel_is_cml()) { /* backward-compatible with existing devices */ switch (ctx->amp_type) { From 0cb6a8134c1cb59d8f41d84968c2e20cfbd7f6cc Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:22 -0500 Subject: [PATCH 384/603] ASoC: Intel: glk_rt5682_max98357a: delete driver Delete this driver and use sof_rt5682 machine driver instead. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-17-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 13 +- sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/glk_rt5682_max98357a.c | 693 ------------------ .../intel/common/soc-acpi-intel-glk-match.c | 2 +- 4 files changed, 4 insertions(+), 706 deletions(-) delete mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index d48b196817b48..e7ea6fe3dcd1b 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -438,18 +438,11 @@ config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH tristate "GLK with RT5682 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_SOC_RT5682_I2C - select SND_SOC_RT5682S - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - select SND_SOC_INTEL_HDA_DSP_COMMON + imply SND_SOC_INTEL_SOF_RT5682_MACH help This adds support for ASoC machine driver for Geminilake platforms - with RT5682 + MAX98357A I2S audio codec. + with RT5682 + MAX98357A I2S audio codec. This option is deprecated + and please use SND_SOC_INTEL_SOF_RT5682_MACH instead. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index e4bd5689eb1c5..949e064908586 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -7,7 +7,6 @@ snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o snd-soc-sst-sof-wm8804-objs := sof_wm8804.o -snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-bytcr-wm5102-objs := bytcr_wm5102.o @@ -55,7 +54,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_ obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o obj-$(CONFIG_SND_SOC_INTEL_SOF_WM8804_MACH) += snd-soc-sst-sof-wm8804.o -obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-bdw-rt286.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c deleted file mode 100644 index 4098b2d32f9bc..0000000000000 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ /dev/null @@ -1,693 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright(c) 2018 Intel Corporation. - -/* - * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs - * - * Modified from: - * Intel Apollolake I2S Machine driver - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/rt5682.h" -#include "../../codecs/rt5682s.h" -#include "../../codecs/hdac_hdmi.h" -#include "hda_dsp_common.h" - -/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ -#define GLK_PLAT_CLK_FREQ 19200000 -#define RT5682_PLL_FREQ (48000 * 512) -#define RT5682_DAI_NAME "rt5682-aif1" -#define RT5682S_DAI_NAME "rt5682s-aif1" -#define GLK_MAXIM_CODEC_DAI "HiFi" -#define RT5682_DEV0_NAME "i2c-10EC5682:00" -#define RT5682S_DEV0_NAME "i2c-RTL5682:00" -#define MAXIM_DEV0_NAME "MX98357A:00" -#define DUAL_CHANNEL 2 -#define QUAD_CHANNEL 4 -#define NAME_SIZE 32 - -static struct snd_soc_jack geminilake_hdmi[3]; - -struct glk_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct glk_card_private { - struct snd_soc_jack geminilake_headset; - struct list_head hdmi_pcm_list; - bool common_hdmi_codec_drv; - int is_rt5682s; -}; - -enum { - GLK_DPCM_AUDIO_PB = 0, - GLK_DPCM_AUDIO_CP, - GLK_DPCM_AUDIO_HS_PB, - GLK_DPCM_AUDIO_ECHO_REF_CP, - GLK_DPCM_AUDIO_REF_CP, - GLK_DPCM_AUDIO_DMIC_CP, - GLK_DPCM_AUDIO_HDMI1_PB, - GLK_DPCM_AUDIO_HDMI2_PB, - GLK_DPCM_AUDIO_HDMI3_PB, -}; - -static const struct snd_kcontrol_new geminilake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), -}; - -static const struct snd_soc_dapm_widget geminilake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SPK("HDMI3", NULL), -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static const struct snd_soc_dapm_route geminilake_map[] = { - /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, - - /* speaker */ - { "Spk", NULL, "Speaker" }, - - /* other jacks */ - { "IN1P", NULL, "Headset Mic" }, - - /* digital mics */ - { "DMic", NULL, "SoC DMIC" }, - - /* CODEC BE connections */ - { "HiFi Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "codec0_out" }, - - { "AIF1 Playback", NULL, "ssp2 Tx" }, - { "ssp2 Tx", NULL, "codec1_out" }, - - { "codec0_in", NULL, "ssp2 Rx" }, - { "ssp2 Rx", NULL, "AIF1 Capture" }, - - { "HDMI1", NULL, "hif5-0 Output" }, - { "HDMI2", NULL, "hif6-0 Output" }, - { "HDMI2", NULL, "hif7-0 Output" }, - - { "hifi3", NULL, "iDisp3 Tx" }, - { "iDisp3 Tx", NULL, "iDisp3_out" }, - { "hifi2", NULL, "iDisp2 Tx" }, - { "iDisp2 Tx", NULL, "iDisp2_out" }, - { "hifi1", NULL, "iDisp1 Tx" }, - { "iDisp1 Tx", NULL, "iDisp1_out" }, - - /* DMIC */ - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "DMIC AIF" }, -}; - -static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will convert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = DUAL_CHANNEL; - - /* set SSP to 24 bit */ - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - - return 0; -} - -static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_jack *jack; - int pll_id, pll_source, clk_id, ret; - - if (ctx->is_rt5682s) { - pll_id = RT5682S_PLL2; - pll_source = RT5682S_PLL_S_MCLK; - clk_id = RT5682S_SCLK_S_PLL2; - } else { - pll_id = RT5682_PLL1; - pll_source = RT5682_PLL1_S_MCLK; - clk_id = RT5682_SCLK_S_PLL1; - } - - ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, - GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec pll: %d\n", ret); - return ret; - } - - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, - RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - /* - * Headset buttons map to the google Reference headset. - * These can be configured by userspace. - */ - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->geminilake_headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); - return ret; - } - - jack = &ctx->geminilake_headset; - - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - - ret = snd_soc_component_set_jack(component, jack, NULL); - - if (ret) { - dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); - return ret; - } - - return ret; -}; - -static int geminilake_rt5682_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int ret; - - /* Set valid bitmask & configuration for I2S in 24 bit */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - - return ret; -} - -static struct snd_soc_ops geminilake_rt5682_ops = { - .hw_params = geminilake_rt5682_hw_params, -}; - -static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) -{ - struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct glk_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - struct snd_soc_dapm_context *dapm; - int ret; - - dapm = snd_soc_component_get_dapm(component); - ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - if (ret) { - dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); - return ret; - } - - return ret; -} - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static unsigned int channels_quad[] = { - QUAD_CHANNEL, -}; - -static struct snd_pcm_hw_constraint_list constraints_channels_quad = { - .count = ARRAY_SIZE(channels_quad), - .list = channels_quad, - .mask = 0, -}; - -static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* - * set BE channel constraint as user FE channels - */ - chan->min = chan->max = 4; - - return 0; -} - -static int geminilake_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels_quad); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops geminilake_dmic_ops = { - .startup = geminilake_dmic_startup, -}; - -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static int geminilake_refcap_startup(struct snd_pcm_substream *substream) -{ - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_16000); -}; - -static const struct snd_soc_ops geminilake_refcap_ops = { - .startup = geminilake_refcap_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(system2, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin2"))); - -SND_SOC_DAILINK_DEF(echoref, - DAILINK_COMP_ARRAY(COMP_CPU("Echoref Pin"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC(MAXIM_DEV0_NAME, - GLK_MAXIM_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP2 Pin"))); -SND_SOC_DAILINK_DEF(ssp2_codec_5682, - DAILINK_COMP_ARRAY(COMP_CODEC(RT5682_DEV0_NAME, - RT5682_DAI_NAME))); -SND_SOC_DAILINK_DEF(ssp2_codec_5682s, - DAILINK_COMP_ARRAY(COMP_CODEC(RT5682S_DEV0_NAME, - RT5682S_DAI_NAME))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:0e.0"))); - -/* geminilake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link geminilake_dais[] = { - /* Front End DAI links */ - [GLK_DPCM_AUDIO_PB] = { - .name = "Glk Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = geminilake_rt5682_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [GLK_DPCM_AUDIO_CP] = { - .name = "Glk Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [GLK_DPCM_AUDIO_HS_PB] = { - .name = "Glk Audio Headset Playback", - .stream_name = "Headset Audio", - .dpcm_playback = 1, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(system2, dummy, platform), - }, - [GLK_DPCM_AUDIO_ECHO_REF_CP] = { - .name = "Glk Audio Echo Reference cap", - .stream_name = "Echoreference Capture", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(echoref, dummy, platform), - }, - [GLK_DPCM_AUDIO_REF_CP] = { - .name = "Glk Audio Reference cap", - .stream_name = "Refcap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &geminilake_refcap_ops, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [GLK_DPCM_AUDIO_DMIC_CP] = { - .name = "Glk Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &geminilake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [GLK_DPCM_AUDIO_HDMI1_PB] = { - .name = "Glk HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [GLK_DPCM_AUDIO_HDMI2_PB] = { - .name = "Glk HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [GLK_DPCM_AUDIO_HDMI3_PB] = { - .name = "Glk HDMI Port3", - .stream_name = "Hdmi3", - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - /* Back End DAI links */ - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = geminilake_ssp_fixup, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - /* SSP2 - Codec */ - .name = "SSP2-Codec", - .id = 1, - .no_pcm = 1, - .init = geminilake_rt5682_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = geminilake_ssp_fixup, - .ops = &geminilake_rt5682_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp2_pin, ssp2_codec_5682, platform), - }, - { - .name = "dmic01", - .id = 2, - .ignore_suspend = 1, - .be_hw_params_fixup = geminilake_dmic_fixup, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .init = geminilake_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = geminilake_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = geminilake_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -static int glk_card_late_probe(struct snd_soc_card *card) -{ - struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_component *component = NULL; - char jack_name[NAME_SIZE]; - struct glk_hdmi_pcm *pcm; - int err; - int i = 0; - - if (list_empty(&ctx->hdmi_pcm_list)) - return -EINVAL; - - if (ctx->common_hdmi_codec_drv) { - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct glk_hdmi_pcm, - head); - component = pcm->codec_dai->component; - return hda_dsp_hdmi_build_controls(card, component); - } - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &geminilake_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &geminilake_hdmi[i]); - if (err < 0) - return err; - - i++; - } - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* geminilake audio machine driver for SPT + RT5682 */ -static struct snd_soc_card glk_audio_card_rt5682_m98357a = { - .name = "glkrt5682max", - .owner = THIS_MODULE, - .dai_link = geminilake_dais, - .num_links = ARRAY_SIZE(geminilake_dais), - .controls = geminilake_controls, - .num_controls = ARRAY_SIZE(geminilake_controls), - .dapm_widgets = geminilake_widgets, - .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), - .dapm_routes = geminilake_map, - .num_dapm_routes = ARRAY_SIZE(geminilake_map), - .fully_routed = true, - .late_probe = glk_card_late_probe, -}; - -static int geminilake_audio_probe(struct platform_device *pdev) -{ - struct glk_card_private *ctx; - struct snd_soc_acpi_mach *mach; - const char *platform_name; - struct snd_soc_card *card; - int ret, i; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - /* Detect the headset codec variant */ - if (acpi_dev_present("RTL5682", NULL, -1)) { - /* ALC5682I-VS is detected */ - ctx->is_rt5682s = 1; - - for (i = 0; i < glk_audio_card_rt5682_m98357a.num_links; i++) { - if (strcmp(geminilake_dais[i].name, "SSP2-Codec")) - continue; - - /* update the dai link to use rt5682s codec */ - geminilake_dais[i].codecs = ssp2_codec_5682s; - geminilake_dais[i].num_codecs = ARRAY_SIZE(ssp2_codec_5682s); - break; - } - } - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - card = &glk_audio_card_rt5682_m98357a; - card->dev = &pdev->dev; - snd_soc_card_set_drvdata(card, ctx); - if (!snd_soc_acpi_sof_parent(&pdev->dev)) - card->disable_route_checks = true; - - /* override platform name, if required */ - mach = pdev->dev.platform_data; - platform_name = mach->mach_params.platform; - - ret = snd_soc_fixup_dai_links_platform_name(card, platform_name); - if (ret) - return ret; - - ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; - - return devm_snd_soc_register_card(&pdev->dev, card); -} - -static const struct platform_device_id glk_board_ids[] = { - { - .name = "glk_rt5682_mx98357a", - .driver_data = - (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, - }, - { } -}; -MODULE_DEVICE_TABLE(platform, glk_board_ids); - -static struct platform_driver geminilake_audio = { - .probe = geminilake_audio_probe, - .driver = { - .name = "glk_rt5682_max98357a", - .pm = &snd_soc_pm_ops, - }, - .id_table = glk_board_ids, -}; -module_platform_driver(geminilake_audio) - -/* Module information */ -MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); -MODULE_AUTHOR("Naveen Manohar "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); -MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 1d30991afdc84..c82c8c93d2004 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -41,7 +41,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { }, { .comp_ids = &glk_rt5682_rt5682s_hp, - .drv_name = "glk_rt5682_mx98357a", + .drv_name = "glk_rt5682_def", .fw_filename = "intel/dsp_fw_glk.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &glk_codecs, From e91d54f8b1bd3393d91fd754a1c40df6f408e84b Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:23 -0500 Subject: [PATCH 385/603] ASoC: Intel: sof-rt5682: add mclk_en to sof_rt5682_private Add a variable mclk_en to sof_rt5682_private structure to reduce global variable access. No functional change. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-18-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.h | 2 ++ sound/soc/intel/boards/sof_rt5682.c | 7 +++++-- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 644f6b4a1b7bc..dfcc2c5c25cc1 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -101,10 +101,12 @@ struct sof_da7219_private { * * @mclk: mclk clock data * @is_legacy_cpu: true for BYT/CHT boards + * @mclk_en: true for mclk pin is connected */ struct sof_rt5682_private { struct clk *mclk; bool is_legacy_cpu; + bool mclk_en; }; /* diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index efca7604040eb..654c2428f625f 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -165,7 +165,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) int extra_jack_data; int ret, mclk_freq; - if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) { + if (ctx->rt5682.mclk_en) { mclk_freq = sof_dai_get_mclk(rtd); if (mclk_freq <= 0) { dev_err(rtd->dev, "invalid mclk freq %d\n", mclk_freq); @@ -278,7 +278,7 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); int pll_id, pll_source, pll_in, pll_out, clk_id, ret; - if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) { + if (ctx->rt5682.mclk_en) { if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ret = clk_prepare_enable(ctx->rt5682.mclk); if (ret < 0) { @@ -728,6 +728,9 @@ static int sof_audio_probe(struct platform_device *pdev) } } + if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) + ctx->rt5682.mclk_en = true; + /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->rt5682.mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); From 207255f3ae4d0cf5034666652668be572d9c5c1e Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:24 -0500 Subject: [PATCH 386/603] ASoC: Intel: sof-rt5682: remove SOF_RT5682_MCLK_BYTCHT_EN We don't need this quirk flag since 'is_legacy_cpu' will be true if this is a BYT/CHT board. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-19-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 39 ++++++++++++++--------------- 1 file changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 654c2428f625f..4c6dfe632edfc 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -30,7 +30,6 @@ /* Driver-specific board quirks: from bit 0 to 7 */ #define SOF_RT5682_MCLK_EN BIT(0) -#define SOF_RT5682_MCLK_BYTCHT_EN BIT(1) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -206,7 +205,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) } } - if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { + if (ctx->rt5682.is_legacy_cpu) { /* * The firmware might enable the clock at * boot (this information may or may not @@ -279,7 +278,7 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, int pll_id, pll_source, pll_in, pll_out, clk_id, ret; if (ctx->rt5682.mclk_en) { - if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { + if (ctx->rt5682.is_legacy_cpu) { ret = clk_prepare_enable(ctx->rt5682.mclk); if (ret < 0) { dev_err(rtd->dev, @@ -661,7 +660,6 @@ static int sof_audio_probe(struct platform_device *pdev) /* default quirk for legacy cpu */ sof_rt5682_quirk = SOF_RT5682_MCLK_EN | - SOF_RT5682_MCLK_BYTCHT_EN | SOF_SSP_PORT_CODEC(2); } @@ -728,26 +726,27 @@ static int sof_audio_probe(struct platform_device *pdev) } } - if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) + if (sof_rt5682_quirk & SOF_RT5682_MCLK_EN) { ctx->rt5682.mclk_en = true; - /* need to get main clock from pmc */ - if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { - ctx->rt5682.mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); - if (IS_ERR(ctx->rt5682.mclk)) { - ret = PTR_ERR(ctx->rt5682.mclk); + /* need to get main clock from pmc */ + if (ctx->rt5682.is_legacy_cpu) { + ctx->rt5682.mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(ctx->rt5682.mclk)) { + ret = PTR_ERR(ctx->rt5682.mclk); - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %d\n", - ret); - return ret; - } + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret); + return ret; + } - ret = clk_prepare_enable(ctx->rt5682.mclk); - if (ret < 0) { - dev_err(&pdev->dev, - "could not configure MCLK state"); - return ret; + ret = clk_prepare_enable(ctx->rt5682.mclk); + if (ret < 0) { + dev_err(&pdev->dev, + "could not configure MCLK state"); + return ret; + } } } From 76fb0d3221833e87b9150ba06728cdde215ec687 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:25 -0500 Subject: [PATCH 387/603] ASoC: Intel: sof_rt5682: add icl_rt5682_def for icl boards Add the board config icl_rt5682_def to rt5682 machine driver for all icl boards using default SSP port allocation (headphone codec on SSP0). Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-20-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 14 +++++--------- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 2 +- 2 files changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4c6dfe632edfc..7a35db5cab35a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -76,15 +76,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1)), }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_SSP_PORT_CODEC(0)), - }, { .callback = sof_rt5682_quirk_cb, .matches = { @@ -808,6 +799,11 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_PORT_CODEC(2) | SOF_SSP_PORT_AMP(1)), }, + { + .name = "icl_rt5682_def", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_SSP_PORT_CODEC(0)), + }, { .name = "cml_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index d0062f2cd2566..39875d67dcd1d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -29,7 +29,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { }, { .id = "10EC5682", - .drv_name = "sof_rt5682", + .drv_name = "icl_rt5682_def", .sof_tplg_filename = "sof-icl-rt5682.tplg", }, { From 76f33e2f93d63eaac93458fdfde3a505b8e73fa2 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:26 -0500 Subject: [PATCH 388/603] ASoC: Intel: sof-rt5682: add driver_data to sof_rt5682 board Move default BYT/CHT quirk to driver_data of sof_rt5682 board. This fixes a problem that DMI quirk of Minnowboard board got overwritten in probe function since it's a BYT board. Fixes: c68e07970eca ("ASoC: intel: sof_rt5682: Add quirk for number of HDMI DAI's") Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-21-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7a35db5cab35a..668b1672f570a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -638,7 +638,6 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct sof_card_private *ctx; char *card_name; - bool is_legacy_cpu = false; int ret; if (pdev->id_entry && pdev->id_entry->driver_data) @@ -646,14 +645,6 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); - if (soc_intel_is_byt() || soc_intel_is_cht()) { - is_legacy_cpu = true; - - /* default quirk for legacy cpu */ - sof_rt5682_quirk = SOF_RT5682_MCLK_EN | - SOF_SSP_PORT_CODEC(2); - } - dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); /* initialize ctx with board quirk */ @@ -676,7 +667,7 @@ static int sof_audio_probe(struct platform_device *pdev) if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - if (is_legacy_cpu) { + if (soc_intel_is_byt() || soc_intel_is_cht()) { ctx->rt5682.is_legacy_cpu = true; ctx->dmic_be_num = 0; /* HDMI is not supported by SOF on Baytrail/CherryTrail */ @@ -792,6 +783,8 @@ static int sof_audio_probe(struct platform_device *pdev) static const struct platform_device_id board_ids[] = { { .name = "sof_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_SSP_PORT_CODEC(2)), }, { .name = "glk_rt5682_def", From b5aaf6a56dcafc2aeefdc7da1f9f86fa5cfa8df7 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:27 -0500 Subject: [PATCH 389/603] ASoC: Intel: sof-rt5682: setup pll_id only when needed The variable 'pll_id' is needed only when we use snd_soc_dai_set_pll() to setup PLL. Move the code segment to improve some readability. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-22-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 58 ++++++++++++++--------------- 1 file changed, 29 insertions(+), 29 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 668b1672f570a..6a79dd5271ccb 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -320,35 +320,6 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, pll_in = params_rate(params) * 50; } - switch (ctx->codec_type) { - case CODEC_RT5650: - pll_id = 0; /* not used in codec driver */ - clk_id = RT5645_SCLK_S_PLL1; - break; - case CODEC_RT5682: - pll_id = RT5682_PLL1; - clk_id = RT5682_SCLK_S_PLL1; - break; - case CODEC_RT5682S: - /* - * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test - * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz - * input, so we have no choice but to use PLL1. Besides, we will not use PLL at - * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz - */ - if (pll_in == 24576000) { - pll_id = RT5682S_PLL1; - clk_id = RT5682S_SCLK_S_PLL1; - } else { - pll_id = RT5682S_PLL2; - clk_id = RT5682S_SCLK_S_PLL2; - } - break; - default: - dev_err(rtd->dev, "invalid codec type %d\n", ctx->codec_type); - return -EINVAL; - } - pll_out = params_rate(params) * 512; /* when MCLK is 512FS, no need to set PLL configuration additionally. */ @@ -369,6 +340,35 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } else { + switch (ctx->codec_type) { + case CODEC_RT5650: + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; + break; + case CODEC_RT5682: + pll_id = RT5682_PLL1; + clk_id = RT5682_SCLK_S_PLL1; + break; + case CODEC_RT5682S: + /* + * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test + * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz + * input, so we have no choice but to use PLL1. Besides, we will not use PLL at + * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz + */ + if (pll_in == 24576000) { + pll_id = RT5682S_PLL1; + clk_id = RT5682S_SCLK_S_PLL1; + } else { + pll_id = RT5682S_PLL2; + clk_id = RT5682S_SCLK_S_PLL2; + } + break; + default: + dev_err(rtd->dev, "invalid codec type %d\n", ctx->codec_type); + return -EINVAL; + } + /* Configure pll for codec */ ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, pll_in, pll_out); From 4524b1e3ef7884e0a54484dce8d921be7a06af13 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:28 -0500 Subject: [PATCH 390/603] ASoC: Intel: sof-rt5682: get bclk frequency from topology A different bclk frequency 3.072MHz was introduced to tgl platform and is used in mtl topologies. Use SOF API to get frequency from topology instead of hardcoding. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-23-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 6a79dd5271ccb..c3b026868653a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -317,7 +317,12 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - pll_in = params_rate(params) * 50; + /* get the tplg configured bclk. */ + pll_in = sof_dai_get_bclk(rtd); + if (pll_in <= 0) { + dev_err(rtd->dev, "invalid bclk freq %d\n", pll_in); + return -EINVAL; + } } pll_out = params_rate(params) * 512; From 3d84e070253eb853e3190a23994aa3074615efd1 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Fri, 26 Apr 2024 10:25:29 -0500 Subject: [PATCH 391/603] ASoC: Intel: sof-rt5682: support bclk as PLL source on rt5682s For rt5682s codec, we could use bclk as PLL source when the frequency is 3.072MHz but no 2.4MHz. Update the code to select correct pll_id and clk_id for 3.072MHz bclk. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152529.38345-24-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index c3b026868653a..e3a2ec6b4c7cd 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -355,18 +355,23 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, clk_id = RT5682_SCLK_S_PLL1; break; case CODEC_RT5682S: - /* - * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test - * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz - * input, so we have no choice but to use PLL1. Besides, we will not use PLL at - * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz - */ - if (pll_in == 24576000) { + /* check plla_table and pllb_table in rt5682s.c */ + switch (pll_in) { + case 3072000: + case 24576000: + /* + * For MCLK = 24.576MHz and sample rate = 96KHz case, use PLL1 We don't test + * pll_out or params_rate() here since rt5682s PLL2 doesn't support 24.576MHz + * input, so we have no choice but to use PLL1. Besides, we will not use PLL at + * all if pll_in == pll_out. ex, MCLK = 24.576Mhz and sample rate = 48KHz + */ pll_id = RT5682S_PLL1; clk_id = RT5682S_SCLK_S_PLL1; - } else { + break; + default: pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + break; } break; default: From 534e0cd89baf74c22ca55b442fb4add8d844e272 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:34 +0200 Subject: [PATCH 392/603] ASoC: doc: dapm: fix typos Fix various typos. Reviewed-by: Alexandre Belloni Reviewed-by: Bagas Sanjaya Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-1-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index c3154ce6e1b27..59f65b181828d 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -50,7 +50,7 @@ Stream domain All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including -internal codec components). All audio components that effect power are called +internal codec components). All audio components that affect power are called widgets hereafter. @@ -221,7 +221,7 @@ when the Mic is inserted:-:: Codec (BIAS) Domain ------------------- -The codec bias power domain has no widgets and is handled by the codecs DAPM +The codec bias power domain has no widgets and is handled by the codec DAPM event handler. This handler is called when the codec powerstate is changed wrt to any stream event or by kernel PM events. @@ -236,7 +236,7 @@ a virtual widget - a widget with no control bits e.g. SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -This can be used to merge to signal paths together in software. +This can be used to merge two signal paths together in software. After all the widgets have been defined, they can then be added to the DAPM subsystem individually with a call to snd_soc_dapm_new_control(). From eaf84bda623e0a034fe03b871addf7dc4f70dfff Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:35 +0200 Subject: [PATCH 393/603] ASoC: doc: dapm: fix struct name struct snd_soc_codec_dai does not exist. Reviewed-by: Alexandre Belloni Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-2-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 59f65b181828d..68ef79f539f25 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -148,7 +148,7 @@ Stream widgets have the following format:- SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert) NOTE: the stream name must match the corresponding stream name in your codec -snd_soc_codec_dai. +snd_soc_dai_driver. e.g. stream widgets for HiFi playback and capture :: From 610277caa77f07997857ab2e31a073d8ad7ed090 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:36 +0200 Subject: [PATCH 394/603] ASoC: doc: dapm: minor rewording Slightly reword for better readability: replace "PM" -> "power management frameworks", add missing comma. Reviewed-by: Alexandre Belloni Reviewed-by: Bagas Sanjaya Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-3-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 68ef79f539f25..479c010d2dac2 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -7,8 +7,8 @@ Description Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices to use the minimum amount of power within the audio -subsystem at all times. It is independent of other kernel PM and as -such, can easily co-exist with the other PM systems. +subsystem at all times. It is independent of other kernel power +management frameworks and, as such, can easily co-exist with them. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or From 3233a68fcb216a2d8615862841c1fca7a940fc5c Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:37 +0200 Subject: [PATCH 395/603] ASoC: doc: dapm: remove dash after colon There is no need for a dash after colons. Also fix an incorrect ":-::" sequence. Reviewed-by: Alexandre Belloni Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-4-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 479c010d2dac2..be849879274d6 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -20,7 +20,7 @@ DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. -There are 4 power domains within DAPM +There are 4 power domains within DAPM: Codec bias domain VREF, VMID (core codec and audio power) @@ -57,7 +57,7 @@ widgets hereafter. DAPM Widgets ============ -Audio DAPM widgets fall into a number of types:- +Audio DAPM widgets fall into a number of types: Mixer Mixes several analog signals into a single analog signal. @@ -141,7 +141,7 @@ Stream Widgets relate to the stream power domain and only consist of ADCs (analog to digital converters), DACs (digital to analog converters), AIF IN and AIF OUT. -Stream widgets have the following format:- +Stream widgets have the following format: :: SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), @@ -167,7 +167,7 @@ Path Domain Widgets ------------------- Path domain widgets have a ability to control or affect the audio signal or -audio paths within the audio subsystem. They have the following form:- +audio paths within the audio subsystem. They have the following form: :: SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) @@ -207,7 +207,7 @@ powered. e.g. A machine widget can have an optional call back. e.g. Jack connector widget for an external Mic that enables Mic Bias -when the Mic is inserted:-:: +when the Mic is inserted:: static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) { @@ -271,7 +271,7 @@ connect the destination widget (wrt audio signal) with its source widgets. {"Output Mixer", "HiFi Playback Switch", "DAC"}, {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, -So we have :- +So we have : * Destination Widget <=== Path Name <=== Source Widget, or * Sink, Path, Source, or @@ -280,7 +280,7 @@ So we have :- When there is no path name connecting widgets (e.g. a direct connection) we pass NULL for the path name. -Interconnections are created with a call to:- +Interconnections are created with a call to: :: snd_soc_dapm_connect_input(codec, sink, path, source); From 7d0e76107894377a14f211c85a821a53416d5ab2 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:39 +0200 Subject: [PATCH 396/603] ASoC: doc: dapm: replace "map" with "graph" DAPM is actually based on a graph, so use this specific term instead of the more generic "map". Reviewed-by: Alexandre Belloni Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-6-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index be849879274d6..eb50400fa3215 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -48,7 +48,7 @@ Stream domain stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio -routing map of the whole machine. This map is specific to each machine and +routing graph of the whole machine. This graph is specific to each machine and consists of the interconnections between every audio component (including internal codec components). All audio components that affect power are called widgets hereafter. @@ -229,7 +229,7 @@ to any stream event or by kernel PM events. Virtual Widgets --------------- -Sometimes widgets exist in the codec or machine audio map that don't have any +Sometimes widgets exist in the codec or machine audio graph that don't have any corresponding soft power control. In this case it is necessary to create a virtual widget - a widget with no control bits e.g. :: @@ -247,7 +247,7 @@ Codec/DSP Widget Interconnections Widgets are connected to each other within the codec, platform and machine by audio paths (called interconnections). Each interconnection must be defined in -order to create a map of all audio paths between widgets. +order to create a graph of all audio paths between widgets. This is easiest with a diagram of the codec or DSP (and schematic of the machine audio system), as it requires joining widgets together via their audio signal From dd5f36e59134654f9fd7b39d4d66327a1a2db82d Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:40 +0200 Subject: [PATCH 397/603] ASoC: doc: dapm: extend initial descrption Extend the initial description providing clearer definitions of "widget" and "route", and to stop using the word "component" to mean "widget". Give more details and clarify wording and add a picture representing a real DAPM graph. Group all the introductory paragraphs before the "DAPM power domains", and split the latter to a specific section. Reviewed-by: Alexandre Belloni Reviewed-by: Bagas Sanjaya Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-7-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm-graph.svg | 375 +++++++++++++++++++++++++ Documentation/sound/soc/dapm.rst | 30 +- 2 files changed, 396 insertions(+), 9 deletions(-) create mode 100644 Documentation/sound/soc/dapm-graph.svg diff --git a/Documentation/sound/soc/dapm-graph.svg b/Documentation/sound/soc/dapm-graph.svg new file mode 100644 index 0000000000000..d783672db8155 --- /dev/null +++ b/Documentation/sound/soc/dapm-graph.svg @@ -0,0 +1,375 @@ + + + + + + +G + + +ROOT + +ROOT + + +4000b000.audio-controller + +4000b000.audio-controller + + +cs42l51.0-004a + +cs42l51.0-004a + + +hdmi-audio-codec.1.auto + +hdmi-audio-codec.1.auto + + + +ROOT_Amplifier + +Amplifier +[out_drv] + + + +4000b000.audio-controller_capture + +capture +[dai_out] + + + +4000b000.audio-controller_playback + +playback +[dai_in] + + + +hdmi-audio-codec.1.auto_I2S Playback + +I2S Playback +[dai_in] + + + +4000b000.audio-controller_playback->hdmi-audio-codec.1.auto_I2S Playback + + + + + +hdmi-audio-codec.1.auto_Capture + +Capture +[dai_out] + + + +hdmi-audio-codec.1.auto_Capture->4000b000.audio-controller_capture + + + + + +cs42l51.0-004a_AIN1L + +AIN1L +[input] + + + +cs42l51.0-004a_PGA-ADC Mux Left + +PGA-ADC Mux Left +[mux] + + + +cs42l51.0-004a_AIN1L->cs42l51.0-004a_PGA-ADC Mux Left + + + + + +cs42l51.0-004a_AIN1R + +AIN1R +[input] + + + +cs42l51.0-004a_PGA-ADC Mux Right + +PGA-ADC Mux Right +[mux] + + + +cs42l51.0-004a_AIN1R->cs42l51.0-004a_PGA-ADC Mux Right + + + + + +cs42l51.0-004a_AIN2L + +AIN2L +[input] + + + +cs42l51.0-004a_AIN2R + +AIN2R +[input] + + + +cs42l51.0-004a_Capture + +Capture +[dai_out] + + + +cs42l51.0-004a_DAC Mux + +DAC Mux +[mux] + + + +cs42l51.0-004a_Left DAC + +Left DAC +[dac] + + + +cs42l51.0-004a_DAC Mux->cs42l51.0-004a_Left DAC + + + + + +cs42l51.0-004a_Right DAC + +Right DAC +[dac] + + + +cs42l51.0-004a_DAC Mux->cs42l51.0-004a_Right DAC + + + + + +cs42l51.0-004a_HPL + +HPL +[output] + + + +cs42l51.0-004a_HPR + +HPR +[output] + + + +cs42l51.0-004a_Left ADC + +Left ADC +[adc] + + + +cs42l51.0-004a_Left ADC->cs42l51.0-004a_Capture + + + + + +cs42l51.0-004a_Left DAC->cs42l51.0-004a_HPL + + + + + +cs42l51.0-004a_Left PGA + +Left PGA +[pga] + + + +cs42l51.0-004a_Left PGA->cs42l51.0-004a_Left ADC + + + + + +cs42l51.0-004a_MCLK + +MCLK +[supply] + + + +cs42l51.0-004a_MCLK->cs42l51.0-004a_Capture + + + + + +cs42l51.0-004a_Playback + +Playback +[dai_in] + + + +cs42l51.0-004a_MCLK->cs42l51.0-004a_Playback + + + + + +cs42l51.0-004a_MICL + +MICL +[input] + + + +cs42l51.0-004a_Mic Preamp Left + +Mic Preamp Left +[mixer] + + + +cs42l51.0-004a_MICL->cs42l51.0-004a_Mic Preamp Left + + + + + +cs42l51.0-004a_MICR + +MICR +[input] + + + +cs42l51.0-004a_Mic Preamp Right + +Mic Preamp Right +[mixer] + + + +cs42l51.0-004a_MICR->cs42l51.0-004a_Mic Preamp Right + + + + + +cs42l51.0-004a_Mic Bias + +Mic Bias +[supply] + + + +cs42l51.0-004a_Mic Bias->cs42l51.0-004a_MICL + + + + + +cs42l51.0-004a_PGA-ADC Mux Left->cs42l51.0-004a_Left PGA + + + + + +cs42l51.0-004a_Right PGA + +Right PGA +[pga] + + + +cs42l51.0-004a_PGA-ADC Mux Right->cs42l51.0-004a_Right PGA + + + + + +cs42l51.0-004a_Playback->cs42l51.0-004a_DAC Mux + + + + + +cs42l51.0-004a_Right ADC + +Right ADC +[adc] + + + +cs42l51.0-004a_Right ADC->cs42l51.0-004a_Capture + + + + + +cs42l51.0-004a_Right DAC->cs42l51.0-004a_HPR + + + + + +cs42l51.0-004a_Right PGA->cs42l51.0-004a_Right ADC + + + + + +hdmi-audio-codec.1.auto_TX + +TX +[output] + + + +hdmi-audio-codec.1.auto_I2S Playback->hdmi-audio-codec.1.auto_TX + + + + + +hdmi-audio-codec.1.auto_RX + +RX +[output] + + + +hdmi-audio-codec.1.auto_RX->hdmi-audio-codec.1.auto_Capture + + + + + diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index eb50400fa3215..8d741295fc40b 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -16,9 +16,27 @@ recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. -DAPM spans the whole machine. It covers power control within the entire -audio subsystem, this includes internal codec power blocks and machine -level power systems. +DAPM is based on two basic elements, called widgets and routes: + + * a **widget** is every part of the audio hardware that can be enabled by + software when in use and disabled to save power when not in use + * a **route** is an interconnection between widgets that exists when sound + can flow from one widget to the other + +All DAPM power switching decisions are made automatically by consulting an +audio routing graph. This graph is specific to each sound card and spans +the whole sound card, so some DAPM routes connect two widgets belonging to +different components (e.g. the LINE OUT pin of a CODEC and the input pin of +an amplifier). + +The graph for the STM32MP1-DK1 sound card is shown in picture: + +.. kernel-figure:: dapm-graph.svg + :alt: Example DAPM graph + :align: center + +DAPM power domains +================== There are 4 power domains within DAPM: @@ -47,12 +65,6 @@ Stream domain Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. -All DAPM power switching decisions are made automatically by consulting an audio -routing graph of the whole machine. This graph is specific to each machine and -consists of the interconnections between every audio component (including -internal codec components). All audio components that affect power are called -widgets hereafter. - DAPM Widgets ============ From 36de8c0e26793beeda1d6a7592f44b4551271469 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:41 +0200 Subject: [PATCH 398/603] ASoC: doc: dapm: describe how widgets and routes are registered The small paragraph describing how to register widgets is incomplete (does not mention routes) and mentions snd_soc_dapm_new_control() which is not really used. Moreover it appears as a part of the "Virtual Widgets" subsection. Replace it with a detailed and current description of how widgets and routes are registered, mentioning both static declaration and runtime registration. Also make this a section on its own. Reviewed-by: Alexandre Belloni Reviewed-by: Bagas Sanjaya Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-8-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 38 ++++++++++++++++++++++++++++++-- 1 file changed, 36 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 8d741295fc40b..57d96a39f1242 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -250,8 +250,42 @@ a virtual widget - a widget with no control bits e.g. This can be used to merge two signal paths together in software. -After all the widgets have been defined, they can then be added to the DAPM -subsystem individually with a call to snd_soc_dapm_new_control(). +Registering DAPM controls +========================= + +In many cases the DAPM widgets implemented statically in a ``static const +struct snd_soc_dapm_widget`` array and the routes connecting them in a +``static const struct snd_soc_dapm_route`` array in a codec driver, and +simply declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of +the ``struct snd_soc_component_driver`` so the driver registration will +take care of populating them:: + + static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("SPKN"), + SND_SOC_DAPM_OUTPUT("SPKP"), + ... + }; + + /* Target, Path, Source */ + static const struct snd_soc_dapm_route wm2000_audio_map[] = { + { "SPKN", NULL, "ANC Engine" }, + { "SPKP", NULL, "ANC Engine" }, + ... + }; + + static const struct snd_soc_component_driver soc_component_dev_wm2000 = { + ... + .dapm_widgets = wm2000_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets), + ... + }; + +In more complex cases the list of DAPM widgets and/or routes can be only +known at build time. This happens for example when a driver supports +different models having a different set of features. In those cases +separate widgets and routes arrays implementing the case-specific features +can be registered programmatically by calling snd_soc_dapm_new_controls() +and snd_soc_dapm_add_routes(). Codec/DSP Widget Interconnections From d40e82194423c9d058bdeff863373b7da9f7045b Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:42 +0200 Subject: [PATCH 399/603] ASoC: doc: dapm: fix and improve section "Registering DAPM controls" Extend the first paragraph to mention the {,num_}dapm_routes fields just like the widget conterparts. Mention the route fields also in the code example. Fix "at build time", this really means "at probe time". Reviewed-by: Alexandre Belloni Reviewed-by: Bagas Sanjaya Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-9-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 57d96a39f1242..50f908f4e8ede 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -253,12 +253,17 @@ This can be used to merge two signal paths together in software. Registering DAPM controls ========================= -In many cases the DAPM widgets implemented statically in a ``static const -struct snd_soc_dapm_widget`` array and the routes connecting them in a -``static const struct snd_soc_dapm_route`` array in a codec driver, and -simply declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of -the ``struct snd_soc_component_driver`` so the driver registration will -take care of populating them:: +In many cases the DAPM widgets are implemented statically in a ``static +const struct snd_soc_dapm_widget`` array in a codec driver, and simply +declared via the ``dapm_widgets`` and ``num_dapm_widgets`` fields of the +``struct snd_soc_component_driver``. + +Similarly, routes connecting them are implemented statically in a ``static +const struct snd_soc_dapm_route`` array and declared via the +``dapm_routes`` and ``num_dapm_routes`` fields of the same struct. + +With the above declared, the driver registration will take care of +populating them:: static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPKN"), @@ -277,11 +282,13 @@ take care of populating them:: ... .dapm_widgets = wm2000_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm2000_dapm_widgets), + .dapm_routes = wm2000_audio_map, + .num_dapm_routes = ARRAY_SIZE(wm2000_audio_map), ... }; In more complex cases the list of DAPM widgets and/or routes can be only -known at build time. This happens for example when a driver supports +known at probe time. This happens for example when a driver supports different models having a different set of features. In those cases separate widgets and routes arrays implementing the case-specific features can be registered programmatically by calling snd_soc_dapm_new_controls() From 057acfb460d6792c7d3ec088f57c411fb0e4d395 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:43 +0200 Subject: [PATCH 400/603] ASoC: doc: dapm: improve section "Codec/DSP Widget Interconnections" Improve wording in a few places, cleanup ReST colon syntax, remove space before colon, and remove the "codec" parameter in the parentheses of snd_soc_dapm_new_widgets() (there should be no parameters in the docs, and that function takes a card, not a codec). Reviewed-by: Alexandre Belloni Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-10-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 50f908f4e8ede..8fccdb0d6454a 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -306,25 +306,23 @@ This is easiest with a diagram of the codec or DSP (and schematic of the machine audio system), as it requires joining widgets together via their audio signal paths. -e.g., from the WM8731 output mixer (wm8731.c) - -The WM8731 output mixer has 3 inputs (sources) +For example the WM8731 output mixer (wm8731.c) has 3 inputs (sources): 1. Line Bypass Input 2. DAC (HiFi playback) 3. Mic Sidetone Input -Each input in this example has a kcontrol associated with it (defined in example -above) and is connected to the output mixer via its kcontrol name. We can now -connect the destination widget (wrt audio signal) with its source widgets. -:: +Each input in this example has a kcontrol associated with it (defined in +the example above) and is connected to the output mixer via its kcontrol +name. We can now connect the destination widget (wrt audio signal) with its +source widgets. :: /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, -So we have : +So we have: * Destination Widget <=== Path Name <=== Source Widget, or * Sink, Path, Source, or @@ -333,12 +331,11 @@ So we have : When there is no path name connecting widgets (e.g. a direct connection) we pass NULL for the path name. -Interconnections are created with a call to: -:: +Interconnections are created with a call to:: snd_soc_dapm_connect_input(codec, sink, path, source); -Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and +Finally, snd_soc_dapm_new_widgets() must be called after all widgets and interconnections have been registered with the core. This causes the core to scan the codec and machine so that the internal DAPM state matches the physical state of the machine. From ce5a4f101cd0dfbda391d4f1815222d82a9d8a5a Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:44 +0200 Subject: [PATCH 401/603] ASoC: doc: dapm: update section "DAPM Widget Events" The example in this section is not in the kernel sources anymore. Replace it with an up to date code fragment. Reword the initial paragraph. Remove "Please" which is not standard practice in documentation. Reviewed-by: Alexandre Belloni Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-11-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 8fccdb0d6454a..2a1cfa2eca83c 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -376,23 +376,29 @@ jacks can also be switched OFF. DAPM Widget Events ================== -Some widgets can register their interest with the DAPM core in PM events. -e.g. A Speaker with an amplifier registers a widget so the amplifier can be -powered only when the spk is in use. -:: +Widgets needing to implement a more complex behaviour than what DAPM can do +can set a custom "event handler" by setting a function pointer. An example +is a power supply needing to enable a GPIO:: - /* turn speaker amplifier on/off depending on use */ - static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); - return 0; + if (SND_SOC_DAPM_EVENT_ON(event)) + gpiod_set_value_cansleep(gpio_pa, true); + else + gpiod_set_value_cansleep(gpio_pa, false); + + return 0; } - /* corgi machine dapm widgets */ - static const struct snd_soc_dapm_widget wm8731_dapm_widgets = - SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); + static const struct snd_soc_dapm_widget st_widgets[] = { + ... + SND_SOC_DAPM_SUPPLY("Speaker Power", SND_SOC_NOPM, 0, 0, + sof_es8316_speaker_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + }; -Please see soc-dapm.h for all other widgets that support events. +See soc-dapm.h for all other widgets that support events. Event types From 4155a82f6af8068473cd66b55da56fd379835fb0 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 26 Apr 2024 18:56:45 +0200 Subject: [PATCH 402/603] ASoC: doc: dapm: update event types Some new event types now exist, so update the code fragment. Reviewed-by: Alexandre Belloni Signed-off-by: Luca Ceresoli Link: https://lore.kernel.org/r/20240426-dapm-docs-v2-12-87b07547eb5b@bootlin.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 2a1cfa2eca83c..14c4dc026e6bd 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -404,13 +404,16 @@ See soc-dapm.h for all other widgets that support events. Event types ----------- -The following event types are supported by event widgets. -:: +The following event types are supported by event widgets:: /* dapm event types */ - #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ - #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ - #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ - #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ - #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ - #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ + #define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ + #define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ + #define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ + #define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ + #define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ + #define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ + #define SND_SOC_DAPM_WILL_PMU 0x40 /* called at start of sequence */ + #define SND_SOC_DAPM_WILL_PMD 0x80 /* called at start of sequence */ + #define SND_SOC_DAPM_PRE_POST_PMD (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD) + #define SND_SOC_DAPM_PRE_POST_PMU (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU) From 31a70a71b3a730aa703bbd05713d21115dd6d33a Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 26 Apr 2024 11:57:26 +0200 Subject: [PATCH 403/603] ASoC: pcm: Reverse iterate DAIs when shutting them down During startup snd_soc_dai_startup() is launched in ascending order and the exact same thing is done during shutdown procedure. Reverse the order in the latter so that it is symmetric to the former. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240426095733.3946951-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++++ sound/soc/soc-pcm.c | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2a1b6c198547c..8aa6ddb7ee1e8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1235,6 +1235,10 @@ struct snd_soc_pcm_runtime { ((i) < (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs) && \ ((dai) = (rtd)->dais[i]); \ (i)++) +#define for_each_rtd_dais_reverse(rtd, i, dai) \ + for ((i) = (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs - 1; \ + (i) >= 0 && ((dai) = (rtd)->dais[i]); \ + (i)--) #define for_each_rtd_ch_maps(rtd, i, ch_maps) for_each_link_ch_maps(rtd->dai_link, i, ch_maps) void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b0e1bd7f588ba..711b2f49ed88d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -724,7 +724,7 @@ static int soc_pcm_clean(struct snd_soc_pcm_runtime *rtd, } } - for_each_rtd_dais(rtd, i, dai) + for_each_rtd_dais_reverse(rtd, i, dai) snd_soc_dai_shutdown(dai, substream, rollback); snd_soc_link_shutdown(substream, rollback); From 140df6d4d5f541e950a35cad2e3dffb49186ed74 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 26 Apr 2024 11:57:27 +0200 Subject: [PATCH 404/603] ASoC: Intel: avs: Relocate HDA BE DAI specific operations DAI's startup()/shutdown() shall deal with allocation and freeing of resources needed to facilitate streaming over it. Currently for HDAudio BE DAIs some of that task is done in component->open()/close(). Relocate the relevant pieces to address that. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240426095733.3946951-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 55 ++++++++++++++++++--------------------- 1 file changed, 26 insertions(+), 29 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 77a7e8f939515..f3cd54f355ef8 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -277,12 +277,36 @@ static const struct snd_soc_dai_ops avs_dai_hda_be_ops; static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - return avs_dai_startup(substream, dai, false, &avs_dai_hda_be_ops); + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct hdac_ext_stream *link_stream; + struct hda_codec *codec; + int ret; + + ret = avs_dai_startup(substream, dai, false, &avs_dai_hda_be_ops); + if (ret) + return ret; + + codec = dev_to_hda_codec(snd_soc_rtd_to_codec(rtd, 0)->dev); + link_stream = snd_hdac_ext_stream_assign(&codec->bus->core, substream, + HDAC_EXT_STREAM_TYPE_LINK); + if (!link_stream) { + avs_dai_nonhda_be_shutdown(substream, dai); + return -EBUSY; + } + + substream->runtime->private_data = link_stream; + return 0; } static void avs_dai_hda_be_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - return avs_dai_nonhda_be_shutdown(substream, dai); + struct hdac_ext_stream *link_stream; + + link_stream = substream->runtime->private_data; + snd_hdac_ext_stream_release(link_stream, HDAC_EXT_STREAM_TYPE_LINK); + substream->runtime->private_data = NULL; + + avs_dai_nonhda_be_shutdown(substream, dai); } static int avs_dai_hda_be_hw_params(struct snd_pcm_substream *substream, @@ -1576,8 +1600,6 @@ static int avs_component_hda_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct hdac_ext_stream *link_stream; - struct hda_codec *codec; if (!rtd->dai_link->no_pcm) { struct snd_pcm_hardware hwparams = avs_pcm_hardware; @@ -1609,30 +1631,6 @@ static int avs_component_hda_open(struct snd_soc_component *component, return snd_soc_set_runtime_hwparams(substream, &hwparams); } - codec = dev_to_hda_codec(snd_soc_rtd_to_codec(rtd, 0)->dev); - link_stream = snd_hdac_ext_stream_assign(&codec->bus->core, substream, - HDAC_EXT_STREAM_TYPE_LINK); - if (!link_stream) - return -EBUSY; - - substream->runtime->private_data = link_stream; - return 0; -} - -static int avs_component_hda_close(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct hdac_ext_stream *link_stream; - - /* only BE DAI links are handled here */ - if (!rtd->dai_link->no_pcm) - return 0; - - link_stream = substream->runtime->private_data; - snd_hdac_ext_stream_release(link_stream, HDAC_EXT_STREAM_TYPE_LINK); - substream->runtime->private_data = NULL; - return 0; } @@ -1643,7 +1641,6 @@ static const struct snd_soc_component_driver avs_hda_component_driver = { .suspend = avs_component_suspend, .resume = avs_component_resume, .open = avs_component_hda_open, - .close = avs_component_hda_close, .pointer = avs_component_pointer, .mmap = avs_component_mmap, .pcm_construct = avs_component_construct, From b9d59f970ea7772957f6da02ca1ba272ef4495b8 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 26 Apr 2024 11:57:28 +0200 Subject: [PATCH 405/603] ASoC: Intel: avs: Remove redundancy around DAI shutdown Move avs_dai_nonhda_be_shutdown() to avs_dai_shutdown() as the function is common for all transfer types, not just non-HDAudio ones. Use it to simplify avs_dai_fe_shutdown(). While at it, fix explicit kfree(data) and use the destructor instead. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240426095733.3946951-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 44 +++++++++++++++++---------------------- 1 file changed, 19 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index f3cd54f355ef8..23f7e0fae8174 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -85,6 +85,21 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d return 0; } +static void avs_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct avs_dev *adev = to_avs_dev(dai->dev); + struct avs_dma_data *data; + + data = snd_soc_dai_get_dma_data(dai, substream); + + if (rtd->dai_link->ignore_suspend) + adev->num_lp_paths--; + + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(data); +} + static int avs_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *fe_hw_params, struct snd_pcm_hw_params *be_hw_params, struct snd_soc_dai *dai, @@ -166,21 +181,6 @@ static int avs_dai_nonhda_be_startup(struct snd_pcm_substream *substream, struct return avs_dai_startup(substream, dai, false, &avs_dai_nonhda_be_ops); } -static void avs_dai_nonhda_be_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct avs_dev *adev = to_avs_dev(dai->dev); - struct avs_dma_data *data; - - if (rtd->dai_link->ignore_suspend) - adev->num_lp_paths--; - - data = snd_soc_dai_get_dma_data(dai, substream); - - snd_soc_dai_set_dma_data(dai, substream, NULL); - kfree(data); -} - static int avs_dai_nonhda_be_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) { @@ -266,7 +266,7 @@ static int avs_dai_nonhda_be_trigger(struct snd_pcm_substream *substream, int cm static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops = { .startup = avs_dai_nonhda_be_startup, - .shutdown = avs_dai_nonhda_be_shutdown, + .shutdown = avs_dai_shutdown, .hw_params = avs_dai_nonhda_be_hw_params, .hw_free = avs_dai_nonhda_be_hw_free, .prepare = avs_dai_nonhda_be_prepare, @@ -290,7 +290,7 @@ static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct sn link_stream = snd_hdac_ext_stream_assign(&codec->bus->core, substream, HDAC_EXT_STREAM_TYPE_LINK); if (!link_stream) { - avs_dai_nonhda_be_shutdown(substream, dai); + avs_dai_shutdown(substream, dai); return -EBUSY; } @@ -306,7 +306,7 @@ static void avs_dai_hda_be_shutdown(struct snd_pcm_substream *substream, struct snd_hdac_ext_stream_release(link_stream, HDAC_EXT_STREAM_TYPE_LINK); substream->runtime->private_data = NULL; - avs_dai_nonhda_be_shutdown(substream, dai); + avs_dai_shutdown(substream, dai); } static int avs_dai_hda_be_hw_params(struct snd_pcm_substream *substream, @@ -558,18 +558,12 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct avs_dev *adev = to_avs_dev(dai->dev); struct avs_dma_data *data; - if (rtd->dai_link->ignore_suspend) - adev->num_lp_paths--; - data = snd_soc_dai_get_dma_data(dai, substream); - snd_soc_dai_set_dma_data(dai, substream, NULL); snd_hdac_ext_stream_release(data->host_stream, HDAC_EXT_STREAM_TYPE_HOST); - kfree(data); + avs_dai_shutdown(substream, dai); } static int avs_dai_fe_hw_params(struct snd_pcm_substream *substream, From c303a994e5d0f7d297cb6ac56052dce8f412ee67 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 26 Apr 2024 11:57:29 +0200 Subject: [PATCH 406/603] ASoC: Intel: avs: Store pointer to adev in DAI dma_data Reduce the number of to_avs_dev() casts by storing the driver context in DAI's dma_data. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240426095733.3946951-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 23f7e0fae8174..a3a04115216c8 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -22,6 +22,7 @@ struct avs_dma_data { struct avs_tplg_path_template *template; struct avs_path *path; + struct avs_dev *adev; /* * link stream is stored within substream's runtime * private_data to fulfill the needs of codec BE path @@ -60,7 +61,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d const struct snd_soc_dai_ops *ops) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct avs_dev *adev = to_avs_dev(dai->dev); + struct avs_dev *adev = to_avs_dev(dai->component->dev); struct avs_tplg_path_template *template; struct avs_dma_data *data; @@ -77,6 +78,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d data->substream = substream; data->template = template; + data->adev = adev; snd_soc_dai_set_dma_data(dai, substream, data); if (rtd->dai_link->ignore_suspend) @@ -88,13 +90,12 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d static void avs_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct avs_dev *adev = to_avs_dev(dai->dev); struct avs_dma_data *data; data = snd_soc_dai_get_dma_data(dai, substream); if (rtd->dai_link->ignore_suspend) - adev->num_lp_paths--; + data->adev->num_lp_paths--; snd_soc_dai_set_dma_data(dai, substream, NULL); kfree(data); @@ -107,7 +108,6 @@ static int avs_dai_hw_params(struct snd_pcm_substream *substream, { struct avs_dma_data *data; struct avs_path *path; - struct avs_dev *adev = to_avs_dev(dai->dev); int ret; data = snd_soc_dai_get_dma_data(dai, substream); @@ -124,7 +124,7 @@ static int avs_dai_hw_params(struct snd_pcm_substream *substream, params_rate(be_hw_params), params_channels(be_hw_params), params_width(be_hw_params), params_physical_width(be_hw_params)); - path = avs_path_create(adev, dma_id, data->template, fe_hw_params, be_hw_params); + path = avs_path_create(data->adev, dma_id, data->template, fe_hw_params, be_hw_params); if (IS_ERR(path)) { ret = PTR_ERR(path); dev_err(dai->dev, "create path failed: %d\n", ret); @@ -505,8 +505,7 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so { struct snd_pcm_runtime *runtime = substream->runtime; struct avs_dma_data *data; - struct avs_dev *adev = to_avs_dev(dai->dev); - struct hdac_bus *bus = &adev->base.core; + struct hdac_bus *bus; struct hdac_ext_stream *host_stream; int ret; @@ -515,6 +514,7 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so return ret; data = snd_soc_dai_get_dma_data(dai, substream); + bus = &data->adev->base.core; host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST); if (!host_stream) { @@ -655,7 +655,6 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_stream *stream_info; struct avs_dma_data *data; - struct avs_dev *adev = to_avs_dev(dai->dev); struct hdac_ext_stream *host_stream; unsigned int format_val; struct hdac_bus *bus; @@ -685,7 +684,7 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so if (ret < 0) return ret; - ret = avs_dai_prepare(adev, substream, dai); + ret = avs_dai_prepare(data->adev, substream, dai); if (ret) return ret; From 3a48d146aa761bc591272bc453eda64743128a31 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 26 Apr 2024 11:57:30 +0200 Subject: [PATCH 407/603] ASoC: Intel: avs: Remove redundancy around DAI startup Half of the arguments in avs_dai_startup() are unused and can be dropped. With the function updated, it matches its template in snd_soc_dai_ops and can be referenced throughout the pcm.c file without need of any wrappers. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240426095733.3946951-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 22 +++++----------------- 1 file changed, 5 insertions(+), 17 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index a3a04115216c8..0771a9716f4bc 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -57,15 +57,14 @@ avs_dai_find_path_template(struct snd_soc_dai *dai, bool is_fe, int direction) return dw->priv; } -static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai, bool is_fe, - const struct snd_soc_dai_ops *ops) +static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct avs_dev *adev = to_avs_dev(dai->component->dev); struct avs_tplg_path_template *template; struct avs_dma_data *data; - template = avs_dai_find_path_template(dai, is_fe, substream->stream); + template = avs_dai_find_path_template(dai, !rtd->dai_link->no_pcm, substream->stream); if (!template) { dev_err(dai->dev, "no %s path for dai %s, invalid tplg?\n", snd_pcm_stream_str(substream), dai->name); @@ -174,13 +173,6 @@ static int avs_dai_prepare(struct avs_dev *adev, struct snd_pcm_substream *subst return ret; } -static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops; - -static int avs_dai_nonhda_be_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) -{ - return avs_dai_startup(substream, dai, false, &avs_dai_nonhda_be_ops); -} - static int avs_dai_nonhda_be_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) { @@ -265,7 +257,7 @@ static int avs_dai_nonhda_be_trigger(struct snd_pcm_substream *substream, int cm } static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops = { - .startup = avs_dai_nonhda_be_startup, + .startup = avs_dai_startup, .shutdown = avs_dai_shutdown, .hw_params = avs_dai_nonhda_be_hw_params, .hw_free = avs_dai_nonhda_be_hw_free, @@ -273,8 +265,6 @@ static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops = { .trigger = avs_dai_nonhda_be_trigger, }; -static const struct snd_soc_dai_ops avs_dai_hda_be_ops; - static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); @@ -282,7 +272,7 @@ static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct sn struct hda_codec *codec; int ret; - ret = avs_dai_startup(substream, dai, false, &avs_dai_hda_be_ops); + ret = avs_dai_startup(substream, dai); if (ret) return ret; @@ -479,8 +469,6 @@ static const struct snd_pcm_hw_constraint_list hw_rates = { .mask = 0, }; -const struct snd_soc_dai_ops avs_dai_fe_ops; - static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval *interval = hw_param_interval(params, rule->var); @@ -509,7 +497,7 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so struct hdac_ext_stream *host_stream; int ret; - ret = avs_dai_startup(substream, dai, true, &avs_dai_fe_ops); + ret = avs_dai_startup(substream, dai); if (ret) return ret; From 0f8843ca4f6cbf0efb8c2d5516a3b92fb2771a04 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 26 Apr 2024 11:57:31 +0200 Subject: [PATCH 408/603] ASoC: Intel: avs: Remove redundancy around DAI prepare Drop unused arguments in the avs_dai_prepare() function. With the function updated, it matches its template in snd_soc_dai_ops and can be referenced throughout the pcm.c file without need of any wrappers. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240426095733.3946951-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 0771a9716f4bc..d4557b7b1c6c2 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -151,8 +151,7 @@ static int avs_dai_be_hw_params(struct snd_pcm_substream *substream, return avs_dai_hw_params(substream, fe_hw_params, be_hw_params, dai, dma_id); } -static int avs_dai_prepare(struct avs_dev *adev, struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int avs_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct avs_dma_data *data; int ret; @@ -201,11 +200,6 @@ static int avs_dai_nonhda_be_hw_free(struct snd_pcm_substream *substream, struct return 0; } -static int avs_dai_nonhda_be_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) -{ - return avs_dai_prepare(to_avs_dev(dai->dev), substream, dai); -} - static int avs_dai_nonhda_be_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -261,7 +255,7 @@ static const struct snd_soc_dai_ops avs_dai_nonhda_be_ops = { .shutdown = avs_dai_shutdown, .hw_params = avs_dai_nonhda_be_hw_params, .hw_free = avs_dai_nonhda_be_hw_free, - .prepare = avs_dai_nonhda_be_prepare, + .prepare = avs_dai_prepare, .trigger = avs_dai_nonhda_be_trigger, }; @@ -381,7 +375,7 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_hdac_ext_bus_link_set_stream_id(link, hdac_stream(link_stream)->stream_tag); - ret = avs_dai_prepare(to_avs_dev(dai->dev), substream, dai); + ret = avs_dai_prepare(substream, dai); if (ret) return ret; @@ -672,7 +666,7 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so if (ret < 0) return ret; - ret = avs_dai_prepare(data->adev, substream, dai); + ret = avs_dai_prepare(substream, dai); if (ret) return ret; From cdcb770a60e8e6b9fbb737ebe21b2daadaba1744 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 26 Apr 2024 11:57:32 +0200 Subject: [PATCH 409/603] ASoC: Intel: avs: Store pointer to link_stream in dma_data While the HDAudio codec driver expectations must be met - store valid pointer to HDAudio LINK stream in substream->runtime->private_data - the code is more readable and easier to maintain if dma_data stores pointers to both HOST and LINK stream. DAI BE operations can refer to the LINK stream with data->link_stream, similarly to how DAI FE operations access the HOST stream with data->host_stream. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240426095733.3946951-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 40 +++++++++++++++++++-------------------- 1 file changed, 19 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index d4557b7b1c6c2..168e16e82116b 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -23,13 +23,12 @@ struct avs_dma_data { struct avs_tplg_path_template *template; struct avs_path *path; struct avs_dev *adev; - /* - * link stream is stored within substream's runtime - * private_data to fulfill the needs of codec BE path - * - * host stream assigned - */ - struct hdac_ext_stream *host_stream; + + /* LINK-stream utilized in BE operations while HOST in FE ones. */ + union { + struct hdac_ext_stream *link_stream; + struct hdac_ext_stream *host_stream; + }; struct snd_pcm_substream *substream; }; @@ -263,6 +262,7 @@ static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct sn { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct hdac_ext_stream *link_stream; + struct avs_dma_data *data; struct hda_codec *codec; int ret; @@ -278,18 +278,18 @@ static int avs_dai_hda_be_startup(struct snd_pcm_substream *substream, struct sn return -EBUSY; } + data = snd_soc_dai_get_dma_data(dai, substream); + data->link_stream = link_stream; substream->runtime->private_data = link_stream; return 0; } static void avs_dai_hda_be_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_stream *link_stream; + struct avs_dma_data *data = snd_soc_dai_get_dma_data(dai, substream); - link_stream = substream->runtime->private_data; - snd_hdac_ext_stream_release(link_stream, HDAC_EXT_STREAM_TYPE_LINK); + snd_hdac_ext_stream_release(data->link_stream, HDAC_EXT_STREAM_TYPE_LINK); substream->runtime->private_data = NULL; - avs_dai_shutdown(substream, dai); } @@ -297,16 +297,13 @@ static int avs_dai_hda_be_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) { struct avs_dma_data *data; - struct hdac_ext_stream *link_stream; data = snd_soc_dai_get_dma_data(dai, substream); if (data->path) return 0; - link_stream = substream->runtime->private_data; - return avs_dai_be_hw_params(substream, hw_params, dai, - hdac_stream(link_stream)->stream_tag - 1); + hdac_stream(data->link_stream)->stream_tag - 1); } static int avs_dai_hda_be_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -323,7 +320,7 @@ static int avs_dai_hda_be_hw_free(struct snd_pcm_substream *substream, struct sn if (!data->path) return 0; - link_stream = substream->runtime->private_data; + link_stream = data->link_stream; link_stream->link_prepared = false; avs_path_free(data->path); data->path = NULL; @@ -347,13 +344,16 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn struct snd_soc_pcm_stream *stream_info; struct hdac_ext_stream *link_stream; struct hdac_ext_link *link; + struct avs_dma_data *data; struct hda_codec *codec; struct hdac_bus *bus; unsigned int format_val; unsigned int bits; int ret; - link_stream = runtime->private_data; + data = snd_soc_dai_get_dma_data(dai, substream); + link_stream = data->link_stream; + if (link_stream->link_prepared) return 0; @@ -387,14 +387,12 @@ static int avs_dai_hda_be_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct hdac_ext_stream *link_stream; struct avs_dma_data *data; int ret = 0; dev_dbg(dai->dev, "entry %s cmd=%d\n", __func__, cmd); data = snd_soc_dai_get_dma_data(dai, substream); - link_stream = substream->runtime->private_data; switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: @@ -403,7 +401,7 @@ static int avs_dai_hda_be_trigger(struct snd_pcm_substream *substream, int cmd, fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - snd_hdac_ext_stream_start(link_stream); + snd_hdac_ext_stream_start(data->link_stream); ret = avs_path_pause(data->path); if (ret < 0) { @@ -426,7 +424,7 @@ static int avs_dai_hda_be_trigger(struct snd_pcm_substream *substream, int cmd, if (ret < 0) dev_err(dai->dev, "pause BE path failed: %d\n", ret); - snd_hdac_ext_stream_clear(link_stream); + snd_hdac_ext_stream_clear(data->link_stream); ret = avs_path_reset(data->path); if (ret < 0) From e85e75b67993c1fb0c80306783c31266261170d4 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 26 Apr 2024 11:57:33 +0200 Subject: [PATCH 410/603] ASoC: Intel: avs: Clean up hw constraints initialization Provide a separate function that initializes all PCM hardware constraints for the driver. No functional changes. Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240426095733.3946951-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/pcm.c | 84 ++++++++++++++++++++------------------- 1 file changed, 44 insertions(+), 40 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 168e16e82116b..845b5ed9eb1ba 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -448,19 +448,6 @@ static const struct snd_soc_dai_ops avs_dai_hda_be_ops = { .trigger = avs_dai_hda_be_trigger, }; -static const unsigned int rates[] = { - 8000, 11025, 12000, 16000, - 22050, 24000, 32000, 44100, - 48000, 64000, 88200, 96000, - 128000, 176400, 192000, -}; - -static const struct snd_pcm_hw_constraint_list hw_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval *interval = hw_param_interval(params, rule->var); @@ -481,40 +468,33 @@ static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_h return snd_interval_refine(interval, &to); } -static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +static int avs_pcm_hw_constraints_init(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct avs_dma_data *data; - struct hdac_bus *bus; - struct hdac_ext_stream *host_stream; + static const unsigned int rates[] = { + 8000, 11025, 12000, 16000, + 22050, 24000, 32000, 44100, + 48000, 64000, 88200, 96000, + 128000, 176400, 192000, + }; + static const struct snd_pcm_hw_constraint_list rate_list = { + .count = ARRAY_SIZE(rates), + .list = rates, + }; int ret; - ret = avs_dai_startup(substream, dai); - if (ret) - return ret; - - data = snd_soc_dai_get_dma_data(dai, substream); - bus = &data->adev->base.core; - - host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST); - if (!host_stream) { - ret = -EBUSY; - goto err; - } - - data->host_stream = host_stream; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) - goto err; + return ret; - /* avoid wrap-around with wall-clock */ + /* Avoid wrap-around with wall-clock. */ ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, 20, 178000000); if (ret < 0) - goto err; + return ret; - ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &rate_list); if (ret < 0) - goto err; + return ret; /* Adjust buffer and period size based on the audio format. */ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, hw_rule_param_size, NULL, @@ -524,16 +504,40 @@ static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_so SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, SNDRV_PCM_HW_PARAM_RATE, -1); + return ret; +} + +static int avs_dai_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct hdac_ext_stream *host_stream; + struct avs_dma_data *data; + struct hdac_bus *bus; + int ret; + + ret = avs_pcm_hw_constraints_init(substream); + if (ret) + return ret; + + ret = avs_dai_startup(substream, dai); + if (ret) + return ret; + + data = snd_soc_dai_get_dma_data(dai, substream); + bus = &data->adev->base.core; + + host_stream = snd_hdac_ext_stream_assign(bus, substream, HDAC_EXT_STREAM_TYPE_HOST); + if (!host_stream) { + avs_dai_shutdown(substream, dai); + return -EBUSY; + } + + data->host_stream = host_stream; snd_pcm_set_sync(substream); dev_dbg(dai->dev, "%s fe STARTUP tag %d str %p", __func__, hdac_stream(host_stream)->stream_tag, substream); return 0; - -err: - kfree(data); - return ret; } static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) From 33e59e50ee7610473c85030edca73ad3df60b5c1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 26 Apr 2024 10:21:12 -0500 Subject: [PATCH 411/603] ASoC: Intel: skl_hda_dsp_generic: Allocate snd_soc_card dynamically The static hda_soc_card might be modified during runtime which might cause issues on next time when the card is created. For example if the dmic_num was set with module parameter then removed for the next module loading then the card's components will still going to point to the previous boot's cfg-dmics:X string. There might be other places where devm allocated memory have been freed but the hda_soc_card still pointing to the now unallocated memory (the memory is freed when the platform device is removed). Fix this issue by moving the snd_soc_card into skl_hda_private and use it for the card registration to ensure that it is correctly initialized every time. Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_common.h | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 42 ++++++++++---------- 2 files changed, 22 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index 4b0b3959182e5..19b814dee4ad6 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -28,6 +28,7 @@ struct skl_hda_hdmi_pcm { }; struct skl_hda_private { + struct snd_soc_card card; struct list_head hdmi_pcm_list; int pcm_count; int dai_index; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 4aa7fd2a05e46..208395872d8b2 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -92,17 +92,6 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) return ret; } -static struct snd_soc_card hda_soc_card = { - .name = "hda-dsp", - .owner = THIS_MODULE, - .dai_link = skl_hda_be_dai_links, - .dapm_widgets = skl_hda_widgets, - .dapm_routes = skl_hda_map, - .add_dai_link = skl_hda_add_dai_link, - .fully_routed = true, - .late_probe = skl_hda_card_late_probe, -}; - static char hda_soc_components[30]; #define IDISP_DAI_COUNT 3 @@ -115,9 +104,9 @@ static char hda_soc_components[30]; #define HDA_CODEC_AUTOSUSPEND_DELAY_MS 1000 -static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params) +static int skl_hda_fill_card_info(struct snd_soc_card *card, + struct snd_soc_acpi_mach_params *mach_params) { - struct snd_soc_card *card = &hda_soc_card; struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *dai_link; u32 codec_count, codec_mask; @@ -199,6 +188,7 @@ static int skl_hda_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach; struct skl_hda_private *ctx; + struct snd_soc_card *card; int ret; dev_dbg(&pdev->dev, "entry\n"); @@ -213,32 +203,42 @@ static int skl_hda_audio_probe(struct platform_device *pdev) if (!mach) return -EINVAL; - snd_soc_card_set_drvdata(&hda_soc_card, ctx); + card = &ctx->card; + card->name = "hda-dsp", + card->owner = THIS_MODULE, + card->dai_link = skl_hda_be_dai_links, + card->dapm_widgets = skl_hda_widgets, + card->dapm_routes = skl_hda_map, + card->add_dai_link = skl_hda_add_dai_link, + card->fully_routed = true, + card->late_probe = skl_hda_card_late_probe, + + snd_soc_card_set_drvdata(card, ctx); - ret = skl_hda_fill_card_info(&mach->mach_params); + ret = skl_hda_fill_card_info(card, &mach->mach_params); if (ret < 0) { dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); return ret; } - ctx->pcm_count = hda_soc_card.num_links; + ctx->pcm_count = card->num_links; ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */ ctx->platform_name = mach->mach_params.platform; ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; - hda_soc_card.dev = &pdev->dev; + card->dev = &pdev->dev; if (!snd_soc_acpi_sof_parent(&pdev->dev)) - hda_soc_card.disable_route_checks = true; + card->disable_route_checks = true; if (mach->mach_params.dmic_num > 0) { snprintf(hda_soc_components, sizeof(hda_soc_components), "cfg-dmics:%d", mach->mach_params.dmic_num); - hda_soc_card.components = hda_soc_components; + card->components = hda_soc_components; } - ret = devm_snd_soc_register_card(&pdev->dev, &hda_soc_card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (!ret) - skl_set_hda_codec_autosuspend_delay(&hda_soc_card); + skl_set_hda_codec_autosuspend_delay(card); return ret; } From 69d0f88b9aebb5749ab0dbaead7414d718994380 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 26 Apr 2024 10:21:13 -0500 Subject: [PATCH 412/603] ASoC: Intel: skl_hda_dsp_generic: Use devm_kasprintf for the components string Instead of using a global char array, allocate the string with devm_kasprintf if needed. Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 208395872d8b2..88d91c0280bbb 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -92,8 +92,6 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) return ret; } -static char hda_soc_components[30]; - #define IDISP_DAI_COUNT 3 #define HDAC_DAI_COUNT 2 #define DMIC_DAI_COUNT 2 @@ -231,9 +229,11 @@ static int skl_hda_audio_probe(struct platform_device *pdev) card->disable_route_checks = true; if (mach->mach_params.dmic_num > 0) { - snprintf(hda_soc_components, sizeof(hda_soc_components), - "cfg-dmics:%d", mach->mach_params.dmic_num); - card->components = hda_soc_components; + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "cfg-dmics:%d", + mach->mach_params.dmic_num); + if (!card->components) + return -ENOMEM; } ret = devm_snd_soc_register_card(&pdev->dev, card); From 02e6f7cb487f18e1171ae6d12ad1066fbd25176d Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:21:14 -0500 Subject: [PATCH 413/603] ASoC: Intel: soc-acpi: mtl: add Dell SKU 0C64 and 0CC6 SKU 0C64 relies on rt713 (jack codec) on link0, rt1318 (single amplifier) on link1 and rt1713 (dmic) on link3. SKU 0CC6 relies on rt713 (jack codec) on link0, rt1318 (two amplifiers) on link 1-2 and rt1713 (dmic) on link3. Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 64 +++++++++++++++++++ 1 file changed, 64 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index f95490a16b551..8c1c430f54829 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -312,6 +312,15 @@ static const struct snd_soc_acpi_adr_device rt1316_3_single_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1318_1_single_adr[] = { + { + .adr = 0x000130025D131801, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt1318" + } +}; + static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = { { .adr = 0x000130025D131801ull, @@ -559,6 +568,49 @@ static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1316_l12_rt1713_l3[] = {} }; +static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1318_l1_rt1713_l3[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt713_0_single_adr), + .adr_d = rt713_0_single_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1318_1_single_adr), + .adr_d = rt1318_1_single_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt1713_3_single_adr), + .adr_d = rt1713_3_single_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1318_l12_rt1713_l3[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt713_0_single_adr), + .adr_d = rt713_0_single_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1318_1_group1_adr), + .adr_d = rt1318_1_group1_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1318_2_group1_adr), + .adr_d = rt1318_2_group1_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt1713_3_single_adr), + .adr_d = rt1713_3_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1316_l12[] = { { .mask = BIT(0), @@ -697,6 +749,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt713-l0-rt1316-l12-rt1713-l3.tplg", }, + { + .link_mask = GENMASK(3, 0), + .links = mtl_rt713_l0_rt1318_l12_rt1713_l3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt713-l0-rt1318-l12-rt1713-l3.tplg", + }, + { + .link_mask = BIT(0) | BIT(1) | BIT(3), + .links = mtl_rt713_l0_rt1318_l1_rt1713_l3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt713-l0-rt1318-l1-rt1713-l3.tplg", + }, { .link_mask = GENMASK(2, 0), .links = mtl_rt713_l0_rt1316_l12, From 64bfd26d982ec29123c65949229fa12c15f7df8f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:21:15 -0500 Subject: [PATCH 414/603] ASoC: Intel: soc-acpi: mtl: add support for Acer Swift Go 14 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This device has an RT712 on link0, but does not rely on RT1712 for the DMIC. PCH-attached DMICs are used instead. Closes: https://github.com/thesofproject/linux/issues/4923 Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 8c1c430f54829..4eeec0bc92dc4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -357,7 +357,7 @@ static const struct snd_soc_acpi_adr_device rt714_1_adr[] = { } }; -static const struct snd_soc_acpi_link_adr mtl_712_only[] = { +static const struct snd_soc_acpi_link_adr mtl_712_l0_1712_l3[] = { { .mask = BIT(0), .num_adr = ARRAY_SIZE(rt712_0_single_adr), @@ -371,6 +371,15 @@ static const struct snd_soc_acpi_link_adr mtl_712_only[] = { {} }; +static const struct snd_soc_acpi_link_adr mtl_712_l0[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt712_0_single_adr), + .adr_d = rt712_0_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { { /* Jack Playback Endpoint */ .num = 0, @@ -769,10 +778,16 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { }, { .link_mask = BIT(3) | BIT(0), - .links = mtl_712_only, + .links = mtl_712_l0_1712_l3, .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt712-l0-rt1712-l3.tplg", }, + { + .link_mask = BIT(0), + .links = mtl_712_l0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt712-l0.tplg", + }, { .link_mask = GENMASK(2, 0), .links = mtl_sdw_rt1318_l12_rt714_l0, From 6d339113df3ab510ce075a18ccb10a20cb325d4e Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Fri, 26 Apr 2024 10:21:16 -0500 Subject: [PATCH 415/603] ASoC: Intel: soc-acpi-intel-lnl-match: adds RT714 and RT1318 support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds support for corresponding codecs on LNL hardware configuration: SDW0: RT714 DMIC SDW1: RT1318 Left Speaker SDW2: RT1318 Right Speaker Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Mac Chiang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-lnl-match.c | 52 +++++++++++++++++++ 1 file changed, 52 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 74d6dcd7471f0..0318c1a46f3c6 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -130,6 +130,33 @@ static const struct snd_soc_acpi_adr_device rt1316_3_group1_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1318_1_group1_adr[] = { + { + .adr = 0x000130025D131801ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "rt1318-1" + } +}; + +static const struct snd_soc_acpi_adr_device rt1318_2_group1_adr[] = { + { + .adr = 0x000232025D131801ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "rt1318-2" + } +}; + +static const struct snd_soc_acpi_adr_device rt714_0_adr[] = { + { + .adr = 0x000030025D071401ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt714" + } +}; + static const struct snd_soc_acpi_adr_device rt714_1_adr[] = { { .adr = 0x000130025D071401ull, @@ -195,6 +222,25 @@ static const struct snd_soc_acpi_link_adr lnl_3_in_1_sdca[] = { {} }; +static const struct snd_soc_acpi_link_adr lnl_sdw_rt1318_l12_rt714_l0[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1318_1_group1_adr), + .adr_d = rt1318_1_group1_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1318_2_group1_adr), + .adr_d = rt1318_2_group1_adr, + }, + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt714_0_adr), + .adr_d = rt714_0_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { /* mockup tests need to be first */ @@ -240,6 +286,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt722-l0.tplg", }, + { + .link_mask = GENMASK(2, 0), + .links = lnl_sdw_rt1318_l12_rt714_l0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-rt1318-l12-rt714-l0.tplg" + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_lnl_sdw_machines); From 38068d91cf3948ffa220d45f738505cc9f6e13d0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 26 Apr 2024 10:21:17 -0500 Subject: [PATCH 416/603] ASoC: Intel: sof_sdw: Allocate snd_soc_card dynamically The static card_sof_sdw struct is modified during runtime and in case the module is not removed, but the card is, then the next time the card is created the card_sof_sdw will contain information from the previous card which might lead to hard to debug issues, side effects. Move the snd_soc_card into mc_private and use that to make sure that the card is initialized correctly. Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Kai Vehmanen Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 20 +++++++++----------- sound/soc/intel/boards/sof_sdw_common.h | 1 + 2 files changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d65c5da49000a..384c3d41a9adc 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1882,12 +1882,6 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card) /* SoC card */ static const char sdw_card_long_name[] = "Intel Soundwire SOF"; -static struct snd_soc_card card_sof_sdw = { - .name = "soundwire", - .owner = THIS_MODULE, - .late_probe = sof_sdw_card_late_probe, -}; - /* helper to get the link that the codec DAI is used */ static struct snd_soc_dai_link *mc_find_codec_dai_used(struct snd_soc_card *card, const char *dai_name) @@ -1939,20 +1933,24 @@ static void mc_dailink_exit_loop(struct snd_soc_card *card) static int mc_probe(struct platform_device *pdev) { - struct snd_soc_card *card = &card_sof_sdw; struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev); + struct snd_soc_card *card; struct mc_private *ctx; int amp_num = 0, i; int ret; - card->dev = &pdev->dev; - - dev_dbg(card->dev, "Entry\n"); + dev_dbg(&pdev->dev, "Entry\n"); - ctx = devm_kzalloc(card->dev, sizeof(*ctx), GFP_KERNEL); + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; + card = &ctx->card; + card->dev = &pdev->dev; + card->name = "soundwire", + card->owner = THIS_MODULE, + card->late_probe = sof_sdw_card_late_probe, + snd_soc_card_set_drvdata(card, ctx); dmi_check_system(sof_sdw_quirk_table); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 89253938ebaaa..853278c6e5250 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -101,6 +101,7 @@ struct sof_sdw_codec_info { }; struct mc_private { + struct snd_soc_card card; struct snd_soc_jack sdw_headset; struct sof_hdmi_private hdmi; struct device *headset_codec_dev; /* only one headset per card */ From 2086b55fd6ddcaa92e473ba7017f6a986870337e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:21:18 -0500 Subject: [PATCH 417/603] ASoC: Intel: sof-sdw: don't set card long_name MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit UCM can load a board-specific file based on the card long_name. Remove the constant "Intel Soundwire SOF" long_name so that the ASoC core can set the long_name based on DMI information. Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 384c3d41a9adc..b9a5fcb42847e 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1879,9 +1879,6 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card) return ret; } -/* SoC card */ -static const char sdw_card_long_name[] = "Intel Soundwire SOF"; - /* helper to get the link that the codec DAI is used */ static struct snd_soc_dai_link *mc_find_codec_dai_used(struct snd_soc_card *card, const char *dai_name) @@ -1999,8 +1996,6 @@ static int mc_probe(struct platform_device *pdev) return -ENOMEM; } - card->long_name = sdw_card_long_name; - /* Register the card */ ret = devm_snd_soc_register_card(card->dev, card); if (ret) { From 6be269d274353d2604bf49b92f703610cb4734e9 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 26 Apr 2024 10:21:19 -0500 Subject: [PATCH 418/603] ASoC: Intel: sof_sdw: add a space before cfg-amp in components UCM parse amp with Regex " cfg-amp:([0-9]+)". The "ASoC: Intel: sof_sdw: remove FOUR_SPEAKER quirks" patch removed "cfg-spk:%d " from components which removed the necessary space as well and cause UCM can't parse the amp number properly. Fixes: 744866d28fe6 ("ASoC: Intel: sof_sdw: remove FOUR_SPEAKER quirks") Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b9a5fcb42847e..f0622aa1b7486 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1983,7 +1983,7 @@ static int mc_probe(struct platform_device *pdev) amp_num += codec_info_list[i].amp_num; card->components = devm_kasprintf(card->dev, GFP_KERNEL, - "cfg-amp:%d", amp_num); + " cfg-amp:%d", amp_num); if (!card->components) return -ENOMEM; From 0bab4cfd7c1560095e29919e2ebe01783b9096dc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:21:20 -0500 Subject: [PATCH 419/603] ASoC: Intel: sof-sdw: really remove FOUR_SPEAKER quirk Two independent GitHub PRs let to the addition of one quirk after it was removed.. Fixes: b10cb955c6c0 ("ASoC: Intel: sof_sdw: add quirk for Dell SKU 0C0F") Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f0622aa1b7486..86bbf1c66ce8e 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -421,8 +421,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0C0F") }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - RT711_JD2 | - SOF_SDW_FOUR_SPK), + RT711_JD2), }, { .callback = sof_sdw_quirk_cb, From 628cc5d0c4bd6a3f70c793968f8e2546afc8c3a3 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 26 Apr 2024 10:21:21 -0500 Subject: [PATCH 420/603] ASoC: Intel: sof_sdw: Delay update of the codec_conf array Move the population of the codec_conf array from endpoint parsing time to link creation time. This is slightly cleaner as the population is done whilst the DAI links are also being populated, putting all population together. However, primarily this facilitates allowing additional non-SoundWire devices to be easily added into the array in future feature additions. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 48 ++++++++++++++++++-------------- 1 file changed, 27 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 86bbf1c66ce8e..bf5f46a4c4aa9 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1279,6 +1279,7 @@ struct sof_sdw_endpoint { u32 link_mask; const char *codec_name; + const char *name_prefix; struct sof_sdw_codec_info *codec_info; const struct sof_sdw_dai_info *dai_info; @@ -1340,7 +1341,6 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; - struct snd_soc_codec_conf *codec_conf = card->codec_conf; const struct snd_soc_acpi_link_adr *adr_link; struct sof_sdw_endpoint *sof_end = sof_ends; int num_dais = 0; @@ -1376,13 +1376,11 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, if (!codec_name) return -ENOMEM; - codec_conf->dlc.name = codec_name; - codec_conf->name_prefix = adr_dev->name_prefix; - codec_conf++; - dev_dbg(dev, "Adding prefix %s for %s\n", adr_dev->name_prefix, codec_name); + sof_end->name_prefix = adr_dev->name_prefix; + for (j = 0; j < adr_dev->num_endpoints; j++) { const struct snd_soc_acpi_endpoint *adr_end; const struct sof_sdw_dai_info *dai_info; @@ -1443,21 +1441,27 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, ctx->append_dai_type |= (num_link_dailinks > 1); } - WARN_ON(codec_conf != card->codec_conf + card->num_configs); - return num_dais; } static int create_sdw_dailink(struct snd_soc_card *card, struct sof_sdw_dailink *sof_dai, struct snd_soc_dai_link **dai_links, - int *be_id) + int *be_id, struct snd_soc_codec_conf **codec_conf) { struct device *dev = card->dev; struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct sof_sdw_endpoint *sof_end; int stream; + list_for_each_entry(sof_end, &sof_dai->endpoints, list) { + if (sof_end->name_prefix) { + (*codec_conf)->dlc.name = sof_end->codec_name; + (*codec_conf)->name_prefix = sof_end->name_prefix; + (*codec_conf)++; + } + } + for_each_pcm_streams(stream) { static const char * const sdw_stream_name[] = { "SDW%d-Playback", @@ -1569,7 +1573,8 @@ static int create_sdw_dailink(struct snd_soc_card *card, static int create_sdw_dailinks(struct snd_soc_card *card, struct snd_soc_dai_link **dai_links, int *be_id, - struct sof_sdw_dailink *sof_dais) + struct sof_sdw_dailink *sof_dais, + struct snd_soc_codec_conf **codec_conf) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); int ret, i; @@ -1581,7 +1586,8 @@ static int create_sdw_dailinks(struct snd_soc_card *card, while (sof_dais->initialised) { int current_be_id; - ret = create_sdw_dailink(card, sof_dais, dai_links, ¤t_be_id); + ret = create_sdw_dailink(card, sof_dais, dai_links, + ¤t_be_id, codec_conf); if (ret) return ret; @@ -1751,16 +1757,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto err_dai; } - /* will be populated when acpi endpoints are parsed */ - codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL); - if (!codec_conf) { - ret = -ENOMEM; - goto err_end; - } - - card->codec_conf = codec_conf; - card->num_configs = num_devs; - ret = parse_sdw_endpoints(card, sof_dais, sof_ends); if (ret < 0) goto err_end; @@ -1798,6 +1794,12 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) sdw_be_num, ssp_num, dmic_num, ctx->hdmi.idisp_codec ? hdmi_num : 0, bt_num); + codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL); + if (!codec_conf) { + ret = -ENOMEM; + goto err_end; + } + /* allocate BE dailinks */ num_links = sdw_be_num + ssp_num + dmic_num + hdmi_num + bt_num; dai_links = devm_kcalloc(dev, num_links, sizeof(*dai_links), GFP_KERNEL); @@ -1806,12 +1808,15 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto err_end; } + card->codec_conf = codec_conf; + card->num_configs = num_devs; card->dai_link = dai_links; card->num_links = num_links; /* SDW */ if (sdw_be_num) { - ret = create_sdw_dailinks(card, &dai_links, &be_id, sof_dais); + ret = create_sdw_dailinks(card, &dai_links, &be_id, + sof_dais, &codec_conf); if (ret) goto err_end; } @@ -1847,6 +1852,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto err_end; } + WARN_ON(codec_conf != card->codec_conf + card->num_configs); WARN_ON(dai_links != card->dai_link + card->num_links); err_end: From da5244180281a18c4c7859674fec308514aaf629 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 26 Apr 2024 10:21:22 -0500 Subject: [PATCH 421/603] ASoC: Intel: sof_sdw: Add callbacks to register sidecar devices Add support for systems that have additional non-SoundWire devices (sidecars) connected to one of the SoundWire devices in the system. This is done through the addition of two callbacks, one used at endpoint parsing time that will return the number of devices and DAI links to be added, and another called later as the DAI links are created that will populate those devices into the appropriate arrays. Reviewed-by: Bard Liao Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 22 ++++++++++++++++++++-- sound/soc/intel/boards/sof_sdw_common.h | 6 ++++++ 2 files changed, 26 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index bf5f46a4c4aa9..eaa79e29f5c2b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1280,6 +1280,7 @@ struct sof_sdw_endpoint { u32 link_mask; const char *codec_name; const char *name_prefix; + bool include_sidecar; struct sof_sdw_codec_info *codec_info; const struct sof_sdw_dai_info *dai_info; @@ -1335,7 +1336,8 @@ static struct sof_sdw_dailink *find_dailink(struct sof_sdw_dailink *dailinks, static int parse_sdw_endpoints(struct snd_soc_card *card, struct sof_sdw_dailink *sof_dais, - struct sof_sdw_endpoint *sof_ends) + struct sof_sdw_endpoint *sof_ends, + int *num_devs) { struct device *dev = card->dev; struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -1345,6 +1347,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, struct sof_sdw_endpoint *sof_end = sof_ends; int num_dais = 0; int i, j; + int ret; for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) { int num_link_dailinks = 0; @@ -1381,6 +1384,14 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, sof_end->name_prefix = adr_dev->name_prefix; + if (codec_info->count_sidecar && codec_info->add_sidecar) { + ret = codec_info->count_sidecar(card, &num_dais, num_devs); + if (ret) + return ret; + + sof_end->include_sidecar = true; + } + for (j = 0; j < adr_dev->num_endpoints; j++) { const struct snd_soc_acpi_endpoint *adr_end; const struct sof_sdw_dai_info *dai_info; @@ -1453,6 +1464,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct sof_sdw_endpoint *sof_end; int stream; + int ret; list_for_each_entry(sof_end, &sof_dai->endpoints, list) { if (sof_end->name_prefix) { @@ -1460,6 +1472,12 @@ static int create_sdw_dailink(struct snd_soc_card *card, (*codec_conf)->name_prefix = sof_end->name_prefix; (*codec_conf)++; } + + if (sof_end->include_sidecar) { + ret = sof_end->codec_info->add_sidecar(card, dai_links, codec_conf); + if (ret) + return ret; + } } for_each_pcm_streams(stream) { @@ -1757,7 +1775,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto err_dai; } - ret = parse_sdw_endpoints(card, sof_dais, sof_ends); + ret = parse_sdw_endpoints(card, sof_dais, sof_ends, &num_devs); if (ret < 0) goto err_end; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 853278c6e5250..9dd42a8da8d74 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -98,6 +98,12 @@ struct sof_sdw_codec_info { const int dai_num; int (*codec_card_late_probe)(struct snd_soc_card *card); + + int (*count_sidecar)(struct snd_soc_card *card, + int *num_dais, int *num_devs); + int (*add_sidecar)(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, + struct snd_soc_codec_conf **codec_conf); }; struct mc_private { From b831b4dca48dbe0f1f7705b44460dd9ca7f2f940 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 26 Apr 2024 10:21:23 -0500 Subject: [PATCH 422/603] ASoC: intel: sof_sdw: Add support for cs42l43-cs35l56 sidecar amps The cs42l43 has both a SPI master and an I2S interface, these can be used to populate 2 cs35l56 amplifiers as sidecar devices along side the cs42l43. Giving a system that looks like: +-----+ +---------+ <- SPI -> +---------+ | CPU | <- SDW -> | CS42L43 | | CS35L56 | +-----+ +---------+ <- I2S -> +---------+ Add a quirk to specify this feature is present and use it to add codec to codec DAI link to connect the amplifiers into the sound card, add appropriate widgets, and setup clocking on the amplifiers. Reviewed-by: Bard Liao Signed-off-by: Maciej Strozek Signed-off-by: Charles Keepax Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426152123.36284-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/bridge_cs35l56.c | 137 +++++++++++++++++++++++ sound/soc/intel/boards/sof_sdw.c | 6 +- sound/soc/intel/boards/sof_sdw_common.h | 20 ++++ sound/soc/intel/boards/sof_sdw_cs42l43.c | 14 ++- 6 files changed, 173 insertions(+), 6 deletions(-) create mode 100644 sound/soc/intel/boards/bridge_cs35l56.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 0ad7b0a1e237a..b122b8aedd9a9 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -690,6 +690,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_CS42L43_SDW select MFD_CS42L43 select MFD_CS42L43_SDW + select SND_SOC_CS35L56_SPI select SND_SOC_CS35L56_SDW select SND_SOC_DMIC select SND_SOC_INTEL_HDA_DSP_COMMON diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 119413c262de7..1c099e717ecad 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -37,6 +37,7 @@ snd-soc-ehl-rt5660-objs := ehl_rt5660.o snd-soc-sof-ssp-amp-objs := sof_ssp_amp.o snd-soc-sof-sdw-objs += sof_sdw.o \ sof_sdw_maxim.o sof_sdw_rt_amp.o \ + bridge_cs35l56.o \ sof_sdw_rt5682.o sof_sdw_rt700.o \ sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ sof_sdw_rt712_sdca.o sof_sdw_rt722_sdca.o \ diff --git a/sound/soc/intel/boards/bridge_cs35l56.c b/sound/soc/intel/boards/bridge_cs35l56.c new file mode 100644 index 0000000000000..c3995e724aed9 --- /dev/null +++ b/sound/soc/intel/boards/bridge_cs35l56.c @@ -0,0 +1,137 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Intel SOF Machine Driver with Cirrus Logic CS35L56 Smart Amp + +#include +#include +#include +#include +#include +#include +#include +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget bridge_widgets[] = { + SND_SOC_DAPM_SPK("Bridge Speaker", NULL), +}; + +static const struct snd_soc_dapm_route bridge_map[] = { + {"Bridge Speaker", NULL, "AMPL SPK"}, + {"Bridge Speaker", NULL, "AMPR SPK"}, +}; + +static const char * const bridge_cs35l56_name_prefixes[] = { + "AMPL", + "AMPR", +}; + +static int bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int i, ret; + unsigned int rx_mask = 3; // ASP RX1, RX2 + unsigned int tx_mask = 3; // ASP TX1, TX2 + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s spk:cs35l56-bridge", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_dapm_new_controls(&card->dapm, bridge_widgets, + ARRAY_SIZE(bridge_widgets)); + if (ret) { + dev_err(card->dev, "widgets addition failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, bridge_map, ARRAY_SIZE(bridge_map)); + if (ret) { + dev_err(card->dev, "map addition failed: %d\n", ret); + return ret; + } + + /* 4 x 16-bit sample slots and FSYNC=48000, BCLK=3.072 MHz */ + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_mask, rx_mask, 4, 16); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 3072000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + } + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_mask, rx_mask, 4, 16); + if (ret < 0) + return ret; + } + + return 0; +} + +static const struct snd_soc_pcm_stream bridge_params = { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +SND_SOC_DAILINK_DEFS(bridge_dai, + DAILINK_COMP_ARRAY(COMP_CODEC("cs42l43-codec", "cs42l43-asp")), + DAILINK_COMP_ARRAY(COMP_CODEC("spi-cs35l56-left", "cs35l56-asp1"), + COMP_CODEC("spi-cs35l56-right", "cs35l56-asp1")), + DAILINK_COMP_ARRAY(COMP_PLATFORM("cs42l43-codec"))); + +static const struct snd_soc_dai_link bridge_dai_template = { + .name = "cs42l43-cs35l56", + .init = bridge_cs35l56_asp_init, + .c2c_params = &bridge_params, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBC_CFC, + SND_SOC_DAILINK_REG(bridge_dai), +}; + +int bridge_cs35l56_count_sidecar(struct snd_soc_card *card, + int *num_dais, int *num_devs) +{ + if (sof_sdw_quirk & SOF_SIDECAR_AMPS) { + (*num_dais)++; + (*num_devs) += ARRAY_SIZE(bridge_cs35l56_name_prefixes); + } + + return 0; +} + +int bridge_cs35l56_add_sidecar(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, + struct snd_soc_codec_conf **codec_conf) +{ + if (sof_sdw_quirk & SOF_SIDECAR_AMPS) { + **dai_links = bridge_dai_template; + + for (int i = 0; i < ARRAY_SIZE(bridge_cs35l56_name_prefixes); i++) { + (*codec_conf)->dlc.name = (*dai_links)->codecs[i].name; + (*codec_conf)->name_prefix = bridge_cs35l56_name_prefixes[i]; + (*codec_conf)++; + } + + (*dai_links)++; + } + + return 0; +} + +int bridge_cs35l56_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + if (sof_sdw_quirk & SOF_SIDECAR_AMPS) + info->amp_num += ARRAY_SIZE(bridge_cs35l56_name_prefixes); + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index eaa79e29f5c2b..b1595fdb500df 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -39,6 +39,8 @@ static void log_quirks(struct device *dev) dev_err(dev, "quirk SOF_SDW_NO_AGGREGATION enabled but no longer supported\n"); if (sof_sdw_quirk & SOF_CODEC_SPKR) dev_dbg(dev, "quirk SOF_CODEC_SPKR enabled\n"); + if (sof_sdw_quirk & SOF_SIDECAR_AMPS) + dev_dbg(dev, "quirk SOF_SIDECAR_AMPS enabled\n"); } static int sof_sdw_quirk_cb(const struct dmi_system_id *id) @@ -995,6 +997,8 @@ static struct sof_sdw_codec_info codec_info_list[] = { { .part_id = 0x4243, .codec_name = "cs42l43-codec", + .count_sidecar = bridge_cs35l56_count_sidecar, + .add_sidecar = bridge_cs35l56_add_sidecar, .dais = { { .direction = {true, false}, @@ -1023,7 +1027,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, .init = sof_sdw_cs42l43_spk_init, .rtd_init = cs42l43_spk_rtd_init, - .quirk = SOF_CODEC_SPKR, + .quirk = SOF_CODEC_SPKR | SOF_SIDECAR_AMPS, }, }, .dai_num = 4, diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 9dd42a8da8d74..94657dd210f56 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -55,6 +55,16 @@ enum { #define SOF_SDW_NO_AGGREGATION BIT(14) /* If a CODEC has an optional speaker output, this quirk will enable it */ #define SOF_CODEC_SPKR BIT(15) +/* + * If the CODEC has additional devices attached directly to it. + * + * For the cs42l43: + * - 0 - No speaker output + * - SOF_CODEC_SPKR - CODEC internal speaker + * - SOF_SIDECAR_AMPS - 2x Sidecar amplifiers + CODEC internal speaker + * - SOF_CODEC_SPKR | SOF_SIDECAR_AMPS - Not currently supported + */ +#define SOF_SIDECAR_AMPS BIT(16) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 15 @@ -177,6 +187,16 @@ int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, bool playback); /* CS AMP support */ +int bridge_cs35l56_count_sidecar(struct snd_soc_card *card, + int *num_dais, int *num_devs); +int bridge_cs35l56_add_sidecar(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, + struct snd_soc_codec_conf **codec_conf); +int bridge_cs35l56_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + int sof_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index 5361249f0f536..0fd5e099bb1a7 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -124,10 +124,14 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; int ret; - card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s spk:cs42l43-spk", - card->components); - if (!card->components) - return -ENOMEM; + if (!(sof_sdw_quirk & SOF_SIDECAR_AMPS)) { + /* Will be set by the bridge code in this case */ + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s spk:cs42l43-spk", + card->components); + if (!card->components) + return -ENOMEM; + } ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_spk_widgets, ARRAY_SIZE(cs42l43_spk_widgets)); @@ -155,7 +159,7 @@ int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, info->amp_num++; - return 0; + return bridge_cs35l56_spk_init(card, dai_links, info, playback); } int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) From b32487ca7b51ce430f15ec785269f11c25a6a560 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 26 Apr 2024 10:38:58 -0500 Subject: [PATCH 423/603] ASoC: SOF: debug: Handle cases when fw_lib_prefix is not set, NULL The firmware libraries are not supported by IPC3, the fw_lib_path is not a valid parameter and it is always NULL. Do not create the debugfs file for IPC3 at all as it is not applicable. With IPC4 some vendors/platforms might not support loadable libraries and the fw_lib_prefix is left to NULL to indicate this. Handle such case with allocating "Not supported" string. Reviewed-by: Marc Herbert Fixes: 17f4041244e6 ("ASoC: SOF: debug: show firmware/topology prefix/names") Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426153902.39560-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/debug.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index cf282c8b18af5..bf635cc96ede0 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -326,8 +326,27 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev) debugfs_create_str("fw_path", 0444, fw_profile, (char **)&plat_data->fw_filename_prefix); - debugfs_create_str("fw_lib_path", 0444, fw_profile, - (char **)&plat_data->fw_lib_prefix); + /* library path is not valid for IPC3 */ + if (plat_data->ipc_type != SOF_IPC_TYPE_3) { + /* + * fw_lib_prefix can be NULL if the vendor/platform does not + * support loadable libraries + */ + if (plat_data->fw_lib_prefix) { + debugfs_create_str("fw_lib_path", 0444, fw_profile, + (char **)&plat_data->fw_lib_prefix); + } else { + static char *fw_lib_path; + + fw_lib_path = devm_kasprintf(sdev->dev, GFP_KERNEL, + "Not supported"); + if (!fw_lib_path) + return -ENOMEM; + + debugfs_create_str("fw_lib_path", 0444, fw_profile, + (char **)&fw_lib_path); + } + } debugfs_create_str("tplg_path", 0444, fw_profile, (char **)&plat_data->tplg_filename_prefix); debugfs_create_str("fw_name", 0444, fw_profile, From 490e6c9b81e0c22087d250246717aee26ac5002e Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Fri, 26 Apr 2024 10:38:59 -0500 Subject: [PATCH 424/603] ASoC: SOF: topology: remove incorrect widget id in error message In sof_widget_ready() function, the shift field of struct snd_soc_tplg_dapm_widget is incorrectly used to print widget id in dev_err(scomp->dev, "error: failed to add widget id %d ..", this patch removes the useless tw->shift from the error output. Reviewed-by: Ranjani Sridharan Signed-off-by: Yong Zhi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426153902.39560-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index bcdb499c96a02..ec931a26b54f0 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1531,10 +1531,9 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, /* check token parsing reply */ if (ret < 0) { dev_err(scomp->dev, - "error: failed to add widget id %d type %d name : %s stream %s\n", - tw->shift, swidget->id, tw->name, - strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0 - ? tw->sname : "none"); + "failed to add widget type %d name : %s stream %s\n", + swidget->id, tw->name, strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0 + ? tw->sname : "none"); goto widget_free; } From 5d4788b2647fabeaeeaf331e53451e0ed6241252 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:39:00 -0500 Subject: [PATCH 425/603] ASoC: SOF: Intel: hda: simplify and reduce indentation for hda_sdw_machine_select() Simplify code to return when no links are enabled. No functional change, just code cleanup before updates. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20240426153902.39560-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 143 +++++++++++++++++++------------------- 1 file changed, 73 insertions(+), 70 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 2fc10bec7331c..d69cafc2d313e 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1653,92 +1653,95 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev hdev = pdata->hw_pdata; link_mask = hdev->info.link_mask; + if (!link_mask) { + dev_info(sdev->dev, "SoundWire links not enabled\n"); + return NULL; + } + /* * Select SoundWire machine driver if needed using the * alternate tables. This case deals with SoundWire-only * machines, for mixed cases with I2C/I2S the detection relies * on the HID list. */ - if (link_mask) { - for (mach = pdata->desc->alt_machines; - mach && mach->link_mask; mach++) { - /* - * On some platforms such as Up Extreme all links - * are enabled but only one link can be used by - * external codec. Instead of exact match of two masks, - * first check whether link_mask of mach is subset of - * link_mask supported by hw and then go on searching - * link_adr - */ - if (~link_mask & mach->link_mask) - continue; - - /* No need to match adr if there is no links defined */ - if (!mach->links) - break; - - link = mach->links; - for (i = 0; i < hdev->info.count && link->num_adr; - i++, link++) { - /* - * Try next machine if any expected Slaves - * are not found on this link. - */ - if (!snd_soc_acpi_sdw_link_slaves_found(sdev->dev, link, - hdev->sdw->ids, - hdev->sdw->num_slaves)) - break; - } - /* Found if all Slaves are checked */ - if (i == hdev->info.count || !link->num_adr) - break; - } - if (mach && mach->link_mask) { - int dmic_num = 0; - bool tplg_fixup; - const char *tplg_filename; - - mach->mach_params.links = mach->links; - mach->mach_params.link_mask = mach->link_mask; - mach->mach_params.platform = dev_name(sdev->dev); + for (mach = pdata->desc->alt_machines; + mach && mach->link_mask; mach++) { + /* + * On some platforms such as Up Extreme all links + * are enabled but only one link can be used by + * external codec. Instead of exact match of two masks, + * first check whether link_mask of mach is subset of + * link_mask supported by hw and then go on searching + * link_adr + */ + if (~link_mask & mach->link_mask) + continue; - if (pdata->tplg_filename) { - tplg_fixup = false; - } else { - tplg_fixup = true; - tplg_filename = mach->sof_tplg_filename; - } + /* No need to match adr if there is no links defined */ + if (!mach->links) + break; + link = mach->links; + for (i = 0; i < hdev->info.count && link->num_adr; + i++, link++) { /* - * DMICs use up to 4 pins and are typically pin-muxed with SoundWire - * link 2 and 3, or link 1 and 2, thus we only try to enable dmics - * if all conditions are true: - * a) 2 or fewer links are used by SoundWire - * b) the NHLT table reports the presence of microphones + * Try next machine if any expected Slaves + * are not found on this link. */ - if (hweight_long(mach->link_mask) <= 2) { - int ret; - - ret = dmic_detect_topology_fixup(sdev, &tplg_filename, "", - &dmic_num, tplg_fixup); - if (ret < 0) - return NULL; - } - if (tplg_fixup) - pdata->tplg_filename = tplg_filename; - mach->mach_params.dmic_num = dmic_num; + if (!snd_soc_acpi_sdw_link_slaves_found(sdev->dev, link, + hdev->sdw->ids, + hdev->sdw->num_slaves)) + break; + } + /* Found if all Slaves are checked */ + if (i == hdev->info.count || !link->num_adr) + break; + } + if (mach && mach->link_mask) { + int dmic_num = 0; + bool tplg_fixup; + const char *tplg_filename; + + mach->mach_params.links = mach->links; + mach->mach_params.link_mask = mach->link_mask; + mach->mach_params.platform = dev_name(sdev->dev); + + if (pdata->tplg_filename) { + tplg_fixup = false; + } else { + tplg_fixup = true; + tplg_filename = mach->sof_tplg_filename; + } - dev_dbg(sdev->dev, - "SoundWire machine driver %s topology %s\n", - mach->drv_name, - pdata->tplg_filename); + /* + * DMICs use up to 4 pins and are typically pin-muxed with SoundWire + * link 2 and 3, or link 1 and 2, thus we only try to enable dmics + * if all conditions are true: + * a) 2 or fewer links are used by SoundWire + * b) the NHLT table reports the presence of microphones + */ + if (hweight_long(mach->link_mask) <= 2) { + int ret; - return mach; + ret = dmic_detect_topology_fixup(sdev, &tplg_filename, "", + &dmic_num, tplg_fixup); + if (ret < 0) + return NULL; } + if (tplg_fixup) + pdata->tplg_filename = tplg_filename; + mach->mach_params.dmic_num = dmic_num; - dev_info(sdev->dev, "No SoundWire machine driver found\n"); + dev_dbg(sdev->dev, + "SoundWire machine driver %s topology %s\n", + mach->drv_name, + pdata->tplg_filename); + + return mach; } + dev_info(sdev->dev, "No SoundWire machine driver found\n"); + return NULL; } #else From 7ff01ca730f240811c13d9c3f8f8030211b3c911 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 26 Apr 2024 10:39:01 -0500 Subject: [PATCH 426/603] ASoC: SOF: Intel: hda: list SoundWire peripherals on mismatch Most of the SoundWire support issues come from bad ACPI information, or configuration reported by ACPI that are not supported by the SOF driver/topology. The users see a "No SoundWire machine driver found" message without any details, and the fallback to HDaudio w/ HDMI is used. We can reduce our support load with a clear dev_info() log that will give us a clear hint on the mismatch and why a machine driver/topology were not found. Example log on a MTL device: [ 13.158599] sof-audio-pci-intel-mtl 0000:00:1f.3: No SoundWire machine driver found for the ACPI-reported configuration: [ 13.158603] sof-audio-pci-intel-mtl 0000:00:1f.3: link 0 mfg_id 0x025d part_id 0x0713 version 0x3 [ 13.158606] sof-audio-pci-intel-mtl 0000:00:1f.3: link 1 mfg_id 0x025d part_id 0x1316 version 0x3 [ 13.158608] sof-audio-pci-intel-mtl 0000:00:1f.3: link 2 mfg_id 0x025d part_id 0x1316 version 0x3 In parallel, we will also provide an update to `alsa-info` to log all SoundWire peripherals found in ACPI tables. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20240426153902.39560-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index d69cafc2d313e..d509cfa38a642 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1645,6 +1645,7 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev { struct snd_sof_pdata *pdata = sdev->pdata; const struct snd_soc_acpi_link_adr *link; + struct sdw_extended_slave_id *ids; struct snd_soc_acpi_mach *mach; struct sof_intel_hda_dev *hdev; u32 link_mask; @@ -1658,6 +1659,16 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev return NULL; } + if (!hdev->sdw) { + dev_dbg(sdev->dev, "SoundWire context not allocated\n"); + return NULL; + } + + if (!hdev->sdw->num_slaves) { + dev_warn(sdev->dev, "No SoundWire peripheral detected in ACPI tables\n"); + return NULL; + } + /* * Select SoundWire machine driver if needed using the * alternate tables. This case deals with SoundWire-only @@ -1740,7 +1751,11 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev return mach; } - dev_info(sdev->dev, "No SoundWire machine driver found\n"); + dev_info(sdev->dev, "No SoundWire machine driver found for the ACPI-reported configuration:\n"); + ids = hdev->sdw->ids; + for (i = 0; i < hdev->sdw->num_slaves; i++) + dev_info(sdev->dev, "link %d mfg_id 0x%04x part_id 0x%04x version %#x\n", + ids[i].link_id, ids[i].id.mfg_id, ids[i].id.part_id, ids[i].id.sdw_version); return NULL; } From eed4872a4220f30de37aeca695d2881630410b66 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 26 Apr 2024 10:39:02 -0500 Subject: [PATCH 427/603] ASoC: SOF: ipc4-topology: Advertise passthrough capable PCMs (using ChainDMA) If a PCM is set to use ChainDMA then add it to the card->components string after a marker of iec61937-pcm:, for example on current HDA platforms where HDMI is set to use ChainDMA: iec61937-pcm:5,4,3 (the order of the PCM ids can differ) UCM is expected to parse and use this property to allow applications to use bytestream passthrough in a standard way. Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240426153902.39560-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 50 +++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index c29c1de4e9258..a972b2b5122a3 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -407,6 +407,52 @@ static void sof_ipc4_widget_update_kcontrol_module_id(struct snd_sof_widget *swi } } +static int +sof_ipc4_update_card_components_string(struct snd_sof_widget *swidget, + struct snd_sof_pcm *spcm, int dir) +{ + struct snd_sof_widget *pipe_widget = swidget->spipe->pipe_widget; + struct sof_ipc4_pipeline *pipeline = pipe_widget->private; + struct snd_soc_component *scomp = spcm->scomp; + struct snd_soc_card *card = scomp->card; + const char *pt_marker = "iec61937-pcm"; + + /* + * Update the card's components list with iec61937-pcm and a list of PCM + * ids where ChainDMA is enabled. + * These PCMs can be used for bytestream passthrough. + */ + if (!pipeline->use_chain_dma) + return 0; + + if (card->components) { + const char *tmp = card->components; + + if (strstr(card->components, pt_marker)) + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s,%d", + card->components, + spcm->pcm.pcm_id); + else + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s %s:%d", + card->components, + pt_marker, + spcm->pcm.pcm_id); + + devm_kfree(card->dev, tmp); + } else { + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s:%d", pt_marker, + spcm->pcm.pcm_id); + } + + if (!card->components) + return -ENOMEM; + + return 0; +} + static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) { struct sof_ipc4_available_audio_format *available_fmt; @@ -452,6 +498,10 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) if (!spcm) goto skip_gtw_cfg; + ret = sof_ipc4_update_card_components_string(swidget, spcm, dir); + if (ret) + goto free_available_fmt; + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { struct snd_sof_pcm_stream *sps = &spcm->stream[dir]; From ffca099bbff1978bc9c97b076f0d35b4fe6ddf27 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:48 +0200 Subject: [PATCH 428/603] ASoC: SOF: debug: Constify local snd_sof_dsp_ops Constify the pointer to 'struct snd_sof_dsp_ops' to annotate that functioon does not modify pointed data. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-1-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/debug.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index cf282c8b18af5..884d85770426b 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -311,8 +311,8 @@ EXPORT_SYMBOL_GPL(snd_sof_dbg_memory_info_init); int snd_sof_dbg_init(struct snd_sof_dev *sdev) { + const struct snd_sof_dsp_ops *ops = sof_ops(sdev); struct snd_sof_pdata *plat_data = sdev->pdata; - struct snd_sof_dsp_ops *ops = sof_ops(sdev); const struct snd_sof_debugfs_map *map; struct dentry *fw_profile; int i; From ee5acc1e035ec5ed5d9f0f63fda9d627220090c2 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:49 +0200 Subject: [PATCH 429/603] ASoC: SOF: ipc3: Constify local snd_sof_dsp_ops Constify the pointer to 'struct snd_sof_dsp_ops' to annotate that functioon does not modify pointed data. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-2-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-priv.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc3-priv.h b/sound/soc/sof/ipc3-priv.h index 0bbca418e67e6..9cd7794f3cb69 100644 --- a/sound/soc/sof/ipc3-priv.h +++ b/sound/soc/sof/ipc3-priv.h @@ -36,7 +36,7 @@ static inline int sof_dtrace_host_init(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmatb, struct sof_ipc_dma_trace_params_ext *dtrace_params) { - struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops; + const struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops; if (dsp_ops->trace_init) return dsp_ops->trace_init(sdev, dmatb, dtrace_params); @@ -46,7 +46,7 @@ static inline int sof_dtrace_host_init(struct snd_sof_dev *sdev, static inline int sof_dtrace_host_release(struct snd_sof_dev *sdev) { - struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops; + const struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops; if (dsp_ops->trace_release) return dsp_ops->trace_release(sdev); @@ -56,7 +56,7 @@ static inline int sof_dtrace_host_release(struct snd_sof_dev *sdev) static inline int sof_dtrace_host_trigger(struct snd_sof_dev *sdev, int cmd) { - struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops; + const struct snd_sof_dsp_ops *dsp_ops = sdev->pdata->desc->ops; if (dsp_ops->trace_trigger) return dsp_ops->trace_trigger(sdev, cmd); From a0db037df9630edad76153c7937c6f5ca04ed44f Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:50 +0200 Subject: [PATCH 430/603] ASoC: SOF: pcm: Constify local snd_sof_dsp_ops Constify the pointer to 'struct snd_sof_dsp_ops' to annotate that functioon does not modify pointed data. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-3-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 8fca4b0d63dec..cb774e011e31a 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -423,7 +423,7 @@ static int sof_pcm_open(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); - struct snd_sof_dsp_ops *ops = sof_ops(sdev); + const struct snd_sof_dsp_ops *ops = sof_ops(sdev); struct snd_sof_pcm *spcm; struct snd_soc_tplg_stream_caps *caps; int ret; From 8bbc692d1abce5bc949dea9acba85fc686601c04 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:51 +0200 Subject: [PATCH 431/603] ASoC: SOF: Constify stored pointer to snd_sof_dsp_ops The core code does not modify the 'struct snd_sof_dsp_ops', stored in 'struct sof_dev_desc'. Make the pointer to 'struct snd_sof_dsp_ops' const to annotate this, make code a bit safer and allow individual drivers to also define this structure as const. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-4-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- include/sound/sof.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/sof.h b/include/sound/sof.h index 05213bb515a38..780c1a755917b 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -166,7 +166,7 @@ struct sof_dev_desc { /* default firmware name */ const char *default_fw_filename[SOF_IPC_TYPE_COUNT]; - struct snd_sof_dsp_ops *ops; + const struct snd_sof_dsp_ops *ops; int (*ops_init)(struct snd_sof_dev *sdev); void (*ops_free)(struct snd_sof_dev *sdev); }; From 8f2b0d55abc44676b076128903a5dc730aab23c6 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:52 +0200 Subject: [PATCH 432/603] ASoC: SOF: intel: pci-tng: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-5-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-tng.c | 2 +- sound/soc/sof/intel/shim.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index c90173003c2ba..d8a36d5a29f76 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -132,7 +132,7 @@ static int tangier_pci_probe(struct snd_sof_dev *sdev) return ret; } -struct snd_sof_dsp_ops sof_tng_ops = { +const struct snd_sof_dsp_ops sof_tng_ops = { /* device init */ .probe = tangier_pci_probe, diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index 9515d753c816e..18ce3d33fa803 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -196,7 +196,7 @@ struct sof_intel_dsp_desc { int (*cl_init)(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot); }; -extern struct snd_sof_dsp_ops sof_tng_ops; +extern const struct snd_sof_dsp_ops sof_tng_ops; extern const struct sof_intel_dsp_desc tng_chip_info; From 6032eefc2c478243a511352dda04495c9a9fec6a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:53 +0200 Subject: [PATCH 433/603] ASoC: SOF: intel: hda: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-6-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-common-ops.c | 2 +- sound/soc/sof/intel/hda.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index d71bb66b99911..466769c68da55 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -14,7 +14,7 @@ #include "hda.h" #include "../sof-audio.h" -struct snd_sof_dsp_ops sof_hda_common_ops = { +const struct snd_sof_dsp_ops sof_hda_common_ops = { /* probe/remove/shutdown */ .probe_early = hda_dsp_probe_early, .probe = hda_dsp_probe, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index b59d1a572bce3..7804c5b50c265 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -874,7 +874,7 @@ int hda_dsp_dais_suspend(struct snd_sof_dev *sdev); /* * Platform Specific HW abstraction Ops. */ -extern struct snd_sof_dsp_ops sof_hda_common_ops; +extern const struct snd_sof_dsp_ops sof_hda_common_ops; extern struct snd_sof_dsp_ops sof_skl_ops; int sof_skl_ops_init(struct snd_sof_dev *sdev); From 04f2f516be09d5493d764e0020a771c46b5470d8 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:54 +0200 Subject: [PATCH 434/603] ASoC: SOF: amd: acp: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-7-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-common.c | 2 +- sound/soc/sof/amd/acp.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c index 0fc4e20ec6737..b26fa471b431f 100644 --- a/sound/soc/sof/amd/acp-common.c +++ b/sound/soc/sof/amd/acp-common.c @@ -193,7 +193,7 @@ struct snd_soc_acpi_mach *amd_sof_machine_select(struct snd_sof_dev *sdev) } /* AMD Common DSP ops */ -struct snd_sof_dsp_ops sof_acp_common_ops = { +const struct snd_sof_dsp_ops sof_acp_common_ops = { /* probe and remove */ .probe = amd_sof_acp_probe, .remove = amd_sof_acp_remove, diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index e229bb6b849d6..87e79d500865a 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -310,7 +310,7 @@ int acp_pcm_hw_params(struct snd_sof_dev *sdev, struct snd_pcm_substream *substr snd_pcm_uframes_t acp_pcm_pointer(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); -extern struct snd_sof_dsp_ops sof_acp_common_ops; +extern const struct snd_sof_dsp_ops sof_acp_common_ops; extern struct snd_sof_dsp_ops sof_renoir_ops; int sof_renoir_ops_init(struct snd_sof_dev *sdev); From ab85c44973298b69eb32795de11ce4d426705532 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:55 +0200 Subject: [PATCH 435/603] ASoC: SOF: imx8: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-8-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index d65e09a927f62..3021dc87ab5a2 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -478,7 +478,7 @@ static int imx8_dsp_set_power_state(struct snd_sof_dev *sdev, } /* i.MX8 ops */ -static struct snd_sof_dsp_ops sof_imx8_ops = { +static const struct snd_sof_dsp_ops sof_imx8_ops = { /* probe and remove */ .probe = imx8_probe, .remove = imx8_remove, @@ -539,7 +539,7 @@ static struct snd_sof_dsp_ops sof_imx8_ops = { }; /* i.MX8X ops */ -static struct snd_sof_dsp_ops sof_imx8x_ops = { +static const struct snd_sof_dsp_ops sof_imx8x_ops = { /* probe and remove */ .probe = imx8_probe, .remove = imx8_remove, From 66d49ab5fb51bb8d1b4c2c9c8fa0fbe8e4c8ca1c Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:56 +0200 Subject: [PATCH 436/603] ASoC: SOF: imx8m: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-9-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8m.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 3e0b88002b4f6..4ed415f043454 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -411,7 +411,7 @@ static int imx8m_dsp_suspend(struct snd_sof_dev *sdev, unsigned int target_state } /* i.MX8 ops */ -static struct snd_sof_dsp_ops sof_imx8m_ops = { +static const struct snd_sof_dsp_ops sof_imx8m_ops = { /* probe and remove */ .probe = imx8m_probe, .remove = imx8m_remove, From 232e0da9fa778233358586617bd22173bcac6bcc Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:57 +0200 Subject: [PATCH 437/603] ASoC: SOF: imx8ulp: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-10-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8ulp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 296fbf11f6d01..8adfdd00413aa 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -408,7 +408,7 @@ static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev, } /* i.MX8 ops */ -static struct snd_sof_dsp_ops sof_imx8ulp_ops = { +static const struct snd_sof_dsp_ops sof_imx8ulp_ops = { /* probe and remove */ .probe = imx8ulp_probe, .remove = imx8ulp_remove, From 936cc56044a87ae7fbd0e4098a7daefa0f2f4e8e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:58 +0200 Subject: [PATCH 438/603] ASoC: SOF: intel: bdw: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-11-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/intel/bdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index e30ca086f3f8d..2e8926cd00f66 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -567,7 +567,7 @@ static struct snd_soc_dai_driver bdw_dai[] = { }; /* broadwell ops */ -static struct snd_sof_dsp_ops sof_bdw_ops = { +static const struct snd_sof_dsp_ops sof_bdw_ops = { /*Device init */ .probe = bdw_probe, From 48d5f1800d0cbda0212c5a58177918c419a24f8a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:03:59 +0200 Subject: [PATCH 439/603] ASoC: SOF: intel: byt: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-12-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/intel/byt.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 373527b206d7b..14a065999a895 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -214,7 +214,7 @@ static int byt_acpi_probe(struct snd_sof_dev *sdev) } /* baytrail ops */ -static struct snd_sof_dsp_ops sof_byt_ops = { +static const struct snd_sof_dsp_ops sof_byt_ops = { /* device init */ .probe = byt_acpi_probe, .remove = byt_remove, @@ -289,7 +289,7 @@ static const struct sof_intel_dsp_desc byt_chip_info = { }; /* cherrytrail and braswell ops */ -static struct snd_sof_dsp_ops sof_cht_ops = { +static const struct snd_sof_dsp_ops sof_cht_ops = { /* device init */ .probe = byt_acpi_probe, .remove = byt_remove, From fe80673f59da01776a1402e4b508a66fca43a24d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:04:00 +0200 Subject: [PATCH 440/603] ASoC: SOF: mediatek: mt8186: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Reviewed-by: AngeloGioacchino Del Regno Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-13-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 0d2d7d697de02..c63e0d2f4b968 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -481,7 +481,7 @@ static struct snd_soc_dai_driver mt8186_dai[] = { }; /* mt8186 ops */ -static struct snd_sof_dsp_ops sof_mt8186_ops = { +static const struct snd_sof_dsp_ops sof_mt8186_ops = { /* probe and remove */ .probe = mt8186_dsp_probe, .remove = mt8186_dsp_remove, From 8b6d678fede700db6466d73f11fcbad496fa515e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 26 Apr 2024 11:04:01 +0200 Subject: [PATCH 441/603] ASoC: SOF: mediatek: mt8195: Constify snd_sof_dsp_ops 'struct snd_sof_dsp_ops' is not modified by core code, so it can be made const for increased code safety. Reviewed-by: AngeloGioacchino Del Regno Tested-by: Pierre-Louis Bossart Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240426-n-const-ops-var-v2-14-e553fe67ae82@kernel.org Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 8ee7ee246344c..fc1c016104aee 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -505,7 +505,7 @@ static struct snd_soc_dai_driver mt8195_dai[] = { }; /* mt8195 ops */ -static struct snd_sof_dsp_ops sof_mt8195_ops = { +static const struct snd_sof_dsp_ops sof_mt8195_ops = { /* probe and remove */ .probe = mt8195_dsp_probe, .remove = mt8195_dsp_remove, From 4a1baeefd1d5a955b5a55a75539244e03e623b0b Mon Sep 17 00:00:00 2001 From: Mohammad Shehar Yaar Tausif Date: Fri, 26 Apr 2024 22:33:18 +0530 Subject: [PATCH 442/603] ASoC: dt-bindings: tegra30-i2s: convert to dt schema Convert NVIDIA Tegra30 I2S binding to DT schema and add "clock-names" property used by multiple tegra i2s blocks in arch/arm64/boot/dts/nvidia/tegra132.dtsi. This is not a required property by the binding. Signed-off-by: Mohammad Shehar Yaar Tausif Reviewed-by: Rob Herring (Arm) Link: https://lore.kernel.org/r/20240426170322.36273-1-sheharyaar48@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/nvidia,tegra30-i2s.txt | 27 -------- .../bindings/sound/nvidia,tegra30-i2s.yaml | 67 +++++++++++++++++++ 2 files changed, 67 insertions(+), 27 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt deleted file mode 100644 index 38caa936f6f8a..0000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.txt +++ /dev/null @@ -1,27 +0,0 @@ -NVIDIA Tegra30 I2S controller - -Required properties: -- compatible : For Tegra30, must contain "nvidia,tegra30-i2s". For Tegra124, - must contain "nvidia,tegra124-i2s". Otherwise, must contain - "nvidia,-i2s" plus at least one of the above, where is - tegra114 or tegra132. -- reg : Should contain I2S registers location and length -- clocks : Must contain one entry, for the module clock. - See ../clocks/clock-bindings.txt for details. -- resets : Must contain an entry for each entry in reset-names. - See ../reset/reset.txt for details. -- reset-names : Must include the following entries: - - i2s -- nvidia,ahub-cif-ids : The list of AHUB CIF IDs for this port, rx (playback) - first, tx (capture) second. See nvidia,tegra30-ahub.txt for values. - -Example: - -i2s@70080300 { - compatible = "nvidia,tegra30-i2s"; - reg = <0x70080300 0x100>; - nvidia,ahub-cif-ids = <4 4>; - clocks = <&tegra_car 11>; - resets = <&tegra_car 11>; - reset-names = "i2s"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml new file mode 100644 index 0000000000000..89c3c6414ab13 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-i2s.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nvidia,tegra30-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NVIDIA Tegra30 I2S controller + +maintainers: + - Thierry Reding + - Jon Hunter + +properties: + compatible: + oneOf: + - enum: + - nvidia,tegra124-i2s + - nvidia,tegra30-i2s + - items: + - const: nvidia,tegra114-i2s + - const: nvidia,tegra30-i2s + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: i2s + + resets: + maxItems: 1 + + reset-names: + const: i2s + + nvidia,ahub-cif-ids: + description: list of AHUB CIF IDs + $ref: /schemas/types.yaml#/definitions/uint32-array + items: + - description: rx (playback) + - description: tx (capture) + +required: + - compatible + - reg + - clocks + - resets + - reset-names + - nvidia,ahub-cif-ids + +additionalProperties: false + +examples: + - | + #include + + i2s@70080300 { + compatible = "nvidia,tegra30-i2s"; + reg = <0x70080300 0x100>; + nvidia,ahub-cif-ids = <4 4>; + clocks = <&tegra_car TEGRA30_CLK_I2S0>; + resets = <&tegra_car 30>; + reset-names = "i2s"; + }; +... From 9a8cadddd9303ae15d1d518c4f2ddf00ee668729 Mon Sep 17 00:00:00 2001 From: Joao Schim Date: Mon, 29 Apr 2024 21:49:20 +0200 Subject: [PATCH 443/603] ASoC: sunxi: DMIC: Add controls for adjusting the mic gains The AllWinner H6 and later SoCs that sport a DMIC block contain a set of registers to control the gain (left + right) of each of the four supported channels. Add ASoC controls for changing each of the stereo channel gains using alsamixer and alike Signed-off-by: Joao Schim Reviewed-by: Jernej Skrabec Link: https://lore.kernel.org/r/20240429194920.1596257-1-joao@schimsalabim.eu Signed-off-by: Mark Brown --- sound/soc/sunxi/sun50i-dmic.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index c76628bc86c65..dd32780fb6a43 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -14,6 +14,7 @@ #include #include #include +#include #define SUN50I_DMIC_EN_CTL (0x00) #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) @@ -43,6 +44,17 @@ #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) #define SUN50I_DMIC_CNT (0x2c) #define SUN50I_DMIC_CNT_N (1 << 0) +#define SUN50I_DMIC_D0D1_VOL_CTR (0x30) + #define SUN50I_DMIC_D0D1_VOL_CTR_0R (0) + #define SUN50I_DMIC_D0D1_VOL_CTR_0L (8) + #define SUN50I_DMIC_D0D1_VOL_CTR_1R (16) + #define SUN50I_DMIC_D0D1_VOL_CTR_1L (24) +#define SUN50I_DMIC_D2D3_VOL_CTR (0x34) + #define SUN50I_DMIC_D2D3_VOL_CTR_2R (0) + #define SUN50I_DMIC_D2D3_VOL_CTR_2L (8) + #define SUN50I_DMIC_D2D3_VOL_CTR_3R (16) + #define SUN50I_DMIC_D2D3_VOL_CTR_3L (24) + #define SUN50I_DMIC_HPF_CTRL (0x38) #define SUN50I_DMIC_VERSION (0x50) @@ -273,8 +285,30 @@ static const struct of_device_id sun50i_dmic_of_match[] = { }; MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(sun50i_dmic_vol_scale, -12000, 75, 1); + +static const struct snd_kcontrol_new sun50i_dmic_controls[] = { + + SOC_DOUBLE_TLV("DMIC Channel 0 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_0L, SUN50I_DMIC_D0D1_VOL_CTR_0R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 1 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_1L, SUN50I_DMIC_D0D1_VOL_CTR_1R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 2 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_2L, SUN50I_DMIC_D2D3_VOL_CTR_2R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 3 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_3L, SUN50I_DMIC_D2D3_VOL_CTR_3R, + 0xFF, 0, sun50i_dmic_vol_scale), + + +}; + static const struct snd_soc_component_driver sun50i_dmic_component = { .name = "sun50i-dmic", + .controls = sun50i_dmic_controls, + .num_controls = ARRAY_SIZE(sun50i_dmic_controls), }; static int sun50i_dmic_runtime_suspend(struct device *dev) From 9723cab054d59d770cd1927f92fe315c26cc6a78 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 1 May 2024 19:51:27 +0200 Subject: [PATCH 444/603] ASoC: Use inline function for type safety in snd_soc_substream_to_rtd() A common pattern in sound drivers is getting 'struct snd_soc_pcm_runtime' from 'struct snd_pcm_substream' opaque pointer private_data field with snd_soc_substream_to_rtd(). However 'private_data' appears in several other structures as well, including 'struct snd_compr_stream'. The field might not hold the same type for every structure, although seems the case at least for 'struct snd_compr_stream', so code can easily make a mistake by using macro for wrong structure passed as argument. Switch from macro to inline function, so such mistake will be build-time detectable. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240501175127.34301-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- include/sound/soc.h | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8aa6ddb7ee1e8..33671437ee896 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1215,8 +1215,12 @@ struct snd_soc_pcm_runtime { /* see soc_new_pcm_runtime() */ #define snd_soc_rtd_to_cpu(rtd, n) (rtd)->dais[n] #define snd_soc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->dai_link->num_cpus] -#define snd_soc_substream_to_rtd(substream) \ - (struct snd_soc_pcm_runtime *)snd_pcm_substream_chip(substream) + +static inline struct snd_soc_pcm_runtime * +snd_soc_substream_to_rtd(const struct snd_pcm_substream *substream) +{ + return snd_pcm_substream_chip(substream); +} #define for_each_rtd_components(rtd, i, component) \ for ((i) = 0, component = NULL; \ From 09068d624c490c0e89f33f963c402f1859964467 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 2 May 2024 19:33:25 +0530 Subject: [PATCH 445/603] ASoC: amd: acp: fix for acp platform device creation failure ACP pin configuration varies based on acp version. ACP PCI driver should read the ACP PIN config value and based on config value, it has to create a platform device in below two conditions. 1) If ACP PDM configuration is selected from BIOS and ACP PDM controller exists. 2) If ACP I2S configuration is selected from BIOS. Other than above scenarios, ACP PCI driver should skip the platform device creation logic, i.e. ACP PCI driver probe sequence should never fail if other acp pin configuration is selected. It should skip platform device creation logic. check_acp_pdm() function was implemented for ACP6.x platforms to check ACP PDM configuration. Previously, this code was safe guarded by FLAG_AMD_LEGACY_ONLY_DMIC flag check. This implementation breaks audio use cases for Huawei Matebooks which are based on ACP3.x varaint uses I2S configuration. In current scenario, check_acp_pdm() function returns -ENODEV value which results in ACP PCI driver probe failure without creating a platform device even in case of valid ACP pin configuration. Implement check_acp_config() as a common function which invokes platform specific acp pin configuration check functions for ACP3.x, ACP6.0 & ACP6.3 & ACP7.0 variants and checks for ACP PDM controller. Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218780 Fixes: 4af565de9f8c ("ASoC: amd: acp: fix for acp pdm configuration check") Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20240502140340.4049021-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-legacy-common.c | 96 ++++++++++++++++++++++----- sound/soc/amd/acp/acp-pci.c | 9 ++- sound/soc/amd/acp/amd.h | 10 ++- sound/soc/amd/acp/chip_offset_byte.h | 1 + 4 files changed, 95 insertions(+), 21 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index b5aff3f230be5..3be7c6d55a6f8 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -358,11 +358,25 @@ int smn_read(struct pci_dev *dev, u32 smn_addr) } EXPORT_SYMBOL_NS_GPL(smn_read, SND_SOC_ACP_COMMON); -int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip) +static void check_acp3x_config(struct acp_chip_info *chip) { - struct acpi_device *pdm_dev; - const union acpi_object *obj; - u32 pdm_addr, val; + u32 val; + + val = readl(chip->base + ACP3X_PIN_CONFIG); + switch (val) { + case ACP_CONFIG_4: + chip->is_i2s_config = true; + chip->is_pdm_config = true; + break; + default: + chip->is_pdm_config = true; + break; + } +} + +static void check_acp6x_config(struct acp_chip_info *chip) +{ + u32 val; val = readl(chip->base + ACP_PIN_CONFIG); switch (val) { @@ -371,42 +385,94 @@ int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip) case ACP_CONFIG_6: case ACP_CONFIG_7: case ACP_CONFIG_8: - case ACP_CONFIG_10: case ACP_CONFIG_11: + case ACP_CONFIG_14: + chip->is_pdm_config = true; + break; + case ACP_CONFIG_9: + chip->is_i2s_config = true; + break; + case ACP_CONFIG_10: case ACP_CONFIG_12: case ACP_CONFIG_13: + chip->is_i2s_config = true; + chip->is_pdm_config = true; + break; + default: + break; + } +} + +static void check_acp70_config(struct acp_chip_info *chip) +{ + u32 val; + + val = readl(chip->base + ACP_PIN_CONFIG); + switch (val) { + case ACP_CONFIG_4: + case ACP_CONFIG_5: + case ACP_CONFIG_6: + case ACP_CONFIG_7: + case ACP_CONFIG_8: + case ACP_CONFIG_11: case ACP_CONFIG_14: + case ACP_CONFIG_17: + case ACP_CONFIG_18: + chip->is_pdm_config = true; + break; + case ACP_CONFIG_9: + chip->is_i2s_config = true; + break; + case ACP_CONFIG_10: + case ACP_CONFIG_12: + case ACP_CONFIG_13: + case ACP_CONFIG_19: + case ACP_CONFIG_20: + chip->is_i2s_config = true; + chip->is_pdm_config = true; break; default: - return -EINVAL; + break; } +} + +void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip) +{ + struct acpi_device *pdm_dev; + const union acpi_object *obj; + u32 pdm_addr; switch (chip->acp_rev) { case ACP3X_DEV: pdm_addr = ACP_RENOIR_PDM_ADDR; + check_acp3x_config(chip); break; case ACP6X_DEV: pdm_addr = ACP_REMBRANDT_PDM_ADDR; + check_acp6x_config(chip); break; case ACP63_DEV: pdm_addr = ACP63_PDM_ADDR; + check_acp6x_config(chip); break; case ACP70_DEV: pdm_addr = ACP70_PDM_ADDR; + check_acp70_config(chip); break; default: - return -EINVAL; + break; } - pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), pdm_addr, 0); - if (pdm_dev) { - if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", - ACPI_TYPE_INTEGER, &obj) && - obj->integer.value == pdm_addr) - return 0; + if (chip->is_pdm_config) { + pdm_dev = acpi_find_child_device(ACPI_COMPANION(&pci->dev), pdm_addr, 0); + if (pdm_dev) { + if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", + ACPI_TYPE_INTEGER, &obj) && + obj->integer.value == pdm_addr) + chip->is_pdm_dev = true; + } } - return -ENODEV; } -EXPORT_SYMBOL_NS_GPL(check_acp_pdm, SND_SOC_ACP_COMMON); +EXPORT_SYMBOL_NS_GPL(check_acp_config, SND_SOC_ACP_COMMON); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 5f35b90eab8d3..ad320b29e87dc 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -100,7 +100,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id ret = -EINVAL; goto release_regions; } - dmic_dev = platform_device_register_data(dev, "dmic-codec", PLATFORM_DEVID_NONE, NULL, 0); if (IS_ERR(dmic_dev)) { dev_err(dev, "failed to create DMIC device\n"); @@ -119,6 +118,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id if (ret) goto unregister_dmic_dev; + check_acp_config(pci, chip); + if (!chip->is_pdm_dev && !chip->is_i2s_config) + goto skip_pdev_creation; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; @@ -136,10 +139,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id } } - ret = check_acp_pdm(pci, chip); - if (ret < 0) - goto skip_pdev_creation; - chip->flag = flag; memset(&pdevinfo, 0, sizeof(pdevinfo)); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 5017e868f39b9..d75b4eb34de8d 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -138,6 +138,9 @@ struct acp_chip_info { void __iomem *base; /* ACP memory PCI base */ struct platform_device *chip_pdev; unsigned int flag; /* Distinguish b/w Legacy or Only PDM */ + bool is_pdm_dev; /* flag set to true when ACP PDM controller exists */ + bool is_pdm_config; /* flag set to true when PDM configuration is selected from BIOS */ + bool is_i2s_config; /* flag set to true when I2S configuration is selected from BIOS */ }; struct acp_stream { @@ -212,6 +215,11 @@ enum acp_config { ACP_CONFIG_13, ACP_CONFIG_14, ACP_CONFIG_15, + ACP_CONFIG_16, + ACP_CONFIG_17, + ACP_CONFIG_18, + ACP_CONFIG_19, + ACP_CONFIG_20, }; extern const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops; @@ -240,7 +248,7 @@ void restore_acp_pdm_params(struct snd_pcm_substream *substream, int restore_acp_i2s_params(struct snd_pcm_substream *substream, struct acp_dev_data *adata, struct acp_stream *stream); -int check_acp_pdm(struct pci_dev *pci, struct acp_chip_info *chip); +void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip); static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int direction) { diff --git a/sound/soc/amd/acp/chip_offset_byte.h b/sound/soc/amd/acp/chip_offset_byte.h index cfd6c4d075944..18da734c0e9e7 100644 --- a/sound/soc/amd/acp/chip_offset_byte.h +++ b/sound/soc/amd/acp/chip_offset_byte.h @@ -20,6 +20,7 @@ #define ACP_SOFT_RESET 0x1000 #define ACP_CONTROL 0x1004 #define ACP_PIN_CONFIG 0x1440 +#define ACP3X_PIN_CONFIG 0x1400 #define ACP_EXTERNAL_INTR_REG_ADDR(adata, offset, ctrl) \ (adata->acp_base + adata->rsrc->irq_reg_offset + offset + (ctrl * 0x04)) From ba2a2c378ada0cd641a1887151ea8af532617c69 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Thu, 2 May 2024 09:47:20 +0200 Subject: [PATCH 446/603] ASoC: codecs: Drop explicit initialization of struct i2c_device_id::driver_data to 0 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit These drivers don't use the driver_data member of struct i2c_device_id, so don't explicitly initialize this member. This prepares putting driver_data in an anonymous union which requires either no initialization or named designators. But it's also a nice cleanup on its own. Signed-off-by: Uwe Kleine-König Reviewed-by: Andy Shevchenko Link: https://lore.kernel.org/r/20240502074722.1103986-2-u.kleine-koenig@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/adau1372-i2c.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 8 ++++---- sound/soc/codecs/adau7118-i2c.c | 2 +- sound/soc/codecs/adav803.c | 2 +- sound/soc/codecs/ak4118.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/cs35l32.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41-i2c.c | 8 ++++---- sound/soc/codecs/cs35l45-i2c.c | 2 +- sound/soc/codecs/cs35l56-i2c.c | 2 +- sound/soc/codecs/cs4265.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271-i2c.c | 2 +- sound/soc/codecs/cs42l42-i2c.c | 2 +- sound/soc/codecs/cs42l51-i2c.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs43130.c | 8 ++++---- sound/soc/codecs/cs4341.c | 2 +- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/cx2072x.c | 4 ++-- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/da7213.c | 2 +- sound/soc/codecs/da732x.c | 2 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8326.c | 2 +- sound/soc/codecs/es8328-i2c.c | 4 ++-- sound/soc/codecs/isabelle.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/lm49453.c | 2 +- sound/soc/codecs/max9768.c | 2 +- sound/soc/codecs/max98371.c | 2 +- sound/soc/codecs/max98373-i2c.c | 2 +- sound/soc/codecs/max98388.c | 2 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/max98520.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max9877.c | 2 +- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/max98927.c | 2 +- sound/soc/codecs/ml26124.c | 2 +- sound/soc/codecs/mt6660.c | 2 +- sound/soc/codecs/nau8325.c | 2 +- sound/soc/codecs/nau8540.c | 2 +- sound/soc/codecs/nau8810.c | 6 +++--- sound/soc/codecs/nau8821.c | 2 +- sound/soc/codecs/nau8822.c | 2 +- sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/pcm1789-i2c.c | 2 +- sound/soc/codecs/pcm179x-i2c.c | 2 +- sound/soc/codecs/rt1011.c | 2 +- sound/soc/codecs/rt1015.c | 2 +- sound/soc/codecs/rt1016.c | 2 +- sound/soc/codecs/rt1019.c | 2 +- sound/soc/codecs/rt1305.c | 4 ++-- sound/soc/codecs/rt1308.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 4 ++-- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5631.c | 4 ++-- sound/soc/codecs/rt5640.c | 6 +++--- sound/soc/codecs/rt5645.c | 4 ++-- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 4 ++-- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5670.c | 6 +++--- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682s.c | 2 +- sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sma1303.c | 2 +- sound/soc/codecs/src4xxx-i2c.c | 2 +- sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/ssm4567.c | 2 +- sound/soc/codecs/sta32x.c | 6 +++--- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tas2764.c | 2 +- sound/soc/codecs/tas2770.c | 2 +- sound/soc/codecs/tas2780.c | 2 +- sound/soc/codecs/tas5086.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/codecs/tda7419.c | 2 +- sound/soc/codecs/tfa9879.c | 2 +- sound/soc/codecs/tlv320aic23-i2c.c | 2 +- sound/soc/codecs/ts3a227e.c | 2 +- sound/soc/codecs/tscs42xx.c | 4 ++-- sound/soc/codecs/tscs454.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm1250-ev1.c | 2 +- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731-i2c.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 4 ++-- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8804-i2c.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 4 ++-- 141 files changed, 168 insertions(+), 168 deletions(-) diff --git a/sound/soc/codecs/adau1372-i2c.c b/sound/soc/codecs/adau1372-i2c.c index 132b9e2cca590..2869325f9aced 100644 --- a/sound/soc/codecs/adau1372-i2c.c +++ b/sound/soc/codecs/adau1372-i2c.c @@ -21,7 +21,7 @@ static int adau1372_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id adau1372_i2c_ids[] = { - { "adau1372", 0 }, + { "adau1372" }, { } }; MODULE_DEVICE_TABLE(i2c, adau1372_i2c_ids); diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 3582c4b968a0f..a910e252aa126 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1496,7 +1496,7 @@ static int adau1373_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id adau1373_i2c_id[] = { - { "adau1373", 0 }, + { "adau1373" }, { } }; MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id); diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d1392d9abccd2..8bd6067df7f75 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -862,10 +862,10 @@ static int adau1701_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id adau1701_i2c_id[] = { - { "adau1401", 0 }, - { "adau1401a", 0 }, - { "adau1701", 0 }, - { "adau1702", 0 }, + { "adau1401" }, + { "adau1401a" }, + { "adau1701" }, + { "adau1702" }, { } }; MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c index b302b28eca7c9..f9dc8f4ef9a4e 100644 --- a/sound/soc/codecs/adau7118-i2c.c +++ b/sound/soc/codecs/adau7118-i2c.c @@ -68,7 +68,7 @@ static const struct of_device_id adau7118_of_match[] = { MODULE_DEVICE_TABLE(of, adau7118_of_match); static const struct i2c_device_id adau7118_id[] = { - {"adau7118", 0}, + {"adau7118"}, {} }; MODULE_DEVICE_TABLE(i2c, adau7118_id); diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c index 78a317947df9b..8b96c41f0354c 100644 --- a/sound/soc/codecs/adav803.c +++ b/sound/soc/codecs/adav803.c @@ -14,7 +14,7 @@ #include "adav80x.h" static const struct i2c_device_id adav803_id[] = { - { "adav803", 0 }, + { "adav803" }, { } }; MODULE_DEVICE_TABLE(i2c, adav803_id); diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c index 74a10108c1d43..9a43235e6a11d 100644 --- a/sound/soc/codecs/ak4118.c +++ b/sound/soc/codecs/ak4118.c @@ -396,7 +396,7 @@ MODULE_DEVICE_TABLE(of, ak4118_of_match); #endif static const struct i2c_device_id ak4118_id_table[] = { - { "ak4118", 0 }, + { "ak4118" }, {} }; MODULE_DEVICE_TABLE(i2c, ak4118_id_table); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 904bf91090aaa..aadc46a472806 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -430,7 +430,7 @@ static int ak4535_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id ak4535_i2c_id[] = { - { "ak4535", 0 }, + { "ak4535" }, { } }; MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id); diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 5b7df2f0dd6a1..ec33e7d73c6c2 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -619,7 +619,7 @@ static void ak4641_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id ak4641_i2c_id[] = { - { "ak4641", 0 }, + { "ak4641" }, { } }; MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id); diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 5b849b390c2a0..d545aa2e0a39c 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -646,7 +646,7 @@ static int ak4671_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id ak4671_i2c_id[] = { - { "ak4671", 0 }, + { "ak4671" }, { } }; MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index d1350ffbf3bdb..96555263e10b8 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -558,7 +558,7 @@ MODULE_DEVICE_TABLE(of, cs35l32_of_match); static const struct i2c_device_id cs35l32_id[] = { - {"cs35l32", 0}, + {"cs35l32"}, {} }; diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index a19a2bafb37c1..b03aab147530c 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1264,7 +1264,7 @@ static const struct of_device_id cs35l33_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l33_of_match); static const struct i2c_device_id cs35l33_id[] = { - {"cs35l33", 0}, + {"cs35l33"}, {} }; diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index cca59de66b73f..4c517231d765e 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -1198,7 +1198,7 @@ static const struct of_device_id cs35l34_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l34_of_match); static const struct i2c_device_id cs35l34_id[] = { - {"cs35l34", 0}, + {"cs35l34"}, {} }; MODULE_DEVICE_TABLE(i2c, cs35l34_id); diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index ddb7d63213a34..c39b3cfe95741 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1639,7 +1639,7 @@ static const struct of_device_id cs35l35_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l35_of_match); static const struct i2c_device_id cs35l35_id[] = { - {"cs35l35", 0}, + {"cs35l35"}, {} }; diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index f5bd32e434a0b..bc79990615e80 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -1930,7 +1930,7 @@ static const struct of_device_id cs35l36_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l36_of_match); static const struct i2c_device_id cs35l36_id[] = { - {"cs35l36", 0}, + {"cs35l36"}, {} }; diff --git a/sound/soc/codecs/cs35l41-i2c.c b/sound/soc/codecs/cs35l41-i2c.c index a0c457c0d04b4..34097996b784f 100644 --- a/sound/soc/codecs/cs35l41-i2c.c +++ b/sound/soc/codecs/cs35l41-i2c.c @@ -20,10 +20,10 @@ #include "cs35l41.h" static const struct i2c_device_id cs35l41_id_i2c[] = { - { "cs35l40", 0 }, - { "cs35l41", 0 }, - { "cs35l51", 0 }, - { "cs35l53", 0 }, + { "cs35l40" }, + { "cs35l41" }, + { "cs35l51" }, + { "cs35l53" }, {} }; diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index bc2af1ed0fe9b..f5fc42dcc8c70 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -53,7 +53,7 @@ static const struct of_device_id cs35l45_of_match[] = { MODULE_DEVICE_TABLE(of, cs35l45_of_match); static const struct i2c_device_id cs35l45_id_i2c[] = { - { "cs35l45", 0 }, + { "cs35l45" }, {} }; MODULE_DEVICE_TABLE(i2c, cs35l45_id_i2c); diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 7063c400e8961..2bd2ff75cd504 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -57,7 +57,7 @@ static void cs35l56_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id cs35l56_id_i2c[] = { - { "cs35l56", 0 }, + { "cs35l56" }, {} }; MODULE_DEVICE_TABLE(i2c, cs35l56_id_i2c); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 1ed1e60d8e536..78ffb7fa7fc5f 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -638,7 +638,7 @@ static const struct of_device_id cs4265_of_match[] = { MODULE_DEVICE_TABLE(of, cs4265_of_match); static const struct i2c_device_id cs4265_id[] = { - { "cs4265", 0 }, + { "cs4265" }, { } }; MODULE_DEVICE_TABLE(i2c, cs4265_id); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 3bbb90c827f22..67e92bfecb567 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -734,7 +734,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client) * cs4270_id - I2C device IDs supported by this driver */ static const struct i2c_device_id cs4270_id[] = { - {"cs4270", 0}, + {"cs4270"}, {} }; MODULE_DEVICE_TABLE(i2c, cs4270_id); diff --git a/sound/soc/codecs/cs4271-i2c.c b/sound/soc/codecs/cs4271-i2c.c index 89fe7d1665df1..1d210b9691738 100644 --- a/sound/soc/codecs/cs4271-i2c.c +++ b/sound/soc/codecs/cs4271-i2c.c @@ -23,7 +23,7 @@ static int cs4271_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id cs4271_i2c_id[] = { - { "cs4271", 0 }, + { "cs4271" }, { } }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); diff --git a/sound/soc/codecs/cs42l42-i2c.c b/sound/soc/codecs/cs42l42-i2c.c index 2552a1e6b82fb..8d10f9328e02d 100644 --- a/sound/soc/codecs/cs42l42-i2c.c +++ b/sound/soc/codecs/cs42l42-i2c.c @@ -78,7 +78,7 @@ static const struct acpi_device_id __maybe_unused cs42l42_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, cs42l42_acpi_match); static const struct i2c_device_id cs42l42_id[] = { - {"cs42l42", 0}, + {"cs42l42"}, {} }; diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 5ed2ef83dcdb7..e7cc500962978 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -14,7 +14,7 @@ #include "cs42l51.h" static struct i2c_device_id cs42l51_i2c_id[] = { - {"cs42l51", 0}, + {"cs42l51"}, {} }; MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id); diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 4fc8a6ae8d92c..7128d4c62f504 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1215,7 +1215,7 @@ MODULE_DEVICE_TABLE(of, cs42l52_of_match); static const struct i2c_device_id cs42l52_id[] = { - { "cs42l52", 0 }, + { "cs42l52" }, { } }; MODULE_DEVICE_TABLE(i2c, cs42l52_id); diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 3e3a86dab4fcc..aaa10c459b52d 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1330,7 +1330,7 @@ MODULE_DEVICE_TABLE(of, cs42l56_of_match); static const struct i2c_device_id cs42l56_id[] = { - { "cs42l56", 0 }, + { "cs42l56" }, { } }; MODULE_DEVICE_TABLE(i2c, cs42l56_id); diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 6ab67d196d102..21ba796a5cd93 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1372,7 +1372,7 @@ static const struct of_device_id cs42l73_of_match[] = { MODULE_DEVICE_TABLE(of, cs42l73_of_match); static const struct i2c_device_id cs42l73_id[] = { - {"cs42l73", 0}, + {"cs42l73"}, {} }; diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index b6d829bbe3cce..be4037890fdb3 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2763,10 +2763,10 @@ MODULE_DEVICE_TABLE(acpi, cs43130_acpi_match); static const struct i2c_device_id cs43130_i2c_id[] = { - {"cs43130", 0}, - {"cs4399", 0}, - {"cs43131", 0}, - {"cs43198", 0}, + {"cs43130"}, + {"cs4399"}, + {"cs43131"}, + {"cs43198"}, {} }; diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index 2ceca5d0e5bf8..d87aae31c5169 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -248,7 +248,7 @@ static int cs4341_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id cs4341_i2c_id[] = { - { "cs4341", 0 }, + { "cs4341" }, { } }; MODULE_DEVICE_TABLE(i2c, cs4341_i2c_id); diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index ca8f21aa4837b..a134ca7228927 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -361,7 +361,7 @@ static const struct of_device_id cs4349_of_match[] = { MODULE_DEVICE_TABLE(of, cs4349_of_match); static const struct i2c_device_id cs4349_i2c_id[] = { - {"cs4349", 0}, + {"cs4349"}, {} }; diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index f4065555c36e6..c0893146423b2 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -1104,7 +1104,7 @@ static const struct of_device_id cs53l30_of_match[] = { MODULE_DEVICE_TABLE(of, cs53l30_of_match); static const struct i2c_device_id cs53l30_id[] = { - { "cs53l30", 0 }, + { "cs53l30" }, {} }; diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index f8b1280840156..e8e22b1a1963b 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -1686,8 +1686,8 @@ static void cx2072x_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id cx2072x_i2c_id[] = { - { "cx20721", 0 }, - { "cx20723", 0 }, + { "cx20721" }, + { "cx20723" }, {} }; MODULE_DEVICE_TABLE(i2c, cx2072x_i2c_id); diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 1e232d01809c4..da2d0242019e2 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1238,7 +1238,7 @@ static int da7210_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id da7210_i2c_id[] = { - { "da7210", 0 }, + { "da7210" }, { } }; MODULE_DEVICE_TABLE(i2c, da7210_i2c_id); diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 369c620787800..a2b328f3b39fa 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -2238,7 +2238,7 @@ static const struct dev_pm_ops da7213_pm = { }; static const struct i2c_device_id da7213_i2c_id[] = { - { "da7213", 0 }, + { "da7213" }, { } }; MODULE_DEVICE_TABLE(i2c, da7213_i2c_id); diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f8ca1afa8af5f..b747f6fa12e4b 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1546,7 +1546,7 @@ static int da732x_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id da732x_i2c_id[] = { - { "da7320", 0}, + { "da7320"}, { } }; MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index c8a34572965d3..8bb8fef2a1d18 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1511,7 +1511,7 @@ static int da9055_i2c_probe(struct i2c_client *i2c) * and PMIC, which must be different to operate together. */ static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055-codec", 0 }, + { "da9055-codec" }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index e53b2856d6250..61729e5b50a8e 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -887,7 +887,7 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client) } static const struct i2c_device_id es8316_i2c_id[] = { - {"es8316", 0 }, + {"es8316" }, {} }; MODULE_DEVICE_TABLE(i2c, es8316_i2c_id); diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 93385f181d2cf..833ea52638abb 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -1263,7 +1263,7 @@ static void es8326_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id es8326_i2c_id[] = { - {"es8326", 0 }, + {"es8326" }, {} }; MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c index 3c4aaa0032a0e..56bfbe9261cee 100644 --- a/sound/soc/codecs/es8328-i2c.c +++ b/sound/soc/codecs/es8328-i2c.c @@ -16,8 +16,8 @@ #include "es8328.h" static const struct i2c_device_id es8328_id[] = { - { "es8328", 0 }, - { "es8388", 0 }, + { "es8328" }, + { "es8388" }, { } }; MODULE_DEVICE_TABLE(i2c, es8328_id); diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index f9456133a89a2..b7a94631d77de 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1133,7 +1133,7 @@ static int isabelle_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id isabelle_i2c_id[] = { - { "isabelle", 0 }, + { "isabelle" }, { } }; MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id); diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index e7542f71323d0..26cdb750cbca6 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -128,7 +128,7 @@ static int lm4857_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id lm4857_i2c_id[] = { - { "lm4857", 0 }, + { "lm4857" }, { } }; MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id); diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index a4094689b3ddf..ab89af7965bfc 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1442,7 +1442,7 @@ static int lm49453_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id lm49453_i2c_id[] = { - { "lm49453", 0 }, + { "lm49453" }, { } }; MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 8d0ca1be99c00..e4793a5d179ef 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -206,7 +206,7 @@ static int max9768_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id max9768_i2c_id[] = { - { "max9768", 0 }, + { "max9768" }, { } }; MODULE_DEVICE_TABLE(i2c, max9768_i2c_id); diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index f0e49179c38f6..852db211ba1e7 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -400,7 +400,7 @@ static int max98371_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98371_i2c_id[] = { - { "max98371", 0 }, + { "max98371" }, { } }; diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index e7ec7875c4a9a..1f7ff3dbcbbeb 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -578,7 +578,7 @@ static int max98373_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98373_i2c_id[] = { - { "max98373", 0}, + { "max98373"}, { }, }; diff --git a/sound/soc/codecs/max98388.c b/sound/soc/codecs/max98388.c index 078adec29312d..b847d7c59ec01 100644 --- a/sound/soc/codecs/max98388.c +++ b/sound/soc/codecs/max98388.c @@ -976,7 +976,7 @@ static int max98388_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98388_i2c_id[] = { - { "max98388", 0}, + { "max98388"}, { }, }; diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 5b8e78e516302..57fa2db1e1489 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -1104,7 +1104,7 @@ static int max98390_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98390_i2c_id[] = { - { "max98390", 0}, + { "max98390"}, {}, }; diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 8b012a85360a6..c395132689b4e 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -320,7 +320,7 @@ static int max9850_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max9850_i2c_id[] = { - { "max9850", 0 }, + { "max9850" }, { } }; MODULE_DEVICE_TABLE(i2c, max9850_i2c_id); diff --git a/sound/soc/codecs/max98520.c b/sound/soc/codecs/max98520.c index edd05253d37c0..479ded22672ec 100644 --- a/sound/soc/codecs/max98520.c +++ b/sound/soc/codecs/max98520.c @@ -734,7 +734,7 @@ static int max98520_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98520_i2c_id[] = { - { "max98520", 0}, + { "max98520"}, { }, }; diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 3b9dd158c34b9..50db88fce904f 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -684,7 +684,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max9867_i2c_id[] = { - { "max9867", 0 }, + { "max9867" }, { } }; MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 2ae64fcf29c74..1bd0d4761ca67 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -151,7 +151,7 @@ static int max9877_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id max9877_i2c_id[] = { - { "max9877", 0 }, + { "max9877" }, { } }; MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index a9c1d85cd0d55..66c78051bd098 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -617,7 +617,7 @@ static int max98925_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98925_i2c_id[] = { - { "max98925", 0 }, + { "max98925" }, { } }; MODULE_DEVICE_TABLE(i2c, max98925_i2c_id); diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 922ce0dc4e609..ae962bda163eb 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -565,7 +565,7 @@ static int max98926_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id max98926_i2c_id[] = { - { "max98926", 0 }, + { "max98926" }, { } }; MODULE_DEVICE_TABLE(i2c, max98926_i2c_id); diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index 70db9d3ff5a5f..747aa6f1d54f0 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -875,7 +875,7 @@ static void max98927_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id max98927_i2c_id[] = { - { "max98927", 0}, + { "max98927"}, { }, }; diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index a45ef9d65703d..c6585e8143a5e 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -572,7 +572,7 @@ static int ml26124_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id ml26124_i2c_id[] = { - { "ml26124", 0 }, + { "ml26124" }, { } }; MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id); diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index 5c50c7de26cd9..39a57f643d814 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -559,7 +559,7 @@ static const struct of_device_id __maybe_unused mt6660_of_id[] = { MODULE_DEVICE_TABLE(of, mt6660_of_id); static const struct i2c_device_id mt6660_i2c_id[] = { - {"mt6660", 0 }, + {"mt6660" }, {}, }; MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); diff --git a/sound/soc/codecs/nau8325.c b/sound/soc/codecs/nau8325.c index d65f731445977..2266f320a8f22 100644 --- a/sound/soc/codecs/nau8325.c +++ b/sound/soc/codecs/nau8325.c @@ -871,7 +871,7 @@ static int nau8325_i2c_probe(struct i2c_client *i2c, } static const struct i2c_device_id nau8325_i2c_ids[] = { - { "nau8325", 0 }, + { "nau8325" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8325_i2c_ids); diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 22251fb2fa1f8..7e59448e7ac63 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -965,7 +965,7 @@ static int nau8540_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8540_i2c_ids[] = { - { "nau8540", 0 }, + { "nau8540" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8540_i2c_ids); diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 97a54059474c6..dc3aaca899195 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -895,9 +895,9 @@ static int nau8810_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8810_i2c_id[] = { - { "nau8810", 0 }, - { "nau8812", 0 }, - { "nau8814", 0 }, + { "nau8810" }, + { "nau8812" }, + { "nau8814" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8810_i2c_id); diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 6818bbd1d3c76..de5c4db05c8f8 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -1922,7 +1922,7 @@ static int nau8821_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8821_i2c_ids[] = { - { "nau8821", 0 }, + { "nau8821" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8821_i2c_ids); diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 7199d734c79f2..e6909e64dfa39 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -1151,7 +1151,7 @@ static int nau8822_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8822_i2c_id[] = { - { "nau8822", 0 }, + { "nau8822" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8822_i2c_id); diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 704af1cf8cbf9..f92b95b21cae4 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -2003,7 +2003,7 @@ static int nau8824_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id nau8824_i2c_ids[] = { - { "nau8824", 0 }, + { "nau8824" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8824_i2c_ids); diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index cd30ad649bae0..bde25bc6909d5 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -2934,7 +2934,7 @@ static void nau8825_i2c_remove(struct i2c_client *client) {} static const struct i2c_device_id nau8825_i2c_ids[] = { - { "nau8825", 0 }, + { "nau8825" }, { } }; MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 316ad53bc66a3..fc152496d5dc5 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -291,7 +291,7 @@ static const struct snd_soc_component_driver soc_component_dev_pcm1681 = { }; static const struct i2c_device_id pcm1681_i2c_id[] = { - {"pcm1681", 0}, + {"pcm1681"}, {} }; MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); diff --git a/sound/soc/codecs/pcm1789-i2c.c b/sound/soc/codecs/pcm1789-i2c.c index f2d0b4d21e417..abadf4f8ed5ee 100644 --- a/sound/soc/codecs/pcm1789-i2c.c +++ b/sound/soc/codecs/pcm1789-i2c.c @@ -41,7 +41,7 @@ MODULE_DEVICE_TABLE(of, pcm1789_of_match); #endif static const struct i2c_device_id pcm1789_i2c_ids[] = { - { "pcm1789", 0 }, + { "pcm1789" }, { } }; MODULE_DEVICE_TABLE(i2c, pcm1789_i2c_ids); diff --git a/sound/soc/codecs/pcm179x-i2c.c b/sound/soc/codecs/pcm179x-i2c.c index 10579681f44b0..effc1dd6df22c 100644 --- a/sound/soc/codecs/pcm179x-i2c.c +++ b/sound/soc/codecs/pcm179x-i2c.c @@ -38,7 +38,7 @@ MODULE_DEVICE_TABLE(of, pcm179x_of_match); #endif static const struct i2c_device_id pcm179x_i2c_ids[] = { - { "pcm179x", 0 }, + { "pcm179x" }, { } }; MODULE_DEVICE_TABLE(i2c, pcm179x_i2c_ids); diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index d5285baad53a3..a0e75b03e9dc9 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2206,7 +2206,7 @@ MODULE_DEVICE_TABLE(acpi, rt1011_acpi_match); #endif static const struct i2c_device_id rt1011_i2c_id[] = { - { "rt1011", 0 }, + { "rt1011" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1011_i2c_id); diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 1250cfaf2adcb..0f806dde9c39c 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1097,7 +1097,7 @@ static const struct regmap_config rt1015_regmap = { }; static const struct i2c_device_id rt1015_i2c_id[] = { - { "rt1015", 0 }, + { "rt1015" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1015_i2c_id); diff --git a/sound/soc/codecs/rt1016.c b/sound/soc/codecs/rt1016.c index 919a1f25e584d..fed4da23cba23 100644 --- a/sound/soc/codecs/rt1016.c +++ b/sound/soc/codecs/rt1016.c @@ -608,7 +608,7 @@ static const struct regmap_config rt1016_regmap = { }; static const struct i2c_device_id rt1016_i2c_id[] = { - { "rt1016", 0 }, + { "rt1016" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1016_i2c_id); diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index ceb8baa6a20d7..d989d06a26147 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -540,7 +540,7 @@ static const struct regmap_config rt1019_regmap = { }; static const struct i2c_device_id rt1019_i2c_id[] = { - { "rt1019", 0 }, + { "rt1019" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1019_i2c_id); diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 80888cbcf49c9..c2b55be8d1653 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -981,8 +981,8 @@ MODULE_DEVICE_TABLE(acpi, rt1305_acpi_match); #endif static const struct i2c_device_id rt1305_i2c_id[] = { - { "rt1305", 0 }, - { "rt1306", 0 }, + { "rt1305" }, + { "rt1306" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1305_i2c_id); diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index 86afb429d4235..b366338cea711 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -795,7 +795,7 @@ MODULE_DEVICE_TABLE(acpi, rt1308_acpi_match); #endif static const struct i2c_device_id rt1308_i2c_id[] = { - { "rt1308", 0 }, + { "rt1308" }, { } }; MODULE_DEVICE_TABLE(i2c, rt1308_i2c_id); diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index 6e78434842509..bd61a257d7b58 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -1097,7 +1097,7 @@ MODULE_DEVICE_TABLE(of, rt274_of_match); #endif static const struct i2c_device_id rt274_i2c_id[] = { - {"rt274", 0}, + {"rt274"}, {} }; MODULE_DEVICE_TABLE(i2c, rt274_i2c_id); diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index f8994f4968c5c..d0f533120c330 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1075,8 +1075,8 @@ static const struct regmap_config rt286_regmap = { }; static const struct i2c_device_id rt286_i2c_id[] = { - {"rt286", 0}, - {"rt288", 0}, + {"rt286"}, + {"rt288"}, {} }; MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 03d9839a5de39..13aef6c5e91c4 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1137,7 +1137,7 @@ static const struct regmap_config rt298_regmap = { }; static const struct i2c_device_id rt298_i2c_id[] = { - {"rt298", 0}, + {"rt298"}, {} }; MODULE_DEVICE_TABLE(i2c, rt298_i2c_id); diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index a8cdc3d6994d5..2b3c0f9e178cc 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1199,7 +1199,7 @@ static const struct regmap_config rt5514_regmap = { }; static const struct i2c_device_id rt5514_i2c_id[] = { - { "rt5514", 0 }, + { "rt5514" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5514_i2c_id); diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index e7aa60e73961c..34461c4620092 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1320,7 +1320,7 @@ static const struct regmap_config rt5616_regmap = { }; static const struct i2c_device_id rt5616_i2c_id[] = { - { "rt5616", 0 }, + { "rt5616" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5616_i2c_id); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index a64e66c2d3c48..12df0c4f2097d 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1669,8 +1669,8 @@ static const struct snd_soc_component_driver soc_component_dev_rt5631 = { }; static const struct i2c_device_id rt5631_i2c_id[] = { - { "rt5631", 0 }, - { "alc5631", 0 }, + { "rt5631" }, + { "alc5631" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 174872ef35d25..16f3425a3e35c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2949,9 +2949,9 @@ static const struct regmap_config rt5640_regmap = { }; static const struct i2c_device_id rt5640_i2c_id[] = { - { "rt5640", 0 }, - { "rt5639", 0 }, - { "rt5642", 0 }, + { "rt5640" }, + { "rt5639" }, + { "rt5642" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d0d24a53df746..05f574bf8b8fb 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3630,8 +3630,8 @@ static const struct regmap_config temp_regmap = { }; static const struct i2c_device_id rt5645_i2c_id[] = { - { "rt5645", 0 }, - { "rt5650", 0 }, + { "rt5645" }, + { "rt5650" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 33a34bd0b4055..00421a1f54bf5 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2199,7 +2199,7 @@ MODULE_DEVICE_TABLE(acpi, rt5651_acpi_match); #endif static const struct i2c_device_id rt5651_i2c_id[] = { - { "rt5651", 0 }, + { "rt5651" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5651_i2c_id); diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index fb094c0fe740f..a2652fa6e1d74 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3815,8 +3815,8 @@ static const struct regmap_config rt5659_regmap = { }; static const struct i2c_device_id rt5659_i2c_id[] = { - { "rt5658", 0 }, - { "rt5659", 0 }, + { "rt5658" }, + { "rt5659" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5659_i2c_id); diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index d5c2f0f2df983..3ac41d2c279bd 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1224,7 +1224,7 @@ static const struct regmap_config rt5660_regmap = { }; static const struct i2c_device_id rt5660_i2c_id[] = { - { "rt5660", 0 }, + { "rt5660" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5660_i2c_id); diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 161dcb3915f9a..9d32debd36895 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3307,7 +3307,7 @@ static const struct regmap_config temp_regmap = { }; static const struct i2c_device_id rt5663_i2c_id[] = { - { "rt5663", 0 }, + { "rt5663" }, {} }; MODULE_DEVICE_TABLE(i2c, rt5663_i2c_id); diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 6f778c8f08328..47df14ba52784 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4635,7 +4635,7 @@ static const struct regmap_config rt5665_regmap = { }; static const struct i2c_device_id rt5665_i2c_id[] = { - {"rt5665", 0}, + {"rt5665"}, {} }; MODULE_DEVICE_TABLE(i2c, rt5665_i2c_id); diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 6d8e228ccb574..494ca3ce9b960 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2378,7 +2378,7 @@ static const struct regmap_config rt5668_regmap = { }; static const struct i2c_device_id rt5668_i2c_id[] = { - {"rt5668b", 0}, + {"rt5668b"}, {} }; MODULE_DEVICE_TABLE(i2c, rt5668_i2c_id); diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0e34293f3395b..30bf96c35b582 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2871,9 +2871,9 @@ static const struct regmap_config rt5670_regmap = { }; static const struct i2c_device_id rt5670_i2c_id[] = { - { "rt5670", 0 }, - { "rt5671", 0 }, - { "rt5672", 0 }, + { "rt5670" }, + { "rt5671" }, + { "rt5672" }, { } }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 62f26ce9d4760..ff9e14fad0cda 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -318,7 +318,7 @@ static const struct acpi_device_id rt5682_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5682_acpi_match); static const struct i2c_device_id rt5682_i2c_id[] = { - {"rt5682", 0}, + {"rt5682"}, {} }; MODULE_DEVICE_TABLE(i2c, rt5682_i2c_id); diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 12741668fdb37..f50f196d700d7 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -3319,7 +3319,7 @@ static const struct acpi_device_id rt5682s_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5682s_acpi_match); static const struct i2c_device_id rt5682s_i2c_id[] = { - {"rt5682s", 0}, + {"rt5682s"}, {} }; MODULE_DEVICE_TABLE(i2c, rt5682s_i2c_id); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 2f468f41b94db..7aa89e34657ea 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1809,7 +1809,7 @@ static void sgtl5000_i2c_shutdown(struct i2c_client *client) } static const struct i2c_device_id sgtl5000_id[] = { - {"sgtl5000", 0}, + {"sgtl5000"}, {}, }; diff --git a/sound/soc/codecs/sma1303.c b/sound/soc/codecs/sma1303.c index 61072e7574a0c..980c48cbc3482 100644 --- a/sound/soc/codecs/sma1303.c +++ b/sound/soc/codecs/sma1303.c @@ -1791,7 +1791,7 @@ static void sma1303_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id sma1303_i2c_id[] = { - {"sma1303", 0}, + {"sma1303"}, {} }; MODULE_DEVICE_TABLE(i2c, sma1303_i2c_id); diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c index 93af8e209b052..55f00ce7c7188 100644 --- a/sound/soc/codecs/src4xxx-i2c.c +++ b/sound/soc/codecs/src4xxx-i2c.c @@ -19,7 +19,7 @@ static int src4xxx_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id src4xxx_i2c_ids[] = { - { "src4392", 0 }, + { "src4392" }, { } }; MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids); diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index d20d897407ebb..06016e88dd270 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -793,7 +793,7 @@ MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); #endif static const struct i2c_device_id ssm2518_i2c_ids[] = { - { "ssm2518", 0 }, + { "ssm2518" }, { } }; MODULE_DEVICE_TABLE(i2c, ssm2518_i2c_ids); diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 0a6f04d8f636a..3e09c85abedbd 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -471,7 +471,7 @@ static int ssm4567_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id ssm4567_i2c_ids[] = { - { "ssm4567", 0 }, + { "ssm4567" }, { } }; MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index fcf0dbfbbbcab..bd8848ea1ec2f 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -1154,9 +1154,9 @@ static int sta32x_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id sta32x_i2c_id[] = { - { "sta326", 0 }, - { "sta328", 0 }, - { "sta329", 0 }, + { "sta326" }, + { "sta328" }, + { "sta329" }, { } }; MODULE_DEVICE_TABLE(i2c, sta32x_i2c_id); diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index 612cc1d7eafe9..d1450de926524 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1238,7 +1238,7 @@ static void sta350_i2c_remove(struct i2c_client *client) {} static const struct i2c_device_id sta350_i2c_id[] = { - { "sta350", 0 }, + { "sta350" }, { } }; MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index eedafef775e53..f7718491c8996 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -363,7 +363,7 @@ static int sta529_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id sta529_i2c_id[] = { - { "sta529", 0 }, + { "sta529" }, { } }; MODULE_DEVICE_TABLE(i2c, sta529_i2c_id); diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 8c9dc318b0e82..40f5f27e74c01 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -742,7 +742,7 @@ static void tas2552_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id tas2552_id[] = { - { "tas2552", 0 }, + { "tas2552" }, { } }; MODULE_DEVICE_TABLE(i2c, tas2552_id); diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index a9838e0738cc1..1dc719d726ab5 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -738,7 +738,7 @@ static int tas2764_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id tas2764_i2c_id[] = { - { "tas2764", 0}, + { "tas2764"}, { } }; MODULE_DEVICE_TABLE(i2c, tas2764_i2c_id); diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 99bf402eb5667..67bc1c8b01317 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -702,7 +702,7 @@ static int tas2770_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id tas2770_i2c_id[] = { - { "tas2770", 0}, + { "tas2770"}, { } }; MODULE_DEVICE_TABLE(i2c, tas2770_i2c_id); diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c index 972e8ea5ebde7..a18ccf5fb7ad2 100644 --- a/sound/soc/codecs/tas2780.c +++ b/sound/soc/codecs/tas2780.c @@ -626,7 +626,7 @@ static int tas2780_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id tas2780_i2c_id[] = { - { "tas2780", 0}, + { "tas2780"}, { } }; MODULE_DEVICE_TABLE(i2c, tas2780_i2c_id); diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index f52c14b43f287..6d45df3b9ba49 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -891,7 +891,7 @@ static const struct snd_soc_component_driver soc_component_dev_tas5086 = { }; static const struct i2c_device_id tas5086_i2c_id[] = { - { "tas5086", 0 }, + { "tas5086" }, { } }; MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index da89e8c681ddf..bb0500e9d3eac 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -792,7 +792,7 @@ static void tas6424_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id tas6424_i2c_ids[] = { - { "tas6424", 0 }, + { "tas6424" }, { } }; MODULE_DEVICE_TABLE(i2c, tas6424_i2c_ids); diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c index e187d74a17376..386b99c8023bd 100644 --- a/sound/soc/codecs/tda7419.c +++ b/sound/soc/codecs/tda7419.c @@ -614,7 +614,7 @@ static int tda7419_probe(struct i2c_client *i2c) } static const struct i2c_device_id tda7419_i2c_id[] = { - { "tda7419", 0 }, + { "tda7419" }, { } }; MODULE_DEVICE_TABLE(i2c, tda7419_i2c_id); diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 8cca2ceadd9e1..ac0c5c3376770 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -296,7 +296,7 @@ static int tfa9879_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id tfa9879_i2c_id[] = { - { "tfa9879", 0 }, + { "tfa9879" }, { } }; MODULE_DEVICE_TABLE(i2c, tfa9879_i2c_id); diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c index 9692ae007c91c..a31fb95048b8f 100644 --- a/sound/soc/codecs/tlv320aic23-i2c.c +++ b/sound/soc/codecs/tlv320aic23-i2c.c @@ -28,7 +28,7 @@ static int tlv320aic23_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id tlv320aic23_id[] = { - {"tlv320aic23", 0}, + {"tlv320aic23"}, {} }; diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 6d9166d9116a6..dbf448dd88649 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -427,7 +427,7 @@ static const struct dev_pm_ops ts3a227e_pm = { }; static const struct i2c_device_id ts3a227e_i2c_ids[] = { - { "ts3a227e", 0 }, + { "ts3a227e" }, { } }; MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 1eefc1fe6ea89..f8a3d1b40990c 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1485,8 +1485,8 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id tscs42xx_i2c_id[] = { - { "tscs42A1", 0 }, - { "tscs42A2", 0 }, + { "tscs42A1" }, + { "tscs42A2" }, { } }; MODULE_DEVICE_TABLE(i2c, tscs42xx_i2c_id); diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 744aef32a21f4..850e5de9271ed 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3457,7 +3457,7 @@ static int tscs454_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id tscs454_i2c_id[] = { - { "tscs454", 0 }, + { "tscs454" }, { } }; MODULE_DEVICE_TABLE(i2c, tscs454_i2c_id); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 5c5751dc14e52..4f8fdd574585b 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -782,7 +782,7 @@ static int uda1380_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id uda1380_i2c_id[] = { - { "uda1380", 0 }, + { "uda1380" }, { } }; MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 9fa6df48799b6..1f59309d8c693 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -204,7 +204,7 @@ static int wm1250_ev1_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm1250_ev1_i2c_id[] = { - { "wm1250-ev1", 0 }, + { "wm1250-ev1" }, { } }; MODULE_DEVICE_TABLE(i2c, wm1250_ev1_i2c_id); diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 9571ea53cb3f0..a07a443ba1963 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -929,7 +929,7 @@ static int wm2000_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm2000_i2c_id[] = { - { "wm2000", 0 }, + { "wm2000" }, { } }; MODULE_DEVICE_TABLE(i2c, wm2000_i2c_id); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 69c9c2bd7e7bb..841247173d98e 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2474,7 +2474,7 @@ static const struct dev_pm_ops wm2200_pm = { }; static const struct i2c_device_id wm2200_i2c_id[] = { - { "wm2200", 0 }, + { "wm2200" }, { } }; MODULE_DEVICE_TABLE(i2c, wm2200_i2c_id); diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 7ee4b45c0834f..11bbc94a282c7 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2670,7 +2670,7 @@ static const struct dev_pm_ops wm5100_pm = { }; static const struct i2c_device_id wm5100_i2c_id[] = { - { "wm5100", 0 }, + { "wm5100" }, { } }; MODULE_DEVICE_TABLE(i2c, wm5100_i2c_id); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 0e671cce84479..4a31d6f895020 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -668,7 +668,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8510_i2c_id[] = { - { "wm8510", 0 }, + { "wm8510" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8510_i2c_id); diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 41b14538b03c7..138eba7e577af 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -517,7 +517,7 @@ static int wm8523_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8523_i2c_id[] = { - { "wm8523", 0 }, + { "wm8523" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 7d339cc65208a..a1c99bbf5aa15 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -454,7 +454,7 @@ static int wm8711_i2c_probe(struct i2c_client *client) } static const struct i2c_device_id wm8711_i2c_id[] = { - { "wm8711", 0 }, + { "wm8711" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index d9cc78fbf1ea9..2cbd6b1894166 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -295,7 +295,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8728_i2c_id[] = { - { "wm8728", 0 }, + { "wm8728" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); diff --git a/sound/soc/codecs/wm8731-i2c.c b/sound/soc/codecs/wm8731-i2c.c index 7f68ad0380e04..1254e583af51e 100644 --- a/sound/soc/codecs/wm8731-i2c.c +++ b/sound/soc/codecs/wm8731-i2c.c @@ -47,7 +47,7 @@ static int wm8731_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8731_i2c_id[] = { - { "wm8731", 0 }, + { "wm8731" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index a0ba1e7dee98d..efdc242c2edea 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -639,7 +639,7 @@ static int wm8737_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8737_i2c_id[] = { - { "wm8737", 0 }, + { "wm8737" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8737_i2c_id); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index a0848774427b4..4863d6ac461bd 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -606,7 +606,7 @@ static int wm8741_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8741_i2c_id[] = { - { "wm8741", 0 }, + { "wm8741" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index b8d76cd001da8..cae97fa3bcb04 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -802,8 +802,8 @@ static int wm8750_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8750_i2c_id[] = { - { "wm8750", 0 }, - { "wm8987", 0 }, + { "wm8750" }, + { "wm8987" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index f42ed24314f3e..38b76b7275e56 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1580,7 +1580,7 @@ static int wm8753_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8753_i2c_id[] = { - { "wm8753", 0 }, + { "wm8753" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c index 7062a8b2f8b5d..e80dad87219b1 100644 --- a/sound/soc/codecs/wm8804-i2c.c +++ b/sound/soc/codecs/wm8804-i2c.c @@ -31,7 +31,7 @@ static void wm8804_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id wm8804_i2c_id[] = { - { "wm8804", 0 }, + { "wm8804" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 84d06c1904119..e44fdf97796f3 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1286,7 +1286,7 @@ static void wm8900_i2c_remove(struct i2c_client *client) {} static const struct i2c_device_id wm8900_i2c_id[] = { - { "wm8900", 0 }, + { "wm8900" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 84ae1102ac883..c643b5377d3a1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2199,7 +2199,7 @@ static const struct of_device_id wm8903_of_match[] = { MODULE_DEVICE_TABLE(of, wm8903_of_match); static const struct i2c_device_id wm8903_i2c_id[] = { - { "wm8903", 0 }, + { "wm8903" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b9432f8b64e5b..8a532f7d750c8 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -844,7 +844,7 @@ static int wm8940_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8940_i2c_id[] = { - { "wm8940", 0 }, + { "wm8940" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 4f43383264386..bae52a8a2e117 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -994,7 +994,7 @@ static int wm8955_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8955_i2c_id[] = { - { "wm8955", 0 }, + { "wm8955" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8955_i2c_id); diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 7689fe3cc86d6..00858b9c95686 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1549,7 +1549,7 @@ static void wm8960_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8960_i2c_id[] = { - { "wm8960", 0 }, + { "wm8960" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 8f8330efb341d..d1c731e25777b 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -966,7 +966,7 @@ static int wm8961_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8961_i2c_id[] = { - { "wm8961", 0 }, + { "wm8961" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7c6ed29831285..3ef95b7dcbdb9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3938,7 +3938,7 @@ static const struct dev_pm_ops wm8962_pm = { }; static const struct i2c_device_id wm8962_i2c_id[] = { - { "wm8962", 0 }, + { "wm8962" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8962_i2c_id); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index e88f323d28b23..b97c7d5bd4e76 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -691,7 +691,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8971_i2c_id[] = { - { "wm8971", 0 }, + { "wm8971" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8971_i2c_id); diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 260bac695b20a..0ee3655cad01f 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -707,7 +707,7 @@ static int wm8974_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8974_i2c_id[] = { - { "wm8974", 0 }, + { "wm8974" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 718bfef302cc6..40d22b36b7a96 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1056,7 +1056,7 @@ static int wm8978_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8978_i2c_id[] = { - { "wm8978", 0 }, + { "wm8978" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id); diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index b26d6a68e8d2c..252b4a6cac04d 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1059,7 +1059,7 @@ static int wm8983_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8983_i2c_id[] = { - { "wm8983", 0 }, + { "wm8983" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8983_i2c_id); diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 76f214f12ce03..f0e9d6e38dc09 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -896,7 +896,7 @@ static int wm8988_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8988_i2c_id[] = { - { "wm8988", 0 }, + { "wm8988" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5a8e765090af4..573bd3d487ba5 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1238,7 +1238,7 @@ static int wm8990_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8990_i2c_id[] = { - { "wm8990", 0 }, + { "wm8990" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8990_i2c_id); diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 590318aafaea1..3bd9b362051b3 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1314,7 +1314,7 @@ static int wm8991_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8991_i2c_id[] = { - { "wm8991", 0 }, + { "wm8991" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 5b788f35e5e4e..98a83f51ab263 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1732,7 +1732,7 @@ static void wm8993_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id wm8993_i2c_id[] = { - { "wm8993", 0 }, + { "wm8993" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 59ef2ef8ce007..1f9a9b6369350 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2258,7 +2258,7 @@ static int wm8995_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm8995_i2c_id[] = { - {"wm8995", 0}, + {"wm8995"}, {} }; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e738326e33eda..f6c5adce164ee 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3069,7 +3069,7 @@ static void wm8996_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8996_i2c_id[] = { - { "wm8996", 0 }, + { "wm8996" }, { } }; MODULE_DEVICE_TABLE(i2c, wm8996_i2c_id); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index e7ec799573d3d..cb9d040b34d64 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1360,7 +1360,7 @@ static void wm9081_i2c_remove(struct i2c_client *client) {} static const struct i2c_device_id wm9081_i2c_id[] = { - { "wm9081", 0 }, + { "wm9081" }, { } }; MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 50c1cbccfdb92..26191bcc161de 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -606,8 +606,8 @@ static int wm9090_i2c_probe(struct i2c_client *i2c) } static const struct i2c_device_id wm9090_id[] = { - { "wm9090", 0 }, - { "wm9093", 0 }, + { "wm9090" }, + { "wm9093" }, { } }; MODULE_DEVICE_TABLE(i2c, wm9090_id); From 5a7543d0ca01d68d992f480d151efd693807e0ce Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:32:51 -0500 Subject: [PATCH 447/603] ASoC: SOF: Intel: discard SoundWire configuration if HDaudio codec is reported MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The machine driver and topology selection starts with I2S, then SoundWire and last uses HDaudio as a fallback. That assumes that the ACPI information is correct but there are of course exceptions to the rule. On a Lenovo platform, an external HDaudio codec is detected, but the ACPI tables expose TWO RT711 jack codecs. This patch skips the SoundWire selection in case an external HDaudio codec is detected - which only works with the additional assumption that no one will mix HDaudio and SoundWire. Closes: https://github.com/thesofproject/linux/issues/4962 Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240503133253.108201-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 9 +++++++-- sound/soc/sof/intel/hda.h | 2 ++ 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index d509cfa38a642..6c10eb2c8241e 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1809,6 +1809,7 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) u32 interface_mask = hda_get_interface_mask(sdev); struct snd_sof_pdata *sof_pdata = sdev->pdata; const struct sof_dev_desc *desc = sof_pdata->desc; + struct hdac_bus *bus = sof_to_bus(sdev); struct snd_soc_acpi_mach *mach = NULL; enum snd_soc_acpi_intel_codec codec_type; const char *tplg_filename; @@ -1981,8 +1982,12 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) } } - /* If I2S fails, try SoundWire if it is supported */ - if (!mach && (interface_mask & BIT(SOF_DAI_INTEL_ALH))) + /* + * If I2S fails and no external HDaudio codec is detected, + * try SoundWire if it is supported + */ + if (!mach && !HDA_EXT_CODEC(bus->codec_mask) && + (interface_mask & BIT(SOF_DAI_INTEL_ALH))) mach = hda_sdw_machine_select(sdev); /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 7804c5b50c265..8b495264f23df 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -454,6 +454,8 @@ #define SSP_SET_SFRM_CONSUMER BIT(24) #define SSP_SET_CBP_CFP (SSP_SET_SCLK_CONSUMER | SSP_SET_SFRM_CONSUMER) +#define HDA_EXT_ADDR 0 +#define HDA_EXT_CODEC(x) ((x) & BIT(HDA_EXT_ADDR)) #define HDA_IDISP_ADDR 2 #define HDA_IDISP_CODEC(x) ((x) & BIT(HDA_IDISP_ADDR)) From b679302526d637700e998da85ed1e06d8f7d3674 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 May 2024 08:32:52 -0500 Subject: [PATCH 448/603] ASoC: SOF: ipc4-topology: Allow selective update in sof_ipc4_update_hw_params Add a bitmask parameter to sof_ipc4_update_hw_params() to be able to select the param to be updated. This feature can be used when not all params should be updated, for example if caller only wants to update the format in the params, leaving the channels and rates untouched. Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240503133253.108201-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 83 +++++++++++++++++++++-------------- 1 file changed, 51 insertions(+), 32 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index a972b2b5122a3..f56b7fbbe10fa 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1119,42 +1119,50 @@ static int sof_ipc4_widget_assign_instance_id(struct snd_sof_dev *sdev, /* update hw_params based on the audio stream format */ static int sof_ipc4_update_hw_params(struct snd_sof_dev *sdev, struct snd_pcm_hw_params *params, - struct sof_ipc4_audio_format *fmt) + struct sof_ipc4_audio_format *fmt, u32 param_to_update) { - snd_pcm_format_t snd_fmt; struct snd_interval *i; - struct snd_mask *m; - int valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); - unsigned int channels, rate; - switch (valid_bits) { - case 16: - snd_fmt = SNDRV_PCM_FORMAT_S16_LE; - break; - case 24: - snd_fmt = SNDRV_PCM_FORMAT_S24_LE; - break; - case 32: - snd_fmt = SNDRV_PCM_FORMAT_S32_LE; - break; - default: - dev_err(sdev->dev, "invalid PCM valid_bits %d\n", valid_bits); - return -EINVAL; + if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_FORMAT)) { + int valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); + snd_pcm_format_t snd_fmt; + struct snd_mask *m; + + switch (valid_bits) { + case 16: + snd_fmt = SNDRV_PCM_FORMAT_S16_LE; + break; + case 24: + snd_fmt = SNDRV_PCM_FORMAT_S24_LE; + break; + case 32: + snd_fmt = SNDRV_PCM_FORMAT_S32_LE; + break; + default: + dev_err(sdev->dev, "invalid PCM valid_bits %d\n", valid_bits); + return -EINVAL; + } + + m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(m); + snd_mask_set_format(m, snd_fmt); } - m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - snd_mask_none(m); - snd_mask_set_format(m, snd_fmt); + if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_RATE)) { + unsigned int rate = fmt->sampling_frequency; - rate = fmt->sampling_frequency; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - i->min = rate; - i->max = rate; + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + i->min = rate; + i->max = rate; + } - channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - i->min = channels; - i->max = channels; + if (param_to_update & BIT(SNDRV_PCM_HW_PARAM_CHANNELS)) { + unsigned int channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + i->min = channels; + i->max = channels; + } return 0; } @@ -1844,7 +1852,11 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, } /* modify the input params for the next widget */ - ret = sof_ipc4_update_hw_params(sdev, pipeline_params, &copier_data->out_format); + ret = sof_ipc4_update_hw_params(sdev, pipeline_params, + &copier_data->out_format, + BIT(SNDRV_PCM_HW_PARAM_FORMAT) | + BIT(SNDRV_PCM_HW_PARAM_CHANNELS) | + BIT(SNDRV_PCM_HW_PARAM_RATE)); if (ret) return ret; @@ -2069,7 +2081,10 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget, src->data.sink_rate = out_audio_fmt->sampling_frequency; /* update pipeline_params for sink widgets */ - return sof_ipc4_update_hw_params(sdev, pipeline_params, out_audio_fmt); + return sof_ipc4_update_hw_params(sdev, pipeline_params, out_audio_fmt, + BIT(SNDRV_PCM_HW_PARAM_FORMAT) | + BIT(SNDRV_PCM_HW_PARAM_CHANNELS) | + BIT(SNDRV_PCM_HW_PARAM_RATE)); } static int @@ -2193,7 +2208,11 @@ static int sof_ipc4_prepare_process_module(struct snd_sof_widget *swidget, sizeof(struct sof_ipc4_audio_format)); /* modify the pipeline params with the pin 0 output format */ - ret = sof_ipc4_update_hw_params(sdev, pipeline_params, &process->output_format); + ret = sof_ipc4_update_hw_params(sdev, pipeline_params, + &process->output_format, + BIT(SNDRV_PCM_HW_PARAM_FORMAT) | + BIT(SNDRV_PCM_HW_PARAM_CHANNELS) | + BIT(SNDRV_PCM_HW_PARAM_RATE)); if (ret) return ret; } From f9209644ae7688e82f629e737417bc8916db7b57 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 May 2024 08:32:53 -0500 Subject: [PATCH 449/603] ASoC: SOF: ipc4-topology: Correct DAI copier config and NHLT blob request In case of capture and when the DAI copier have single bit depth supported on it's input side we should use this format instead of the one in fe_params. Regardless of the stream direction for the NHLT blob lookup when the DAI copier only supports single bit depth on the DAI side we should only look for a blob which matches with this single configuration. For DMIC if the DAI copier supports multiple bit depths, try to request 32-bit blob first if the requested bit depth is 16-bit. If the 32-bit blob is available then look for marching (32-bit) copier format to make sure that both the blob and copier have correct parameters. Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240503133253.108201-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 162 ++++++++++++++++++++++++++++------ 1 file changed, 136 insertions(+), 26 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f56b7fbbe10fa..d0125fd1c09ff 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1420,13 +1420,16 @@ static int snd_sof_get_hw_config_params(struct snd_sof_dev *sdev, struct snd_sof return 0; } -static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, - struct snd_pcm_hw_params *params, u32 dai_index, - u32 linktype, u8 dir, u32 **dst, u32 *len) +static int +snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + bool single_format, + struct snd_pcm_hw_params *params, u32 dai_index, + u32 linktype, u8 dir, u32 **dst, u32 *len) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; struct nhlt_specific_cfg *cfg; int sample_rate, channel_count; + bool format_change = false; int bit_depth, ret; u32 nhlt_type; int dev_type = 0; @@ -1435,9 +1438,18 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s switch (linktype) { case SOF_DAI_INTEL_DMIC: nhlt_type = NHLT_LINK_DMIC; - bit_depth = params_width(params); channel_count = params_channels(params); sample_rate = params_rate(params); + bit_depth = params_width(params); + /* + * Look for 32-bit blob first instead of 16-bit if copier + * supports multiple formats + */ + if (bit_depth == 16 && !single_format) { + dev_dbg(sdev->dev, "Looking for 32-bit blob first for DMIC\n"); + format_change = true; + bit_depth = 32; + } break; case SOF_DAI_INTEL_SSP: nhlt_type = NHLT_LINK_SSP; @@ -1471,22 +1483,56 @@ static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_s dir, dev_type); if (!cfg) { + if (format_change) { + /* + * The 32-bit blob was not found in NHLT table, try to + * look for one based on the params + */ + bit_depth = params_width(params); + format_change = false; + + cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, + dai_index, nhlt_type, + bit_depth, bit_depth, + channel_count, sample_rate, + dir, dev_type); + if (cfg) + goto out; + } + dev_err(sdev->dev, "no matching blob for sample rate: %d sample width: %d channels: %d\n", sample_rate, bit_depth, channel_count); return -EINVAL; } +out: /* config length should be in dwords */ *len = cfg->size >> 2; *dst = (u32 *)cfg->caps; + if (format_change) { + /* + * Update the params to reflect that we have loaded 32-bit blob + * instead of the 16-bit. + * This information is going to be used by the caller to find + * matching copier format on the dai side. + */ + struct snd_mask *m; + + m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(m); + snd_mask_set_format(m, SNDRV_PCM_FORMAT_S32_LE); + } + return 0; } #else -static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, - struct snd_pcm_hw_params *params, u32 dai_index, - u32 linktype, u8 dir, u32 **dst, u32 *len) +static int +snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + bool single_format, + struct snd_pcm_hw_params *params, u32 dai_index, + u32 linktype, u8 dir, u32 **dst, u32 *len) { return 0; } @@ -1517,6 +1563,68 @@ bool sof_ipc4_copier_is_single_format(struct snd_sof_dev *sdev, return true; } +static int +sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, + struct snd_pcm_hw_params *params, int dir) +{ + struct sof_ipc4_available_audio_format *available_fmt; + struct snd_pcm_hw_params dai_params = *params; + struct sof_ipc4_copier_data *copier_data; + struct sof_ipc4_copier *ipc4_copier; + bool single_format; + int ret; + + ipc4_copier = dai->private; + copier_data = &ipc4_copier->data; + available_fmt = &ipc4_copier->available_fmt; + + /* + * If the copier on the DAI side supports only single bit depth then + * this depth (format) should be used to look for the NHLT blob (if + * needed) and in case of capture this should be used for the input + * format lookup + */ + if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + single_format = sof_ipc4_copier_is_single_format(sdev, + available_fmt->output_pin_fmts, + available_fmt->num_output_formats); + + /* Update the dai_params with the only supported format */ + if (single_format) { + ret = sof_ipc4_update_hw_params(sdev, &dai_params, + &available_fmt->output_pin_fmts[0].audio_fmt, + BIT(SNDRV_PCM_HW_PARAM_FORMAT)); + if (ret) + return ret; + } + } else { + single_format = sof_ipc4_copier_is_single_format(sdev, + available_fmt->input_pin_fmts, + available_fmt->num_input_formats); + + /* Update the dai_params with the only supported format */ + if (single_format) { + ret = sof_ipc4_update_hw_params(sdev, &dai_params, + &available_fmt->input_pin_fmts[0].audio_fmt, + BIT(SNDRV_PCM_HW_PARAM_FORMAT)); + if (ret) + return ret; + } + } + + ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_format, + &dai_params, + ipc4_copier->dai_index, + ipc4_copier->dai_type, dir, + &ipc4_copier->copier_config, + &copier_data->gtw_cfg.config_length); + /* Update the params to reflect the changes made in this function */ + if (!ret) + *params = dai_params; + + return ret; +} + static int sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct snd_pcm_hw_params *fe_params, @@ -1527,7 +1635,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, struct snd_soc_component *scomp = swidget->scomp; struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct sof_ipc4_copier_data *copier_data; - struct snd_pcm_hw_params *ref_params; + struct snd_pcm_hw_params ref_params; struct sof_ipc4_copier *ipc4_copier; struct snd_sof_dai *dai; u32 gtw_cfg_config_length; @@ -1605,9 +1713,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, * for capture. */ if (dir == SNDRV_PCM_STREAM_PLAYBACK) - ref_params = fe_params; + ref_params = *fe_params; else - ref_params = pipeline_params; + ref_params = *pipeline_params; copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; copier_data->gtw_cfg.node_id |= @@ -1633,23 +1741,25 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, available_fmt = &ipc4_copier->available_fmt; /* - * When there is format conversion within a pipeline, the number of supported - * output formats is typically limited to just 1 for the DAI copiers. But when there - * is no format conversion, the DAI copiers input format must match that of the - * FE hw_params for capture and the pipeline params for playback. + * Use the fe_params as a base for the copier configuration. + * The ref_params might get updated to reflect what format is + * supported by the copier on the DAI side. + * + * In case of capture the ref_params returned will be used to + * find the input configuration of the copier. */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK) - ref_params = pipeline_params; - else - ref_params = fe_params; - - ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, fe_params, ipc4_copier->dai_index, - ipc4_copier->dai_type, dir, - &ipc4_copier->copier_config, - &copier_data->gtw_cfg.config_length); + ref_params = *fe_params; + ret = sof_ipc4_prepare_dai_copier(sdev, dai, &ref_params, dir); if (ret < 0) return ret; + /* + * For playback the pipeline_params needs to be used to find the + * input configuration of the copier. + */ + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + ref_params = *pipeline_params; + break; } case snd_soc_dapm_buffer: @@ -1657,7 +1767,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, ipc4_copier = (struct sof_ipc4_copier *)swidget->private; copier_data = &ipc4_copier->data; available_fmt = &ipc4_copier->available_fmt; - ref_params = pipeline_params; + ref_params = *pipeline_params; break; } @@ -1668,8 +1778,8 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, } /* set input and output audio formats */ - ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &copier_data->base_config, ref_params, - available_fmt); + ret = sof_ipc4_init_input_audio_fmt(sdev, swidget, &copier_data->base_config, + &ref_params, available_fmt); if (ret < 0) return ret; From 9e7fd21ab03e6fadc4856f4c06e642daa9ccf7c6 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:52:15 -0500 Subject: [PATCH 450/603] ASoC: SOF: Intel: hda-stream: export stream_get_position() helper Export this helper so that we can report the DPIB position if the BPT DMA do not complete - this is very useful to see if the DMA started or gets stuck somehow with invalid bandwidth configurations. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Link: https://lore.kernel.org/r/20240503135221.229202-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 91b1371745e66..c5afcda183593 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -1079,6 +1079,7 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, return pos; } +EXPORT_SYMBOL_NS(hda_dsp_stream_get_position, SND_SOC_SOF_INTEL_HDA_COMMON); #define merge_u64(u32_u, u32_l) (((u64)(u32_u) << 32) | (u32_l)) From e4c6eba86d9618d99ba9892bc5d6e47715038530 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:52:16 -0500 Subject: [PATCH 451/603] ASoC: SOF: Intel: regroup all SoundWire/Intel functions in hda.c To avoid circular dependencies between SOF/Intel and SoundWire/Intel, we need to split the top-level hda.c from the rest of the code. This patch first regroups all SoundWire related code in hda.c. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20240503135221.229202-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 40 -------------------------------- sound/soc/sof/intel/hda.c | 39 +++++++++++++++++++++++++++++++ 2 files changed, 39 insertions(+), 40 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index dc88b7ea599ea..d0474522ea760 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -15,7 +15,6 @@ * Hardware interface for HDA DSP code loader */ -#include #include #include #include @@ -651,45 +650,6 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, return ret; } -/* pre fw run operations */ -int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) -{ - /* disable clock gating and power gating */ - return hda_dsp_ctrl_clock_power_gating(sdev, false); -} - -/* post fw run operations */ -int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) -{ - int ret; - - if (sdev->first_boot) { - struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; - - ret = hda_sdw_startup(sdev); - if (ret < 0) { - dev_err(sdev->dev, - "error: could not startup SoundWire links\n"); - return ret; - } - - /* Check if IMR boot is usable */ - if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) && - (sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT || - sdev->pdata->ipc_type == SOF_IPC_TYPE_4)) { - hdev->imrboot_supported = true; - debugfs_create_bool("skip_imr_boot", - 0644, sdev->debugfs_root, - &hdev->skip_imr_boot); - } - } - - hda_sdw_int_enable(sdev, true); - - /* re-enable clock gating and power gating */ - return hda_dsp_ctrl_clock_power_gating(sdev, true); -} - int hda_dsp_ext_man_get_cavs_config_data(struct snd_sof_dev *sdev, const struct sof_ext_man_elem_header *hdr) { diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index d509cfa38a642..93f316300043c 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -545,6 +545,45 @@ static inline bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) #endif /* IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) */ +/* pre fw run operations */ +int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) +{ + /* disable clock gating and power gating */ + return hda_dsp_ctrl_clock_power_gating(sdev, false); +} + +/* post fw run operations */ +int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) +{ + int ret; + + if (sdev->first_boot) { + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; + + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: could not startup SoundWire links\n"); + return ret; + } + + /* Check if IMR boot is usable */ + if (!sof_debug_check_flag(SOF_DBG_IGNORE_D3_PERSISTENT) && + (sdev->fw_ready.flags & SOF_IPC_INFO_D3_PERSISTENT || + sdev->pdata->ipc_type == SOF_IPC_TYPE_4)) { + hdev->imrboot_supported = true; + debugfs_create_bool("skip_imr_boot", + 0644, sdev->debugfs_root, + &hdev->skip_imr_boot); + } + } + + hda_sdw_int_enable(sdev, true); + + /* re-enable clock gating and power gating */ + return hda_dsp_ctrl_clock_power_gating(sdev, true); +} + /* * Debug */ From 456644cbeab14394fff1308ffdaf87d4e0e6fb01 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:52:17 -0500 Subject: [PATCH 452/603] ASoC: SOF: Intel: start splitting top-level from common parts The existing code relies on the 'HDA_COMMON' module and namespace. We need to start splitting top-level parts from the low-level ones, otherwise we will not be able to reuse the low-level parts DMA support for SoundWire/BPT. In the end the dependencies will be: +----------------------------------------------+ | | | v sof-pci-intel-xxx --> sof-intel-hda ------------> sof-hda-common | ^ | | +-> soundwire_intel --> sof_hda_sdw_bpt This patch adds the initial split between the sof-pci-intel-xxx modules and the common parts, in a follow-up patch we will further split the HDA_COMMON parts Since the PCI modules are not all independent, i.e. the CNL parts are also used in JSL and TGL, additional Kconfig and namespace modules were added. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20240503135221.229202-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 3 +++ sound/soc/sof/intel/Makefile | 17 ++++++++--------- sound/soc/sof/intel/apl.c | 3 --- sound/soc/sof/intel/cnl.c | 13 +++++++++---- sound/soc/sof/intel/hda-common-ops.c | 1 + sound/soc/sof/intel/hda-ctrl.c | 1 + sound/soc/sof/intel/hda-dai.c | 1 + sound/soc/sof/intel/hda-dsp.c | 16 ++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 11 +++++++++++ sound/soc/sof/intel/hda-loader.c | 4 ++++ sound/soc/sof/intel/hda.c | 18 ++++++++++++++++++ sound/soc/sof/intel/icl.c | 3 --- sound/soc/sof/intel/lnl.c | 3 --- sound/soc/sof/intel/mtl.c | 19 +++++++++++++++---- sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-lnl.c | 2 ++ sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/intel/telemetry.c | 1 + sound/soc/sof/intel/tgl.c | 6 ------ 19 files changed, 92 insertions(+), 32 deletions(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 9de86aaa8d074..b804be7321be5 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -188,6 +188,7 @@ config SND_SOC_SOF_INTEL_ICL select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE select SND_SOC_SOF_IPC3 select SND_SOC_SOF_IPC4 + select SND_SOC_SOF_INTEL_CNL config SND_SOC_SOF_ICELAKE tristate "SOF support for Icelake" @@ -215,6 +216,7 @@ config SND_SOC_SOF_INTEL_TGL select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE select SND_SOC_SOF_IPC3 select SND_SOC_SOF_IPC4 + select SND_SOC_SOF_INTEL_CNL config SND_SOC_SOF_TIGERLAKE tristate "SOF support for Tigerlake" @@ -267,6 +269,7 @@ config SND_SOC_SOF_INTEL_LNL select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE select SND_SOC_SOF_IPC4 + select SND_SOC_SOF_INTEL_MTL config SND_SOC_SOF_LUNARLAKE tristate "SOF support for Lunarlake" diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index 6489d0660d582..c895bae0efdd7 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -6,8 +6,7 @@ snd-sof-acpi-intel-bdw-objs := bdw.o snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \ hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ hda-dai.o hda-dai-ops.o hda-bus.o \ - skl.o hda-loader-skl.o \ - apl.o cnl.o tgl.o icl.o mtl.o lnl.o hda-common-ops.o \ + hda-common-ops.o \ telemetry.o snd-sof-intel-hda-mlink-objs := hda-mlink.o @@ -26,13 +25,13 @@ obj-$(CONFIG_SND_SOC_SOF_HDA_MLINK) += snd-sof-intel-hda-mlink.o obj-$(CONFIG_SND_SOC_SOF_HDA) += snd-sof-intel-hda.o snd-sof-pci-intel-tng-objs := pci-tng.o -snd-sof-pci-intel-skl-objs := pci-skl.o -snd-sof-pci-intel-apl-objs := pci-apl.o -snd-sof-pci-intel-cnl-objs := pci-cnl.o -snd-sof-pci-intel-icl-objs := pci-icl.o -snd-sof-pci-intel-tgl-objs := pci-tgl.o -snd-sof-pci-intel-mtl-objs := pci-mtl.o -snd-sof-pci-intel-lnl-objs := pci-lnl.o +snd-sof-pci-intel-skl-objs := pci-skl.o skl.o hda-loader-skl.o +snd-sof-pci-intel-apl-objs := pci-apl.o apl.o +snd-sof-pci-intel-cnl-objs := pci-cnl.o cnl.o +snd-sof-pci-intel-icl-objs := pci-icl.o icl.o +snd-sof-pci-intel-tgl-objs := pci-tgl.o tgl.o +snd-sof-pci-intel-mtl-objs := pci-mtl.o mtl.o +snd-sof-pci-intel-lnl-objs := pci-lnl.o lnl.o obj-$(CONFIG_SND_SOC_SOF_MERRIFIELD) += snd-sof-pci-intel-tng.o obj-$(CONFIG_SND_SOC_SOF_INTEL_SKL) += snd-sof-pci-intel-skl.o diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index dee6c7f73e80a..bc364c7cb3b6b 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -29,7 +29,6 @@ static const struct snd_sof_debugfs_map apl_dsp_debugfs[] = { /* apollolake ops */ struct snd_sof_dsp_ops sof_apl_ops; -EXPORT_SYMBOL_NS(sof_apl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); int sof_apl_ops_init(struct snd_sof_dev *sdev) { @@ -97,7 +96,6 @@ int sof_apl_ops_init(struct snd_sof_dev *sdev) return 0; }; -EXPORT_SYMBOL_NS(sof_apl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON); const struct sof_intel_dsp_desc apl_chip_info = { /* Apollolake */ @@ -121,4 +119,3 @@ const struct sof_intel_dsp_desc apl_chip_info = { .disable_interrupts = hda_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_CAVS_1_5_PLUS, }; -EXPORT_SYMBOL_NS(apl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 85e1e4760d0e5..460bc404ddb69 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -110,6 +110,7 @@ irqreturn_t cnl_ipc4_irq_thread(int irq, void *context) return IRQ_HANDLED; } +EXPORT_SYMBOL_NS(cnl_ipc4_irq_thread, SND_SOC_SOF_INTEL_CNL); irqreturn_t cnl_ipc_irq_thread(int irq, void *context) { @@ -202,6 +203,7 @@ irqreturn_t cnl_ipc_irq_thread(int irq, void *context) return IRQ_HANDLED; } +EXPORT_SYMBOL_NS(cnl_ipc_irq_thread, SND_SOC_SOF_INTEL_CNL); static void cnl_ipc_host_done(struct snd_sof_dev *sdev) { @@ -284,6 +286,7 @@ int cnl_ipc4_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) return 0; } +EXPORT_SYMBOL_NS(cnl_ipc4_send_msg, SND_SOC_SOF_INTEL_CNL); int cnl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) { @@ -331,6 +334,7 @@ int cnl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) return 0; } +EXPORT_SYMBOL_NS(cnl_ipc_send_msg, SND_SOC_SOF_INTEL_CNL); void cnl_ipc_dump(struct snd_sof_dev *sdev) { @@ -351,6 +355,7 @@ void cnl_ipc_dump(struct snd_sof_dev *sdev) "error: host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n", hipcida, hipctdr, hipcctl); } +EXPORT_SYMBOL_NS(cnl_ipc_dump, SND_SOC_SOF_INTEL_CNL); void cnl_ipc4_dump(struct snd_sof_dev *sdev) { @@ -372,10 +377,11 @@ void cnl_ipc4_dump(struct snd_sof_dev *sdev) "Host IPC initiator: %#x|%#x|%#x, target: %#x|%#x|%#x, ctl: %#x\n", hipcidr, hipcidd, hipcida, hipctdr, hipctdd, hipctda, hipcctl); } +EXPORT_SYMBOL_NS(cnl_ipc4_dump, SND_SOC_SOF_INTEL_CNL); /* cannonlake ops */ struct snd_sof_dsp_ops sof_cnl_ops; -EXPORT_SYMBOL_NS(sof_cnl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(sof_cnl_ops, SND_SOC_SOF_INTEL_CNL); int sof_cnl_ops_init(struct snd_sof_dev *sdev) { @@ -444,7 +450,7 @@ int sof_cnl_ops_init(struct snd_sof_dev *sdev) return 0; }; -EXPORT_SYMBOL_NS(sof_cnl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(sof_cnl_ops_init, SND_SOC_SOF_INTEL_CNL); const struct sof_intel_dsp_desc cnl_chip_info = { /* Cannonlake */ @@ -473,7 +479,6 @@ const struct sof_intel_dsp_desc cnl_chip_info = { .disable_interrupts = hda_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_CAVS_1_8, }; -EXPORT_SYMBOL_NS(cnl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); /* * JasperLake is technically derived from IceLake, and should be in @@ -509,4 +514,4 @@ const struct sof_intel_dsp_desc jsl_chip_info = { .disable_interrupts = hda_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_CAVS_2_0, }; -EXPORT_SYMBOL_NS(jsl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(jsl_chip_info, SND_SOC_SOF_INTEL_CNL); diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 466769c68da55..6a863ceba8da0 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -105,3 +105,4 @@ const struct snd_sof_dsp_ops sof_hda_common_ops = { .dsp_arch_ops = &sof_xtensa_arch_ops, }; +EXPORT_SYMBOL_NS(sof_hda_common_ops, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 6d941209847f5..56c0f493042c8 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -178,6 +178,7 @@ int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable) return 0; } +EXPORT_SYMBOL_NS(hda_dsp_ctrl_clock_power_gating, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev) { diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 3f2fd84907d2f..650e3a37326ea 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -690,6 +690,7 @@ void hda_set_dai_drv_ops(struct snd_sof_dev *sdev, struct snd_sof_dsp_ops *ops) ipc4_data->nhlt = intel_nhlt_init(sdev->dev); } } +EXPORT_SYMBOL_NS(hda_set_dai_drv_ops, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_ops_free(struct snd_sof_dev *sdev) { diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index ef5c915db8ffb..32f0340746686 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -126,6 +126,7 @@ int hda_dsp_core_stall_reset(struct snd_sof_dev *sdev, unsigned int core_mask) /* set reset state */ return hda_dsp_core_reset_enter(sdev, core_mask); } +EXPORT_SYMBOL_NS(hda_dsp_core_stall_reset, SND_SOC_SOF_INTEL_HDA_COMMON); bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev, unsigned int core_mask) { @@ -151,6 +152,7 @@ bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev, unsigned int core_mask) return is_enable; } +EXPORT_SYMBOL_NS(hda_dsp_core_is_enabled, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask) { @@ -178,6 +180,7 @@ int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_core_run, SND_SOC_SOF_INTEL_HDA_COMMON); /* * Power Management. @@ -229,6 +232,7 @@ int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_core_power_up, SND_SOC_SOF_INTEL_HDA_COMMON); static int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask) { @@ -276,6 +280,7 @@ int hda_dsp_enable_core(struct snd_sof_dev *sdev, unsigned int core_mask) return hda_dsp_core_run(sdev, core_mask); } +EXPORT_SYMBOL_NS(hda_dsp_enable_core, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev, unsigned int core_mask) @@ -316,6 +321,7 @@ int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev, return ret; } +EXPORT_SYMBOL_NS(hda_dsp_core_reset_power_down, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev) { @@ -334,6 +340,7 @@ void hda_dsp_ipc_int_enable(struct snd_sof_dev *sdev) snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); } +EXPORT_SYMBOL_NS(hda_dsp_ipc_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev) { @@ -351,6 +358,7 @@ void hda_dsp_ipc_int_disable(struct snd_sof_dev *sdev) snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, chip->ipc_ctl, HDA_DSP_REG_HIPCCTL_BUSY | HDA_DSP_REG_HIPCCTL_DONE, 0); } +EXPORT_SYMBOL_NS(hda_dsp_ipc_int_disable, SND_SOC_SOF_INTEL_HDA_COMMON); static int hda_dsp_wait_d0i3c_done(struct snd_sof_dev *sdev) { @@ -634,6 +642,7 @@ int hda_dsp_set_power_state_ipc3(struct snd_sof_dev *sdev, return hda_dsp_set_power_state(sdev, target_state); } +EXPORT_SYMBOL_NS(hda_dsp_set_power_state_ipc3, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_set_power_state_ipc4(struct snd_sof_dev *sdev, const struct sof_dsp_power_state *target_state) @@ -645,6 +654,7 @@ int hda_dsp_set_power_state_ipc4(struct snd_sof_dev *sdev, return hda_dsp_set_power_state(sdev, target_state); } +EXPORT_SYMBOL_NS(hda_dsp_set_power_state_ipc4, SND_SOC_SOF_INTEL_HDA_COMMON); /* * Audio DSP states may transform as below:- @@ -853,6 +863,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) return snd_sof_dsp_set_power_state(sdev, &target_state); } +EXPORT_SYMBOL_NS(hda_dsp_resume, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) { @@ -868,6 +879,7 @@ int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) return snd_sof_dsp_set_power_state(sdev, &target_state); } +EXPORT_SYMBOL_NS(hda_dsp_runtime_resume, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) { @@ -1034,12 +1046,14 @@ int hda_dsp_shutdown_dma_flush(struct snd_sof_dev *sdev) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_shutdown_dma_flush, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_shutdown(struct snd_sof_dev *sdev) { sdev->system_suspend_target = SOF_SUSPEND_S3; return snd_sof_suspend(sdev->dev); } +EXPORT_SYMBOL_NS(hda_dsp_shutdown, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) { @@ -1118,6 +1132,7 @@ int hda_dsp_core_get(struct snd_sof_dev *sdev, int core) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_core_get, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev) { @@ -1126,3 +1141,4 @@ int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev) return 0; } +EXPORT_SYMBOL_NS(hda_dsp_disable_interrupts, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index a838dddb1d327..6919729cef52b 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -20,6 +20,10 @@ #include "../ops.h" #include "hda.h" +EXPORT_TRACEPOINT_SYMBOL(sof_intel_ipc_firmware_initiated); +EXPORT_TRACEPOINT_SYMBOL(sof_intel_ipc_firmware_response); +EXPORT_TRACEPOINT_SYMBOL(sof_intel_hda_irq_ipc_check); + static void hda_dsp_ipc_host_done(struct snd_sof_dev *sdev) { /* @@ -66,6 +70,7 @@ int hda_dsp_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) return 0; } +EXPORT_SYMBOL_NS(hda_dsp_ipc_send_msg, SND_SOC_SOF_INTEL_HDA_COMMON); static inline bool hda_dsp_ipc4_pm_msg(u32 primary) { @@ -92,6 +97,7 @@ void hda_dsp_ipc4_schedule_d0i3_work(struct sof_intel_hda_dev *hdev, mod_delayed_work(system_wq, &hdev->d0i3_work, msecs_to_jiffies(SOF_HDA_D0I3_WORK_DELAY_MS)); } +EXPORT_SYMBOL_NS(hda_dsp_ipc4_schedule_d0i3_work, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_ipc4_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) { @@ -118,6 +124,7 @@ int hda_dsp_ipc4_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) return 0; } +EXPORT_SYMBOL_NS(hda_dsp_ipc4_send_msg, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) { @@ -153,6 +160,7 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) snd_sof_ipc_get_reply(sdev); } } +EXPORT_SYMBOL_NS(hda_dsp_ipc_get_reply, SND_SOC_SOF_INTEL_HDA_COMMON); irqreturn_t hda_dsp_ipc4_irq_thread(int irq, void *context) { @@ -235,6 +243,7 @@ irqreturn_t hda_dsp_ipc4_irq_thread(int irq, void *context) return IRQ_HANDLED; } +EXPORT_SYMBOL_NS(hda_dsp_ipc4_irq_thread, SND_SOC_SOF_INTEL_HDA_COMMON); /* IPC handler thread */ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) @@ -347,6 +356,7 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) return IRQ_HANDLED; } +EXPORT_SYMBOL_NS(hda_dsp_ipc_irq_thread, SND_SOC_SOF_INTEL_HDA_COMMON); /* Check if an IPC IRQ occurred */ bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev) @@ -380,6 +390,7 @@ bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev) out: return ret; } +EXPORT_SYMBOL_NS(hda_dsp_check_ipc_irq, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev) { diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index d0474522ea760..34c18275c949a 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -219,6 +219,7 @@ int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) kfree(dump_msg); return ret; } +EXPORT_SYMBOL_NS(cl_dsp_init, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_cl_trigger(struct device *dev, struct hdac_ext_stream *hext_stream, int cmd) { @@ -393,6 +394,7 @@ int hda_dsp_cl_boot_firmware_iccmax(struct snd_sof_dev *sdev) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_cl_boot_firmware_iccmax, SND_SOC_SOF_INTEL_CNL); static int hda_dsp_boot_imr(struct snd_sof_dev *sdev) { @@ -546,6 +548,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_cl_boot_firmware, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, struct sof_ipc4_fw_library *fw_lib, bool reload) @@ -649,6 +652,7 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, return ret; } +EXPORT_SYMBOL_NS(hda_dsp_ipc4_load_library, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_ext_man_get_cavs_config_data(struct snd_sof_dev *sdev, const struct sof_ext_man_elem_header *hdr) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 93f316300043c..e4ee2918614e6 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -223,6 +223,7 @@ void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable) HDA_DSP_REG_ADSPIC2_SNDW, enable ? HDA_DSP_REG_ADSPIC2_SNDW : 0); } +EXPORT_SYMBOL_NS(hda_common_enable_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { @@ -236,6 +237,7 @@ void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) if (chip && chip->enable_sdw_irq) chip->enable_sdw_irq(sdev, enable); } +EXPORT_SYMBOL_NS(hda_sdw_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON); static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) { @@ -350,6 +352,7 @@ int hda_sdw_check_lcount_common(struct snd_sof_dev *sdev) return 0; } +EXPORT_SYMBOL_NS(hda_sdw_check_lcount_common, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev) { @@ -375,6 +378,7 @@ int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev) return 0; } +EXPORT_SYMBOL_NS(hda_sdw_check_lcount_ext, SND_SOC_SOF_INTEL_HDA_COMMON); static int hda_sdw_check_lcount(struct snd_sof_dev *sdev) { @@ -407,6 +411,7 @@ int hda_sdw_startup(struct snd_sof_dev *sdev) return sdw_intel_startup(hdev->sdw); } +EXPORT_SYMBOL_NS(hda_sdw_startup, SND_SOC_SOF_INTEL_HDA_COMMON); static int hda_sdw_exit(struct snd_sof_dev *sdev) { @@ -448,6 +453,7 @@ bool hda_common_check_sdw_irq(struct snd_sof_dev *sdev) out: return ret; } +EXPORT_SYMBOL_NS(hda_common_check_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON); static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) { @@ -481,6 +487,7 @@ bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev) return false; } +EXPORT_SYMBOL_NS(hda_sdw_check_wakeen_irq_common, SND_SOC_SOF_INTEL_HDA_COMMON); static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) { @@ -583,6 +590,7 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) /* re-enable clock gating and power gating */ return hda_dsp_ctrl_clock_power_gating(sdev, true); } +EXPORT_SYMBOL_NS(hda_dsp_post_fw_run, SND_SOC_SOF_INTEL_HDA_COMMON); /* * Debug @@ -826,6 +834,7 @@ void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level) dev_printk(level, sdev->dev, "error code: %#x (%s)\n", error_code, error_text); } +EXPORT_SYMBOL_NS(hda_dsp_get_state, SND_SOC_SOF_INTEL_HDA_COMMON); static void hda_dsp_get_registers(struct snd_sof_dev *sdev, struct sof_ipc_dsp_oops_xtensa *xoops, @@ -911,6 +920,7 @@ void hda_ipc4_dsp_dump(struct snd_sof_dev *sdev, u32 flags) else hda_dsp_dump_ext_rom_status(sdev, level, flags); } +EXPORT_SYMBOL_NS(hda_ipc4_dsp_dump, SND_SOC_SOF_INTEL_HDA_COMMON); static bool hda_check_ipc_irq(struct snd_sof_dev *sdev) { @@ -942,6 +952,7 @@ void hda_ipc_irq_dump(struct snd_sof_dev *sdev) intsts, intctl, rirbsts); dev_err(sdev->dev, "dsp irq ppsts 0x%8.8x adspis 0x%8.8x\n", ppsts, adspis); } +EXPORT_SYMBOL_NS(hda_ipc_irq_dump, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_ipc_dump(struct snd_sof_dev *sdev) { @@ -961,6 +972,7 @@ void hda_ipc_dump(struct snd_sof_dev *sdev) dev_err(sdev->dev, "host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n", hipcie, hipct, hipcctl); } +EXPORT_SYMBOL_NS(hda_ipc_dump, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_ipc4_dump(struct snd_sof_dev *sdev) { @@ -979,6 +991,7 @@ void hda_ipc4_dump(struct snd_sof_dev *sdev) dev_err(sdev->dev, "Host IPC initiator: %#x|%#x, target: %#x|%#x, ctl: %#x\n", hipci, hipcie, hipct, hipcte, hipcctl); } +EXPORT_SYMBOL_NS(hda_ipc4_dump, SND_SOC_SOF_INTEL_HDA_COMMON); bool hda_ipc4_tx_is_busy(struct snd_sof_dev *sdev) { @@ -990,6 +1003,7 @@ bool hda_ipc4_tx_is_busy(struct snd_sof_dev *sdev) return !!(val & chip->ipc_req_mask); } +EXPORT_SYMBOL_NS(hda_ipc4_tx_is_busy, SND_SOC_SOF_INTEL_HDA_COMMON); static int hda_init(struct snd_sof_dev *sdev) { @@ -1354,6 +1368,7 @@ int hda_dsp_probe_early(struct snd_sof_dev *sdev) err: return ret; } +EXPORT_SYMBOL_NS(hda_dsp_probe_early, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_probe(struct snd_sof_dev *sdev) { @@ -1510,6 +1525,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_probe, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_remove(struct snd_sof_dev *sdev) { @@ -1563,6 +1579,7 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) if (!sdev->dspless_mode_selected) iounmap(sdev->bar[HDA_DSP_BAR]); } +EXPORT_SYMBOL_NS(hda_dsp_remove, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_remove_late(struct snd_sof_dev *sdev) { @@ -1578,6 +1595,7 @@ int hda_power_down_dsp(struct snd_sof_dev *sdev) return hda_dsp_core_reset_power_down(sdev, chip->host_managed_cores_mask); } +EXPORT_SYMBOL_NS(hda_power_down_dsp, SND_SOC_SOF_INTEL_HDA_COMMON); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) static void hda_generic_machine_select(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c index 0406985919921..115866225e053 100644 --- a/sound/soc/sof/intel/icl.c +++ b/sound/soc/sof/intel/icl.c @@ -97,7 +97,6 @@ static int icl_dsp_post_fw_run(struct snd_sof_dev *sdev) /* Icelake ops */ struct snd_sof_dsp_ops sof_icl_ops; -EXPORT_SYMBOL_NS(sof_icl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); int sof_icl_ops_init(struct snd_sof_dev *sdev) { @@ -166,7 +165,6 @@ int sof_icl_ops_init(struct snd_sof_dev *sdev) return 0; }; -EXPORT_SYMBOL_NS(sof_icl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON); const struct sof_intel_dsp_desc icl_chip_info = { /* Icelake */ @@ -195,4 +193,3 @@ const struct sof_intel_dsp_desc icl_chip_info = { .disable_interrupts = hda_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_CAVS_2_0, }; -EXPORT_SYMBOL_NS(icl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 82c975823ab32..ca0cb91d2b2c3 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -22,7 +22,6 @@ /* LunarLake ops */ struct snd_sof_dsp_ops sof_lnl_ops; -EXPORT_SYMBOL_NS(sof_lnl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = { {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS}, @@ -182,7 +181,6 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) return 0; }; -EXPORT_SYMBOL_NS(sof_lnl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON); /* Check if an SDW IRQ occurred */ static bool lnl_dsp_check_sdw_irq(struct snd_sof_dev *sdev) @@ -246,4 +244,3 @@ const struct sof_intel_dsp_desc lnl_chip_info = { .disable_interrupts = lnl_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_ACE_2_0, }; -EXPORT_SYMBOL_NS(lnl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 5d3b2d4be78ae..13f780f33c223 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -77,6 +77,7 @@ bool mtl_dsp_check_ipc_irq(struct snd_sof_dev *sdev) return false; } +EXPORT_SYMBOL_NS(mtl_dsp_check_ipc_irq, SND_SOC_SOF_INTEL_MTL); /* Check if an SDW IRQ occurred */ static bool mtl_dsp_check_sdw_irq(struct snd_sof_dev *sdev) @@ -120,6 +121,7 @@ int mtl_ipc_send_msg(struct snd_sof_dev *sdev, struct snd_sof_ipc_msg *msg) return 0; } +EXPORT_SYMBOL_NS(mtl_ipc_send_msg, SND_SOC_SOF_INTEL_MTL); void mtl_enable_ipc_interrupts(struct snd_sof_dev *sdev) { @@ -147,6 +149,7 @@ void mtl_disable_ipc_interrupts(struct snd_sof_dev *sdev) snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, chip->ipc_ctl, MTL_DSP_REG_HFIPCXCTL_BUSY | MTL_DSP_REG_HFIPCXCTL_DONE, 0); } +EXPORT_SYMBOL_NS(mtl_disable_ipc_interrupts, SND_SOC_SOF_INTEL_MTL); static void mtl_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable) { @@ -231,6 +234,7 @@ int mtl_enable_interrupts(struct snd_sof_dev *sdev, bool enable) return ret; } +EXPORT_SYMBOL_NS(mtl_enable_interrupts, SND_SOC_SOF_INTEL_MTL); /* pre fw run operations */ int mtl_dsp_pre_fw_run(struct snd_sof_dev *sdev) @@ -281,6 +285,7 @@ int mtl_dsp_pre_fw_run(struct snd_sof_dev *sdev) return ret; } +EXPORT_SYMBOL_NS(mtl_dsp_pre_fw_run, SND_SOC_SOF_INTEL_MTL); int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev) { @@ -307,6 +312,7 @@ int mtl_dsp_post_fw_run(struct snd_sof_dev *sdev) hda_sdw_int_enable(sdev, true); return 0; } +EXPORT_SYMBOL_NS(mtl_dsp_post_fw_run, SND_SOC_SOF_INTEL_MTL); void mtl_dsp_dump(struct snd_sof_dev *sdev, u32 flags) { @@ -324,6 +330,7 @@ void mtl_dsp_dump(struct snd_sof_dev *sdev, u32 flags) sof_ipc4_intel_dump_telemetry_state(sdev, flags); } +EXPORT_SYMBOL_NS(mtl_dsp_dump, SND_SOC_SOF_INTEL_MTL); static bool mtl_dsp_primary_core_is_enabled(struct snd_sof_dev *sdev) { @@ -434,6 +441,7 @@ int mtl_power_down_dsp(struct snd_sof_dev *sdev) (dsphfdsscs & cpa) == 0, HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_RESET_TIMEOUT_US); } +EXPORT_SYMBOL_NS(mtl_power_down_dsp, SND_SOC_SOF_INTEL_MTL); int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) { @@ -536,6 +544,7 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) kfree(dump_msg); return ret; } +EXPORT_SYMBOL_NS(mtl_dsp_cl_init, SND_SOC_SOF_INTEL_MTL); irqreturn_t mtl_ipc_irq_thread(int irq, void *context) { @@ -619,16 +628,19 @@ irqreturn_t mtl_ipc_irq_thread(int irq, void *context) return IRQ_HANDLED; } +EXPORT_SYMBOL_NS(mtl_ipc_irq_thread, SND_SOC_SOF_INTEL_MTL); int mtl_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev) { return MTL_DSP_MBOX_UPLINK_OFFSET; } +EXPORT_SYMBOL_NS(mtl_dsp_ipc_get_mailbox_offset, SND_SOC_SOF_INTEL_MTL); int mtl_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id) { return MTL_SRAM_WINDOW_OFFSET(id); } +EXPORT_SYMBOL_NS(mtl_dsp_ipc_get_window_offset, SND_SOC_SOF_INTEL_MTL); void mtl_ipc_dump(struct snd_sof_dev *sdev) { @@ -646,6 +658,7 @@ void mtl_ipc_dump(struct snd_sof_dev *sdev) "Host IPC initiator: %#x|%#x|%#x, target: %#x|%#x|%#x, ctl: %#x\n", hipcidr, hipcidd, hipcida, hipctdr, hipctdd, hipctda, hipcctl); } +EXPORT_SYMBOL_NS(mtl_ipc_dump, SND_SOC_SOF_INTEL_MTL); static int mtl_dsp_disable_interrupts(struct snd_sof_dev *sdev) { @@ -666,6 +679,7 @@ int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core) return 0; } +EXPORT_SYMBOL_NS(mtl_dsp_core_get, SND_SOC_SOF_INTEL_MTL); int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core) { @@ -683,10 +697,10 @@ int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core) return 0; } +EXPORT_SYMBOL_NS(mtl_dsp_core_put, SND_SOC_SOF_INTEL_MTL); /* Meteorlake ops */ struct snd_sof_dsp_ops sof_mtl_ops; -EXPORT_SYMBOL_NS(sof_mtl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); int sof_mtl_ops_init(struct snd_sof_dev *sdev) { @@ -744,7 +758,6 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev) return 0; }; -EXPORT_SYMBOL_NS(sof_mtl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON); const struct sof_intel_dsp_desc mtl_chip_info = { .cores_num = 3, @@ -772,7 +785,6 @@ const struct sof_intel_dsp_desc mtl_chip_info = { .disable_interrupts = mtl_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_ACE_1_0, }; -EXPORT_SYMBOL_NS(mtl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); const struct sof_intel_dsp_desc arl_s_chip_info = { .cores_num = 2, @@ -800,4 +812,3 @@ const struct sof_intel_dsp_desc arl_s_chip_info = { .disable_interrupts = mtl_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_ACE_1_0, }; -EXPORT_SYMBOL_NS(arl_s_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c index b99c7c9aad7dd..70689b8b431b3 100644 --- a/sound/soc/sof/intel/pci-icl.c +++ b/sound/soc/sof/intel/pci-icl.c @@ -109,4 +109,5 @@ module_pci_driver(snd_sof_pci_intel_icl_driver); MODULE_LICENSE("Dual BSD/GPL"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-lnl.c b/sound/soc/sof/intel/pci-lnl.c index b14e508f1f315..38ae649d1defa 100644 --- a/sound/soc/sof/intel/pci-lnl.c +++ b/sound/soc/sof/intel/pci-lnl.c @@ -71,4 +71,6 @@ module_pci_driver(snd_sof_pci_intel_lnl_driver); MODULE_LICENSE("Dual BSD/GPL"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_MTL); +MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index a361ee9d1107f..d7ecb892550f2 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -318,4 +318,5 @@ module_pci_driver(snd_sof_pci_intel_tgl_driver); MODULE_LICENSE("Dual BSD/GPL"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/telemetry.c b/sound/soc/sof/intel/telemetry.c index 1a3b5c28a6f04..042360f0d03f2 100644 --- a/sound/soc/sof/intel/telemetry.c +++ b/sound/soc/sof/intel/telemetry.c @@ -93,3 +93,4 @@ void sof_ipc4_intel_dump_telemetry_state(struct snd_sof_dev *sdev, u32 flags) free_telemetry_data: kfree(telemetry_data); } +EXPORT_SYMBOL_NS(sof_ipc4_intel_dump_telemetry_state, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index 6cda47a78118a..6acc659238342 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -63,7 +63,6 @@ static int tgl_dsp_core_put(struct snd_sof_dev *sdev, int core) /* Tigerlake ops */ struct snd_sof_dsp_ops sof_tgl_ops; -EXPORT_SYMBOL_NS(sof_tgl_ops, SND_SOC_SOF_INTEL_HDA_COMMON); int sof_tgl_ops_init(struct snd_sof_dev *sdev) { @@ -135,7 +134,6 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev) return 0; }; -EXPORT_SYMBOL_NS(sof_tgl_ops_init, SND_SOC_SOF_INTEL_HDA_COMMON); const struct sof_intel_dsp_desc tgl_chip_info = { /* Tigerlake , Alderlake */ @@ -164,7 +162,6 @@ const struct sof_intel_dsp_desc tgl_chip_info = { .disable_interrupts = hda_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_CAVS_2_5, }; -EXPORT_SYMBOL_NS(tgl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); const struct sof_intel_dsp_desc tglh_chip_info = { /* Tigerlake-H */ @@ -193,7 +190,6 @@ const struct sof_intel_dsp_desc tglh_chip_info = { .disable_interrupts = hda_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_CAVS_2_5, }; -EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); const struct sof_intel_dsp_desc ehl_chip_info = { /* Elkhartlake */ @@ -222,7 +218,6 @@ const struct sof_intel_dsp_desc ehl_chip_info = { .disable_interrupts = hda_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_CAVS_2_5, }; -EXPORT_SYMBOL_NS(ehl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); const struct sof_intel_dsp_desc adls_chip_info = { /* Alderlake-S */ @@ -251,4 +246,3 @@ const struct sof_intel_dsp_desc adls_chip_info = { .disable_interrupts = hda_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_CAVS_2_5, }; -EXPORT_SYMBOL_NS(adls_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); From 0bfbe91a2dbba31d41add146ab173721dee85ab3 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:52:18 -0500 Subject: [PATCH 453/603] ASoC: SOF: Intel: move common code from hda.c To avoid circular dependencies when moving hda.c to a separate module, we need to move the common code to hda-ipc.c and hda-dsp.c No functionality change, just code move. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20240503135221.229202-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 462 +++++++++++++++++++++++++ sound/soc/sof/intel/hda-ipc.c | 99 ++++++ sound/soc/sof/intel/hda-stream.c | 4 + sound/soc/sof/intel/hda.c | 561 ------------------------------- sound/soc/sof/intel/hda.h | 10 + 5 files changed, 575 insertions(+), 561 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 32f0340746686..974ac1cd8c6e6 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -20,11 +20,21 @@ #include #include #include +#include #include "../sof-audio.h" #include "../ops.h" #include "hda.h" +#include "mtl.h" #include "hda-ipc.h" +#define EXCEPT_MAX_HDR_SIZE 0x400 +#define HDA_EXT_ROM_STATUS_SIZE 8 + +struct hda_dsp_msg_code { + u32 code; + const char *text; +}; + static bool hda_enable_trace_D0I3_S0; #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG) module_param_named(enable_trace_D0I3_S0, hda_enable_trace_D0I3_S0, bool, 0444); @@ -32,6 +42,83 @@ MODULE_PARM_DESC(enable_trace_D0I3_S0, "SOF HDA enable trace when the DSP is in D0I3 in S0"); #endif +static void hda_get_interfaces(struct snd_sof_dev *sdev, u32 *interface_mask) +{ + const struct sof_intel_dsp_desc *chip; + + chip = get_chip_info(sdev->pdata); + switch (chip->hw_ip_version) { + case SOF_INTEL_TANGIER: + case SOF_INTEL_BAYTRAIL: + case SOF_INTEL_BROADWELL: + interface_mask[SOF_DAI_DSP_ACCESS] = BIT(SOF_DAI_INTEL_SSP); + break; + case SOF_INTEL_CAVS_1_5: + case SOF_INTEL_CAVS_1_5_PLUS: + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | BIT(SOF_DAI_INTEL_HDA); + interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); + break; + case SOF_INTEL_CAVS_1_8: + case SOF_INTEL_CAVS_2_0: + case SOF_INTEL_CAVS_2_5: + case SOF_INTEL_ACE_1_0: + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | + BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); + interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); + break; + case SOF_INTEL_ACE_2_0: + interface_mask[SOF_DAI_DSP_ACCESS] = + BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | + BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); + /* all interfaces accessible without DSP */ + interface_mask[SOF_DAI_HOST_ACCESS] = + interface_mask[SOF_DAI_DSP_ACCESS]; + break; + default: + break; + } +} + +u32 hda_get_interface_mask(struct snd_sof_dev *sdev) +{ + u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 }; + + hda_get_interfaces(sdev, interface_mask); + + return interface_mask[sdev->dspless_mode_selected]; +} + +bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type) +{ + u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 }; + const struct sof_intel_dsp_desc *chip; + + if (sdev->dspless_mode_selected) + return false; + + hda_get_interfaces(sdev, interface_mask); + + if (!(interface_mask[SOF_DAI_DSP_ACCESS] & BIT(dai_type))) + return false; + + if (dai_type == SOF_DAI_INTEL_HDA) + return true; + + switch (dai_type) { + case SOF_DAI_INTEL_SSP: + case SOF_DAI_INTEL_DMIC: + case SOF_DAI_INTEL_ALH: + chip = get_chip_info(sdev->pdata); + if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) + return false; + return true; + default: + return false; + } +} + /* * DSP Core control. */ @@ -1134,6 +1221,98 @@ int hda_dsp_core_get(struct snd_sof_dev *sdev, int core) } EXPORT_SYMBOL_NS(hda_dsp_core_get, SND_SOC_SOF_INTEL_HDA_COMMON); +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + if (!hdev->sdw) + return; + + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC2, + HDA_DSP_REG_ADSPIC2_SNDW, + enable ? HDA_DSP_REG_ADSPIC2_SNDW : 0); +} +EXPORT_SYMBOL_NS(hda_common_enable_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON); + +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ + u32 interface_mask = hda_get_interface_mask(sdev); + const struct sof_intel_dsp_desc *chip; + + if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH))) + return; + + chip = get_chip_info(sdev->pdata); + if (chip && chip->enable_sdw_irq) + chip->enable_sdw_irq(sdev, enable); +} +EXPORT_SYMBOL_NS(hda_sdw_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON); + +int hda_sdw_check_lcount_common(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + struct sdw_intel_ctx *ctx; + u32 caps; + + hdev = sdev->pdata->hw_pdata; + ctx = hdev->sdw; + + caps = snd_sof_dsp_read(sdev, HDA_DSP_BAR, ctx->shim_base + SDW_SHIM_LCAP); + caps &= SDW_SHIM_LCAP_LCOUNT_MASK; + + /* Check HW supported vs property value */ + if (caps < ctx->count) { + dev_err(sdev->dev, + "%s: BIOS master count %d is larger than hardware capabilities %d\n", + __func__, ctx->count, caps); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_NS(hda_sdw_check_lcount_common, SND_SOC_SOF_INTEL_HDA_COMMON); + +int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + struct sdw_intel_ctx *ctx; + struct hdac_bus *bus; + u32 slcount; + + bus = sof_to_bus(sdev); + + hdev = sdev->pdata->hw_pdata; + ctx = hdev->sdw; + + slcount = hdac_bus_eml_get_count(bus, true, AZX_REG_ML_LEPTR_ID_SDW); + + /* Check HW supported vs property value */ + if (slcount < ctx->count) { + dev_err(sdev->dev, + "%s: BIOS master count %d is larger than hardware capabilities %d\n", + __func__, ctx->count, slcount); + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_NS(hda_sdw_check_lcount_ext, SND_SOC_SOF_INTEL_HDA_COMMON); + +int hda_sdw_check_lcount(struct snd_sof_dev *sdev) +{ + const struct sof_intel_dsp_desc *chip; + + chip = get_chip_info(sdev->pdata); + if (chip && chip->read_sdw_lcount) + return chip->read_sdw_lcount(sdev); + + return 0; +} +#endif + int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev) { hda_sdw_int_enable(sdev, false); @@ -1142,3 +1321,286 @@ int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev) return 0; } EXPORT_SYMBOL_NS(hda_dsp_disable_interrupts, SND_SOC_SOF_INTEL_HDA_COMMON); + +static const struct hda_dsp_msg_code hda_dsp_rom_fw_error_texts[] = { + {HDA_DSP_ROM_CSE_ERROR, "error: cse error"}, + {HDA_DSP_ROM_CSE_WRONG_RESPONSE, "error: cse wrong response"}, + {HDA_DSP_ROM_IMR_TO_SMALL, "error: IMR too small"}, + {HDA_DSP_ROM_BASE_FW_NOT_FOUND, "error: base fw not found"}, + {HDA_DSP_ROM_CSE_VALIDATION_FAILED, "error: signature verification failed"}, + {HDA_DSP_ROM_IPC_FATAL_ERROR, "error: ipc fatal error"}, + {HDA_DSP_ROM_L2_CACHE_ERROR, "error: L2 cache error"}, + {HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL, "error: load offset too small"}, + {HDA_DSP_ROM_API_PTR_INVALID, "error: API ptr invalid"}, + {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatible"}, + {HDA_DSP_ROM_UNHANDLED_INTERRUPT, "error: unhandled interrupt"}, + {HDA_DSP_ROM_MEMORY_HOLE_ECC, "error: ECC memory hole"}, + {HDA_DSP_ROM_KERNEL_EXCEPTION, "error: kernel exception"}, + {HDA_DSP_ROM_USER_EXCEPTION, "error: user exception"}, + {HDA_DSP_ROM_UNEXPECTED_RESET, "error: unexpected reset"}, + {HDA_DSP_ROM_NULL_FW_ENTRY, "error: null FW entry point"}, +}; + +#define FSR_ROM_STATE_ENTRY(state) {FSR_STATE_ROM_##state, #state} +static const struct hda_dsp_msg_code cavs_fsr_rom_state_names[] = { + FSR_ROM_STATE_ENTRY(INIT), + FSR_ROM_STATE_ENTRY(INIT_DONE), + FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED), + FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED), + FSR_ROM_STATE_ENTRY(FW_FW_LOADED), + FSR_ROM_STATE_ENTRY(FW_ENTERED), + FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK), + FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET), + FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT), + FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT_DONE), + /* CSE states */ + FSR_ROM_STATE_ENTRY(CSE_IMR_REQUEST), + FSR_ROM_STATE_ENTRY(CSE_IMR_GRANTED), + FSR_ROM_STATE_ENTRY(CSE_VALIDATE_IMAGE_REQUEST), + FSR_ROM_STATE_ENTRY(CSE_IMAGE_VALIDATED), + FSR_ROM_STATE_ENTRY(CSE_IPC_IFACE_INIT), + FSR_ROM_STATE_ENTRY(CSE_IPC_RESET_PHASE_1), + FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL_ENTRY), + FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL), + FSR_ROM_STATE_ENTRY(CSE_IPC_DOWN), +}; + +static const struct hda_dsp_msg_code ace_fsr_rom_state_names[] = { + FSR_ROM_STATE_ENTRY(INIT), + FSR_ROM_STATE_ENTRY(INIT_DONE), + FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED), + FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED), + FSR_ROM_STATE_ENTRY(FW_FW_LOADED), + FSR_ROM_STATE_ENTRY(FW_ENTERED), + FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK), + FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET), + FSR_ROM_STATE_ENTRY(RESET_VECTOR_DONE), + FSR_ROM_STATE_ENTRY(PURGE_BOOT), + FSR_ROM_STATE_ENTRY(RESTORE_BOOT), + FSR_ROM_STATE_ENTRY(FW_ENTRY_POINT), + FSR_ROM_STATE_ENTRY(VALIDATE_PUB_KEY), + FSR_ROM_STATE_ENTRY(POWER_DOWN_HPSRAM), + FSR_ROM_STATE_ENTRY(POWER_DOWN_ULPSRAM), + FSR_ROM_STATE_ENTRY(POWER_UP_ULPSRAM_STACK), + FSR_ROM_STATE_ENTRY(POWER_UP_HPSRAM_DMA), + FSR_ROM_STATE_ENTRY(BEFORE_EP_POINTER_READ), + FSR_ROM_STATE_ENTRY(VALIDATE_MANIFEST), + FSR_ROM_STATE_ENTRY(VALIDATE_FW_MODULE), + FSR_ROM_STATE_ENTRY(PROTECT_IMR_REGION), + FSR_ROM_STATE_ENTRY(PUSH_MODEL_ROUTINE), + FSR_ROM_STATE_ENTRY(PULL_MODEL_ROUTINE), + FSR_ROM_STATE_ENTRY(VALIDATE_PKG_DIR), + FSR_ROM_STATE_ENTRY(VALIDATE_CPD), + FSR_ROM_STATE_ENTRY(VALIDATE_CSS_MAN_HEADER), + FSR_ROM_STATE_ENTRY(VALIDATE_BLOB_SVN), + FSR_ROM_STATE_ENTRY(VERIFY_IFWI_PARTITION), + FSR_ROM_STATE_ENTRY(REMOVE_ACCESS_CONTROL), + FSR_ROM_STATE_ENTRY(AUTH_BYPASS), + FSR_ROM_STATE_ENTRY(AUTH_ENABLED), + FSR_ROM_STATE_ENTRY(INIT_DMA), + FSR_ROM_STATE_ENTRY(PURGE_FW_ENTRY), + FSR_ROM_STATE_ENTRY(PURGE_FW_END), + FSR_ROM_STATE_ENTRY(CLEAN_UP_BSS_DONE), + FSR_ROM_STATE_ENTRY(IMR_RESTORE_ENTRY), + FSR_ROM_STATE_ENTRY(IMR_RESTORE_END), + FSR_ROM_STATE_ENTRY(FW_MANIFEST_IN_DMA_BUFF), + FSR_ROM_STATE_ENTRY(LOAD_CSE_MAN_TO_IMR), + FSR_ROM_STATE_ENTRY(LOAD_FW_MAN_TO_IMR), + FSR_ROM_STATE_ENTRY(LOAD_FW_CODE_TO_IMR), + FSR_ROM_STATE_ENTRY(FW_LOADING_DONE), + FSR_ROM_STATE_ENTRY(FW_CODE_LOADED), + FSR_ROM_STATE_ENTRY(VERIFY_IMAGE_TYPE), + FSR_ROM_STATE_ENTRY(AUTH_API_INIT), + FSR_ROM_STATE_ENTRY(AUTH_API_PROC), + FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_BUSY), + FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_RESULT), + FSR_ROM_STATE_ENTRY(AUTH_API_CLEANUP), +}; + +#define FSR_BRINGUP_STATE_ENTRY(state) {FSR_STATE_BRINGUP_##state, #state} +static const struct hda_dsp_msg_code fsr_bringup_state_names[] = { + FSR_BRINGUP_STATE_ENTRY(INIT), + FSR_BRINGUP_STATE_ENTRY(INIT_DONE), + FSR_BRINGUP_STATE_ENTRY(HPSRAM_LOAD), + FSR_BRINGUP_STATE_ENTRY(UNPACK_START), + FSR_BRINGUP_STATE_ENTRY(IMR_RESTORE), + FSR_BRINGUP_STATE_ENTRY(FW_ENTERED), +}; + +#define FSR_WAIT_STATE_ENTRY(state) {FSR_WAIT_FOR_##state, #state} +static const struct hda_dsp_msg_code fsr_wait_state_names[] = { + FSR_WAIT_STATE_ENTRY(IPC_BUSY), + FSR_WAIT_STATE_ENTRY(IPC_DONE), + FSR_WAIT_STATE_ENTRY(CACHE_INVALIDATION), + FSR_WAIT_STATE_ENTRY(LP_SRAM_OFF), + FSR_WAIT_STATE_ENTRY(DMA_BUFFER_FULL), + FSR_WAIT_STATE_ENTRY(CSE_CSR), +}; + +#define FSR_MODULE_NAME_ENTRY(mod) [FSR_MOD_##mod] = #mod +static const char * const fsr_module_names[] = { + FSR_MODULE_NAME_ENTRY(ROM), + FSR_MODULE_NAME_ENTRY(ROM_BYP), + FSR_MODULE_NAME_ENTRY(BASE_FW), + FSR_MODULE_NAME_ENTRY(LP_BOOT), + FSR_MODULE_NAME_ENTRY(BRNGUP), + FSR_MODULE_NAME_ENTRY(ROM_EXT), +}; + +static const char * +hda_dsp_get_state_text(u32 code, const struct hda_dsp_msg_code *msg_code, + size_t array_size) +{ + int i; + + for (i = 0; i < array_size; i++) { + if (code == msg_code[i].code) + return msg_code[i].text; + } + + return NULL; +} + +void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level) +{ + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); + const char *state_text, *error_text, *module_text; + u32 fsr, state, wait_state, module, error_code; + + fsr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg); + state = FSR_TO_STATE_CODE(fsr); + wait_state = FSR_TO_WAIT_STATE_CODE(fsr); + module = FSR_TO_MODULE_CODE(fsr); + + if (module > FSR_MOD_ROM_EXT) + module_text = "unknown"; + else + module_text = fsr_module_names[module]; + + if (module == FSR_MOD_BRNGUP) { + state_text = hda_dsp_get_state_text(state, fsr_bringup_state_names, + ARRAY_SIZE(fsr_bringup_state_names)); + } else { + if (chip->hw_ip_version < SOF_INTEL_ACE_1_0) + state_text = hda_dsp_get_state_text(state, + cavs_fsr_rom_state_names, + ARRAY_SIZE(cavs_fsr_rom_state_names)); + else + state_text = hda_dsp_get_state_text(state, + ace_fsr_rom_state_names, + ARRAY_SIZE(ace_fsr_rom_state_names)); + } + + /* not for us, must be generic sof message */ + if (!state_text) { + dev_printk(level, sdev->dev, "%#010x: unknown ROM status value\n", fsr); + return; + } + + if (wait_state) { + const char *wait_state_text; + + wait_state_text = hda_dsp_get_state_text(wait_state, fsr_wait_state_names, + ARRAY_SIZE(fsr_wait_state_names)); + if (!wait_state_text) + wait_state_text = "unknown"; + + dev_printk(level, sdev->dev, + "%#010x: module: %s, state: %s, waiting for: %s, %s\n", + fsr, module_text, state_text, wait_state_text, + fsr & FSR_HALTED ? "not running" : "running"); + } else { + dev_printk(level, sdev->dev, "%#010x: module: %s, state: %s, %s\n", + fsr, module_text, state_text, + fsr & FSR_HALTED ? "not running" : "running"); + } + + error_code = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + 4); + if (!error_code) + return; + + error_text = hda_dsp_get_state_text(error_code, hda_dsp_rom_fw_error_texts, + ARRAY_SIZE(hda_dsp_rom_fw_error_texts)); + if (!error_text) + error_text = "unknown"; + + if (state == FSR_STATE_FW_ENTERED) + dev_printk(level, sdev->dev, "status code: %#x (%s)\n", error_code, + error_text); + else + dev_printk(level, sdev->dev, "error code: %#x (%s)\n", error_code, + error_text); +} +EXPORT_SYMBOL_NS(hda_dsp_get_state, SND_SOC_SOF_INTEL_HDA_COMMON); + +static void hda_dsp_get_registers(struct snd_sof_dev *sdev, + struct sof_ipc_dsp_oops_xtensa *xoops, + struct sof_ipc_panic_info *panic_info, + u32 *stack, size_t stack_words) +{ + u32 offset = sdev->dsp_oops_offset; + + /* first read registers */ + sof_mailbox_read(sdev, offset, xoops, sizeof(*xoops)); + + /* note: variable AR register array is not read */ + + /* then get panic info */ + if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) { + dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n", + xoops->arch_hdr.totalsize); + return; + } + offset += xoops->arch_hdr.totalsize; + sof_block_read(sdev, sdev->mmio_bar, offset, + panic_info, sizeof(*panic_info)); + + /* then get the stack */ + offset += sizeof(*panic_info); + sof_block_read(sdev, sdev->mmio_bar, offset, stack, + stack_words * sizeof(u32)); +} + +/* dump the first 8 dwords representing the extended ROM status */ +void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *level, + u32 flags) +{ + const struct sof_intel_dsp_desc *chip; + char msg[128]; + int len = 0; + u32 value; + int i; + + chip = get_chip_info(sdev->pdata); + for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { + value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4); + len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); + } + + dev_printk(level, sdev->dev, "extended rom status: %s", msg); + +} + +void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) +{ + char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR; + struct sof_ipc_dsp_oops_xtensa xoops; + struct sof_ipc_panic_info panic_info; + u32 stack[HDA_DSP_STACK_DUMP_SIZE]; + + /* print ROM/FW status */ + hda_dsp_get_state(sdev, level); + + /* The firmware register dump only available with IPC3 */ + if (flags & SOF_DBG_DUMP_REGS && sdev->pdata->ipc_type == SOF_IPC_TYPE_3) { + u32 status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_STATUS); + u32 panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); + + hda_dsp_get_registers(sdev, &xoops, &panic_info, stack, + HDA_DSP_STACK_DUMP_SIZE); + sof_print_oops_and_stack(sdev, level, status, panic, &xoops, + &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE); + } else { + hda_dsp_dump_ext_rom_status(sdev, level, flags); + } +} diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 6919729cef52b..2252c9198cc02 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -15,10 +15,12 @@ * Hardware interface for generic Intel audio DSP HDA IP */ +#include #include #include #include "../ops.h" #include "hda.h" +#include "telemetry.h" EXPORT_TRACEPOINT_SYMBOL(sof_intel_ipc_firmware_initiated); EXPORT_TRACEPOINT_SYMBOL(sof_intel_ipc_firmware_response); @@ -450,3 +452,100 @@ int hda_set_stream_data_offset(struct snd_sof_dev *sdev, return 0; } + +void hda_ipc4_dsp_dump(struct snd_sof_dev *sdev, u32 flags) +{ + char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR; + + /* print ROM/FW status */ + hda_dsp_get_state(sdev, level); + + if (flags & SOF_DBG_DUMP_REGS) + sof_ipc4_intel_dump_telemetry_state(sdev, flags); + else + hda_dsp_dump_ext_rom_status(sdev, level, flags); +} +EXPORT_SYMBOL_NS(hda_ipc4_dsp_dump, SND_SOC_SOF_INTEL_HDA_COMMON); + +bool hda_check_ipc_irq(struct snd_sof_dev *sdev) +{ + const struct sof_intel_dsp_desc *chip; + + chip = get_chip_info(sdev->pdata); + if (chip && chip->check_ipc_irq) + return chip->check_ipc_irq(sdev); + + return false; +} + +void hda_ipc_irq_dump(struct snd_sof_dev *sdev) +{ + u32 adspis; + u32 intsts; + u32 intctl; + u32 ppsts; + u8 rirbsts; + + /* read key IRQ stats and config registers */ + adspis = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS); + intsts = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS); + intctl = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL); + ppsts = snd_sof_dsp_read(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPSTS); + rirbsts = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, AZX_REG_RIRBSTS); + + dev_err(sdev->dev, "hda irq intsts 0x%8.8x intlctl 0x%8.8x rirb %2.2x\n", + intsts, intctl, rirbsts); + dev_err(sdev->dev, "dsp irq ppsts 0x%8.8x adspis 0x%8.8x\n", ppsts, adspis); +} +EXPORT_SYMBOL_NS(hda_ipc_irq_dump, SND_SOC_SOF_INTEL_HDA_COMMON); + +void hda_ipc_dump(struct snd_sof_dev *sdev) +{ + u32 hipcie; + u32 hipct; + u32 hipcctl; + + hda_ipc_irq_dump(sdev); + + /* read IPC status */ + hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE); + hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT); + hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL); + + /* dump the IPC regs */ + /* TODO: parse the raw msg */ + dev_err(sdev->dev, "host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n", + hipcie, hipct, hipcctl); +} +EXPORT_SYMBOL_NS(hda_ipc_dump, SND_SOC_SOF_INTEL_HDA_COMMON); + +void hda_ipc4_dump(struct snd_sof_dev *sdev) +{ + u32 hipci, hipcie, hipct, hipcte, hipcctl; + + hda_ipc_irq_dump(sdev); + + hipci = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCI); + hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE); + hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT); + hipcte = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCTE); + hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL); + + /* dump the IPC regs */ + /* TODO: parse the raw msg */ + dev_err(sdev->dev, "Host IPC initiator: %#x|%#x, target: %#x|%#x, ctl: %#x\n", + hipci, hipcie, hipct, hipcte, hipcctl); +} +EXPORT_SYMBOL_NS(hda_ipc4_dump, SND_SOC_SOF_INTEL_HDA_COMMON); + +bool hda_ipc4_tx_is_busy(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; + u32 val; + + val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->ipc_req); + + return !!(val & chip->ipc_req_mask); +} +EXPORT_SYMBOL_NS(hda_ipc4_tx_is_busy, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index c5afcda183593..4fef1964b5cdb 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -24,6 +24,10 @@ #include "../ipc4-priv.h" #include "hda.h" +int sof_hda_position_quirk = SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS; +module_param_named(position_quirk, sof_hda_position_quirk, int, 0444); +MODULE_PARM_DESC(position_quirk, "SOF HDaudio position quirk"); + #define HDA_LTRP_GB_VALUE_US 95 static inline const char *hda_hstream_direction_str(struct hdac_stream *hstream) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index e4ee2918614e6..a96926fb45b65 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -34,8 +34,6 @@ #include "../ops.h" #include "../ipc4-topology.h" #include "hda.h" -#include "telemetry.h" -#include "mtl.h" #define CREATE_TRACE_POINTS #include @@ -47,86 +45,6 @@ /* platform specific devices */ #include "shim.h" -#define EXCEPT_MAX_HDR_SIZE 0x400 -#define HDA_EXT_ROM_STATUS_SIZE 8 - -static void hda_get_interfaces(struct snd_sof_dev *sdev, u32 *interface_mask) -{ - const struct sof_intel_dsp_desc *chip; - - chip = get_chip_info(sdev->pdata); - switch (chip->hw_ip_version) { - case SOF_INTEL_TANGIER: - case SOF_INTEL_BAYTRAIL: - case SOF_INTEL_BROADWELL: - interface_mask[SOF_DAI_DSP_ACCESS] = BIT(SOF_DAI_INTEL_SSP); - break; - case SOF_INTEL_CAVS_1_5: - case SOF_INTEL_CAVS_1_5_PLUS: - interface_mask[SOF_DAI_DSP_ACCESS] = - BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | BIT(SOF_DAI_INTEL_HDA); - interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); - break; - case SOF_INTEL_CAVS_1_8: - case SOF_INTEL_CAVS_2_0: - case SOF_INTEL_CAVS_2_5: - case SOF_INTEL_ACE_1_0: - interface_mask[SOF_DAI_DSP_ACCESS] = - BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | - BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); - interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); - break; - case SOF_INTEL_ACE_2_0: - interface_mask[SOF_DAI_DSP_ACCESS] = - BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | - BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); - /* all interfaces accessible without DSP */ - interface_mask[SOF_DAI_HOST_ACCESS] = - interface_mask[SOF_DAI_DSP_ACCESS]; - break; - default: - break; - } -} - -static u32 hda_get_interface_mask(struct snd_sof_dev *sdev) -{ - u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 }; - - hda_get_interfaces(sdev, interface_mask); - - return interface_mask[sdev->dspless_mode_selected]; -} - -bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type) -{ - u32 interface_mask[SOF_DAI_ACCESS_NUM] = { 0 }; - const struct sof_intel_dsp_desc *chip; - - if (sdev->dspless_mode_selected) - return false; - - hda_get_interfaces(sdev, interface_mask); - - if (!(interface_mask[SOF_DAI_DSP_ACCESS] & BIT(dai_type))) - return false; - - if (dai_type == SOF_DAI_INTEL_HDA) - return true; - - switch (dai_type) { - case SOF_DAI_INTEL_SSP: - case SOF_DAI_INTEL_DMIC: - case SOF_DAI_INTEL_ALH: - chip = get_chip_info(sdev->pdata); - if (chip->hw_ip_version < SOF_INTEL_ACE_2_0) - return false; - return true; - default: - return false; - } -} - #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) /* @@ -210,35 +128,6 @@ static struct sdw_intel_ops sdw_ace2x_callback = { .trigger = sdw_ace2x_trigger, }; -void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable) -{ - struct sof_intel_hda_dev *hdev; - - hdev = sdev->pdata->hw_pdata; - - if (!hdev->sdw) - return; - - snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC2, - HDA_DSP_REG_ADSPIC2_SNDW, - enable ? HDA_DSP_REG_ADSPIC2_SNDW : 0); -} -EXPORT_SYMBOL_NS(hda_common_enable_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON); - -void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) -{ - u32 interface_mask = hda_get_interface_mask(sdev); - const struct sof_intel_dsp_desc *chip; - - if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH))) - return; - - chip = get_chip_info(sdev->pdata); - if (chip && chip->enable_sdw_irq) - chip->enable_sdw_irq(sdev, enable); -} -EXPORT_SYMBOL_NS(hda_sdw_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON); - static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) { u32 interface_mask = hda_get_interface_mask(sdev); @@ -330,67 +219,6 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) return 0; } -int hda_sdw_check_lcount_common(struct snd_sof_dev *sdev) -{ - struct sof_intel_hda_dev *hdev; - struct sdw_intel_ctx *ctx; - u32 caps; - - hdev = sdev->pdata->hw_pdata; - ctx = hdev->sdw; - - caps = snd_sof_dsp_read(sdev, HDA_DSP_BAR, ctx->shim_base + SDW_SHIM_LCAP); - caps &= SDW_SHIM_LCAP_LCOUNT_MASK; - - /* Check HW supported vs property value */ - if (caps < ctx->count) { - dev_err(sdev->dev, - "%s: BIOS master count %d is larger than hardware capabilities %d\n", - __func__, ctx->count, caps); - return -EINVAL; - } - - return 0; -} -EXPORT_SYMBOL_NS(hda_sdw_check_lcount_common, SND_SOC_SOF_INTEL_HDA_COMMON); - -int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev) -{ - struct sof_intel_hda_dev *hdev; - struct sdw_intel_ctx *ctx; - struct hdac_bus *bus; - u32 slcount; - - bus = sof_to_bus(sdev); - - hdev = sdev->pdata->hw_pdata; - ctx = hdev->sdw; - - slcount = hdac_bus_eml_get_count(bus, true, AZX_REG_ML_LEPTR_ID_SDW); - - /* Check HW supported vs property value */ - if (slcount < ctx->count) { - dev_err(sdev->dev, - "%s: BIOS master count %d is larger than hardware capabilities %d\n", - __func__, ctx->count, slcount); - return -EINVAL; - } - - return 0; -} -EXPORT_SYMBOL_NS(hda_sdw_check_lcount_ext, SND_SOC_SOF_INTEL_HDA_COMMON); - -static int hda_sdw_check_lcount(struct snd_sof_dev *sdev) -{ - const struct sof_intel_dsp_desc *chip; - - chip = get_chip_info(sdev->pdata); - if (chip && chip->read_sdw_lcount) - return chip->read_sdw_lcount(sdev); - - return 0; -} - int hda_sdw_startup(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hdev; @@ -596,11 +424,6 @@ EXPORT_SYMBOL_NS(hda_dsp_post_fw_run, SND_SOC_SOF_INTEL_HDA_COMMON); * Debug */ -struct hda_dsp_msg_code { - u32 code; - const char *text; -}; - #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG) static bool hda_use_msi = true; module_param_named(use_msi, hda_use_msi, bool, 0444); @@ -609,10 +432,6 @@ MODULE_PARM_DESC(use_msi, "SOF HDA use PCI MSI mode"); #define hda_use_msi (1) #endif -int sof_hda_position_quirk = SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS; -module_param_named(position_quirk, sof_hda_position_quirk, int, 0444); -MODULE_PARM_DESC(position_quirk, "SOF HDaudio position quirk"); - static char *hda_model; module_param(hda_model, charp, 0444); MODULE_PARM_DESC(hda_model, "Use the given HDA board model."); @@ -625,386 +444,6 @@ static int mclk_id_override = -1; module_param_named(mclk_id, mclk_id_override, int, 0444); MODULE_PARM_DESC(mclk_id, "SOF SSP mclk_id"); -static const struct hda_dsp_msg_code hda_dsp_rom_fw_error_texts[] = { - {HDA_DSP_ROM_CSE_ERROR, "error: cse error"}, - {HDA_DSP_ROM_CSE_WRONG_RESPONSE, "error: cse wrong response"}, - {HDA_DSP_ROM_IMR_TO_SMALL, "error: IMR too small"}, - {HDA_DSP_ROM_BASE_FW_NOT_FOUND, "error: base fw not found"}, - {HDA_DSP_ROM_CSE_VALIDATION_FAILED, "error: signature verification failed"}, - {HDA_DSP_ROM_IPC_FATAL_ERROR, "error: ipc fatal error"}, - {HDA_DSP_ROM_L2_CACHE_ERROR, "error: L2 cache error"}, - {HDA_DSP_ROM_LOAD_OFFSET_TO_SMALL, "error: load offset too small"}, - {HDA_DSP_ROM_API_PTR_INVALID, "error: API ptr invalid"}, - {HDA_DSP_ROM_BASEFW_INCOMPAT, "error: base fw incompatible"}, - {HDA_DSP_ROM_UNHANDLED_INTERRUPT, "error: unhandled interrupt"}, - {HDA_DSP_ROM_MEMORY_HOLE_ECC, "error: ECC memory hole"}, - {HDA_DSP_ROM_KERNEL_EXCEPTION, "error: kernel exception"}, - {HDA_DSP_ROM_USER_EXCEPTION, "error: user exception"}, - {HDA_DSP_ROM_UNEXPECTED_RESET, "error: unexpected reset"}, - {HDA_DSP_ROM_NULL_FW_ENTRY, "error: null FW entry point"}, -}; - -#define FSR_ROM_STATE_ENTRY(state) {FSR_STATE_ROM_##state, #state} -static const struct hda_dsp_msg_code cavs_fsr_rom_state_names[] = { - FSR_ROM_STATE_ENTRY(INIT), - FSR_ROM_STATE_ENTRY(INIT_DONE), - FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED), - FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED), - FSR_ROM_STATE_ENTRY(FW_FW_LOADED), - FSR_ROM_STATE_ENTRY(FW_ENTERED), - FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK), - FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET), - FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT), - FSR_ROM_STATE_ENTRY(FETCH_ROM_EXT_DONE), - /* CSE states */ - FSR_ROM_STATE_ENTRY(CSE_IMR_REQUEST), - FSR_ROM_STATE_ENTRY(CSE_IMR_GRANTED), - FSR_ROM_STATE_ENTRY(CSE_VALIDATE_IMAGE_REQUEST), - FSR_ROM_STATE_ENTRY(CSE_IMAGE_VALIDATED), - FSR_ROM_STATE_ENTRY(CSE_IPC_IFACE_INIT), - FSR_ROM_STATE_ENTRY(CSE_IPC_RESET_PHASE_1), - FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL_ENTRY), - FSR_ROM_STATE_ENTRY(CSE_IPC_OPERATIONAL), - FSR_ROM_STATE_ENTRY(CSE_IPC_DOWN), -}; - -static const struct hda_dsp_msg_code ace_fsr_rom_state_names[] = { - FSR_ROM_STATE_ENTRY(INIT), - FSR_ROM_STATE_ENTRY(INIT_DONE), - FSR_ROM_STATE_ENTRY(CSE_MANIFEST_LOADED), - FSR_ROM_STATE_ENTRY(FW_MANIFEST_LOADED), - FSR_ROM_STATE_ENTRY(FW_FW_LOADED), - FSR_ROM_STATE_ENTRY(FW_ENTERED), - FSR_ROM_STATE_ENTRY(VERIFY_FEATURE_MASK), - FSR_ROM_STATE_ENTRY(GET_LOAD_OFFSET), - FSR_ROM_STATE_ENTRY(RESET_VECTOR_DONE), - FSR_ROM_STATE_ENTRY(PURGE_BOOT), - FSR_ROM_STATE_ENTRY(RESTORE_BOOT), - FSR_ROM_STATE_ENTRY(FW_ENTRY_POINT), - FSR_ROM_STATE_ENTRY(VALIDATE_PUB_KEY), - FSR_ROM_STATE_ENTRY(POWER_DOWN_HPSRAM), - FSR_ROM_STATE_ENTRY(POWER_DOWN_ULPSRAM), - FSR_ROM_STATE_ENTRY(POWER_UP_ULPSRAM_STACK), - FSR_ROM_STATE_ENTRY(POWER_UP_HPSRAM_DMA), - FSR_ROM_STATE_ENTRY(BEFORE_EP_POINTER_READ), - FSR_ROM_STATE_ENTRY(VALIDATE_MANIFEST), - FSR_ROM_STATE_ENTRY(VALIDATE_FW_MODULE), - FSR_ROM_STATE_ENTRY(PROTECT_IMR_REGION), - FSR_ROM_STATE_ENTRY(PUSH_MODEL_ROUTINE), - FSR_ROM_STATE_ENTRY(PULL_MODEL_ROUTINE), - FSR_ROM_STATE_ENTRY(VALIDATE_PKG_DIR), - FSR_ROM_STATE_ENTRY(VALIDATE_CPD), - FSR_ROM_STATE_ENTRY(VALIDATE_CSS_MAN_HEADER), - FSR_ROM_STATE_ENTRY(VALIDATE_BLOB_SVN), - FSR_ROM_STATE_ENTRY(VERIFY_IFWI_PARTITION), - FSR_ROM_STATE_ENTRY(REMOVE_ACCESS_CONTROL), - FSR_ROM_STATE_ENTRY(AUTH_BYPASS), - FSR_ROM_STATE_ENTRY(AUTH_ENABLED), - FSR_ROM_STATE_ENTRY(INIT_DMA), - FSR_ROM_STATE_ENTRY(PURGE_FW_ENTRY), - FSR_ROM_STATE_ENTRY(PURGE_FW_END), - FSR_ROM_STATE_ENTRY(CLEAN_UP_BSS_DONE), - FSR_ROM_STATE_ENTRY(IMR_RESTORE_ENTRY), - FSR_ROM_STATE_ENTRY(IMR_RESTORE_END), - FSR_ROM_STATE_ENTRY(FW_MANIFEST_IN_DMA_BUFF), - FSR_ROM_STATE_ENTRY(LOAD_CSE_MAN_TO_IMR), - FSR_ROM_STATE_ENTRY(LOAD_FW_MAN_TO_IMR), - FSR_ROM_STATE_ENTRY(LOAD_FW_CODE_TO_IMR), - FSR_ROM_STATE_ENTRY(FW_LOADING_DONE), - FSR_ROM_STATE_ENTRY(FW_CODE_LOADED), - FSR_ROM_STATE_ENTRY(VERIFY_IMAGE_TYPE), - FSR_ROM_STATE_ENTRY(AUTH_API_INIT), - FSR_ROM_STATE_ENTRY(AUTH_API_PROC), - FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_BUSY), - FSR_ROM_STATE_ENTRY(AUTH_API_FIRST_RESULT), - FSR_ROM_STATE_ENTRY(AUTH_API_CLEANUP), -}; - -#define FSR_BRINGUP_STATE_ENTRY(state) {FSR_STATE_BRINGUP_##state, #state} -static const struct hda_dsp_msg_code fsr_bringup_state_names[] = { - FSR_BRINGUP_STATE_ENTRY(INIT), - FSR_BRINGUP_STATE_ENTRY(INIT_DONE), - FSR_BRINGUP_STATE_ENTRY(HPSRAM_LOAD), - FSR_BRINGUP_STATE_ENTRY(UNPACK_START), - FSR_BRINGUP_STATE_ENTRY(IMR_RESTORE), - FSR_BRINGUP_STATE_ENTRY(FW_ENTERED), -}; - -#define FSR_WAIT_STATE_ENTRY(state) {FSR_WAIT_FOR_##state, #state} -static const struct hda_dsp_msg_code fsr_wait_state_names[] = { - FSR_WAIT_STATE_ENTRY(IPC_BUSY), - FSR_WAIT_STATE_ENTRY(IPC_DONE), - FSR_WAIT_STATE_ENTRY(CACHE_INVALIDATION), - FSR_WAIT_STATE_ENTRY(LP_SRAM_OFF), - FSR_WAIT_STATE_ENTRY(DMA_BUFFER_FULL), - FSR_WAIT_STATE_ENTRY(CSE_CSR), -}; - -#define FSR_MODULE_NAME_ENTRY(mod) [FSR_MOD_##mod] = #mod -static const char * const fsr_module_names[] = { - FSR_MODULE_NAME_ENTRY(ROM), - FSR_MODULE_NAME_ENTRY(ROM_BYP), - FSR_MODULE_NAME_ENTRY(BASE_FW), - FSR_MODULE_NAME_ENTRY(LP_BOOT), - FSR_MODULE_NAME_ENTRY(BRNGUP), - FSR_MODULE_NAME_ENTRY(ROM_EXT), -}; - -static const char * -hda_dsp_get_state_text(u32 code, const struct hda_dsp_msg_code *msg_code, - size_t array_size) -{ - int i; - - for (i = 0; i < array_size; i++) { - if (code == msg_code[i].code) - return msg_code[i].text; - } - - return NULL; -} - -void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level) -{ - const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); - const char *state_text, *error_text, *module_text; - u32 fsr, state, wait_state, module, error_code; - - fsr = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg); - state = FSR_TO_STATE_CODE(fsr); - wait_state = FSR_TO_WAIT_STATE_CODE(fsr); - module = FSR_TO_MODULE_CODE(fsr); - - if (module > FSR_MOD_ROM_EXT) - module_text = "unknown"; - else - module_text = fsr_module_names[module]; - - if (module == FSR_MOD_BRNGUP) { - state_text = hda_dsp_get_state_text(state, fsr_bringup_state_names, - ARRAY_SIZE(fsr_bringup_state_names)); - } else { - if (chip->hw_ip_version < SOF_INTEL_ACE_1_0) - state_text = hda_dsp_get_state_text(state, - cavs_fsr_rom_state_names, - ARRAY_SIZE(cavs_fsr_rom_state_names)); - else - state_text = hda_dsp_get_state_text(state, - ace_fsr_rom_state_names, - ARRAY_SIZE(ace_fsr_rom_state_names)); - } - - /* not for us, must be generic sof message */ - if (!state_text) { - dev_printk(level, sdev->dev, "%#010x: unknown ROM status value\n", fsr); - return; - } - - if (wait_state) { - const char *wait_state_text; - - wait_state_text = hda_dsp_get_state_text(wait_state, fsr_wait_state_names, - ARRAY_SIZE(fsr_wait_state_names)); - if (!wait_state_text) - wait_state_text = "unknown"; - - dev_printk(level, sdev->dev, - "%#010x: module: %s, state: %s, waiting for: %s, %s\n", - fsr, module_text, state_text, wait_state_text, - fsr & FSR_HALTED ? "not running" : "running"); - } else { - dev_printk(level, sdev->dev, "%#010x: module: %s, state: %s, %s\n", - fsr, module_text, state_text, - fsr & FSR_HALTED ? "not running" : "running"); - } - - error_code = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + 4); - if (!error_code) - return; - - error_text = hda_dsp_get_state_text(error_code, hda_dsp_rom_fw_error_texts, - ARRAY_SIZE(hda_dsp_rom_fw_error_texts)); - if (!error_text) - error_text = "unknown"; - - if (state == FSR_STATE_FW_ENTERED) - dev_printk(level, sdev->dev, "status code: %#x (%s)\n", error_code, - error_text); - else - dev_printk(level, sdev->dev, "error code: %#x (%s)\n", error_code, - error_text); -} -EXPORT_SYMBOL_NS(hda_dsp_get_state, SND_SOC_SOF_INTEL_HDA_COMMON); - -static void hda_dsp_get_registers(struct snd_sof_dev *sdev, - struct sof_ipc_dsp_oops_xtensa *xoops, - struct sof_ipc_panic_info *panic_info, - u32 *stack, size_t stack_words) -{ - u32 offset = sdev->dsp_oops_offset; - - /* first read registers */ - sof_mailbox_read(sdev, offset, xoops, sizeof(*xoops)); - - /* note: variable AR register array is not read */ - - /* then get panic info */ - if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) { - dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n", - xoops->arch_hdr.totalsize); - return; - } - offset += xoops->arch_hdr.totalsize; - sof_block_read(sdev, sdev->mmio_bar, offset, - panic_info, sizeof(*panic_info)); - - /* then get the stack */ - offset += sizeof(*panic_info); - sof_block_read(sdev, sdev->mmio_bar, offset, stack, - stack_words * sizeof(u32)); -} - -/* dump the first 8 dwords representing the extended ROM status */ -static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *level, - u32 flags) -{ - const struct sof_intel_dsp_desc *chip; - char msg[128]; - int len = 0; - u32 value; - int i; - - chip = get_chip_info(sdev->pdata); - for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { - value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4); - len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); - } - - dev_printk(level, sdev->dev, "extended rom status: %s", msg); - -} - -void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) -{ - char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR; - struct sof_ipc_dsp_oops_xtensa xoops; - struct sof_ipc_panic_info panic_info; - u32 stack[HDA_DSP_STACK_DUMP_SIZE]; - - /* print ROM/FW status */ - hda_dsp_get_state(sdev, level); - - /* The firmware register dump only available with IPC3 */ - if (flags & SOF_DBG_DUMP_REGS && sdev->pdata->ipc_type == SOF_IPC_TYPE_3) { - u32 status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_STATUS); - u32 panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP); - - hda_dsp_get_registers(sdev, &xoops, &panic_info, stack, - HDA_DSP_STACK_DUMP_SIZE); - sof_print_oops_and_stack(sdev, level, status, panic, &xoops, - &panic_info, stack, HDA_DSP_STACK_DUMP_SIZE); - } else { - hda_dsp_dump_ext_rom_status(sdev, level, flags); - } -} - -void hda_ipc4_dsp_dump(struct snd_sof_dev *sdev, u32 flags) -{ - char *level = (flags & SOF_DBG_DUMP_OPTIONAL) ? KERN_DEBUG : KERN_ERR; - - /* print ROM/FW status */ - hda_dsp_get_state(sdev, level); - - if (flags & SOF_DBG_DUMP_REGS) - sof_ipc4_intel_dump_telemetry_state(sdev, flags); - else - hda_dsp_dump_ext_rom_status(sdev, level, flags); -} -EXPORT_SYMBOL_NS(hda_ipc4_dsp_dump, SND_SOC_SOF_INTEL_HDA_COMMON); - -static bool hda_check_ipc_irq(struct snd_sof_dev *sdev) -{ - const struct sof_intel_dsp_desc *chip; - - chip = get_chip_info(sdev->pdata); - if (chip && chip->check_ipc_irq) - return chip->check_ipc_irq(sdev); - - return false; -} - -void hda_ipc_irq_dump(struct snd_sof_dev *sdev) -{ - u32 adspis; - u32 intsts; - u32 intctl; - u32 ppsts; - u8 rirbsts; - - /* read key IRQ stats and config registers */ - adspis = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS); - intsts = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS); - intctl = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL); - ppsts = snd_sof_dsp_read(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPSTS); - rirbsts = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, AZX_REG_RIRBSTS); - - dev_err(sdev->dev, "hda irq intsts 0x%8.8x intlctl 0x%8.8x rirb %2.2x\n", - intsts, intctl, rirbsts); - dev_err(sdev->dev, "dsp irq ppsts 0x%8.8x adspis 0x%8.8x\n", ppsts, adspis); -} -EXPORT_SYMBOL_NS(hda_ipc_irq_dump, SND_SOC_SOF_INTEL_HDA_COMMON); - -void hda_ipc_dump(struct snd_sof_dev *sdev) -{ - u32 hipcie; - u32 hipct; - u32 hipcctl; - - hda_ipc_irq_dump(sdev); - - /* read IPC status */ - hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE); - hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT); - hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL); - - /* dump the IPC regs */ - /* TODO: parse the raw msg */ - dev_err(sdev->dev, "host status 0x%8.8x dsp status 0x%8.8x mask 0x%8.8x\n", - hipcie, hipct, hipcctl); -} -EXPORT_SYMBOL_NS(hda_ipc_dump, SND_SOC_SOF_INTEL_HDA_COMMON); - -void hda_ipc4_dump(struct snd_sof_dev *sdev) -{ - u32 hipci, hipcie, hipct, hipcte, hipcctl; - - hda_ipc_irq_dump(sdev); - - hipci = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCI); - hipcie = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCIE); - hipct = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCT); - hipcte = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCTE); - hipcctl = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_HIPCCTL); - - /* dump the IPC regs */ - /* TODO: parse the raw msg */ - dev_err(sdev->dev, "Host IPC initiator: %#x|%#x, target: %#x|%#x, ctl: %#x\n", - hipci, hipcie, hipct, hipcte, hipcctl); -} -EXPORT_SYMBOL_NS(hda_ipc4_dump, SND_SOC_SOF_INTEL_HDA_COMMON); - -bool hda_ipc4_tx_is_busy(struct snd_sof_dev *sdev) -{ - struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; - const struct sof_intel_dsp_desc *chip = hda->desc; - u32 val; - - val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->ipc_req); - - return !!(val & chip->ipc_req_mask); -} -EXPORT_SYMBOL_NS(hda_ipc4_tx_is_busy, SND_SOC_SOF_INTEL_HDA_COMMON); - static int hda_init(struct snd_sof_dev *sdev) { struct hda_bus *hbus; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 7804c5b50c265..809a1be0d388a 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -617,6 +617,8 @@ void hda_ipc_dump(struct snd_sof_dev *sdev); void hda_ipc_irq_dump(struct snd_sof_dev *sdev); void hda_dsp_d0i3_work(struct work_struct *work); int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev); +bool hda_check_ipc_irq(struct snd_sof_dev *sdev); +u32 hda_get_interface_mask(struct snd_sof_dev *sdev); /* * DSP PCM Operations. @@ -698,6 +700,8 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context); int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir); void hda_dsp_get_state(struct snd_sof_dev *sdev, const char *level); +void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *level, + u32 flags); /* * DSP Code loader. @@ -806,6 +810,7 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); int hda_sdw_check_lcount_common(struct snd_sof_dev *sdev); int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev); +int hda_sdw_check_lcount(struct snd_sof_dev *sdev); int hda_sdw_startup(struct snd_sof_dev *sdev); void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable); void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); @@ -825,6 +830,11 @@ static inline int hda_sdw_check_lcount_ext(struct snd_sof_dev *sdev) return 0; } +static inline int hda_sdw_check_lcount(struct snd_sof_dev *sdev) +{ + return 0; +} + static inline int hda_sdw_startup(struct snd_sof_dev *sdev) { return 0; From 136b37369cc4f044139b886133f3b9a721ecacd1 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:52:19 -0500 Subject: [PATCH 454/603] ASoC: SOF: Intel: move tracepoint creation CREATE_TRACEPOINTS is supposed to be used once. To avoid modpost issues when creating modules, let's move the tracepoint creation in a single object file. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20240503135221.229202-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/hda.c | 1 - sound/soc/sof/intel/tracepoints.c | 3 +++ 3 files changed, 4 insertions(+), 2 deletions(-) create mode 100644 sound/soc/sof/intel/tracepoints.c diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index c895bae0efdd7..cf74548f87ade 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -7,7 +7,7 @@ snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \ hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ hda-dai.o hda-dai-ops.o hda-bus.o \ hda-common-ops.o \ - telemetry.o + telemetry.o tracepoints.o snd-sof-intel-hda-mlink-objs := hda-mlink.o diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index a96926fb45b65..ea7dcda90f2a9 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -35,7 +35,6 @@ #include "../ipc4-topology.h" #include "hda.h" -#define CREATE_TRACE_POINTS #include #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) diff --git a/sound/soc/sof/intel/tracepoints.c b/sound/soc/sof/intel/tracepoints.c new file mode 100644 index 0000000000000..c223c96fe0e39 --- /dev/null +++ b/sound/soc/sof/intel/tracepoints.c @@ -0,0 +1,3 @@ +// SPDX-License-Identifier: GPL-2.0 +#define CREATE_TRACE_POINTS +#include From 3b7bd0c139849e95981961525b7f6dd20097a563 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:52:20 -0500 Subject: [PATCH 455/603] ASoC: SOF: Intel: remove circular dependency on hda_sdw_process_wakeen() hda_sdw_process_wakeen() is used in hda-loader.c, but defined in hda.c. This code split will create a circular dependency when hda.c is moved to a different module. Rather than an invasive code change, this patch follows the model used for sdw_check_wakeen_irq() with an abstraction. For now all abstractions point to the same common routine, which is arguably not great, but this also provides us with a future-proof way of addressing platform-specific wake processing. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20240503135221.229202-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/cnl.c | 2 ++ sound/soc/sof/intel/hda-dsp.c | 15 +++++++++++++++ sound/soc/sof/intel/hda.c | 3 ++- sound/soc/sof/intel/hda.h | 5 +++++ sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/lnl.c | 1 + sound/soc/sof/intel/mtl.c | 2 ++ sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/intel/tgl.c | 4 ++++ 9 files changed, 33 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 460bc404ddb69..1936865b2937a 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -473,6 +473,7 @@ const struct sof_intel_dsp_desc cnl_chip_info = { .enable_sdw_irq = hda_common_enable_sdw_irq, .check_sdw_irq = hda_common_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = hda_dsp_check_ipc_irq, .cl_init = cl_dsp_init, .power_down_dsp = hda_power_down_dsp, @@ -508,6 +509,7 @@ const struct sof_intel_dsp_desc jsl_chip_info = { .enable_sdw_irq = hda_common_enable_sdw_irq, .check_sdw_irq = hda_common_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = hda_dsp_check_ipc_irq, .cl_init = cl_dsp_init, .power_down_dsp = hda_power_down_dsp, diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 974ac1cd8c6e6..780f4c33e6784 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -1311,6 +1311,21 @@ int hda_sdw_check_lcount(struct snd_sof_dev *sdev) return 0; } + +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ + u32 interface_mask = hda_get_interface_mask(sdev); + const struct sof_intel_dsp_desc *chip; + + if (!(interface_mask & BIT(SOF_DAI_INTEL_ALH))) + return; + + chip = get_chip_info(sdev->pdata); + if (chip && chip->sdw_process_wakeen) + chip->sdw_process_wakeen(sdev); +} +EXPORT_SYMBOL_NS(hda_sdw_process_wakeen, SND_SOC_SOF_INTEL_HDA_COMMON); + #endif int hda_dsp_disable_interrupts(struct snd_sof_dev *sdev) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ea7dcda90f2a9..87b0375250a25 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -331,7 +331,7 @@ static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) return false; } -void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +void hda_sdw_process_wakeen_common(struct snd_sof_dev *sdev) { u32 interface_mask = hda_get_interface_mask(sdev); struct sof_intel_hda_dev *hdev; @@ -345,6 +345,7 @@ void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) sdw_intel_process_wakeen_event(hdev->sdw); } +EXPORT_SYMBOL_NS(hda_sdw_process_wakeen_common, SND_SOC_SOF_INTEL_HDA_COMMON); #else /* IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) */ static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 809a1be0d388a..ef78c9b5c2503 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -815,6 +815,7 @@ int hda_sdw_startup(struct snd_sof_dev *sdev); void hda_common_enable_sdw_irq(struct snd_sof_dev *sdev, bool enable); void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev); +void hda_sdw_process_wakeen_common(struct snd_sof_dev *sdev); void hda_sdw_process_wakeen(struct snd_sof_dev *sdev); bool hda_common_check_sdw_irq(struct snd_sof_dev *sdev); @@ -853,6 +854,10 @@ static inline bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev) return false; } +static inline void hda_sdw_process_wakeen_common(struct snd_sof_dev *sdev) +{ +} + static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) { } diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c index 115866225e053..9b916eb4fc21e 100644 --- a/sound/soc/sof/intel/icl.c +++ b/sound/soc/sof/intel/icl.c @@ -187,6 +187,7 @@ const struct sof_intel_dsp_desc icl_chip_info = { .enable_sdw_irq = hda_common_enable_sdw_irq, .check_sdw_irq = hda_common_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = hda_dsp_check_ipc_irq, .cl_init = cl_dsp_init, .power_down_dsp = hda_power_down_dsp, diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index ca0cb91d2b2c3..a7446be36627f 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -238,6 +238,7 @@ const struct sof_intel_dsp_desc lnl_chip_info = { .enable_sdw_irq = lnl_enable_sdw_irq, .check_sdw_irq = lnl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 13f780f33c223..b9fc123be75df 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -779,6 +779,7 @@ const struct sof_intel_dsp_desc mtl_chip_info = { .enable_sdw_irq = mtl_enable_sdw_irq, .check_sdw_irq = mtl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, @@ -806,6 +807,7 @@ const struct sof_intel_dsp_desc arl_s_chip_info = { .enable_sdw_irq = mtl_enable_sdw_irq, .check_sdw_irq = mtl_dsp_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = mtl_dsp_check_ipc_irq, .cl_init = mtl_dsp_cl_init, .power_down_dsp = mtl_power_down_dsp, diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index 18ce3d33fa803..448ec54a6aa94 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -190,6 +190,7 @@ struct sof_intel_dsp_desc { void (*enable_sdw_irq)(struct snd_sof_dev *sdev, bool enable); bool (*check_sdw_irq)(struct snd_sof_dev *sdev); bool (*check_sdw_wakeen_irq)(struct snd_sof_dev *sdev); + void (*sdw_process_wakeen)(struct snd_sof_dev *sdev); bool (*check_ipc_irq)(struct snd_sof_dev *sdev); int (*power_down_dsp)(struct snd_sof_dev *sdev); int (*disable_interrupts)(struct snd_sof_dev *sdev); diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index 6acc659238342..062327d4f068f 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -156,6 +156,7 @@ const struct sof_intel_dsp_desc tgl_chip_info = { .enable_sdw_irq = hda_common_enable_sdw_irq, .check_sdw_irq = hda_common_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = hda_dsp_check_ipc_irq, .cl_init = cl_dsp_init, .power_down_dsp = hda_power_down_dsp, @@ -184,6 +185,7 @@ const struct sof_intel_dsp_desc tglh_chip_info = { .enable_sdw_irq = hda_common_enable_sdw_irq, .check_sdw_irq = hda_common_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = hda_dsp_check_ipc_irq, .cl_init = cl_dsp_init, .power_down_dsp = hda_power_down_dsp, @@ -212,6 +214,7 @@ const struct sof_intel_dsp_desc ehl_chip_info = { .enable_sdw_irq = hda_common_enable_sdw_irq, .check_sdw_irq = hda_common_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = hda_dsp_check_ipc_irq, .cl_init = cl_dsp_init, .power_down_dsp = hda_power_down_dsp, @@ -240,6 +243,7 @@ const struct sof_intel_dsp_desc adls_chip_info = { .enable_sdw_irq = hda_common_enable_sdw_irq, .check_sdw_irq = hda_common_check_sdw_irq, .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .sdw_process_wakeen = hda_sdw_process_wakeen_common, .check_ipc_irq = hda_dsp_check_ipc_irq, .cl_init = cl_dsp_init, .power_down_dsp = hda_power_down_dsp, From 6fe61f31eab1ec84c385786cd052415d966e5235 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 08:52:21 -0500 Subject: [PATCH 456/603] ASoC: SOF: Intel: move hda.c to different module Now that most of the code moves are done, we can add a new module and the required EXPORT_SYMBOL definitions. No functionality change, just a new module added. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20240503135221.229202-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 22 +++++++++++++--------- sound/soc/sof/intel/Makefile | 6 ++++-- sound/soc/sof/intel/hda-bus.c | 2 ++ sound/soc/sof/intel/hda-common-ops.c | 2 +- sound/soc/sof/intel/hda-ctrl.c | 9 +++++++++ sound/soc/sof/intel/hda-dai.c | 5 +++++ sound/soc/sof/intel/hda-dsp.c | 9 +++++++++ sound/soc/sof/intel/hda-ipc.c | 5 +++++ sound/soc/sof/intel/hda-loader.c | 1 + sound/soc/sof/intel/hda-pcm.c | 6 ++++++ sound/soc/sof/intel/hda-probes.c | 2 ++ sound/soc/sof/intel/hda-stream.c | 8 ++++++++ sound/soc/sof/intel/hda-trace.c | 3 +++ sound/soc/sof/intel/hda.c | 21 +++++++++++---------- sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-lnl.c | 1 + sound/soc/sof/intel/pci-mtl.c | 1 + sound/soc/sof/intel/pci-skl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/intel/tracepoints.c | 2 ++ 22 files changed, 88 insertions(+), 22 deletions(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index b804be7321be5..3396bd46b7788 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -97,7 +97,7 @@ config SND_SOC_SOF_MERRIFIELD config SND_SOC_SOF_INTEL_SKL tristate - select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_HDA_GENERIC select SND_SOC_SOF_IPC4 config SND_SOC_SOF_SKYLAKE @@ -122,7 +122,7 @@ config SND_SOC_SOF_KABYLAKE config SND_SOC_SOF_INTEL_APL tristate - select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_HDA_GENERIC select SND_SOC_SOF_IPC3 select SND_SOC_SOF_IPC4 @@ -148,7 +148,7 @@ config SND_SOC_SOF_GEMINILAKE config SND_SOC_SOF_INTEL_CNL tristate - select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_HDA_GENERIC select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE select SND_SOC_SOF_IPC3 select SND_SOC_SOF_IPC4 @@ -184,7 +184,7 @@ config SND_SOC_SOF_COMETLAKE config SND_SOC_SOF_INTEL_ICL tristate - select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_HDA_GENERIC select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE select SND_SOC_SOF_IPC3 select SND_SOC_SOF_IPC4 @@ -212,7 +212,7 @@ config SND_SOC_SOF_JASPERLAKE config SND_SOC_SOF_INTEL_TGL tristate - select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_HDA_GENERIC select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE select SND_SOC_SOF_IPC3 select SND_SOC_SOF_IPC4 @@ -250,7 +250,7 @@ config SND_SOC_SOF_ALDERLAKE config SND_SOC_SOF_INTEL_MTL tristate - select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_HDA_GENERIC select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE select SND_SOC_SOF_IPC4 @@ -266,7 +266,7 @@ config SND_SOC_SOF_METEORLAKE config SND_SOC_SOF_INTEL_LNL tristate - select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_HDA_GENERIC select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE select SND_SOC_SOF_IPC4 select SND_SOC_SOF_INTEL_MTL @@ -283,6 +283,10 @@ config SND_SOC_SOF_LUNARLAKE config SND_SOC_SOF_HDA_COMMON tristate + +config SND_SOC_SOF_HDA_GENERIC + tristate + select SND_SOC_SOF_HDA_COMMON select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_PCI_DEV select SND_INTEL_DSP_CONFIG @@ -299,7 +303,7 @@ config SND_SOC_SOF_HDA_MLINK This option is not user-selectable but automagically handled by 'select' statements at a higher level. -if SND_SOC_SOF_HDA_COMMON +if SND_SOC_SOF_HDA_GENERIC config SND_SOC_SOF_HDA_LINK bool "SOF support for HDA Links(HDA/HDMI)" @@ -319,7 +323,7 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC Say Y if you want to enable HDAudio codecs with SOF. If unsure select "N". -endif ## SND_SOC_SOF_HDA_COMMON +endif ## SND_SOC_SOF_HDA_GENERIC config SND_SOC_SOF_HDA_LINK_BASELINE tristate diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index cf74548f87ade..806df08e3fd58 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -3,12 +3,13 @@ snd-sof-acpi-intel-byt-objs := byt.o snd-sof-acpi-intel-bdw-objs := bdw.o -snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \ +snd-sof-intel-hda-common-objs := hda-loader.o hda-stream.o hda-trace.o \ hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ hda-dai.o hda-dai-ops.o hda-bus.o \ - hda-common-ops.o \ telemetry.o tracepoints.o +snd-sof-intel-hda-generic-objs := hda.o hda-common-ops.o + snd-sof-intel-hda-mlink-objs := hda-mlink.o snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-probes.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_SOC_SOF_INTEL_ATOM_HIFI_EP) += snd-sof-intel-atom.o obj-$(CONFIG_SND_SOC_SOF_BAYTRAIL) += snd-sof-acpi-intel-byt.o obj-$(CONFIG_SND_SOC_SOF_BROADWELL) += snd-sof-acpi-intel-bdw.o obj-$(CONFIG_SND_SOC_SOF_HDA_COMMON) += snd-sof-intel-hda-common.o +obj-$(CONFIG_SND_SOC_SOF_HDA_GENERIC) += snd-sof-intel-hda-generic.o obj-$(CONFIG_SND_SOC_SOF_HDA_MLINK) += snd-sof-intel-hda-mlink.o obj-$(CONFIG_SND_SOC_SOF_HDA) += snd-sof-intel-hda.o diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index fc63085d2d743..e9f65386e4806 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -94,6 +94,7 @@ void sof_hda_bus_init(struct snd_sof_dev *sdev, struct device *dev) spin_lock_init(&bus->reg_lock); #endif /* CONFIG_SND_SOC_SOF_HDA_LINK */ } +EXPORT_SYMBOL_NS(sof_hda_bus_init, SND_SOC_SOF_INTEL_HDA_COMMON); void sof_hda_bus_exit(struct snd_sof_dev *sdev) { @@ -103,3 +104,4 @@ void sof_hda_bus_exit(struct snd_sof_dev *sdev) snd_hdac_ext_bus_exit(bus); #endif } +EXPORT_SYMBOL_NS(sof_hda_bus_exit, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 6a863ceba8da0..e4c5031d227ea 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -105,4 +105,4 @@ const struct snd_sof_dsp_ops sof_hda_common_ops = { .dsp_arch_ops = &sof_xtensa_arch_ops, }; -EXPORT_SYMBOL_NS(sof_hda_common_ops, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(sof_hda_common_ops, SND_SOC_SOF_INTEL_HDA_GENERIC); diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 56c0f493042c8..810a2ba4fbe40 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -128,6 +128,7 @@ int hda_dsp_ctrl_get_caps(struct snd_sof_dev *sdev) return 0; } +EXPORT_SYMBOL_NS(hda_dsp_ctrl_get_caps, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_ctrl_ppcap_enable(struct snd_sof_dev *sdev, bool enable) { @@ -136,6 +137,7 @@ void hda_dsp_ctrl_ppcap_enable(struct snd_sof_dev *sdev, bool enable) snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, SOF_HDA_PPCTL_GPROCEN, val); } +EXPORT_SYMBOL_NS(hda_dsp_ctrl_ppcap_enable, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_ctrl_ppcap_int_enable(struct snd_sof_dev *sdev, bool enable) { @@ -144,6 +146,7 @@ void hda_dsp_ctrl_ppcap_int_enable(struct snd_sof_dev *sdev, bool enable) snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, SOF_HDA_PPCTL_PIE, val); } +EXPORT_SYMBOL_NS(hda_dsp_ctrl_ppcap_int_enable, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable) { @@ -263,6 +266,7 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_ctrl_init_chip, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev) { @@ -322,3 +326,8 @@ void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev) bus->chip_init = false; } + +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK); +MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC); +MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC_I915); diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 650e3a37326ea..f494d8701345e 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -54,6 +54,7 @@ int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, return 0; } +EXPORT_SYMBOL_NS(hda_dai_config, SND_SOC_SOF_INTEL_HDA_COMMON); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_LINK) @@ -542,6 +543,7 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream, } return 0; } +EXPORT_SYMBOL_NS(sdw_hda_dai_hw_params, SND_SOC_SOF_INTEL_HDA_COMMON); int sdw_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai, @@ -570,12 +572,14 @@ int sdw_hda_dai_hw_free(struct snd_pcm_substream *substream, return 0; } +EXPORT_SYMBOL_NS(sdw_hda_dai_hw_free, SND_SOC_SOF_INTEL_HDA_COMMON); int sdw_hda_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { return hda_dai_trigger(substream, cmd, cpu_dai); } +EXPORT_SYMBOL_NS(sdw_hda_dai_trigger, SND_SOC_SOF_INTEL_HDA_COMMON); static int hda_dai_suspend(struct hdac_bus *bus) { @@ -856,6 +860,7 @@ struct snd_soc_dai_driver skl_dai[] = { }, #endif }; +EXPORT_SYMBOL_NS(skl_dai, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_dais_suspend(struct snd_sof_dev *sdev) { diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 780f4c33e6784..4a27e1dfca392 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -89,6 +89,7 @@ u32 hda_get_interface_mask(struct snd_sof_dev *sdev) return interface_mask[sdev->dspless_mode_selected]; } +EXPORT_SYMBOL_NS(hda_get_interface_mask, SND_SOC_SOF_INTEL_HDA_COMMON); bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type) { @@ -118,6 +119,7 @@ bool hda_is_chain_dma_supported(struct snd_sof_dev *sdev, u32 dai_type) return false; } } +EXPORT_SYMBOL_NS(hda_is_chain_dma_supported, SND_SOC_SOF_INTEL_HDA_COMMON); /* * DSP Core control. @@ -980,6 +982,7 @@ int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) return 0; } +EXPORT_SYMBOL_NS(hda_dsp_runtime_idle, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) { @@ -1001,6 +1004,7 @@ int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) return snd_sof_dsp_set_power_state(sdev, &target_state); } +EXPORT_SYMBOL_NS(hda_dsp_runtime_suspend, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) { @@ -1061,6 +1065,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); } +EXPORT_SYMBOL_NS(hda_dsp_suspend, SND_SOC_SOF_INTEL_HDA_COMMON); static unsigned int hda_dsp_check_for_dma_streams(struct snd_sof_dev *sdev) { @@ -1153,6 +1158,7 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_set_hw_params_upon_resume, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_d0i3_work(struct work_struct *work) { @@ -1179,6 +1185,7 @@ void hda_dsp_d0i3_work(struct work_struct *work) "error: failed to set DSP state %d substate %d\n", target_state.state, target_state.substate); } +EXPORT_SYMBOL_NS(hda_dsp_d0i3_work, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_core_get(struct snd_sof_dev *sdev, int core) { @@ -1311,6 +1318,7 @@ int hda_sdw_check_lcount(struct snd_sof_dev *sdev) return 0; } +EXPORT_SYMBOL_NS(hda_sdw_check_lcount, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) { @@ -1619,3 +1627,4 @@ void hda_dsp_dump(struct snd_sof_dev *sdev, u32 flags) hda_dsp_dump_ext_rom_status(sdev, level, flags); } } +EXPORT_SYMBOL_NS(hda_dsp_dump, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 2252c9198cc02..6b7f2337567ce 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -398,11 +398,13 @@ int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev) { return HDA_DSP_MBOX_UPLINK_OFFSET; } +EXPORT_SYMBOL_NS(hda_dsp_ipc_get_mailbox_offset, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id) { return SRAM_WINDOW_OFFSET(id); } +EXPORT_SYMBOL_NS(hda_dsp_ipc_get_window_offset, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_ipc_msg_data(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sps, @@ -428,6 +430,7 @@ int hda_ipc_msg_data(struct snd_sof_dev *sdev, return 0; } +EXPORT_SYMBOL_NS(hda_ipc_msg_data, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_set_stream_data_offset(struct snd_sof_dev *sdev, struct snd_sof_pcm_stream *sps, @@ -452,6 +455,7 @@ int hda_set_stream_data_offset(struct snd_sof_dev *sdev, return 0; } +EXPORT_SYMBOL_NS(hda_set_stream_data_offset, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_ipc4_dsp_dump(struct snd_sof_dev *sdev, u32 flags) { @@ -477,6 +481,7 @@ bool hda_check_ipc_irq(struct snd_sof_dev *sdev) return false; } +EXPORT_SYMBOL_NS(hda_check_ipc_irq, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_ipc_irq_dump(struct snd_sof_dev *sdev) { diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 34c18275c949a..df668806a8c4c 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -691,3 +691,4 @@ int hda_dsp_ext_man_get_cavs_config_data(struct snd_sof_dev *sdev, return 0; } +EXPORT_SYMBOL_NS(hda_dsp_ext_man_get_cavs_config_data, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index d7b446f3f973e..27dc134f04a84 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -142,6 +142,7 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, return 0; } +EXPORT_SYMBOL_NS(hda_dsp_pcm_hw_params, SND_SOC_SOF_INTEL_HDA_COMMON); /* update SPIB register with appl position */ int hda_dsp_pcm_ack(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream) @@ -164,6 +165,7 @@ int hda_dsp_pcm_ack(struct snd_sof_dev *sdev, struct snd_pcm_substream *substrea return 0; } +EXPORT_SYMBOL_NS(hda_dsp_pcm_ack, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, int cmd) @@ -173,6 +175,7 @@ int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev, return hda_dsp_stream_trigger(sdev, hext_stream, cmd); } +EXPORT_SYMBOL_NS(hda_dsp_pcm_trigger, SND_SOC_SOF_INTEL_HDA_COMMON); snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream) @@ -204,6 +207,7 @@ snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, trace_sof_intel_hda_dsp_pcm(sdev, hstream, substream, pos); return pos; } +EXPORT_SYMBOL_NS(hda_dsp_pcm_pointer, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_pcm_open(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream) @@ -292,6 +296,7 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, return 0; } +EXPORT_SYMBOL_NS(hda_dsp_pcm_open, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_pcm_close(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream) @@ -311,3 +316,4 @@ int hda_dsp_pcm_close(struct snd_sof_dev *sdev, substream->runtime->private_data = NULL; return 0; } +EXPORT_SYMBOL_NS(hda_dsp_pcm_close, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-probes.c b/sound/soc/sof/intel/hda-probes.c index 56a533c63cb00..be0fd56ddb276 100644 --- a/sound/soc/sof/intel/hda-probes.c +++ b/sound/soc/sof/intel/hda-probes.c @@ -139,10 +139,12 @@ int hda_probes_register(struct snd_sof_dev *sdev) return sof_client_dev_register(sdev, "hda-probes", 0, &hda_probes_ops, sizeof(hda_probes_ops)); } +EXPORT_SYMBOL_NS(hda_probes_register, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_probes_unregister(struct snd_sof_dev *sdev) { sof_client_dev_unregister(sdev, "hda-probes", 0); } +EXPORT_SYMBOL_NS(hda_probes_unregister, SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_CLIENT); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 4fef1964b5cdb..7a38bea663fe2 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -27,6 +27,7 @@ int sof_hda_position_quirk = SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS; module_param_named(position_quirk, sof_hda_position_quirk, int, 0444); MODULE_PARM_DESC(position_quirk, "SOF HDaudio position quirk"); +EXPORT_SYMBOL_NS(sof_hda_position_quirk, SND_SOC_SOF_INTEL_HDA_COMMON); #define HDA_LTRP_GB_VALUE_US 95 @@ -713,6 +714,7 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, return 0; } +EXPORT_SYMBOL_NS(hda_dsp_stream_hw_free, SND_SOC_SOF_INTEL_HDA_COMMON); bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) { @@ -735,6 +737,7 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) return ret; } +EXPORT_SYMBOL_NS(hda_dsp_check_stream_irq, SND_SOC_SOF_INTEL_HDA_COMMON); static void hda_dsp_compr_bytes_transferred(struct hdac_stream *hstream, int direction) @@ -831,6 +834,7 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) return IRQ_HANDLED; } +EXPORT_SYMBOL_NS(hda_dsp_stream_threaded_handler, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_stream_init(struct snd_sof_dev *sdev) { @@ -968,6 +972,7 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) return 0; } +EXPORT_SYMBOL_NS(hda_dsp_stream_init, SND_SOC_SOF_INTEL_HDA_COMMON); void hda_dsp_stream_free(struct snd_sof_dev *sdev) { @@ -997,6 +1002,7 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev) devm_kfree(sdev->dev, hda_stream); } } +EXPORT_SYMBOL_NS(hda_dsp_stream_free, SND_SOC_SOF_INTEL_HDA_COMMON); snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, int direction, bool can_sleep) @@ -1123,6 +1129,7 @@ u64 hda_dsp_get_stream_llp(struct snd_sof_dev *sdev, return merge_u64(llp_u, llp_l); } +EXPORT_SYMBOL_NS(hda_dsp_get_stream_llp, SND_SOC_SOF_INTEL_HDA_COMMON); /** * hda_dsp_get_stream_ldp - Retrieve the LDP (Linear DMA Position) of the stream @@ -1154,3 +1161,4 @@ u64 hda_dsp_get_stream_ldp(struct snd_sof_dev *sdev, return ((u64)ldp_u << 32) | ldp_l; } +EXPORT_SYMBOL_NS(hda_dsp_get_stream_ldp, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-trace.c b/sound/soc/sof/intel/hda-trace.c index cbb9bd7770e69..f0d959ba50c42 100644 --- a/sound/soc/sof/intel/hda-trace.c +++ b/sound/soc/sof/intel/hda-trace.c @@ -68,6 +68,7 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, return ret; } +EXPORT_SYMBOL_NS(hda_dsp_trace_init, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_trace_release(struct snd_sof_dev *sdev) { @@ -86,6 +87,7 @@ int hda_dsp_trace_release(struct snd_sof_dev *sdev) dev_dbg(sdev->dev, "DMA trace stream is not opened!\n"); return -ENODEV; } +EXPORT_SYMBOL_NS(hda_dsp_trace_release, SND_SOC_SOF_INTEL_HDA_COMMON); int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd) { @@ -93,3 +95,4 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd) return hda_dsp_stream_trigger(sdev, hda->dtrace_stream, cmd); } +EXPORT_SYMBOL_NS(hda_dsp_trace_trigger, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 87b0375250a25..87b2be025f40f 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -238,7 +238,7 @@ int hda_sdw_startup(struct snd_sof_dev *sdev) return sdw_intel_startup(hdev->sdw); } -EXPORT_SYMBOL_NS(hda_sdw_startup, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_sdw_startup, SND_SOC_SOF_INTEL_HDA_GENERIC); static int hda_sdw_exit(struct snd_sof_dev *sdev) { @@ -280,7 +280,7 @@ bool hda_common_check_sdw_irq(struct snd_sof_dev *sdev) out: return ret; } -EXPORT_SYMBOL_NS(hda_common_check_sdw_irq, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_common_check_sdw_irq, SND_SOC_SOF_INTEL_HDA_GENERIC); static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) { @@ -314,7 +314,7 @@ bool hda_sdw_check_wakeen_irq_common(struct snd_sof_dev *sdev) return false; } -EXPORT_SYMBOL_NS(hda_sdw_check_wakeen_irq_common, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_sdw_check_wakeen_irq_common, SND_SOC_SOF_INTEL_HDA_GENERIC); static bool hda_sdw_check_wakeen_irq(struct snd_sof_dev *sdev) { @@ -345,7 +345,7 @@ void hda_sdw_process_wakeen_common(struct snd_sof_dev *sdev) sdw_intel_process_wakeen_event(hdev->sdw); } -EXPORT_SYMBOL_NS(hda_sdw_process_wakeen_common, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_sdw_process_wakeen_common, SND_SOC_SOF_INTEL_HDA_GENERIC); #else /* IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) */ static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) @@ -418,7 +418,7 @@ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) /* re-enable clock gating and power gating */ return hda_dsp_ctrl_clock_power_gating(sdev, true); } -EXPORT_SYMBOL_NS(hda_dsp_post_fw_run, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_dsp_post_fw_run, SND_SOC_SOF_INTEL_HDA_GENERIC); /* * Debug @@ -807,7 +807,7 @@ int hda_dsp_probe_early(struct snd_sof_dev *sdev) err: return ret; } -EXPORT_SYMBOL_NS(hda_dsp_probe_early, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_dsp_probe_early, SND_SOC_SOF_INTEL_HDA_GENERIC); int hda_dsp_probe(struct snd_sof_dev *sdev) { @@ -964,7 +964,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) return ret; } -EXPORT_SYMBOL_NS(hda_dsp_probe, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_dsp_probe, SND_SOC_SOF_INTEL_HDA_GENERIC); void hda_dsp_remove(struct snd_sof_dev *sdev) { @@ -1018,7 +1018,7 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) if (!sdev->dspless_mode_selected) iounmap(sdev->bar[HDA_DSP_BAR]); } -EXPORT_SYMBOL_NS(hda_dsp_remove, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_dsp_remove, SND_SOC_SOF_INTEL_HDA_GENERIC); void hda_dsp_remove_late(struct snd_sof_dev *sdev) { @@ -1034,7 +1034,7 @@ int hda_power_down_dsp(struct snd_sof_dev *sdev) return hda_dsp_core_reset_power_down(sdev, chip->host_managed_cores_mask); } -EXPORT_SYMBOL_NS(hda_power_down_dsp, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_power_down_dsp, SND_SOC_SOF_INTEL_HDA_GENERIC); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) static void hda_generic_machine_select(struct snd_sof_dev *sdev, @@ -1504,7 +1504,7 @@ int hda_pci_intel_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return sof_pci_probe(pci, pci_id); } -EXPORT_SYMBOL_NS(hda_pci_intel_probe, SND_SOC_SOF_INTEL_HDA_COMMON); +EXPORT_SYMBOL_NS(hda_pci_intel_probe, SND_SOC_SOF_INTEL_HDA_GENERIC); int hda_register_clients(struct snd_sof_dev *sdev) { @@ -1525,4 +1525,5 @@ MODULE_IMPORT_NS(SND_INTEL_SOUNDWIRE_ACPI); MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT); MODULE_IMPORT_NS(SOUNDWIRE_INTEL); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_ACPI_INTEL_MATCH); diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c index 4b287b5e9077c..7551d4eb150df 100644 --- a/sound/soc/sof/intel/pci-apl.c +++ b/sound/soc/sof/intel/pci-apl.c @@ -105,5 +105,6 @@ static struct pci_driver snd_sof_pci_intel_apl_driver = { module_pci_driver(snd_sof_pci_intel_apl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c index 9fa0cd2eae79a..f2584a314711a 100644 --- a/sound/soc/sof/intel/pci-cnl.c +++ b/sound/soc/sof/intel/pci-cnl.c @@ -143,5 +143,6 @@ static struct pci_driver snd_sof_pci_intel_cnl_driver = { module_pci_driver(snd_sof_pci_intel_cnl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c index 70689b8b431b3..a2826073ad3b7 100644 --- a/sound/soc/sof/intel/pci-icl.c +++ b/sound/soc/sof/intel/pci-icl.c @@ -108,6 +108,7 @@ static struct pci_driver snd_sof_pci_intel_icl_driver = { module_pci_driver(snd_sof_pci_intel_icl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-lnl.c b/sound/soc/sof/intel/pci-lnl.c index 38ae649d1defa..5ad4978269830 100644 --- a/sound/soc/sof/intel/pci-lnl.c +++ b/sound/soc/sof/intel/pci-lnl.c @@ -70,6 +70,7 @@ static struct pci_driver snd_sof_pci_intel_lnl_driver = { module_pci_driver(snd_sof_pci_intel_lnl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_MTL); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK); diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index cacc985d80f41..ad26158083948 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -133,5 +133,6 @@ static struct pci_driver snd_sof_pci_intel_mtl_driver = { module_pci_driver(snd_sof_pci_intel_mtl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c index 9dde439a0b0f5..0bbfc4f55ff3d 100644 --- a/sound/soc/sof/intel/pci-skl.c +++ b/sound/soc/sof/intel/pci-skl.c @@ -89,5 +89,6 @@ static struct pci_driver snd_sof_pci_intel_skl_driver = { module_pci_driver(snd_sof_pci_intel_skl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index d7ecb892550f2..dc40ec8c62562 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -317,6 +317,7 @@ static struct pci_driver snd_sof_pci_intel_tgl_driver = { module_pci_driver(snd_sof_pci_intel_tgl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/tracepoints.c b/sound/soc/sof/intel/tracepoints.c index c223c96fe0e39..9e3260a062c21 100644 --- a/sound/soc/sof/intel/tracepoints.c +++ b/sound/soc/sof/intel/tracepoints.c @@ -1,3 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 #define CREATE_TRACE_POINTS #include + +EXPORT_TRACEPOINT_SYMBOL(sof_intel_hda_irq); From 1ae14f3520b1a0ad144610a3f592c81a3e81cd1b Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sun, 5 May 2024 20:23:45 +0800 Subject: [PATCH 457/603] ASoC: tas2781: Fix a warning reported by robot kernel test Fix a warning reported by robot kernel test that 'fw_entry' in function 'tas2781_load_calibration' is used uninitialized with compiler sh4-linux-gcc (GCC) 13.2.0, an update of copyright and a correction of the comments. Fixes: ef3bcde75d06 ("ASoc: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding Link: https://lore.kernel.org/r/20240505122346.1326-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-fmwlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 45760fe195237..a6be81adcb839 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1,8 +1,8 @@ // SPDX-License-Identifier: GPL-2.0 // -// tasdevice-fmw.c -- TASDEVICE firmware support +// tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 Texas Instruments, Inc. +// Copyright 2023 - 2024 Texas Instruments, Inc. // // Author: Shenghao Ding @@ -1878,7 +1878,7 @@ int tas2781_load_calibration(void *context, char *file_name, { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; struct tasdevice_fw *tas_fmw; struct firmware fmw; int offset = 0; From 6490bec6d5bf1001032c5efea94bdf5b5104bce9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Mon, 6 May 2024 14:11:06 +0200 Subject: [PATCH 458/603] ASoC: Intel: avs: boards: Properly name input device MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Machine boards expose input device for use with userspace. Current name in some cases is incorrect, fix it. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Link: https://lore.kernel.org/r/20240506121106.3792340-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/es8336.c | 2 +- sound/soc/intel/avs/boards/nau8825.c | 2 +- sound/soc/intel/avs/boards/rt274.c | 3 ++- sound/soc/intel/avs/boards/rt286.c | 4 ++-- sound/soc/intel/avs/boards/rt298.c | 4 ++-- 5 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index 5c90a60075773..b824e8dca9db0 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -113,7 +113,7 @@ static int avs_es8336_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, &data->jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 55db75efae414..4e249cae73990 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -96,7 +96,7 @@ static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) * 4 buttons here map to the google Reference headset. * The use of these buttons can be decided by the user space. */ - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack, pins, num_pins); if (ret) diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index 1cf5242160875..4120d4492b6e8 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -102,7 +102,8 @@ static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET, jack, pins, + num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index 4740bba105703..db554836c7a8e 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -63,8 +63,8 @@ static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 6e409e29f6974..c131da8f65cef 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -74,8 +74,8 @@ static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) if (!pins) return -ENOMEM; - ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, - pins, num_pins); + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_BTN_0, + jack, pins, num_pins); if (ret) return ret; From 293ad28116e2c3e4d3eb28bd0378558edc897f55 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:51 -0500 Subject: [PATCH 459/603] ASoC: SOF: Intel: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Reviewed-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/control.c | 2 +- sound/soc/sof/core.c | 2 +- sound/soc/sof/debug.c | 2 +- sound/soc/sof/fw-file-profile.c | 2 +- sound/soc/sof/intel/apl.c | 2 +- sound/soc/sof/intel/atom.c | 2 +- sound/soc/sof/intel/atom.h | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/intel/cnl.c | 2 +- sound/soc/sof/intel/ext_manifest.h | 2 +- sound/soc/sof/intel/hda-bus.c | 2 +- sound/soc/sof/intel/hda-codec.c | 2 +- sound/soc/sof/intel/hda-common-ops.c | 2 +- sound/soc/sof/intel/hda-ctrl.c | 2 +- sound/soc/sof/intel/hda-dai-ops.c | 2 +- sound/soc/sof/intel/hda-dai.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-ipc.c | 2 +- sound/soc/sof/intel/hda-ipc.h | 2 +- sound/soc/sof/intel/hda-loader-skl.c | 2 +- sound/soc/sof/intel/hda-loader.c | 2 +- sound/soc/sof/intel/hda-mlink.c | 2 +- sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/intel/hda-probes.c | 2 +- sound/soc/sof/intel/hda-stream.c | 2 +- sound/soc/sof/intel/hda-trace.c | 2 +- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/intel/icl.c | 2 +- sound/soc/sof/intel/lnl.c | 2 +- sound/soc/sof/intel/lnl.h | 2 +- sound/soc/sof/intel/mtl.c | 2 +- sound/soc/sof/intel/mtl.h | 2 +- sound/soc/sof/intel/pci-apl.c | 2 +- sound/soc/sof/intel/pci-cnl.c | 2 +- sound/soc/sof/intel/pci-icl.c | 2 +- sound/soc/sof/intel/pci-lnl.c | 2 +- sound/soc/sof/intel/pci-mtl.c | 2 +- sound/soc/sof/intel/pci-skl.c | 2 +- sound/soc/sof/intel/pci-tgl.c | 2 +- sound/soc/sof/intel/pci-tng.c | 2 +- sound/soc/sof/intel/shim.h | 2 +- sound/soc/sof/intel/skl.c | 2 +- sound/soc/sof/intel/telemetry.c | 2 +- sound/soc/sof/intel/telemetry.h | 2 +- sound/soc/sof/intel/tgl.c | 2 +- sound/soc/sof/iomem-utils.c | 2 +- sound/soc/sof/ipc.c | 2 +- sound/soc/sof/ipc3-control.c | 2 +- sound/soc/sof/ipc3-dtrace.c | 2 +- sound/soc/sof/ipc3-loader.c | 2 +- sound/soc/sof/ipc3-pcm.c | 2 +- sound/soc/sof/ipc3-priv.h | 2 +- sound/soc/sof/ipc3-topology.c | 2 +- sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/ipc4-control.c | 2 +- sound/soc/sof/ipc4-fw-reg.h | 2 +- sound/soc/sof/ipc4-loader.c | 2 +- sound/soc/sof/ipc4-mtrace.c | 2 +- sound/soc/sof/ipc4-pcm.c | 2 +- sound/soc/sof/ipc4-priv.h | 2 +- sound/soc/sof/ipc4-telemetry.c | 2 +- sound/soc/sof/ipc4-telemetry.h | 2 +- sound/soc/sof/ipc4-topology.c | 2 +- sound/soc/sof/ipc4-topology.h | 2 +- sound/soc/sof/ipc4.c | 2 +- sound/soc/sof/loader.c | 2 +- sound/soc/sof/nocodec.c | 2 +- sound/soc/sof/ops.c | 2 +- sound/soc/sof/ops.h | 2 +- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/pm.c | 2 +- sound/soc/sof/sof-acpi-dev.c | 2 +- sound/soc/sof/sof-acpi-dev.h | 2 +- sound/soc/sof/sof-audio.c | 2 +- sound/soc/sof/sof-audio.h | 2 +- sound/soc/sof/sof-client-ipc-flood-test.c | 2 +- sound/soc/sof/sof-client-ipc-kernel-injector.c | 2 +- sound/soc/sof/sof-client-ipc-msg-injector.c | 2 +- sound/soc/sof/sof-client-probes-ipc3.c | 2 +- sound/soc/sof/sof-client-probes-ipc4.c | 2 +- sound/soc/sof/sof-client-probes.c | 2 +- sound/soc/sof/sof-client.c | 2 +- sound/soc/sof/sof-pci-dev.c | 2 +- sound/soc/sof/sof-pci-dev.h | 2 +- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sof/sof-utils.c | 2 +- sound/soc/sof/sof-utils.h | 2 +- sound/soc/sof/stream-ipc.c | 2 +- sound/soc/sof/topology.c | 2 +- sound/soc/sof/trace.c | 2 +- sound/soc/sof/xtensa/core.c | 2 +- 93 files changed, 93 insertions(+), 93 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 75e13f4fd1eb8..463d418e72001 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 238bda5f6b76f..0a4917136ff97 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index a620d1bcd49da..d0ffa1d711450 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/fw-file-profile.c b/sound/soc/sof/fw-file-profile.c index b56b14232444c..1c0eb13ae5576 100644 --- a/sound/soc/sof/fw-file-profile.c +++ b/sound/soc/sof/fw-file-profile.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2023 Intel Corporation. All rights reserved. +// Copyright(c) 2023 Intel Corporation // #include diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index dee6c7f73e80a..68e1ede270eb8 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/atom.c b/sound/soc/sof/intel/atom.c index bd9789b483b17..86af4e9a716ef 100644 --- a/sound/soc/sof/intel/atom.c +++ b/sound/soc/sof/intel/atom.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2021 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2021 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/intel/atom.h b/sound/soc/sof/intel/atom.h index b965e5e080a67..20fb19102cb07 100644 --- a/sound/soc/sof/intel/atom.h +++ b/sound/soc/sof/intel/atom.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2017-2021 Intel Corporation. All rights reserved. + * Copyright(c) 2017-2021 Intel Corporation * * Author: Liam Girdwood */ diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 2e8926cd00f66..3262286a9a9d7 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 14a065999a895..d78d11d4cfbf6 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 85e1e4760d0e5..108a13ce5c5c6 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/ext_manifest.h b/sound/soc/sof/intel/ext_manifest.h index 2dfae9285d3c9..1ca19c6918522 100644 --- a/sound/soc/sof/intel/ext_manifest.h +++ b/sound/soc/sof/intel/ext_manifest.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2020 Intel Corporation. All rights reserved. + * Copyright(c) 2020 Intel Corporation */ /* diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index fc63085d2d743..96315613f7d95 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Keyon Jie diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6a13f38a8d210..da3db3ed379ec 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Keyon Jie // diff --git a/sound/soc/sof/intel/hda-common-ops.c b/sound/soc/sof/intel/hda-common-ops.c index 466769c68da55..79d1f8ca4f846 100644 --- a/sound/soc/sof/intel/hda-common-ops.c +++ b/sound/soc/sof/intel/hda-common-ops.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // /* diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 6d941209847f5..243dc6df911aa 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 4571442039994..484c761478853 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation #include #include diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 3f2fd84907d2f..ee159d239b407 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Keyon Jie // diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index ef5c915db8ffb..3ef9b98bcf639 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index a838dddb1d327..de0426ab8021f 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/hda-ipc.h b/sound/soc/sof/intel/hda-ipc.h index 8ec5e9f6f8d7a..ad9478b8c3906 100644 --- a/sound/soc/sof/intel/hda-ipc.h +++ b/sound/soc/sof/intel/hda-ipc.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2019 Intel Corporation. All rights reserved. + * Copyright(c) 2019 Intel Corporation * * Author: Keyon Jie */ diff --git a/sound/soc/sof/intel/hda-loader-skl.c b/sound/soc/sof/intel/hda-loader-skl.c index 1e77ca936f806..f38178c904dec 100644 --- a/sound/soc/sof/intel/hda-loader-skl.c +++ b/sound/soc/sof/intel/hda-loader-skl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2022 Intel Corporation // #include diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index dc88b7ea599ea..3065ebe62ed34 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/hda-mlink.c b/sound/soc/sof/intel/hda-mlink.c index b592e687a87a1..04bbc5c9904ce 100644 --- a/sound/soc/sof/intel/hda-mlink.c +++ b/sound/soc/sof/intel/hda-mlink.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // /* diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index d7b446f3f973e..d4628527165d7 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/hda-probes.c b/sound/soc/sof/intel/hda-probes.c index 56a533c63cb00..2c15975c2a72a 100644 --- a/sound/soc/sof/intel/hda-probes.c +++ b/sound/soc/sof/intel/hda-probes.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2019-2021 Intel Corporation. All rights reserved. +// Copyright(c) 2019-2021 Intel Corporation // // Author: Cezary Rojewski // Converted to SOF client: diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 91b1371745e66..0ff05b1006a79 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/hda-trace.c b/sound/soc/sof/intel/hda-trace.c index cbb9bd7770e69..fed4e4c08fdf6 100644 --- a/sound/soc/sof/intel/hda-trace.c +++ b/sound/soc/sof/intel/hda-trace.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index d509cfa38a642..72354748bedea 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Authors: Liam Girdwood // Ranjani Sridharan diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 7804c5b50c265..88d63ecc9f92e 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2017 Intel Corporation. All rights reserved. + * Copyright(c) 2017 Intel Corporation * * Author: Liam Girdwood */ diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c index 0406985919921..a8dde7b37b1b2 100644 --- a/sound/soc/sof/intel/icl.c +++ b/sound/soc/sof/intel/icl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Author: Fred Oh // diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 82c975823ab32..219c46ddbf6a2 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // -// Copyright(c) 2023 Intel Corporation. All rights reserved. +// Copyright(c) 2023 Intel Corporation /* * Hardware interface for audio DSP on LunarLake. diff --git a/sound/soc/sof/intel/lnl.h b/sound/soc/sof/intel/lnl.h index 4f4734fe7e089..79101af84b2e5 100644 --- a/sound/soc/sof/intel/lnl.h +++ b/sound/soc/sof/intel/lnl.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2024 Intel Corporation. All rights reserved. + * Copyright(c) 2024 Intel Corporation */ #ifndef __SOF_INTEL_LNL_H diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 5d3b2d4be78ae..9fd19ad3f03dc 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // Authors: Ranjani Sridharan // diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index d2d709fb4f069..7acaa7e724f44 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2020-2022 Intel Corporation. All rights reserved. + * Copyright(c) 2020-2022 Intel Corporation */ /* DSP Registers */ diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c index 4b287b5e9077c..0b076ff70d80c 100644 --- a/sound/soc/sof/intel/pci-apl.c +++ b/sound/soc/sof/intel/pci-apl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2021 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2021 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c index 9fa0cd2eae79a..fcaee9ec790de 100644 --- a/sound/soc/sof/intel/pci-cnl.c +++ b/sound/soc/sof/intel/pci-cnl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c index b99c7c9aad7dd..284e7c8732879 100644 --- a/sound/soc/sof/intel/pci-icl.c +++ b/sound/soc/sof/intel/pci-icl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2021 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2021 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/intel/pci-lnl.c b/sound/soc/sof/intel/pci-lnl.c index b14e508f1f315..6daa5acb8645b 100644 --- a/sound/soc/sof/intel/pci-lnl.c +++ b/sound/soc/sof/intel/pci-lnl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2023 Intel Corporation. All rights reserved. +// Copyright(c) 2023 Intel Corporation // // Author: Ranjani Sridharan // diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index cacc985d80f41..bd1590282f672 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2022 Intel Corporation // // Author: Ranjani Sridharan // diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c index 9dde439a0b0f5..d9f5f4a7adbdc 100644 --- a/sound/soc/sof/intel/pci-skl.c +++ b/sound/soc/sof/intel/pci-skl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2022 Intel Corporation // #include diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index a361ee9d1107f..aa0afe4112d07 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2021 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2021 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index d8a36d5a29f76..5c3069588bb77 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2021 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2021 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index 18ce3d33fa803..9d2de2e8f746c 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2017 Intel Corporation. All rights reserved. + * Copyright(c) 2017 Intel Corporation * * Author: Liam Girdwood */ diff --git a/sound/soc/sof/intel/skl.c b/sound/soc/sof/intel/skl.c index 93824e6ce5730..9a002811e9ffd 100644 --- a/sound/soc/sof/intel/skl.c +++ b/sound/soc/sof/intel/skl.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2022 Intel Corporation // /* diff --git a/sound/soc/sof/intel/telemetry.c b/sound/soc/sof/intel/telemetry.c index 1a3b5c28a6f04..5053075028a09 100644 --- a/sound/soc/sof/intel/telemetry.c +++ b/sound/soc/sof/intel/telemetry.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2023 Intel Corporation. All rights reserved. +// Copyright(c) 2023 Intel Corporation /* telemetry data queried from debug window */ diff --git a/sound/soc/sof/intel/telemetry.h b/sound/soc/sof/intel/telemetry.h index 3c2b23c75f5de..e4e91943a41ae 100644 --- a/sound/soc/sof/intel/telemetry.h +++ b/sound/soc/sof/intel/telemetry.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2023 Intel Corporation. All rights reserved. + * Copyright(c) 2023 Intel Corporation * * telemetry data in debug windows */ diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index 6cda47a78118a..d17c4e549c480 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Authors: Ranjani Sridharan // diff --git a/sound/soc/sof/iomem-utils.c b/sound/soc/sof/iomem-utils.c index 3f57f6cf65428..cd9cb54e7b23f 100644 --- a/sound/soc/sof/iomem-utils.c +++ b/sound/soc/sof/iomem-utils.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2022 Intel Corporation // // Author: Keyon Jie // diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index febe372f9aa87..3fb8d3e9dc6ae 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/ipc3-control.c b/sound/soc/sof/ipc3-control.c index a8deec7dc0214..2b1befad6d5c0 100644 --- a/sound/soc/sof/ipc3-control.c +++ b/sound/soc/sof/ipc3-control.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Intel Corporation. All rights reserved. +// Copyright(c) 2021 Intel Corporation // // diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 0dca139322f3d..744a91a150bc9 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // Author: Liam Girdwood diff --git a/sound/soc/sof/ipc3-loader.c b/sound/soc/sof/ipc3-loader.c index 6e3ef06721106..35b89c2b9d4c0 100644 --- a/sound/soc/sof/ipc3-loader.c +++ b/sound/soc/sof/ipc3-loader.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation #include #include "sof-priv.h" diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index af0bf354cb209..1c1b8f5953674 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Intel Corporation. All rights reserved. +// Copyright(c) 2021 Intel Corporation // // diff --git a/sound/soc/sof/ipc3-priv.h b/sound/soc/sof/ipc3-priv.h index 9cd7794f3cb69..866c5f67b91a7 100644 --- a/sound/soc/sof/ipc3-priv.h +++ b/sound/soc/sof/ipc3-priv.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2021 Intel Corporation. All rights reserved. + * Copyright(c) 2021 Intel Corporation */ #ifndef __SOUND_SOC_SOF_IPC3_PRIV_H diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index ab7f46a162da7..32c7d1f3b528d 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Intel Corporation. All rights reserved. +// Copyright(c) 2021 Intel Corporation // // diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index c03dd513fbff1..83c22d4a48304 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Intel Corporation. All rights reserved. +// Copyright(c) 2021 Intel Corporation // // diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c index 1be9519de909b..576f407cd456a 100644 --- a/sound/soc/sof/ipc4-control.c +++ b/sound/soc/sof/ipc4-control.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // diff --git a/sound/soc/sof/ipc4-fw-reg.h b/sound/soc/sof/ipc4-fw-reg.h index 7226161e57e17..7b85a364a6a6c 100644 --- a/sound/soc/sof/ipc4-fw-reg.h +++ b/sound/soc/sof/ipc4-fw-reg.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2022 Intel Corporation. All rights reserved. + * Copyright(c) 2022 Intel Corporation */ #ifndef __IPC4_FW_REG_H__ diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index 641c4f24cca98..bcdb33d036820 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation #include #include diff --git a/sound/soc/sof/ipc4-mtrace.c b/sound/soc/sof/ipc4-mtrace.c index 0e04bea9432dd..aa5b78604db69 100644 --- a/sound/soc/sof/ipc4-mtrace.c +++ b/sound/soc/sof/ipc4-mtrace.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation #include #include diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 4594470ed08b1..307bee63756b5 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // #include diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index e157ab80a1032..ea3323b90343d 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2022 Intel Corporation. All rights reserved. + * Copyright(c) 2022 Intel Corporation */ #ifndef __SOUND_SOC_SOF_IPC4_PRIV_H diff --git a/sound/soc/sof/ipc4-telemetry.c b/sound/soc/sof/ipc4-telemetry.c index ec4ae96743643..ddc3bc494ffef 100644 --- a/sound/soc/sof/ipc4-telemetry.c +++ b/sound/soc/sof/ipc4-telemetry.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2023 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2023 Intel Corporation // #include diff --git a/sound/soc/sof/ipc4-telemetry.h b/sound/soc/sof/ipc4-telemetry.h index ab3599e3d87da..9298f8acc648b 100644 --- a/sound/soc/sof/ipc4-telemetry.h +++ b/sound/soc/sof/ipc4-telemetry.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2023 Intel Corporation. All rights reserved. + * Copyright(c) 2023 Intel Corporation */ #ifndef __SOUND_SOC_SOF_IPC4_TELEMETRY_H diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index a972b2b5122a3..b85715d33c865 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // #include diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 6e33208a357f5..4488762f6a71b 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2022 Intel Corporation. All rights reserved. + * Copyright(c) 2022 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_IPC4_TOPOLOGY_H__ diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index ac5c6bc66d2ab..4386cbae16d4e 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // Authors: Rander Wang // Peter Ujfalusi diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2f8555f11c036..0baf316b0064b 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index 34aa8a7cfc7db..fdcbe33d3dcfb 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/ops.c b/sound/soc/sof/ops.c index ff066de4ceb94..bd52e7ec68836 100644 --- a/sound/soc/sof/ops.c +++ b/sound/soc/sof/ops.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 3cd748e134609..2584621c3b2d4 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation * * Author: Liam Girdwood */ diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index cb774e011e31a..baad4c1445aa7 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 704b21413c719..8e3bcf602beb3 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/sof-acpi-dev.c b/sound/soc/sof/sof-acpi-dev.c index 2977f0a63fba5..2d96d00f1c449 100644 --- a/sound/soc/sof/sof-acpi-dev.c +++ b/sound/soc/sof/sof-acpi-dev.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/sof-acpi-dev.h b/sound/soc/sof/sof-acpi-dev.h index 9bf8f75ceaaed..89adfa5070357 100644 --- a/sound/soc/sof/sof-acpi-dev.h +++ b/sound/soc/sof/sof-acpi-dev.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2021 Intel Corporation. All rights reserved. + * Copyright(c) 2021 Intel Corporation */ #ifndef __SOUND_SOC_SOF_ACPI_H diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 32fef64ef10dd..b3ac040811e79 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2019 Intel Corporation. All rights reserved. +// Copyright(c) 2019 Intel Corporation // // Author: Ranjani Sridharan // diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 2aeb269af752a..ec2a3bb644d2b 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2019 Intel Corporation. All rights reserved. + * Copyright(c) 2019 Intel Corporation * * Author: Ranjani Sridharan */ diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c index 1b2e9e25a8366..4356149260925 100644 --- a/sound/soc/sof/sof-client-ipc-flood-test.c +++ b/sound/soc/sof/sof-client-ipc-flood-test.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // Authors: Ranjani Sridharan // Peter Ujfalusi diff --git a/sound/soc/sof/sof-client-ipc-kernel-injector.c b/sound/soc/sof/sof-client-ipc-kernel-injector.c index ad0ed2d570a91..6973b6690df4b 100644 --- a/sound/soc/sof/sof-client-ipc-kernel-injector.c +++ b/sound/soc/sof/sof-client-ipc-kernel-injector.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2023 Google Inc. All rights reserved. +// Copyright(c) 2023 Google Inc // // Author: Curtis Malainey // diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index e249d3a9afb59..af22e6421029c 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // Author: Peter Ujfalusi // diff --git a/sound/soc/sof/sof-client-probes-ipc3.c b/sound/soc/sof/sof-client-probes-ipc3.c index 5e8eb19582a86..816df745c9afc 100644 --- a/sound/soc/sof/sof-client-probes-ipc3.c +++ b/sound/soc/sof/sof-client-probes-ipc3.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2019-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2019-2022 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/sof/sof-client-probes-ipc4.c b/sound/soc/sof/sof-client-probes-ipc4.c index c56a85854d92c..796eac0a2e74f 100644 --- a/sound/soc/sof/sof-client-probes-ipc4.c +++ b/sound/soc/sof/sof-client-probes-ipc4.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2019-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2019-2022 Intel Corporation // // Author: Jyri Sarha // diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 30f771ac7bbf9..b8f297307565c 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2019-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2019-2022 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/sof/sof-client.c b/sound/soc/sof/sof-client.c index 54dca91255a0a..99f74def4ab60 100644 --- a/sound/soc/sof/sof-client.c +++ b/sound/soc/sof/sof-client.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // Authors: Ranjani Sridharan // Peter Ujfalusi diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index aab5c900cecf8..4365405783e61 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/sof-pci-dev.h b/sound/soc/sof/sof-pci-dev.h index 81155a59e63ae..c90e6276c83b0 100644 --- a/sound/soc/sof/sof-pci-dev.h +++ b/sound/soc/sof/sof-pci-dev.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2021 Intel Corporation. All rights reserved. + * Copyright(c) 2021 Intel Corporation */ #ifndef __SOUND_SOC_SOF_PCI_H diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index d3c436f826046..4d6a1517f9b35 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation * * Author: Liam Girdwood */ diff --git a/sound/soc/sof/sof-utils.c b/sound/soc/sof/sof-utils.c index b6345a7345af7..cad041bf56ccf 100644 --- a/sound/soc/sof/sof-utils.c +++ b/sound/soc/sof/sof-utils.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2018-2022 Intel Corporation // // Author: Keyon Jie // diff --git a/sound/soc/sof/sof-utils.h b/sound/soc/sof/sof-utils.h index 6f902893807e6..9ac6de9a6d6a3 100644 --- a/sound/soc/sof/sof-utils.h +++ b/sound/soc/sof/sof-utils.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2022 Intel Corporation. All rights reserved. + * Copyright(c) 2022 Intel Corporation */ #ifndef __SOC_SOF_UTILS_H diff --git a/sound/soc/sof/stream-ipc.c b/sound/soc/sof/stream-ipc.c index 216b454f6b94e..eb71303aa24c6 100644 --- a/sound/soc/sof/stream-ipc.c +++ b/sound/soc/sof/stream-ipc.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2019 Intel Corporation. All rights reserved. +// Copyright(c) 2019 Intel Corporation // // Authors: Guennadi Liakhovetski diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index ec931a26b54f0..da182314aa874 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Liam Girdwood // diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index b2ab51e5214a9..fe6c67c01b0d7 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation #include "sof-priv.h" diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c index 7c91a919eadc7..ccbc3fcdadd5e 100644 --- a/sound/soc/sof/xtensa/core.c +++ b/sound/soc/sof/xtensa/core.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2018 Intel Corporation. All rights reserved. +// Copyright(c) 2018 Intel Corporation // // Author: Pan Xiuli // From 0e0440535de8e7c5c0ae0bd469b6ae184f9c732c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:52 -0500 Subject: [PATCH 460/603] ASoC: Intel: boards: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Acked-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bdw_rt286.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/hsw_rt5640.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 ++-- sound/soc/intel/boards/skl_rt286.c | 4 ++-- sound/soc/intel/boards/sof_board_helpers.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 2 +- sound/soc/intel/boards/sof_ssp_amp.c | 2 +- sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 2 +- 15 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c index 7f20159c23e54..58db09d9b6e16 100644 --- a/sound/soc/intel/boards/bdw_rt286.c +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -2,7 +2,7 @@ /* * Sound card driver for Intel Broadwell Wildcat Point with Realtek 286 * - * Copyright (C) 2013, Intel Corporation. All rights reserved. + * Copyright (C) 2013, Intel Corporation */ #include diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 57e75f87bea62..e1082bfe5ca9a 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -2,7 +2,7 @@ /* * Intel Broxton-P I2S Machine Driver * - * Copyright (C) 2016, Intel Corporation. All rights reserved. + * Copyright (C) 2016, Intel Corporation * * Modified from: * Intel Skylake I2S Machine driver diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index afc499be8db26..dce6a2086f2a4 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -2,7 +2,7 @@ /* * Intel Broxton-P I2S Machine Driver * - * Copyright (C) 2014-2016, Intel Corporation. All rights reserved. + * Copyright (C) 2014-2016, Intel Corporation * * Modified from: * Intel Skylake I2S Machine driver diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c index 04b7d4f7f9e24..fda5a92b00063 100644 --- a/sound/soc/intel/boards/hda_dsp_common.c +++ b/sound/soc/intel/boards/hda_dsp_common.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2019 Intel Corporation. All rights reserved. +// Copyright(c) 2019 Intel Corporation #include #include diff --git a/sound/soc/intel/boards/hsw_rt5640.c b/sound/soc/intel/boards/hsw_rt5640.c index 2a2fe27dff0e8..1826a4dfd0f35 100644 --- a/sound/soc/intel/boards/hsw_rt5640.c +++ b/sound/soc/intel/boards/hsw_rt5640.c @@ -2,7 +2,7 @@ /* * Sound card driver for Intel Haswell Lynx Point with Realtek 5640 * - * Copyright (C) 2013, Intel Corporation. All rights reserved. + * Copyright (C) 2013, Intel Corporation */ #include diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 646e8ff8e9619..e16c42e81ecaa 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -3,7 +3,7 @@ * Intel Kabylake I2S Machine Driver with MAXIM98927 * and RT5663 Codecs * - * Copyright (C) 2017, Intel Corporation. All rights reserved. + * Copyright (C) 2017, Intel Corporation * * Modified from: * Intel Skylake I2S Machine driver diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 924d5d1de03ac..a9501cd106ff2 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -3,7 +3,7 @@ * Intel Kabylake I2S Machine Driver with MAXIM98927 * RT5514 and RT5663 Codecs * - * Copyright (C) 2017, Intel Corporation. All rights reserved. + * Copyright (C) 2017, Intel Corporation * * Modified from: * Intel Kabylake I2S Machine driver supporting MAXIM98927 and diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index e4630c33176e2..91fe9834aab42 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -3,7 +3,7 @@ * Intel Skylake I2S Machine Driver with MAXIM98357A * and NAU88L25 * - * Copyright (C) 2015, Intel Corporation. All rights reserved. + * Copyright (C) 2015, Intel Corporation */ #include diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index fadc25a536b4b..d53bf3516c0d3 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -2,12 +2,12 @@ /* * Intel Skylake I2S Machine Driver for NAU88L25+SSM4567 * - * Copyright (C) 2015, Intel Corporation. All rights reserved. + * Copyright (C) 2015, Intel Corporation * * Modified from: * Intel Skylake I2S Machine Driver for NAU88L25 and SSM4567 * - * Copyright (C) 2015, Intel Corporation. All rights reserved. + * Copyright (C) 2015, Intel Corporation */ #include diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 9a80442749081..3ea03f8144036 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -2,12 +2,12 @@ /* * Intel Skylake I2S Machine Driver * - * Copyright (C) 2014-2015, Intel Corporation. All rights reserved. + * Copyright (C) 2014-2015, Intel Corporation * * Modified from: * Intel Broadwell Wildcatpoint SST Audio * - * Copyright (C) 2013, Intel Corporation. All rights reserved. + * Copyright (C) 2013, Intel Corporation */ #include diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index eb140e13153f8..7519c545cbe29 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2023 Intel Corporation. All rights reserved. +// Copyright(c) 2023 Intel Corporation #include #include "../common/soc-intel-quirks.h" diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index bb12aab15ad5e..7e1f485dce7ab 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation #include #include #include diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index cb1c21d822f6c..6915ff8b8716b 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation #include #include diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 206c9b7238059..f51f1008e0169 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation /* * sof_ssp_amp.c - ASoc Machine driver for Intel platforms diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c index a887f019afe25..75d0b931d895d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c +++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2023 Intel Corporation. All rights reserved. +// Copyright(c) 2023 Intel Corporation #include #include From 5cf4ffa4113da2e797c8281bac2838d29f5a03bf Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:53 -0500 Subject: [PATCH 461/603] ASoC: Intel: common: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Acked-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 2 +- sound/soc/intel/common/sst-dsp-priv.h | 2 +- sound/soc/intel/common/sst-dsp.c | 2 +- sound/soc/intel/common/sst-dsp.h | 2 +- sound/soc/intel/common/sst-ipc.c | 2 +- sound/soc/intel/common/sst-ipc.h | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 0318c1a46f3c6..0c08859c4773c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -2,7 +2,7 @@ /* * soc-acpi-intel-lnl-match.c - tables and support for LNL ACPI enumeration. * - * Copyright (c) 2023, Intel Corporation. All rights reserved. + * Copyright (c) 2023, Intel Corporation * */ diff --git a/sound/soc/intel/common/sst-dsp-priv.h b/sound/soc/intel/common/sst-dsp-priv.h index de2b44568feb4..de32bb9afccbd 100644 --- a/sound/soc/intel/common/sst-dsp-priv.h +++ b/sound/soc/intel/common/sst-dsp-priv.h @@ -2,7 +2,7 @@ /* * Intel Smart Sound Technology * - * Copyright (C) 2013, Intel Corporation. All rights reserved. + * Copyright (C) 2013, Intel Corporation */ #ifndef __SOUND_SOC_SST_DSP_PRIV_H diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index 229d09d61afbd..cdd2f7cf50ae5 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -2,7 +2,7 @@ /* * Intel Smart Sound Technology (SST) DSP Core Driver * - * Copyright (C) 2013, Intel Corporation. All rights reserved. + * Copyright (C) 2013, Intel Corporation */ #include diff --git a/sound/soc/intel/common/sst-dsp.h b/sound/soc/intel/common/sst-dsp.h index f111fd3f334b1..998b1a0522816 100644 --- a/sound/soc/intel/common/sst-dsp.h +++ b/sound/soc/intel/common/sst-dsp.h @@ -2,7 +2,7 @@ /* * Intel Smart Sound Technology (SST) Core * - * Copyright (C) 2013, Intel Corporation. All rights reserved. + * Copyright (C) 2013, Intel Corporation */ #ifndef __SOUND_SOC_SST_DSP_H diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 89c10724d2a30..6b2c83f9f0107 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -2,7 +2,7 @@ /* * Intel SST generic IPC Support * - * Copyright (C) 2015, Intel Corporation. All rights reserved. + * Copyright (C) 2015, Intel Corporation */ #include diff --git a/sound/soc/intel/common/sst-ipc.h b/sound/soc/intel/common/sst-ipc.h index 77d651e888f91..86d44ceadc926 100644 --- a/sound/soc/intel/common/sst-ipc.h +++ b/sound/soc/intel/common/sst-ipc.h @@ -2,7 +2,7 @@ /* * Intel SST generic IPC Support * - * Copyright (C) 2015, Intel Corporation. All rights reserved. + * Copyright (C) 2015, Intel Corporation */ #ifndef __SST_GENERIC_IPC_H From 884077bd7110db61eddff086f9ab7f2c6748d169 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:54 -0500 Subject: [PATCH 462/603] ASoC: codecs: Intel: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Acked-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/hda-dai.c | 2 +- sound/soc/codecs/hda.c | 2 +- sound/soc/codecs/hda.h | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hda-dai.c b/sound/soc/codecs/hda-dai.c index 7bd7ddcd810f2..b9caae7e48179 100644 --- a/sound/soc/codecs/hda-dai.c +++ b/sound/soc/codecs/hda-dai.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index 5a58723dc0e96..ddc00927313cf 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/codecs/hda.h b/sound/soc/codecs/hda.h index 78a2be4945b13..59308cc6afef4 100644 --- a/sound/soc/codecs/hda.h +++ b/sound/soc/codecs/hda.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* - * Copyright(c) 2021-2022 Intel Corporation. All rights reserved. + * Copyright(c) 2021-2022 Intel Corporation * * Author: Cezary Rojewski */ From 618ae0d7e740d212044ba25d0f1013374eda448a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:55 -0500 Subject: [PATCH 463/603] ASoC: Intel: catpt: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Acked-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/catpt/core.h | 2 +- sound/soc/intel/catpt/device.c | 2 +- sound/soc/intel/catpt/dsp.c | 2 +- sound/soc/intel/catpt/ipc.c | 2 +- sound/soc/intel/catpt/loader.c | 2 +- sound/soc/intel/catpt/messages.c | 2 +- sound/soc/intel/catpt/messages.h | 2 +- sound/soc/intel/catpt/pcm.c | 2 +- sound/soc/intel/catpt/registers.h | 2 +- sound/soc/intel/catpt/sysfs.c | 2 +- sound/soc/intel/catpt/trace.h | 2 +- 11 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/catpt/core.h b/sound/soc/intel/catpt/core.h index a64a0a77dcb77..c01d27e9fd88f 100644 --- a/sound/soc/intel/catpt/core.h +++ b/sound/soc/intel/catpt/core.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2020 Intel Corporation. All rights reserved. + * Copyright(c) 2020 Intel Corporation * * Author: Cezary Rojewski */ diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index cac3dffbd0d9f..2e1fa79a04d4e 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c index 5454c6d9ab5b1..5993819cc58a2 100644 --- a/sound/soc/intel/catpt/dsp.c +++ b/sound/soc/intel/catpt/dsp.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/catpt/ipc.c b/sound/soc/intel/catpt/ipc.c index 5b718a846fda5..d26863249097f 100644 --- a/sound/soc/intel/catpt/ipc.c +++ b/sound/soc/intel/catpt/ipc.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/catpt/loader.c b/sound/soc/intel/catpt/loader.c index ff7b8f0d34ac7..696d84314eeb5 100644 --- a/sound/soc/intel/catpt/loader.c +++ b/sound/soc/intel/catpt/loader.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/catpt/messages.c b/sound/soc/intel/catpt/messages.c index a793d114afa4f..30eec2de4dc13 100644 --- a/sound/soc/intel/catpt/messages.c +++ b/sound/soc/intel/catpt/messages.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/catpt/messages.h b/sound/soc/intel/catpt/messages.h index c17e948d9f52f..a634943eb6693 100644 --- a/sound/soc/intel/catpt/messages.h +++ b/sound/soc/intel/catpt/messages.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2020 Intel Corporation. All rights reserved. + * Copyright(c) 2020 Intel Corporation * * Author: Cezary Rojewski */ diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index 3daf5eb37f7ba..81a2f0339e055 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/catpt/registers.h b/sound/soc/intel/catpt/registers.h index 47280d82842eb..6c1ad28c6d692 100644 --- a/sound/soc/intel/catpt/registers.h +++ b/sound/soc/intel/catpt/registers.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2020 Intel Corporation. All rights reserved. + * Copyright(c) 2020 Intel Corporation * * Author: Cezary Rojewski */ diff --git a/sound/soc/intel/catpt/sysfs.c b/sound/soc/intel/catpt/sysfs.c index 9b6d2d93a2e7e..936ac9d503ff3 100644 --- a/sound/soc/intel/catpt/sysfs.c +++ b/sound/soc/intel/catpt/sysfs.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2020 Intel Corporation. All rights reserved. +// Copyright(c) 2020 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/catpt/trace.h b/sound/soc/intel/catpt/trace.h index bb3d627dbeafe..010f57b6a7a89 100644 --- a/sound/soc/intel/catpt/trace.h +++ b/sound/soc/intel/catpt/trace.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2020 Intel Corporation. All rights reserved. + * Copyright(c) 2020 Intel Corporation * * Author: Cezary Rojewski */ From 94001147a09ffeaf0657db7c189af77cda427f30 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:56 -0500 Subject: [PATCH 464/603] ASoC: Intel: avs: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Acked-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/apl.c | 2 +- sound/soc/intel/avs/avs.h | 2 +- sound/soc/intel/avs/board_selection.c | 2 +- sound/soc/intel/avs/boards/da7219.c | 2 +- sound/soc/intel/avs/boards/dmic.c | 2 +- sound/soc/intel/avs/boards/es8336.c | 2 +- sound/soc/intel/avs/boards/hdaudio.c | 2 +- sound/soc/intel/avs/boards/i2s_test.c | 2 +- sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 +- sound/soc/intel/avs/boards/max98927.c | 2 +- sound/soc/intel/avs/boards/nau8825.c | 2 +- sound/soc/intel/avs/boards/probe.c | 2 +- sound/soc/intel/avs/boards/rt274.c | 2 +- sound/soc/intel/avs/boards/rt286.c | 2 +- sound/soc/intel/avs/boards/rt298.c | 2 +- sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 +- sound/soc/intel/avs/boards/rt5682.c | 2 +- sound/soc/intel/avs/boards/ssm4567.c | 2 +- sound/soc/intel/avs/cldma.c | 2 +- sound/soc/intel/avs/cldma.h | 2 +- sound/soc/intel/avs/cnl.c | 2 +- sound/soc/intel/avs/control.c | 2 +- sound/soc/intel/avs/control.h | 2 +- sound/soc/intel/avs/core.c | 2 +- sound/soc/intel/avs/debugfs.c | 2 +- sound/soc/intel/avs/dsp.c | 2 +- sound/soc/intel/avs/icl.c | 2 +- sound/soc/intel/avs/ipc.c | 2 +- sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/messages.c | 2 +- sound/soc/intel/avs/messages.h | 2 +- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/avs/path.h | 2 +- sound/soc/intel/avs/pcm.c | 2 +- sound/soc/intel/avs/probes.c | 2 +- sound/soc/intel/avs/registers.h | 2 +- sound/soc/intel/avs/skl.c | 2 +- sound/soc/intel/avs/sysfs.c | 2 +- sound/soc/intel/avs/tgl.c | 2 +- sound/soc/intel/avs/topology.c | 2 +- sound/soc/intel/avs/topology.h | 2 +- sound/soc/intel/avs/trace.c | 2 +- sound/soc/intel/avs/utils.c | 2 +- sound/soc/intel/avs/utils.h | 2 +- 46 files changed, 46 insertions(+), 46 deletions(-) diff --git a/sound/soc/intel/avs/apl.c b/sound/soc/intel/avs/apl.c index bf97e4e428a49..27516ef571859 100644 --- a/sound/soc/intel/avs/apl.c +++ b/sound/soc/intel/avs/apl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index 101b1902a7153..eca6ec0428bbb 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2021-2022 Intel Corporation. All rights reserved. + * Copyright(c) 2021-2022 Intel Corporation * * Authors: Cezary Rojewski * Amadeusz Slawinski diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 290ea314ace82..0266edeafc19d 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index fc072dc58968c..80c0a1a956542 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c index d9e5e85f52335..a31aa471a1c21 100644 --- a/sound/soc/intel/avs/boards/dmic.c +++ b/sound/soc/intel/avs/boards/dmic.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index ff3bd15132694..e19ad8deed5c3 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2023 Intel Corporation. All rights reserved. +// Copyright(c) 2023 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c index 4433175814f80..430c070a1a0ee 100644 --- a/sound/soc/intel/avs/boards/hdaudio.c +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index d419556856643..7e6c8d9c900bd 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index 1ff85e4d8e160..8d550e82b46a8 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c index 8d31586b73eae..fdef5a008daff 100644 --- a/sound/soc/intel/avs/boards/max98373.c +++ b/sound/soc/intel/avs/boards/max98373.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c index 572ec58073d06..082f311d8b840 100644 --- a/sound/soc/intel/avs/boards/max98927.c +++ b/sound/soc/intel/avs/boards/max98927.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022 Intel Corporation. All rights reserved. +// Copyright(c) 2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index 3164745b15167..8dee6801fe516 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c index 8be6887bbc6e8..1cdc285ab8101 100644 --- a/sound/soc/intel/avs/boards/probe.c +++ b/sound/soc/intel/avs/boards/probe.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index e3aa28780df5a..68d6fcb215bb7 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index d24316fc4db9c..a3d0c5da5dfe5 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index 7fd28544f7869..f819f8dd447ba 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c index 097ae5f73241e..cfa146b6cf087 100644 --- a/sound/soc/intel/avs/boards/rt5514.c +++ b/sound/soc/intel/avs/boards/rt5514.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2023 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2023 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c index ba8c42d026da2..44f857e909691 100644 --- a/sound/soc/intel/avs/boards/rt5663.c +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2022-2023 Intel Corporation. All rights reserved. +// Copyright(c) 2022-2023 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 823f04dba2f76..0dcc6392a0cc8 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index 9e78a3b912087..63bbfc30f35e9 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/cldma.c b/sound/soc/intel/avs/cldma.c index 945ea376d0312..61326d7059b1c 100644 --- a/sound/soc/intel/avs/cldma.c +++ b/sound/soc/intel/avs/cldma.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Author: Cezary Rojewski // diff --git a/sound/soc/intel/avs/cldma.h b/sound/soc/intel/avs/cldma.h index 7d95e2747f527..7f9b2b1c566e3 100644 --- a/sound/soc/intel/avs/cldma.h +++ b/sound/soc/intel/avs/cldma.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2021-2022 Intel Corporation. All rights reserved. + * Copyright(c) 2021-2022 Intel Corporation * * Author: Cezary Rojewski */ diff --git a/sound/soc/intel/avs/cnl.c b/sound/soc/intel/avs/cnl.c index 0d03e1e03c112..bd3c4bb8bf5a1 100644 --- a/sound/soc/intel/avs/cnl.c +++ b/sound/soc/intel/avs/cnl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2024 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/control.c b/sound/soc/intel/avs/control.c index 3dfa2e9816db0..dc7dc45e0a0a9 100644 --- a/sound/soc/intel/avs/control.c +++ b/sound/soc/intel/avs/control.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Amadeusz Slawinski // Cezary Rojewski diff --git a/sound/soc/intel/avs/control.h b/sound/soc/intel/avs/control.h index 08631bde13c34..d9fac3569e8de 100644 --- a/sound/soc/intel/avs/control.h +++ b/sound/soc/intel/avs/control.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2021-2022 Intel Corporation. All rights reserved. + * Copyright(c) 2021-2022 Intel Corporation * * Authors: Amadeusz Slawinski * Cezary Rojewski diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index f56f9e19276ae..d9816d0701a8e 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/debugfs.c b/sound/soc/intel/avs/debugfs.c index 4dfbff0ce5083..3fc2bbb633691 100644 --- a/sound/soc/intel/avs/debugfs.c +++ b/sound/soc/intel/avs/debugfs.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c index aa03af4473e94..7b47e52c2b395 100644 --- a/sound/soc/intel/avs/dsp.c +++ b/sound/soc/intel/avs/dsp.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/icl.c b/sound/soc/intel/avs/icl.c index 3a96c4304ca65..f8d327ea2656e 100644 --- a/sound/soc/intel/avs/icl.c +++ b/sound/soc/intel/avs/icl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2024 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/ipc.c b/sound/soc/intel/avs/ipc.c index f9b302215c378..4fba46e77c470 100644 --- a/sound/soc/intel/avs/ipc.c +++ b/sound/soc/intel/avs/ipc.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c index c255c898b7a86..890efd2f1feab 100644 --- a/sound/soc/intel/avs/loader.c +++ b/sound/soc/intel/avs/loader.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/messages.c b/sound/soc/intel/avs/messages.c index f874e4f0d95f4..ec458bd51b10b 100644 --- a/sound/soc/intel/avs/messages.c +++ b/sound/soc/intel/avs/messages.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/messages.h b/sound/soc/intel/avs/messages.h index 285d89607b6a8..d0bdb7d9266c4 100644 --- a/sound/soc/intel/avs/messages.h +++ b/sound/soc/intel/avs/messages.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2021-2022 Intel Corporation. All rights reserved. + * Copyright(c) 2021-2022 Intel Corporation * * Authors: Cezary Rojewski * Amadeusz Slawinski diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 0a34d63e262eb..f31d5e2caa7b0 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021 Intel Corporation. All rights reserved. +// Copyright(c) 2021 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h index 657f7b093e805..bfd253c9fa951 100644 --- a/sound/soc/intel/avs/path.h +++ b/sound/soc/intel/avs/path.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2021 Intel Corporation. All rights reserved. + * Copyright(c) 2021 Intel Corporation * * Authors: Cezary Rojewski * Amadeusz Slawinski diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 845b5ed9eb1ba..88e7118750048 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c index 7e781a3156909..f0b0109563038 100644 --- a/sound/soc/intel/avs/probes.c +++ b/sound/soc/intel/avs/probes.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/registers.h b/sound/soc/intel/avs/registers.h index 6126adca500ce..f76e91cff2a9a 100644 --- a/sound/soc/intel/avs/registers.h +++ b/sound/soc/intel/avs/registers.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2021-2022 Intel Corporation. All rights reserved. + * Copyright(c) 2021-2022 Intel Corporation * * Authors: Cezary Rojewski * Amadeusz Slawinski diff --git a/sound/soc/intel/avs/skl.c b/sound/soc/intel/avs/skl.c index fceed353449a3..34f859d6e5a49 100644 --- a/sound/soc/intel/avs/skl.c +++ b/sound/soc/intel/avs/skl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/sysfs.c b/sound/soc/intel/avs/sysfs.c index cce21636fbc02..74b2e6f38d76e 100644 --- a/sound/soc/intel/avs/sysfs.c +++ b/sound/soc/intel/avs/sysfs.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2024 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/tgl.c b/sound/soc/intel/avs/tgl.c index b985a8299b721..a9019ff5e3afb 100644 --- a/sound/soc/intel/avs/tgl.c +++ b/sound/soc/intel/avs/tgl.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2024 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2024 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 42b42903ae9de..02bae207f6ece 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021 Intel Corporation. All rights reserved. +// Copyright(c) 2021 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/topology.h b/sound/soc/intel/avs/topology.h index 6a59dd766603f..7892e3797f63c 100644 --- a/sound/soc/intel/avs/topology.h +++ b/sound/soc/intel/avs/topology.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2021 Intel Corporation. All rights reserved. + * Copyright(c) 2021 Intel Corporation * * Authors: Cezary Rojewski * Amadeusz Slawinski diff --git a/sound/soc/intel/avs/trace.c b/sound/soc/intel/avs/trace.c index c63eea909b5e8..a98da521db0fd 100644 --- a/sound/soc/intel/avs/trace.c +++ b/sound/soc/intel/avs/trace.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Author: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/utils.c b/sound/soc/intel/avs/utils.c index 8100c2fa0a7ed..81f9b67d8e293 100644 --- a/sound/soc/intel/avs/utils.c +++ b/sound/soc/intel/avs/utils.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0-only // -// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// Copyright(c) 2021-2022 Intel Corporation // // Authors: Cezary Rojewski // Amadeusz Slawinski diff --git a/sound/soc/intel/avs/utils.h b/sound/soc/intel/avs/utils.h index 0b82a98ed0245..5ee569c39380a 100644 --- a/sound/soc/intel/avs/utils.h +++ b/sound/soc/intel/avs/utils.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2023 Intel Corporation. All rights reserved. + * Copyright(c) 2023 Intel Corporation * * Authors: Cezary Rojewski * Amadeusz Slawinski From 83e495d7b29b663861c68a1ad77710b6315472ad Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:57 -0500 Subject: [PATCH 465/603] ASoC: soc-topology-test: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Reviewed-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index 70cbccc42a42a..02aa2448f0e67 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -2,7 +2,7 @@ /* * soc-topology-test.c -- ALSA SoC Topology Kernel Unit Tests * - * Copyright(c) 2021 Intel Corporation. All rights reserved. + * Copyright(c) 2021 Intel Corporation. */ #include From 77678a25d1ecf70dc1d7ea2c0ab7609af15b83d3 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:10 +0200 Subject: [PATCH 466/603] ASoC: qcom: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-1-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/apq8016_sbc.c | 4 ++-- sound/soc/qcom/qdsp6/q6apm-dai.c | 2 +- sound/soc/qcom/sc7180.c | 10 +++++----- sound/soc/qcom/sc7280.c | 12 ++++++------ sound/soc/qcom/sc8280xp.c | 8 ++++---- sound/soc/qcom/sdw.c | 8 ++++---- sound/soc/qcom/sm8250.c | 10 +++++----- sound/soc/qcom/x1e80100.c | 8 ++++---- 8 files changed, 31 insertions(+), 31 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 4834a56eaa88a..3023cf180a755 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -192,7 +192,7 @@ static int msm8916_qdsp6_dai_init(struct snd_soc_pcm_runtime *rtd) static int msm8916_qdsp6_startup(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); @@ -213,7 +213,7 @@ static int msm8916_qdsp6_startup(struct snd_pcm_substream *substream) static void msm8916_qdsp6_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 00bbd291be5ce..8a66c05ef2099 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -331,7 +331,7 @@ static int q6apm_dai_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; + struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(soc_prtd, 0); struct device *dev = component->dev; struct q6apm_dai_data *pdata; diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 029780d6fe6d0..bc030ce296803 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -200,7 +200,7 @@ static int sc7180_startup_realtek_codec(struct snd_soc_pcm_runtime *rtd) static int sc7180_snd_startup(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); @@ -234,7 +234,7 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) static int sc7180_qdsp_snd_startup(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); @@ -307,7 +307,7 @@ static int dmic_set(struct snd_kcontrol *kcontrol, static void sc7180_snd_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); @@ -334,7 +334,7 @@ static void sc7180_snd_shutdown(struct snd_pcm_substream *substream) static void sc7180_qdsp_snd_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); @@ -389,7 +389,7 @@ static int sc7180_adau7002_init(struct snd_soc_pcm_runtime *rtd) static int sc7180_adau7002_snd_startup(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); struct snd_pcm_runtime *runtime = substream->runtime; diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index d36f029b78889..207ac5da4dd43 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -205,7 +205,7 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai; const struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sc7280_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card); @@ -237,7 +237,7 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream, static int sc7280_snd_swr_prepare(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); const struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sc7280_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; @@ -268,7 +268,7 @@ static int sc7280_snd_swr_prepare(struct snd_pcm_substream *substream) static int sc7280_snd_prepare(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); const struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); switch (cpu_dai->id) { @@ -287,7 +287,7 @@ static int sc7280_snd_prepare(struct snd_pcm_substream *substream) static int sc7280_snd_hw_free(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sc7280_snd_data *data = snd_soc_card_get_drvdata(rtd->card); const struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; @@ -312,7 +312,7 @@ static int sc7280_snd_hw_free(struct snd_pcm_substream *substream) static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); @@ -339,7 +339,7 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream) { unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); int ret = 0; diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 878bd50ad4a73..06fd47c4178ff 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -50,7 +50,7 @@ static int sc8280xp_snd_init(struct snd_soc_pcm_runtime *rtd) static void sc8280xp_snd_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sc8280xp_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = pdata->sruntime[cpu_dai->id]; @@ -89,7 +89,7 @@ static int sc8280xp_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int sc8280xp_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sc8280xp_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card); @@ -98,7 +98,7 @@ static int sc8280xp_snd_hw_params(struct snd_pcm_substream *substream, static int sc8280xp_snd_prepare(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sc8280xp_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; @@ -109,7 +109,7 @@ static int sc8280xp_snd_prepare(struct snd_pcm_substream *substream) static int sc8280xp_snd_hw_free(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sc8280xp_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c index 7f5089bbe0223..eaa8bb016e509 100644 --- a/sound/soc/qcom/sdw.c +++ b/sound/soc/qcom/sdw.c @@ -21,7 +21,7 @@ */ int qcom_snd_sdw_startup(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sdw_stream_runtime *sruntime; struct snd_soc_dai *codec_dai; @@ -54,7 +54,7 @@ int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sruntime, bool *stream_prepared) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); int ret; @@ -105,7 +105,7 @@ int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct sdw_stream_runtime **psruntime) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sdw_stream_runtime *sruntime; @@ -135,7 +135,7 @@ EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params); int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sruntime, bool *stream_prepared) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); switch (cpu_dai->id) { diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index d70df72c0160d..a15dafb99b337 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -50,7 +50,7 @@ static int sm8250_snd_startup(struct snd_pcm_substream *substream) { unsigned int fmt = SND_SOC_DAIFMT_BP_FP; unsigned int codec_dai_fmt = SND_SOC_DAIFMT_BC_FC; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); @@ -72,7 +72,7 @@ static int sm8250_snd_startup(struct snd_pcm_substream *substream) static void sm2450_snd_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sm8250_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; @@ -84,7 +84,7 @@ static void sm2450_snd_shutdown(struct snd_pcm_substream *substream) static int sm8250_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sm8250_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card); @@ -93,7 +93,7 @@ static int sm8250_snd_hw_params(struct snd_pcm_substream *substream, static int sm8250_snd_prepare(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sm8250_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; @@ -104,7 +104,7 @@ static int sm8250_snd_prepare(struct snd_pcm_substream *substream) static int sm8250_snd_hw_free(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sm8250_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index c3c8bf7ffb5bb..0e0773a858099 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -31,7 +31,7 @@ static int x1e80100_snd_init(struct snd_soc_pcm_runtime *rtd) static void x1e80100_snd_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; @@ -67,7 +67,7 @@ static int x1e80100_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); @@ -76,7 +76,7 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; @@ -87,7 +87,7 @@ static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) static int x1e80100_snd_hw_free(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; From 3beb985abbf29e660edd1708f8a120ae9bbbddc3 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:11 +0200 Subject: [PATCH 467/603] ASoC: tegra: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-2-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_machine.c | 2 +- sound/soc/tegra/tegra_pcm.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index 192e9692bdf2e..775ce433fdbfd 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -290,7 +290,7 @@ static unsigned int tegra_machine_mclk_rate_6mhz(unsigned int srate) static int tegra_machine_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); struct snd_soc_card *card = rtd->card; struct tegra_machine *machine = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 42acb56543db6..4bdbcd2635ef5 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -76,7 +76,7 @@ EXPORT_SYMBOL_GPL(tegra_pcm_platform_unregister); int tegra_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_dmaengine_dai_dma_data *dmap; struct dma_chan *chan; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); @@ -127,7 +127,7 @@ EXPORT_SYMBOL_GPL(tegra_pcm_open); int tegra_pcm_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); if (rtd->dai_link->no_pcm) return 0; @@ -142,7 +142,7 @@ int tegra_pcm_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_dmaengine_dai_dma_data *dmap; struct dma_slave_config slave_config; struct dma_chan *chan; From 72a666f47f958a57db16b6bdd9ed385674069693 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:12 +0200 Subject: [PATCH 468/603] ASoC: ti: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-3-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/ti/omap-hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index 4513b527ab970..639bc83f42639 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -40,7 +40,7 @@ struct hdmi_audio_data { static struct hdmi_audio_data *card_drvdata_substream(struct snd_pcm_substream *ss) { - struct snd_soc_pcm_runtime *rtd = ss->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(ss); return snd_soc_card_get_drvdata(rtd->card); } From a80f2f8443a4ae10c568566f57fe704ea52c5bdb Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:13 +0200 Subject: [PATCH 469/603] ASoC: arm: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-4-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/arm/pxa2xx-pcm-lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 51d2ff80df165..571e9d909cdf0 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -33,7 +33,7 @@ int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_dmaengine_dai_dma_data *dma_params; struct dma_slave_config config; int ret; @@ -79,7 +79,7 @@ EXPORT_SYMBOL(pxa2xx_pcm_prepare); int pxa2xx_pcm_open(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dmaengine_dai_dma_data *dma_params; int ret; From a84d84077512fc64cf1fc2292a3638690a026737 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:14 +0200 Subject: [PATCH 470/603] ASoC: amd: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-5-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 665a6ea0a2a8c..a36300a4ed8a6 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -217,7 +217,7 @@ static void acp_card_shutdown(struct snd_pcm_substream *substream) static int acp_card_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index d6cdb6d9fdd63..357dfd016bafd 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -143,7 +143,7 @@ static int rt5682_clk_enable(struct snd_pcm_substream *substream) static int acp3x_1015_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai; int srate, i, ret; diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c index 66b800962f8cc..2f630753278dc 100644 --- a/sound/soc/amd/ps/ps-sdw-dma.c +++ b/sound/soc/amd/ps/ps-sdw-dma.c @@ -218,7 +218,7 @@ static int acp63_sdw_dma_open(struct snd_soc_component *component, struct acp_sdw_dma_stream *stream; struct snd_soc_dai *cpu_dai; struct amd_sdw_manager *amd_manager; - struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct snd_soc_pcm_runtime *prtd = snd_soc_substream_to_rtd(substream); int ret; runtime = substream->runtime; From b695d8be5bba9897ee670ec102ca608ecaf625c4 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:15 +0200 Subject: [PATCH 471/603] ASoC: fsl: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-6-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/imx-card.c | 6 +++--- sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 6 +++--- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index f6d2564864c60..5ddc0c2fe53ff 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -242,7 +242,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = rtd->card->dev; diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index cb8723965f2f8..0e18ccabe28c3 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -252,7 +252,7 @@ static unsigned long akcodec_get_mclk_rate(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, int slots, int slot_width) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct imx_card_data *data = snd_soc_card_get_drvdata(rtd->card); const struct imx_card_plat_data *plat_data = data->plat_data; struct dai_link_data *link_data = &data->link_data[rtd->num]; @@ -289,7 +289,7 @@ static unsigned long akcodec_get_mclk_rate(struct snd_pcm_substream *substream, static int imx_aif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_card *card = rtd->card; struct imx_card_data *data = snd_soc_card_get_drvdata(card); @@ -405,7 +405,7 @@ static int ak5558_hw_rule_rate(struct snd_pcm_hw_params *p, struct snd_pcm_hw_ru static int imx_aif_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct imx_card_data *data = snd_soc_card_get_drvdata(card); struct dai_link_data *link_data = &data->link_data[rtd->num]; diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index e454085c6e5cf..fe47b439a8183 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -32,7 +32,7 @@ struct imx_hdmi_data { static int imx_hdmi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct imx_hdmi_data *data = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index b84d1dfddba2f..ba491cbb99300 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -316,7 +316,7 @@ static int imx_rpmsg_pcm_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev); @@ -461,7 +461,7 @@ static int imx_rpmsg_pcm_trigger(struct snd_soc_component *component, struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev); int ret = 0; @@ -515,7 +515,7 @@ static int imx_rpmsg_pcm_ack(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev); struct rpmsg_info *info = dev_get_drvdata(component->dev); From 3b62178720594e08bdf8a87515ccca0328fe41fe Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:16 +0200 Subject: [PATCH 472/603] ASoC: img: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-7-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index dacc29fcf24b3..b69a364d619e7 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -395,7 +395,7 @@ static int img_i2s_in_dma_prepare_slave_config(struct snd_pcm_substream *st, struct snd_pcm_hw_params *params, struct dma_slave_config *sc) { unsigned int i2s_channels = params_channels(params) / 2; - struct snd_soc_pcm_runtime *rtd = st->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(st); struct snd_dmaengine_dai_dma_data *dma_data; int ret; diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index f442d985ab87e..6f9831c6d6e06 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -401,7 +401,7 @@ static int img_i2s_out_dma_prepare_slave_config(struct snd_pcm_substream *st, struct snd_pcm_hw_params *params, struct dma_slave_config *sc) { unsigned int i2s_channels = params_channels(params) / 2; - struct snd_soc_pcm_runtime *rtd = st->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(st); struct snd_dmaengine_dai_dma_data *dma_data; int ret; From fe42c3b75b93dee9a4010e2297f1783e48684af7 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:17 +0200 Subject: [PATCH 473/603] ASoC: kirkwood: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-8-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index ef00792e1d49a..036b42058272f 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -19,7 +19,7 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) { - struct snd_soc_pcm_runtime *soc_runtime = subs->private_data; + struct snd_soc_pcm_runtime *soc_runtime = snd_soc_substream_to_rtd(subs); return snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(soc_runtime, 0)); } From ffad75cebb865fef6f8e40f921c08c79a8faf7e3 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:18 +0200 Subject: [PATCH 474/603] ASoC: loongson: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-9-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 2 +- sound/soc/loongson/loongson_dma.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index e8432d466f60f..fae5e9312bf08 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -23,7 +23,7 @@ struct loongson_card_data { static int loongson_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); struct loongson_card_data *ls_card = snd_soc_card_get_drvdata(rtd->card); diff --git a/sound/soc/loongson/loongson_dma.c b/sound/soc/loongson/loongson_dma.c index 8090662e8ff24..4fcc2868160bb 100644 --- a/sound/soc/loongson/loongson_dma.c +++ b/sound/soc/loongson/loongson_dma.c @@ -226,7 +226,7 @@ static int loongson_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_card *card = substream->pcm->card; struct loongson_runtime_data *prtd; struct loongson_dma_data *dma_data; From 410a45140fb76709cf2bbad84bc8a731acf632c8 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:19 +0200 Subject: [PATCH 475/603] ASoC: mediatek: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-10-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 14 +++++++------- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 2 +- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 ++++---- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 6 +++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 10 +++++----- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 4 ++-- 7 files changed, 24 insertions(+), 24 deletions(-) diff --git a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c index c1c486e275b9d..572ded279b534 100644 --- a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c +++ b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c @@ -106,7 +106,7 @@ static const struct snd_pcm_hardware mt7986_afe_hardware = { static int mt7986_memif_fs(struct snd_pcm_substream *substream, unsigned int rate) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); @@ -116,7 +116,7 @@ static int mt7986_memif_fs(struct snd_pcm_substream *substream, static int mt7986_irq_fs(struct snd_pcm_substream *substream, unsigned int rate) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c index bfcfc68ac64d9..bafbef96a42da 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c @@ -40,7 +40,7 @@ static const struct snd_pcm_hardware mt8186_afe_hardware = { static int mt8186_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct snd_pcm_runtime *runtime = substream->runtime; int id = snd_soc_rtd_to_cpu(rtd, 0)->id; @@ -82,7 +82,7 @@ static int mt8186_fe_startup(struct snd_pcm_substream *substream, static void mt8186_fe_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mt8186_afe_private *afe_priv = afe->platform_priv; int id = snd_soc_rtd_to_cpu(rtd, 0)->id; @@ -104,7 +104,7 @@ static int mt8186_fe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); int id = snd_soc_rtd_to_cpu(rtd, 0)->id; unsigned int channels = params_channels(params); @@ -153,7 +153,7 @@ static int mt8186_fe_hw_free(struct snd_pcm_substream *substream, static int mt8186_fe_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime * const runtime = substream->runtime; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mt8186_afe_private *afe_priv = afe->platform_priv; @@ -252,7 +252,7 @@ static int mt8186_fe_trigger(struct snd_pcm_substream *substream, int cmd, static int mt8186_memif_fs(struct snd_pcm_substream *substream, unsigned int rate) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); @@ -269,7 +269,7 @@ static int mt8186_get_dai_fs(struct mtk_base_afe *afe, static int mt8186_irq_fs(struct snd_pcm_substream *substream, unsigned int rate) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); @@ -290,7 +290,7 @@ static int mt8186_get_memif_pbuf_size(struct snd_pcm_substream *substream) static int mt8186_fe_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime * const runtime = substream->runtime; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); int id = snd_soc_rtd_to_cpu(rtd, 0)->id; diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366.c b/sound/soc/mediatek/mt8186/mt8186-mt6366.c index 8fd31c52b077e..771d53611c2a4 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366.c @@ -297,7 +297,7 @@ static const struct snd_soc_ops mt8186_da7219_i2s_ops = { static int mt8186_rt5682s_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c index 9647fe133dc82..ccb6c1f3adc7d 100644 --- a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c +++ b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c @@ -91,7 +91,7 @@ int mt8188_afe_fs_timing(unsigned int rate) static int mt8188_memif_fs(struct snd_pcm_substream *substream, unsigned int rate) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_component *component = NULL; struct mtk_base_afe *afe = NULL; struct mt8188_afe_private *afe_priv = NULL; @@ -300,7 +300,7 @@ static int mt8188_afe_enable_cm(struct mtk_base_afe *afe, static int mt8188_afe_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); int id = snd_soc_rtd_to_cpu(rtd, 0)->id; @@ -334,7 +334,7 @@ static int mt8188_afe_fe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); int id = snd_soc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[id]; @@ -358,7 +358,7 @@ static int mt8188_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, { struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); const struct mt8188_afe_channel_merge *cm = mt8188_afe_found_cm(dai); - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime * const runtime = substream->runtime; int id = snd_soc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[id]; diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index f629fc6bbb531..eba6f4c445ffb 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -557,7 +557,7 @@ enum { static int mt8188_dptx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); unsigned int rate = params_rate(params); unsigned int mclk_fs_ratio = 256; unsigned int mclk_fs = rate * mclk_fs_ratio; @@ -658,7 +658,7 @@ static int mt8188_dumb_amp_init(struct snd_soc_pcm_runtime *rtd) static int mt8188_max98390_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); unsigned int bit_width = params_width(params); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai; @@ -821,7 +821,7 @@ static const struct snd_soc_ops mt8188_nau8825_ops = { static int mt8188_rt5682s_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 64af9bf363fda..38891d1bd18a5 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -84,7 +84,7 @@ int mt8195_afe_fs_timing(unsigned int rate) static int mt8195_memif_fs(struct snd_pcm_substream *substream, unsigned int rate) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); @@ -281,7 +281,7 @@ mt8195_afe_paired_memif_clk_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai, int enable) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mt8195_afe_private *afe_priv = afe->platform_priv; int id = snd_soc_rtd_to_cpu(rtd, 0)->id; @@ -310,7 +310,7 @@ mt8195_afe_paired_memif_clk_enable(struct snd_pcm_substream *substream, struct snd_soc_dai *dai, int enable) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mt8195_afe_private *afe_priv = afe->platform_priv; int id = snd_soc_rtd_to_cpu(rtd, 0)->id; @@ -342,7 +342,7 @@ mt8195_afe_paired_memif_clk_enable(struct snd_pcm_substream *substream, static int mt8195_afe_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); int id = snd_soc_rtd_to_cpu(rtd, 0)->id; @@ -380,7 +380,7 @@ static int mt8195_afe_fe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); int id = snd_soc_rtd_to_cpu(rtd, 0)->id; struct mtk_base_afe_memif *memif = &afe->memif[id]; diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 62e4e34cffbbf..ca87511905203 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -338,7 +338,7 @@ static const struct snd_soc_ops mt8195_hdmitx_dptx_playback_ops = { static int mt8195_dptx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); return snd_soc_dai_set_sysclk(cpu_dai, 0, params_rate(params) * 256, @@ -394,7 +394,7 @@ static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8195_rt5682_etdm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); From 22f5680a9cbc7388f97e5386c15c325d6961b958 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:20 +0200 Subject: [PATCH 476/603] ASoC: meson: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-11-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-fifo.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c index 4041ff8e437fe..b222bde1f61bb 100644 --- a/sound/soc/meson/aiu-fifo.c +++ b/sound/soc/meson/aiu-fifo.c @@ -25,7 +25,7 @@ static struct snd_soc_dai *aiu_fifo_dai(struct snd_pcm_substream *ss) { - struct snd_soc_pcm_runtime *rtd = ss->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(ss); return snd_soc_rtd_to_cpu(rtd, 0); } diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index bebee0ca8e388..1ead5ebc84c46 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -46,7 +46,7 @@ static struct snd_pcm_hardware axg_fifo_hw = { static struct snd_soc_dai *axg_fifo_dai(struct snd_pcm_substream *ss) { - struct snd_soc_pcm_runtime *rtd = ss->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(ss); return snd_soc_rtd_to_cpu(rtd, 0); } From 3e726593107d134221f666b4f2be612b278c3ddb Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:21 +0200 Subject: [PATCH 477/603] ASoC: samsung: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-12-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/samsung/midas_wm1811.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index f31244156ff62..0841e2e6f8ce4 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -127,7 +127,7 @@ static int midas_stop_fll1(struct snd_soc_pcm_runtime *rtd) static int midas_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); unsigned int pll_out; /* AIF1CLK should be at least 3MHz for "optimal performance" */ From 47aa51677c975a5f66bc93d1c527e8878cf34d6c Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:02:22 +0200 Subject: [PATCH 478/603] ASoC: sunxi: Use snd_soc_substream_to_rtd() for accessing private_data Do not open-code snd_soc_substream_to_rtd(). Signed-off-by: Krzysztof Kozlowski Reviewed-by: Jernej Skrabec Link: https://lore.kernel.org/r/20240430-asoc-snd-substream-clean-v1-13-6f8a8902b479@linaro.org Signed-off-by: Mark Brown --- sound/soc/sunxi/sun50i-dmic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index dd32780fb6a43..884394ddaf86b 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -86,7 +86,7 @@ static const struct dmic_rate dmic_rate_s[] = { static int sun50i_dmic_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(rtd, 0)); /* only support capture */ From cfcd957e63506273dc54f34b320172c8709244c7 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 30 Apr 2024 13:54:34 +0200 Subject: [PATCH 479/603] ASoC: codecs: wm8962: use 'time_left' variable with wait_for_completion_timeout() There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_completion_timeout() causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code self explaining. Signed-off-by: Wolfram Sang Link: https://lore.kernel.org/r/20240430115438.29134-2-wsa+renesas@sang-engineering.com Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3ef95b7dcbdb9..08d164ce3e49c 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2886,7 +2886,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s { struct wm8962_priv *wm8962 = snd_soc_component_get_drvdata(component); struct _fll_div fll_div; - unsigned long timeout; + unsigned long time_left; int ret; int fll1 = 0; @@ -2974,14 +2974,14 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s * higher if we'll error out */ if (wm8962->irq) - timeout = msecs_to_jiffies(5); + time_left = msecs_to_jiffies(5); else - timeout = msecs_to_jiffies(1); + time_left = msecs_to_jiffies(1); - timeout = wait_for_completion_timeout(&wm8962->fll_lock, - timeout); + time_left = wait_for_completion_timeout(&wm8962->fll_lock, + time_left); - if (timeout == 0 && wm8962->irq) { + if (time_left == 0 && wm8962->irq) { dev_err(component->dev, "FLL lock timed out"); snd_soc_component_update_bits(component, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, 0); From 0800660d8c59539b628f5a6646bb63091d58152f Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 30 Apr 2024 13:54:35 +0200 Subject: [PATCH 480/603] ASoC: codecs: wm8993: use 'time_left' variable with wait_for_completion_timeout() There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_completion_timeout() causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code self explaining. Fix to the proper variable type 'unsigned long' while here. Signed-off-by: Wolfram Sang Link: https://lore.kernel.org/r/20240430115438.29134-3-wsa+renesas@sang-engineering.com Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 98a83f51ab263..f257980f9b56e 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -470,7 +470,7 @@ static int _wm8993_set_fll(struct snd_soc_component *component, int fll_id, int struct i2c_client *i2c = to_i2c_client(component->dev); u16 reg1, reg4, reg5; struct _fll_div fll_div; - unsigned int timeout; + unsigned long time_left; int ret; /* Any change? */ @@ -543,19 +543,19 @@ static int _wm8993_set_fll(struct snd_soc_component *component, int fll_id, int /* If we've got an interrupt wired up make sure we get it */ if (i2c->irq) - timeout = msecs_to_jiffies(20); + time_left = msecs_to_jiffies(20); else if (Fref < 1000000) - timeout = msecs_to_jiffies(3); + time_left = msecs_to_jiffies(3); else - timeout = msecs_to_jiffies(1); + time_left = msecs_to_jiffies(1); try_wait_for_completion(&wm8993->fll_lock); /* Enable the FLL */ snd_soc_component_write(component, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); - timeout = wait_for_completion_timeout(&wm8993->fll_lock, timeout); - if (i2c->irq && !timeout) + time_left = wait_for_completion_timeout(&wm8993->fll_lock, time_left); + if (i2c->irq && !time_left) dev_warn(component->dev, "Timed out waiting for FLL\n"); dev_dbg(component->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); From 19c70b4668306632d3cbbecdf5fea98b528e873e Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 30 Apr 2024 13:54:36 +0200 Subject: [PATCH 481/603] ASoC: codecs: wm8994: use 'time_left' variable with wait_for_completion_timeout() There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_completion_timeout() causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code self explaining. Signed-off-by: Wolfram Sang Link: https://lore.kernel.org/r/20240430115438.29134-4-wsa+renesas@sang-engineering.com Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index fc9894975a1d5..a99908582a50a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2210,7 +2210,7 @@ static int _wm8994_set_fll(struct snd_soc_component *component, int id, int src, int reg_offset, ret; struct fll_div fll; u16 reg, clk1, aif_reg, aif_src; - unsigned long timeout; + unsigned long time_left; bool was_enabled; struct clk *mclk; @@ -2403,9 +2403,9 @@ static int _wm8994_set_fll(struct snd_soc_component *component, int id, int src, WM8994_FLL1_FRAC, reg); if (wm8994->fll_locked_irq) { - timeout = wait_for_completion_timeout(&wm8994->fll_locked[id], - msecs_to_jiffies(10)); - if (timeout == 0) + time_left = wait_for_completion_timeout(&wm8994->fll_locked[id], + msecs_to_jiffies(10)); + if (time_left == 0) dev_warn(component->dev, "Timed out waiting for FLL lock\n"); } else { From 4e1f953a4a447b5e001655b453505c4c15904c61 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 30 Apr 2024 13:54:37 +0200 Subject: [PATCH 482/603] ASoC: codecs: wm8996: use 'time_left' variable with wait_for_completion_timeout() There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_completion_timeout() causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code self explaining. Signed-off-by: Wolfram Sang Link: https://lore.kernel.org/r/20240430115438.29134-5-wsa+renesas@sang-engineering.com Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f6c5adce164ee..5c06cea09bd18 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -655,28 +655,28 @@ static void wait_for_dc_servo(struct snd_soc_component *component, u16 mask) struct i2c_client *i2c = to_i2c_client(component->dev); struct wm8996_priv *wm8996 = snd_soc_component_get_drvdata(component); int ret; - unsigned long timeout = 200; + unsigned long time_left = 200; snd_soc_component_write(component, WM8996_DC_SERVO_2, mask); /* Use the interrupt if possible */ do { if (i2c->irq) { - timeout = wait_for_completion_timeout(&wm8996->dcs_done, - msecs_to_jiffies(200)); - if (timeout == 0) + time_left = wait_for_completion_timeout(&wm8996->dcs_done, + msecs_to_jiffies(200)); + if (time_left == 0) dev_err(component->dev, "DC servo timed out\n"); } else { msleep(1); - timeout--; + time_left--; } ret = snd_soc_component_read(component, WM8996_DC_SERVO_2); dev_dbg(component->dev, "DC servo state: %x\n", ret); - } while (timeout && ret & mask); + } while (time_left && ret & mask); - if (timeout == 0) + if (time_left == 0) dev_err(component->dev, "DC servo timed out for %x\n", mask); else dev_dbg(component->dev, "DC servo complete for %x\n", mask); From e6fa3509cb32df373b15212a99f69a6595efd1c3 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 29 Apr 2024 13:48:46 +0200 Subject: [PATCH 483/603] ASoC: qcom: Constify static snd_pcm_hardware Static 'struct snd_pcm_hardware' is not modified by the driver and its copy is passed to the core, so it can be made const for increased code safety. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240429-n-asoc-const-snd-pcm-hardware-v1-1-c6ce60989834@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6apm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6asm-dai.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 00bbd291be5ce..3c78440a2f0f0 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -85,7 +85,7 @@ struct q6apm_dai_data { long long sid; }; -static struct snd_pcm_hardware q6apm_dai_hardware_capture = { +static const struct snd_pcm_hardware q6apm_dai_hardware_capture = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | @@ -104,7 +104,7 @@ static struct snd_pcm_hardware q6apm_dai_hardware_capture = { .fifo_size = 0, }; -static struct snd_pcm_hardware q6apm_dai_hardware_playback = { +static const struct snd_pcm_hardware q6apm_dai_hardware_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index aeb6a9d479ab4..3913706ccdc5f 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -103,7 +103,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { .fifo_size = 0, }; -static struct snd_pcm_hardware q6asm_dai_hardware_playback = { +static const struct snd_pcm_hardware q6asm_dai_hardware_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | From ed90156037659473ee95eafe3f72d8498e5384ff Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 29 Apr 2024 13:48:47 +0200 Subject: [PATCH 484/603] ASoC: fsl: Constify static snd_pcm_hardware Static 'struct snd_pcm_hardware' is not modified by the driver and its copy is passed to the core, so it can be made const for increased code safety. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240429-n-asoc-const-snd-pcm-hardware-v1-2-c6ce60989834@linaro.org Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index b84d1dfddba2f..128728b65a3ac 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -16,7 +16,7 @@ #include "fsl_rpmsg.h" #include "imx-pcm-rpmsg.h" -static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = { +static const struct snd_pcm_hardware imx_rpmsg_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_BATCH | From 7b5ce9f0c52a5885d34d46bba62e9eaedc3dd459 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 29 Apr 2024 13:48:48 +0200 Subject: [PATCH 485/603] ASoC: meson: Constify static snd_pcm_hardware Static 'struct snd_pcm_hardware' is not modified by the driver and its copy is passed to the core, so it can be made const for increased code safety. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240429-n-asoc-const-snd-pcm-hardware-v1-3-c6ce60989834@linaro.org Signed-off-by: Mark Brown --- sound/soc/meson/aiu-fifo-i2s.c | 2 +- sound/soc/meson/aiu-fifo-spdif.c | 2 +- sound/soc/meson/aiu-fifo.h | 2 +- sound/soc/meson/axg-fifo.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c index 7d833500c7993..eccbc16b293ad 100644 --- a/sound/soc/meson/aiu-fifo-i2s.c +++ b/sound/soc/meson/aiu-fifo-i2s.c @@ -25,7 +25,7 @@ #define AIU_FIFO_I2S_BLOCK 256 -static struct snd_pcm_hardware fifo_i2s_pcm = { +static const struct snd_pcm_hardware fifo_i2s_pcm = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c index fa91f3c53fa46..e0e00ec026dcc 100644 --- a/sound/soc/meson/aiu-fifo-spdif.c +++ b/sound/soc/meson/aiu-fifo-spdif.c @@ -27,7 +27,7 @@ #define AIU_FIFO_SPDIF_BLOCK 8 -static struct snd_pcm_hardware fifo_spdif_pcm = { +static const struct snd_pcm_hardware fifo_spdif_pcm = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | diff --git a/sound/soc/meson/aiu-fifo.h b/sound/soc/meson/aiu-fifo.h index 42ce266677cc8..84ab4577815a4 100644 --- a/sound/soc/meson/aiu-fifo.h +++ b/sound/soc/meson/aiu-fifo.h @@ -18,7 +18,7 @@ struct snd_pcm_hw_params; struct platform_device; struct aiu_fifo { - struct snd_pcm_hardware *pcm; + const struct snd_pcm_hardware *pcm; unsigned int mem_offset; unsigned int fifo_block; struct clk *pclk; diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index bebee0ca8e388..633fd971b4447 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -23,7 +23,7 @@ * These differences are handled in the respective DAI drivers */ -static struct snd_pcm_hardware axg_fifo_hw = { +static const struct snd_pcm_hardware axg_fifo_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | From 74a15fabd271d0fd82ceecbbfa1b98ea0a4709dd Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 29 Apr 2024 13:48:49 +0200 Subject: [PATCH 486/603] ASoC: uniphier: Constify static snd_pcm_hardware Static 'struct snd_pcm_hardware' is not modified by the driver and its copy is passed to the core, so it can be made const for increased code safety. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240429-n-asoc-const-snd-pcm-hardware-v1-4-c6ce60989834@linaro.org Signed-off-by: Mark Brown --- sound/soc/uniphier/aio-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c index fe272befd9676..265d61723e99a 100644 --- a/sound/soc/uniphier/aio-dma.c +++ b/sound/soc/uniphier/aio-dma.c @@ -14,7 +14,7 @@ #include "aio.h" -static struct snd_pcm_hardware uniphier_aiodma_hw = { +static const struct snd_pcm_hardware uniphier_aiodma_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED, From bd381c9d151467e784988bbacf22bd7ca02455d6 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 30 Apr 2024 16:09:54 +0200 Subject: [PATCH 487/603] ASoC: qcom: q6apm-dai: drop unused 'q6apm_dai_rtd' fields Remove few unused fields from 'struct q6apm_dai_rtd'. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20240430140954.328127-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6apm-dai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 00bbd291be5ce..cee23d4bda6fa 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -70,14 +70,10 @@ struct q6apm_dai_rtd { unsigned int bytes_received; unsigned int copied_total; uint16_t bits_per_sample; - uint16_t source; /* Encoding source bit mask */ - uint16_t session_id; bool next_track; enum stream_state state; struct q6apm_graph *graph; spinlock_t lock; - uint32_t initial_samples_drop; - uint32_t trailing_samples_drop; bool notify_on_drain; }; @@ -720,14 +716,12 @@ static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component, switch (metadata->key) { case SNDRV_COMPRESS_ENCODER_PADDING: - prtd->trailing_samples_drop = metadata->value[0]; q6apm_remove_trailing_silence(component->dev, prtd->graph, - prtd->trailing_samples_drop); + metadata->value[0]); break; case SNDRV_COMPRESS_ENCODER_DELAY: - prtd->initial_samples_drop = metadata->value[0]; q6apm_remove_initial_silence(component->dev, prtd->graph, - prtd->initial_samples_drop); + metadata->value[0]); break; default: ret = -EINVAL; From fd25e1c1328f40c6eb0bfa5ebee757e0e1dfb308 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:05 +0200 Subject: [PATCH 488/603] ASoC: core: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-2-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/Makefile | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/Makefile b/sound/soc/Makefile index f90f5300b36e5..fd61847dd1eb6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,14 +1,14 @@ # SPDX-License-Identifier: GPL-2.0 -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o -snd-soc-core-objs += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o +snd-soc-core-y := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o +snd-soc-core-y += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) -snd-soc-core-objs += soc-topology.o +snd-soc-core-y += soc-topology.o endif ifneq ($(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST),) -# snd-soc-test-objs := soc-topology-test.o +# snd-soc-test-y := soc-topology-test.o obj-$(CONFIG_SND_SOC_TOPOLOGY_KUNIT_TEST) += soc-topology-test.o endif @@ -17,20 +17,20 @@ obj-$(CONFIG_SND_SOC_CARD_KUNIT_TEST) += soc-card-test.o endif ifneq ($(CONFIG_SND_SOC_UTILS_KUNIT_TEST),) -# snd-soc-test-objs := soc-utils-test.o +# snd-soc-test-y := soc-utils-test.o obj-$(CONFIG_SND_SOC_UTILS_KUNIT_TEST) += soc-utils-test.o endif ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) -snd-soc-core-objs += soc-generic-dmaengine-pcm.o +snd-soc-core-y += soc-generic-dmaengine-pcm.o endif ifneq ($(CONFIG_SND_SOC_AC97_BUS),) -snd-soc-core-objs += soc-ac97.o +snd-soc-core-y += soc-ac97.o endif ifneq ($(CONFIG_SND_SOC_ACPI),) -snd-soc-acpi-objs := soc-acpi.o +snd-soc-acpi-y := soc-acpi.o endif obj-$(CONFIG_SND_SOC_ACPI) += snd-soc-acpi.o From 1c3fa6662832f1498b78817317796bc37df7da40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:06 +0200 Subject: [PATCH 489/603] ASoC: codecs: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-3-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 784 +++++++++++++++++++------------------- 1 file changed, 392 insertions(+), 392 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cddb16cd6a4c2..b4df22186e255 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,399 +1,399 @@ # SPDX-License-Identifier: GPL-2.0 -snd-soc-88pm860x-objs := 88pm860x-codec.o -snd-soc-ab8500-codec-objs := ab8500-codec.o -snd-soc-ac97-objs := ac97.o -snd-soc-ad1836-objs := ad1836.o -snd-soc-ad193x-objs := ad193x.o -snd-soc-ad193x-spi-objs := ad193x-spi.o -snd-soc-ad193x-i2c-objs := ad193x-i2c.o -snd-soc-ad1980-objs := ad1980.o -snd-soc-ad73311-objs := ad73311.o -snd-soc-adau-utils-objs := adau-utils.o -snd-soc-adau1372-objs := adau1372.o -snd-soc-adau1372-i2c-objs := adau1372-i2c.o -snd-soc-adau1372-spi-objs := adau1372-spi.o -snd-soc-adau1373-objs := adau1373.o -snd-soc-adau1701-objs := adau1701.o -snd-soc-adau17x1-objs := adau17x1.o -snd-soc-adau1761-objs := adau1761.o -snd-soc-adau1761-i2c-objs := adau1761-i2c.o -snd-soc-adau1761-spi-objs := adau1761-spi.o -snd-soc-adau1781-objs := adau1781.o -snd-soc-adau1781-i2c-objs := adau1781-i2c.o -snd-soc-adau1781-spi-objs := adau1781-spi.o -snd-soc-adau1977-objs := adau1977.o -snd-soc-adau1977-spi-objs := adau1977-spi.o -snd-soc-adau1977-i2c-objs := adau1977-i2c.o -snd-soc-adau7002-objs := adau7002.o -snd-soc-adau7118-objs := adau7118.o -snd-soc-adau7118-i2c-objs := adau7118-i2c.o -snd-soc-adau7118-hw-objs := adau7118-hw.o -snd-soc-adav80x-objs := adav80x.o -snd-soc-adav801-objs := adav801.o -snd-soc-adav803-objs := adav803.o -snd-soc-ads117x-objs := ads117x.o -snd-soc-ak4104-objs := ak4104.o -snd-soc-ak4118-objs := ak4118.o -snd-soc-ak4375-objs := ak4375.o -snd-soc-ak4458-objs := ak4458.o -snd-soc-ak4535-objs := ak4535.o -snd-soc-ak4554-objs := ak4554.o -snd-soc-ak4613-objs := ak4613.o -snd-soc-ak4641-objs := ak4641.o -snd-soc-ak4642-objs := ak4642.o -snd-soc-ak4671-objs := ak4671.o -snd-soc-ak5386-objs := ak5386.o -snd-soc-ak5558-objs := ak5558.o -snd-soc-arizona-objs := arizona.o arizona-jack.o -snd-soc-audio-iio-aux-objs := audio-iio-aux.o -snd-soc-aw8738-objs := aw8738.o -snd-soc-aw87390-objs := aw87390.o -snd-soc-aw88395-lib-objs := aw88395/aw88395_lib.o -snd-soc-aw88395-objs := aw88395/aw88395.o \ +snd-soc-88pm860x-y := 88pm860x-codec.o +snd-soc-ab8500-codec-y := ab8500-codec.o +snd-soc-ac97-y := ac97.o +snd-soc-ad1836-y := ad1836.o +snd-soc-ad193x-y := ad193x.o +snd-soc-ad193x-spi-y := ad193x-spi.o +snd-soc-ad193x-i2c-y := ad193x-i2c.o +snd-soc-ad1980-y := ad1980.o +snd-soc-ad73311-y := ad73311.o +snd-soc-adau-utils-y := adau-utils.o +snd-soc-adau1372-y := adau1372.o +snd-soc-adau1372-i2c-y := adau1372-i2c.o +snd-soc-adau1372-spi-y := adau1372-spi.o +snd-soc-adau1373-y := adau1373.o +snd-soc-adau1701-y := adau1701.o +snd-soc-adau17x1-y := adau17x1.o +snd-soc-adau1761-y := adau1761.o +snd-soc-adau1761-i2c-y := adau1761-i2c.o +snd-soc-adau1761-spi-y := adau1761-spi.o +snd-soc-adau1781-y := adau1781.o +snd-soc-adau1781-i2c-y := adau1781-i2c.o +snd-soc-adau1781-spi-y := adau1781-spi.o +snd-soc-adau1977-y := adau1977.o +snd-soc-adau1977-spi-y := adau1977-spi.o +snd-soc-adau1977-i2c-y := adau1977-i2c.o +snd-soc-adau7002-y := adau7002.o +snd-soc-adau7118-y := adau7118.o +snd-soc-adau7118-i2c-y := adau7118-i2c.o +snd-soc-adau7118-hw-y := adau7118-hw.o +snd-soc-adav80x-y := adav80x.o +snd-soc-adav801-y := adav801.o +snd-soc-adav803-y := adav803.o +snd-soc-ads117x-y := ads117x.o +snd-soc-ak4104-y := ak4104.o +snd-soc-ak4118-y := ak4118.o +snd-soc-ak4375-y := ak4375.o +snd-soc-ak4458-y := ak4458.o +snd-soc-ak4535-y := ak4535.o +snd-soc-ak4554-y := ak4554.o +snd-soc-ak4613-y := ak4613.o +snd-soc-ak4641-y := ak4641.o +snd-soc-ak4642-y := ak4642.o +snd-soc-ak4671-y := ak4671.o +snd-soc-ak5386-y := ak5386.o +snd-soc-ak5558-y := ak5558.o +snd-soc-arizona-y := arizona.o arizona-jack.o +snd-soc-audio-iio-aux-y := audio-iio-aux.o +snd-soc-aw8738-y := aw8738.o +snd-soc-aw87390-y := aw87390.o +snd-soc-aw88395-lib-y := aw88395/aw88395_lib.o +snd-soc-aw88395-y := aw88395/aw88395.o \ aw88395/aw88395_device.o -snd-soc-aw88261-objs := aw88261.o -snd-soc-aw88399-objs := aw88399.o -snd-soc-bd28623-objs := bd28623.o -snd-soc-bt-sco-objs := bt-sco.o -snd-soc-chv3-codec-objs := chv3-codec.o -snd-soc-cpcap-objs := cpcap.o -snd-soc-cq93vc-objs := cq93vc.o -snd-soc-cros-ec-codec-objs := cros_ec_codec.o -snd-soc-cs-amp-lib-objs := cs-amp-lib.o -snd-soc-cs-amp-lib-test-objs := cs-amp-lib-test.o -snd-soc-cs35l32-objs := cs35l32.o -snd-soc-cs35l33-objs := cs35l33.o -snd-soc-cs35l34-objs := cs35l34.o -snd-soc-cs35l35-objs := cs35l35.o -snd-soc-cs35l36-objs := cs35l36.o -snd-soc-cs35l41-lib-objs := cs35l41-lib.o -snd-soc-cs35l41-objs := cs35l41.o -snd-soc-cs35l41-spi-objs := cs35l41-spi.o -snd-soc-cs35l41-i2c-objs := cs35l41-i2c.o -snd-soc-cs35l45-objs := cs35l45.o cs35l45-tables.o -snd-soc-cs35l45-spi-objs := cs35l45-spi.o -snd-soc-cs35l45-i2c-objs := cs35l45-i2c.o -snd-soc-cs35l56-objs := cs35l56.o -snd-soc-cs35l56-shared-objs := cs35l56-shared.o -snd-soc-cs35l56-i2c-objs := cs35l56-i2c.o -snd-soc-cs35l56-spi-objs := cs35l56-spi.o -snd-soc-cs35l56-sdw-objs := cs35l56-sdw.o -snd-soc-cs42l42-objs := cs42l42.o -snd-soc-cs42l42-i2c-objs := cs42l42-i2c.o -snd-soc-cs42l42-sdw-objs := cs42l42-sdw.o -snd-soc-cs42l43-objs := cs42l43.o cs42l43-jack.o -snd-soc-cs42l43-sdw-objs := cs42l43-sdw.o -snd-soc-cs42l51-objs := cs42l51.o -snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o -snd-soc-cs42l52-objs := cs42l52.o -snd-soc-cs42l56-objs := cs42l56.o -snd-soc-cs42l73-objs := cs42l73.o -snd-soc-cs42l83-i2c-objs := cs42l83-i2c.o -snd-soc-cs4234-objs := cs4234.o -snd-soc-cs4265-objs := cs4265.o -snd-soc-cs4270-objs := cs4270.o -snd-soc-cs4271-objs := cs4271.o -snd-soc-cs4271-i2c-objs := cs4271-i2c.o -snd-soc-cs4271-spi-objs := cs4271-spi.o -snd-soc-cs42xx8-objs := cs42xx8.o -snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o -snd-soc-cs43130-objs := cs43130.o -snd-soc-cs4341-objs := cs4341.o -snd-soc-cs4349-objs := cs4349.o -snd-soc-cs47l15-objs := cs47l15.o -snd-soc-cs47l24-objs := cs47l24.o -snd-soc-cs47l35-objs := cs47l35.o -snd-soc-cs47l85-objs := cs47l85.o -snd-soc-cs47l90-objs := cs47l90.o -snd-soc-cs47l92-objs := cs47l92.o -snd-soc-cs53l30-objs := cs53l30.o -snd-soc-cx20442-objs := cx20442.o -snd-soc-cx2072x-objs := cx2072x.o -snd-soc-da7210-objs := da7210.o -snd-soc-da7213-objs := da7213.o -snd-soc-da7218-objs := da7218.o -snd-soc-da7219-objs := da7219.o da7219-aad.o -snd-soc-da732x-objs := da732x.o -snd-soc-da9055-objs := da9055.o -snd-soc-dmic-objs := dmic.o -snd-soc-es7134-objs := es7134.o -snd-soc-es7241-objs := es7241.o -snd-soc-es83xx-dsm-common-objs := es83xx-dsm-common.o -snd-soc-es8316-objs := es8316.o -snd-soc-es8326-objs := es8326.o -snd-soc-es8328-objs := es8328.o -snd-soc-es8328-i2c-objs := es8328-i2c.o -snd-soc-es8328-spi-objs := es8328-spi.o -snd-soc-framer-objs := framer-codec.o -snd-soc-gtm601-objs := gtm601.o -snd-soc-hdac-hdmi-objs := hdac_hdmi.o -snd-soc-hdac-hda-objs := hdac_hda.o -snd-soc-hda-codec-objs := hda.o hda-dai.o -snd-soc-ics43432-objs := ics43432.o -snd-soc-idt821034-objs := idt821034.o -snd-soc-inno-rk3036-objs := inno_rk3036.o -snd-soc-isabelle-objs := isabelle.o -snd-soc-jz4740-codec-objs := jz4740.o -snd-soc-jz4725b-codec-objs := jz4725b.o -snd-soc-jz4760-codec-objs := jz4760.o -snd-soc-jz4770-codec-objs := jz4770.o -snd-soc-lm4857-objs := lm4857.o -snd-soc-lm49453-objs := lm49453.o -snd-soc-lochnagar-sc-objs := lochnagar-sc.o -snd-soc-lpass-macro-common-objs := lpass-macro-common.o -snd-soc-lpass-rx-macro-objs := lpass-rx-macro.o -snd-soc-lpass-tx-macro-objs := lpass-tx-macro.o -snd-soc-lpass-wsa-macro-objs := lpass-wsa-macro.o -snd-soc-lpass-va-macro-objs := lpass-va-macro.o -snd-soc-madera-objs := madera.o -snd-soc-max9759-objs := max9759.o -snd-soc-max9768-objs := max9768.o -snd-soc-max98088-objs := max98088.o -snd-soc-max98090-objs := max98090.o -snd-soc-max98095-objs := max98095.o -snd-soc-max98357a-objs := max98357a.o -snd-soc-max98371-objs := max98371.o -snd-soc-max9867-objs := max9867.o -snd-soc-max98925-objs := max98925.o -snd-soc-max98926-objs := max98926.o -snd-soc-max98927-objs := max98927.o -snd-soc-max98520-objs := max98520.o -snd-soc-max98363-objs := max98363.o -snd-soc-max98373-objs := max98373.o -snd-soc-max98373-i2c-objs := max98373-i2c.o -snd-soc-max98373-sdw-objs := max98373-sdw.o -snd-soc-max98388-objs := max98388.o -snd-soc-max98390-objs := max98390.o -snd-soc-max98396-objs := max98396.o -snd-soc-max9850-objs := max9850.o -snd-soc-max9860-objs := max9860.o -snd-soc-mc13783-objs := mc13783.o -snd-soc-ml26124-objs := ml26124.o -snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o -snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o -snd-soc-mt6351-objs := mt6351.o -snd-soc-mt6358-objs := mt6358.o -snd-soc-mt6359-objs := mt6359.o -snd-soc-mt6359-accdet-objs := mt6359-accdet.o -snd-soc-mt6660-objs := mt6660.o -snd-soc-nau8315-objs := nau8315.o -snd-soc-nau8540-objs := nau8540.o -snd-soc-nau8810-objs := nau8810.o -snd-soc-nau8821-objs := nau8821.o -snd-soc-nau8822-objs := nau8822.o -snd-soc-nau8824-objs := nau8824.o -snd-soc-nau8825-objs := nau8825.o -snd-soc-hdmi-codec-objs := hdmi-codec.o -snd-soc-pcm1681-objs := pcm1681.o -snd-soc-pcm1789-codec-objs := pcm1789.o -snd-soc-pcm1789-i2c-objs := pcm1789-i2c.o -snd-soc-pcm179x-codec-objs := pcm179x.o -snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o -snd-soc-pcm179x-spi-objs := pcm179x-spi.o -snd-soc-pcm186x-objs := pcm186x.o -snd-soc-pcm186x-i2c-objs := pcm186x-i2c.o -snd-soc-pcm186x-spi-objs := pcm186x-spi.o -snd-soc-pcm3008-objs := pcm3008.o -snd-soc-pcm3060-objs := pcm3060.o -snd-soc-pcm3060-i2c-objs := pcm3060-i2c.o -snd-soc-pcm3060-spi-objs := pcm3060-spi.o -snd-soc-pcm3168a-objs := pcm3168a.o -snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o -snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o -snd-soc-pcm5102a-objs := pcm5102a.o -snd-soc-pcm512x-objs := pcm512x.o -snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o -snd-soc-pcm512x-spi-objs := pcm512x-spi.o -snd-soc-pcm6240-objs := pcm6240.o -snd-soc-peb2466-objs := peb2466.o -snd-soc-rk3308-objs := rk3308_codec.o -snd-soc-rk3328-objs := rk3328_codec.o -snd-soc-rk817-objs := rk817_codec.o -snd-soc-rl6231-objs := rl6231.o -snd-soc-rl6347a-objs := rl6347a.o -snd-soc-rt1011-objs := rt1011.o -snd-soc-rt1015-objs := rt1015.o -snd-soc-rt1015p-objs := rt1015p.o -snd-soc-rt1016-objs := rt1016.o -snd-soc-rt1017-sdca-objs := rt1017-sdca-sdw.o -snd-soc-rt1019-objs := rt1019.o -snd-soc-rt1305-objs := rt1305.o -snd-soc-rt1308-objs := rt1308.o -snd-soc-rt1308-sdw-objs := rt1308-sdw.o -snd-soc-rt1316-sdw-objs := rt1316-sdw.o -snd-soc-rt1318-sdw-objs := rt1318-sdw.o -snd-soc-rt274-objs := rt274.o -snd-soc-rt286-objs := rt286.o -snd-soc-rt298-objs := rt298.o -snd-soc-rt5514-objs := rt5514.o -snd-soc-rt5514-spi-objs := rt5514-spi.o -snd-soc-rt5616-objs := rt5616.o -snd-soc-rt5631-objs := rt5631.o -snd-soc-rt5640-objs := rt5640.o -snd-soc-rt5645-objs := rt5645.o -snd-soc-rt5651-objs := rt5651.o -snd-soc-rt5659-objs := rt5659.o -snd-soc-rt5660-objs := rt5660.o -snd-soc-rt5663-objs := rt5663.o -snd-soc-rt5665-objs := rt5665.o -snd-soc-rt5668-objs := rt5668.o -snd-soc-rt5670-objs := rt5670.o -snd-soc-rt5677-objs := rt5677.o -snd-soc-rt5677-spi-objs := rt5677-spi.o -snd-soc-rt5682-objs := rt5682.o -snd-soc-rt5682-sdw-objs := rt5682-sdw.o -snd-soc-rt5682-i2c-objs := rt5682-i2c.o -snd-soc-rt5682s-objs := rt5682s.o -snd-soc-rt700-objs := rt700.o rt700-sdw.o -snd-soc-rt711-objs := rt711.o rt711-sdw.o -snd-soc-rt711-sdca-objs := rt711-sdca.o rt711-sdca-sdw.o -snd-soc-rt712-sdca-objs := rt712-sdca.o rt712-sdca-sdw.o -snd-soc-rt712-sdca-dmic-objs := rt712-sdca-dmic.o -snd-soc-rt715-objs := rt715.o rt715-sdw.o -snd-soc-rt715-sdca-objs := rt715-sdca.o rt715-sdca-sdw.o -snd-soc-rt722-sdca-objs := rt722-sdca.o rt722-sdca-sdw.o -snd-soc-rt9120-objs := rt9120.o -snd-soc-rtq9128-objs := rtq9128.o -snd-soc-sdw-mockup-objs := sdw-mockup.o -snd-soc-sgtl5000-objs := sgtl5000.o -snd-soc-alc5623-objs := alc5623.o -snd-soc-alc5632-objs := alc5632.o -snd-soc-sigmadsp-objs := sigmadsp.o -snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o -snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o -snd-soc-si476x-objs := si476x.o -snd-soc-sma1303-objs := sma1303.o -snd-soc-spdif-tx-objs := spdif_transmitter.o -snd-soc-spdif-rx-objs := spdif_receiver.o -snd-soc-src4xxx-objs := src4xxx.o -snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o -snd-soc-ssm2305-objs := ssm2305.o -snd-soc-ssm2518-objs := ssm2518.o -snd-soc-ssm2602-objs := ssm2602.o -snd-soc-ssm2602-spi-objs := ssm2602-spi.o -snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o -snd-soc-ssm3515-objs := ssm3515.o -snd-soc-ssm4567-objs := ssm4567.o -snd-soc-sta32x-objs := sta32x.o -snd-soc-sta350-objs := sta350.o -snd-soc-sta529-objs := sta529.o -snd-soc-stac9766-objs := stac9766.o -snd-soc-sti-sas-objs := sti-sas.o -snd-soc-tas5086-objs := tas5086.o -snd-soc-tas571x-objs := tas571x.o -snd-soc-tas5720-objs := tas5720.o -snd-soc-tas5805m-objs := tas5805m.o -snd-soc-tas6424-objs := tas6424.o -snd-soc-tda7419-objs := tda7419.o -snd-soc-tas2770-objs := tas2770.o -snd-soc-tas2781-comlib-objs := tas2781-comlib.o -snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o -snd-soc-tas2781-i2c-objs := tas2781-i2c.o -snd-soc-tfa9879-objs := tfa9879.o -snd-soc-tfa989x-objs := tfa989x.o -snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o -snd-soc-tlv320aic23-objs := tlv320aic23.o -snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o -snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o -snd-soc-tlv320aic26-objs := tlv320aic26.o -snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o -snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o tlv320aic32x4-clk.o -snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o -snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o -snd-soc-tlv320aic3x-objs := tlv320aic3x.o -snd-soc-tlv320aic3x-i2c-objs := tlv320aic3x-i2c.o -snd-soc-tlv320aic3x-spi-objs := tlv320aic3x-spi.o -snd-soc-tlv320dac33-objs := tlv320dac33.o -snd-soc-tlv320adcx140-objs := tlv320adcx140.o -snd-soc-tscs42xx-objs := tscs42xx.o -snd-soc-tscs454-objs := tscs454.o -snd-soc-ts3a227e-objs := ts3a227e.o -snd-soc-twl4030-objs := twl4030.o -snd-soc-twl6040-objs := twl6040.o -snd-soc-uda1334-objs := uda1334.o -snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd-classh-objs := wcd-clsh-v2.o -snd-soc-wcd-mbhc-objs := wcd-mbhc-v2.o -snd-soc-wcd9335-objs := wcd9335.o -snd-soc-wcd934x-objs := wcd934x.o -snd-soc-wcd938x-objs := wcd938x.o -snd-soc-wcd938x-sdw-objs := wcd938x-sdw.o -snd-soc-wcd939x-objs := wcd939x.o -snd-soc-wcd939x-sdw-objs := wcd939x-sdw.o -snd-soc-wl1273-objs := wl1273.o -snd-soc-wm-adsp-objs := wm_adsp.o -snd-soc-wm0010-objs := wm0010.o -snd-soc-wm1250-ev1-objs := wm1250-ev1.o -snd-soc-wm2000-objs := wm2000.o -snd-soc-wm2200-objs := wm2200.o -snd-soc-wm5100-objs := wm5100.o wm5100-tables.o -snd-soc-wm5102-objs := wm5102.o -snd-soc-wm5110-objs := wm5110.o -snd-soc-wm8350-objs := wm8350.o -snd-soc-wm8400-objs := wm8400.o -snd-soc-wm8510-objs := wm8510.o -snd-soc-wm8523-objs := wm8523.o -snd-soc-wm8524-objs := wm8524.o -snd-soc-wm8580-objs := wm8580.o -snd-soc-wm8711-objs := wm8711.o -snd-soc-wm8727-objs := wm8727.o -snd-soc-wm8728-objs := wm8728.o -snd-soc-wm8731-objs := wm8731.o -snd-soc-wm8731-i2c-objs := wm8731-i2c.o -snd-soc-wm8731-spi-objs := wm8731-spi.o -snd-soc-wm8737-objs := wm8737.o -snd-soc-wm8741-objs := wm8741.o -snd-soc-wm8750-objs := wm8750.o -snd-soc-wm8753-objs := wm8753.o -snd-soc-wm8770-objs := wm8770.o -snd-soc-wm8776-objs := wm8776.o -snd-soc-wm8782-objs := wm8782.o -snd-soc-wm8804-objs := wm8804.o -snd-soc-wm8804-i2c-objs := wm8804-i2c.o -snd-soc-wm8804-spi-objs := wm8804-spi.o -snd-soc-wm8900-objs := wm8900.o -snd-soc-wm8903-objs := wm8903.o -snd-soc-wm8904-objs := wm8904.o -snd-soc-wm8996-objs := wm8996.o -snd-soc-wm8940-objs := wm8940.o -snd-soc-wm8955-objs := wm8955.o -snd-soc-wm8960-objs := wm8960.o -snd-soc-wm8961-objs := wm8961.o -snd-soc-wm8962-objs := wm8962.o -snd-soc-wm8971-objs := wm8971.o -snd-soc-wm8974-objs := wm8974.o -snd-soc-wm8978-objs := wm8978.o -snd-soc-wm8983-objs := wm8983.o -snd-soc-wm8985-objs := wm8985.o -snd-soc-wm8988-objs := wm8988.o -snd-soc-wm8990-objs := wm8990.o -snd-soc-wm8991-objs := wm8991.o -snd-soc-wm8993-objs := wm8993.o -snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o -snd-soc-wm8995-objs := wm8995.o -snd-soc-wm8997-objs := wm8997.o -snd-soc-wm8998-objs := wm8998.o -snd-soc-wm9081-objs := wm9081.o -snd-soc-wm9090-objs := wm9090.o -snd-soc-wm9705-objs := wm9705.o -snd-soc-wm9712-objs := wm9712.o -snd-soc-wm9713-objs := wm9713.o -snd-soc-wm-hubs-objs := wm_hubs.o -snd-soc-wsa881x-objs := wsa881x.o -snd-soc-wsa883x-objs := wsa883x.o -snd-soc-wsa884x-objs := wsa884x.o -snd-soc-zl38060-objs := zl38060.o +snd-soc-aw88261-y := aw88261.o +snd-soc-aw88399-y := aw88399.o +snd-soc-bd28623-y := bd28623.o +snd-soc-bt-sco-y := bt-sco.o +snd-soc-chv3-codec-y := chv3-codec.o +snd-soc-cpcap-y := cpcap.o +snd-soc-cq93vc-y := cq93vc.o +snd-soc-cros-ec-codec-y := cros_ec_codec.o +snd-soc-cs-amp-lib-y := cs-amp-lib.o +snd-soc-cs-amp-lib-test-y := cs-amp-lib-test.o +snd-soc-cs35l32-y := cs35l32.o +snd-soc-cs35l33-y := cs35l33.o +snd-soc-cs35l34-y := cs35l34.o +snd-soc-cs35l35-y := cs35l35.o +snd-soc-cs35l36-y := cs35l36.o +snd-soc-cs35l41-lib-y := cs35l41-lib.o +snd-soc-cs35l41-y := cs35l41.o +snd-soc-cs35l41-spi-y := cs35l41-spi.o +snd-soc-cs35l41-i2c-y := cs35l41-i2c.o +snd-soc-cs35l45-y := cs35l45.o cs35l45-tables.o +snd-soc-cs35l45-spi-y := cs35l45-spi.o +snd-soc-cs35l45-i2c-y := cs35l45-i2c.o +snd-soc-cs35l56-y := cs35l56.o +snd-soc-cs35l56-shared-y := cs35l56-shared.o +snd-soc-cs35l56-i2c-y := cs35l56-i2c.o +snd-soc-cs35l56-spi-y := cs35l56-spi.o +snd-soc-cs35l56-sdw-y := cs35l56-sdw.o +snd-soc-cs42l42-y := cs42l42.o +snd-soc-cs42l42-i2c-y := cs42l42-i2c.o +snd-soc-cs42l42-sdw-y := cs42l42-sdw.o +snd-soc-cs42l43-y := cs42l43.o cs42l43-jack.o +snd-soc-cs42l43-sdw-y := cs42l43-sdw.o +snd-soc-cs42l51-y := cs42l51.o +snd-soc-cs42l51-i2c-y := cs42l51-i2c.o +snd-soc-cs42l52-y := cs42l52.o +snd-soc-cs42l56-y := cs42l56.o +snd-soc-cs42l73-y := cs42l73.o +snd-soc-cs42l83-i2c-y := cs42l83-i2c.o +snd-soc-cs4234-y := cs4234.o +snd-soc-cs4265-y := cs4265.o +snd-soc-cs4270-y := cs4270.o +snd-soc-cs4271-y := cs4271.o +snd-soc-cs4271-i2c-y := cs4271-i2c.o +snd-soc-cs4271-spi-y := cs4271-spi.o +snd-soc-cs42xx8-y := cs42xx8.o +snd-soc-cs42xx8-i2c-y := cs42xx8-i2c.o +snd-soc-cs43130-y := cs43130.o +snd-soc-cs4341-y := cs4341.o +snd-soc-cs4349-y := cs4349.o +snd-soc-cs47l15-y := cs47l15.o +snd-soc-cs47l24-y := cs47l24.o +snd-soc-cs47l35-y := cs47l35.o +snd-soc-cs47l85-y := cs47l85.o +snd-soc-cs47l90-y := cs47l90.o +snd-soc-cs47l92-y := cs47l92.o +snd-soc-cs53l30-y := cs53l30.o +snd-soc-cx20442-y := cx20442.o +snd-soc-cx2072x-y := cx2072x.o +snd-soc-da7210-y := da7210.o +snd-soc-da7213-y := da7213.o +snd-soc-da7218-y := da7218.o +snd-soc-da7219-y := da7219.o da7219-aad.o +snd-soc-da732x-y := da732x.o +snd-soc-da9055-y := da9055.o +snd-soc-dmic-y := dmic.o +snd-soc-es7134-y := es7134.o +snd-soc-es7241-y := es7241.o +snd-soc-es83xx-dsm-common-y := es83xx-dsm-common.o +snd-soc-es8316-y := es8316.o +snd-soc-es8326-y := es8326.o +snd-soc-es8328-y := es8328.o +snd-soc-es8328-i2c-y := es8328-i2c.o +snd-soc-es8328-spi-y := es8328-spi.o +snd-soc-framer-y := framer-codec.o +snd-soc-gtm601-y := gtm601.o +snd-soc-hdac-hdmi-y := hdac_hdmi.o +snd-soc-hdac-hda-y := hdac_hda.o +snd-soc-hda-codec-y := hda.o hda-dai.o +snd-soc-ics43432-y := ics43432.o +snd-soc-idt821034-y := idt821034.o +snd-soc-inno-rk3036-y := inno_rk3036.o +snd-soc-isabelle-y := isabelle.o +snd-soc-jz4740-codec-y := jz4740.o +snd-soc-jz4725b-codec-y := jz4725b.o +snd-soc-jz4760-codec-y := jz4760.o +snd-soc-jz4770-codec-y := jz4770.o +snd-soc-lm4857-y := lm4857.o +snd-soc-lm49453-y := lm49453.o +snd-soc-lochnagar-sc-y := lochnagar-sc.o +snd-soc-lpass-macro-common-y := lpass-macro-common.o +snd-soc-lpass-rx-macro-y := lpass-rx-macro.o +snd-soc-lpass-tx-macro-y := lpass-tx-macro.o +snd-soc-lpass-wsa-macro-y := lpass-wsa-macro.o +snd-soc-lpass-va-macro-y := lpass-va-macro.o +snd-soc-madera-y := madera.o +snd-soc-max9759-y := max9759.o +snd-soc-max9768-y := max9768.o +snd-soc-max98088-y := max98088.o +snd-soc-max98090-y := max98090.o +snd-soc-max98095-y := max98095.o +snd-soc-max98357a-y := max98357a.o +snd-soc-max98371-y := max98371.o +snd-soc-max9867-y := max9867.o +snd-soc-max98925-y := max98925.o +snd-soc-max98926-y := max98926.o +snd-soc-max98927-y := max98927.o +snd-soc-max98520-y := max98520.o +snd-soc-max98363-y := max98363.o +snd-soc-max98373-y := max98373.o +snd-soc-max98373-i2c-y := max98373-i2c.o +snd-soc-max98373-sdw-y := max98373-sdw.o +snd-soc-max98388-y := max98388.o +snd-soc-max98390-y := max98390.o +snd-soc-max98396-y := max98396.o +snd-soc-max9850-y := max9850.o +snd-soc-max9860-y := max9860.o +snd-soc-mc13783-y := mc13783.o +snd-soc-ml26124-y := ml26124.o +snd-soc-msm8916-analog-y := msm8916-wcd-analog.o +snd-soc-msm8916-digital-y := msm8916-wcd-digital.o +snd-soc-mt6351-y := mt6351.o +snd-soc-mt6358-y := mt6358.o +snd-soc-mt6359-y := mt6359.o +snd-soc-mt6359-accdet-y := mt6359-accdet.o +snd-soc-mt6660-y := mt6660.o +snd-soc-nau8315-y := nau8315.o +snd-soc-nau8540-y := nau8540.o +snd-soc-nau8810-y := nau8810.o +snd-soc-nau8821-y := nau8821.o +snd-soc-nau8822-y := nau8822.o +snd-soc-nau8824-y := nau8824.o +snd-soc-nau8825-y := nau8825.o +snd-soc-hdmi-codec-y := hdmi-codec.o +snd-soc-pcm1681-y := pcm1681.o +snd-soc-pcm1789-codec-y := pcm1789.o +snd-soc-pcm1789-i2c-y := pcm1789-i2c.o +snd-soc-pcm179x-codec-y := pcm179x.o +snd-soc-pcm179x-i2c-y := pcm179x-i2c.o +snd-soc-pcm179x-spi-y := pcm179x-spi.o +snd-soc-pcm186x-y := pcm186x.o +snd-soc-pcm186x-i2c-y := pcm186x-i2c.o +snd-soc-pcm186x-spi-y := pcm186x-spi.o +snd-soc-pcm3008-y := pcm3008.o +snd-soc-pcm3060-y := pcm3060.o +snd-soc-pcm3060-i2c-y := pcm3060-i2c.o +snd-soc-pcm3060-spi-y := pcm3060-spi.o +snd-soc-pcm3168a-y := pcm3168a.o +snd-soc-pcm3168a-i2c-y := pcm3168a-i2c.o +snd-soc-pcm3168a-spi-y := pcm3168a-spi.o +snd-soc-pcm5102a-y := pcm5102a.o +snd-soc-pcm512x-y := pcm512x.o +snd-soc-pcm512x-i2c-y := pcm512x-i2c.o +snd-soc-pcm512x-spi-y := pcm512x-spi.o +snd-soc-pcm6240-y := pcm6240.o +snd-soc-peb2466-y := peb2466.o +snd-soc-rk3308-y := rk3308_codec.o +snd-soc-rk3328-y := rk3328_codec.o +snd-soc-rk817-y := rk817_codec.o +snd-soc-rl6231-y := rl6231.o +snd-soc-rl6347a-y := rl6347a.o +snd-soc-rt1011-y := rt1011.o +snd-soc-rt1015-y := rt1015.o +snd-soc-rt1015p-y := rt1015p.o +snd-soc-rt1016-y := rt1016.o +snd-soc-rt1017-sdca-y := rt1017-sdca-sdw.o +snd-soc-rt1019-y := rt1019.o +snd-soc-rt1305-y := rt1305.o +snd-soc-rt1308-y := rt1308.o +snd-soc-rt1308-sdw-y := rt1308-sdw.o +snd-soc-rt1316-sdw-y := rt1316-sdw.o +snd-soc-rt1318-sdw-y := rt1318-sdw.o +snd-soc-rt274-y := rt274.o +snd-soc-rt286-y := rt286.o +snd-soc-rt298-y := rt298.o +snd-soc-rt5514-y := rt5514.o +snd-soc-rt5514-spi-y := rt5514-spi.o +snd-soc-rt5616-y := rt5616.o +snd-soc-rt5631-y := rt5631.o +snd-soc-rt5640-y := rt5640.o +snd-soc-rt5645-y := rt5645.o +snd-soc-rt5651-y := rt5651.o +snd-soc-rt5659-y := rt5659.o +snd-soc-rt5660-y := rt5660.o +snd-soc-rt5663-y := rt5663.o +snd-soc-rt5665-y := rt5665.o +snd-soc-rt5668-y := rt5668.o +snd-soc-rt5670-y := rt5670.o +snd-soc-rt5677-y := rt5677.o +snd-soc-rt5677-spi-y := rt5677-spi.o +snd-soc-rt5682-y := rt5682.o +snd-soc-rt5682-sdw-y := rt5682-sdw.o +snd-soc-rt5682-i2c-y := rt5682-i2c.o +snd-soc-rt5682s-y := rt5682s.o +snd-soc-rt700-y := rt700.o rt700-sdw.o +snd-soc-rt711-y := rt711.o rt711-sdw.o +snd-soc-rt711-sdca-y := rt711-sdca.o rt711-sdca-sdw.o +snd-soc-rt712-sdca-y := rt712-sdca.o rt712-sdca-sdw.o +snd-soc-rt712-sdca-dmic-y := rt712-sdca-dmic.o +snd-soc-rt715-y := rt715.o rt715-sdw.o +snd-soc-rt715-sdca-y := rt715-sdca.o rt715-sdca-sdw.o +snd-soc-rt722-sdca-y := rt722-sdca.o rt722-sdca-sdw.o +snd-soc-rt9120-y := rt9120.o +snd-soc-rtq9128-y := rtq9128.o +snd-soc-sdw-mockup-y := sdw-mockup.o +snd-soc-sgtl5000-y := sgtl5000.o +snd-soc-alc5623-y := alc5623.o +snd-soc-alc5632-y := alc5632.o +snd-soc-sigmadsp-y := sigmadsp.o +snd-soc-sigmadsp-i2c-y := sigmadsp-i2c.o +snd-soc-sigmadsp-regmap-y := sigmadsp-regmap.o +snd-soc-si476x-y := si476x.o +snd-soc-sma1303-y := sma1303.o +snd-soc-spdif-tx-y := spdif_transmitter.o +snd-soc-spdif-rx-y := spdif_receiver.o +snd-soc-src4xxx-y := src4xxx.o +snd-soc-src4xxx-i2c-y := src4xxx-i2c.o +snd-soc-ssm2305-y := ssm2305.o +snd-soc-ssm2518-y := ssm2518.o +snd-soc-ssm2602-y := ssm2602.o +snd-soc-ssm2602-spi-y := ssm2602-spi.o +snd-soc-ssm2602-i2c-y := ssm2602-i2c.o +snd-soc-ssm3515-y := ssm3515.o +snd-soc-ssm4567-y := ssm4567.o +snd-soc-sta32x-y := sta32x.o +snd-soc-sta350-y := sta350.o +snd-soc-sta529-y := sta529.o +snd-soc-stac9766-y := stac9766.o +snd-soc-sti-sas-y := sti-sas.o +snd-soc-tas5086-y := tas5086.o +snd-soc-tas571x-y := tas571x.o +snd-soc-tas5720-y := tas5720.o +snd-soc-tas5805m-y := tas5805m.o +snd-soc-tas6424-y := tas6424.o +snd-soc-tda7419-y := tda7419.o +snd-soc-tas2770-y := tas2770.o +snd-soc-tas2781-comlib-y := tas2781-comlib.o +snd-soc-tas2781-fmwlib-y := tas2781-fmwlib.o +snd-soc-tas2781-i2c-y := tas2781-i2c.o +snd-soc-tfa9879-y := tfa9879.o +snd-soc-tfa989x-y := tfa989x.o +snd-soc-tlv320adc3xxx-y := tlv320adc3xxx.o +snd-soc-tlv320aic23-y := tlv320aic23.o +snd-soc-tlv320aic23-i2c-y := tlv320aic23-i2c.o +snd-soc-tlv320aic23-spi-y := tlv320aic23-spi.o +snd-soc-tlv320aic26-y := tlv320aic26.o +snd-soc-tlv320aic31xx-y := tlv320aic31xx.o +snd-soc-tlv320aic32x4-y := tlv320aic32x4.o tlv320aic32x4-clk.o +snd-soc-tlv320aic32x4-i2c-y := tlv320aic32x4-i2c.o +snd-soc-tlv320aic32x4-spi-y := tlv320aic32x4-spi.o +snd-soc-tlv320aic3x-y := tlv320aic3x.o +snd-soc-tlv320aic3x-i2c-y := tlv320aic3x-i2c.o +snd-soc-tlv320aic3x-spi-y := tlv320aic3x-spi.o +snd-soc-tlv320dac33-y := tlv320dac33.o +snd-soc-tlv320adcx140-y := tlv320adcx140.o +snd-soc-tscs42xx-y := tscs42xx.o +snd-soc-tscs454-y := tscs454.o +snd-soc-ts3a227e-y := ts3a227e.o +snd-soc-twl4030-y := twl4030.o +snd-soc-twl6040-y := twl6040.o +snd-soc-uda1334-y := uda1334.o +snd-soc-uda1380-y := uda1380.o +snd-soc-wcd-classh-y := wcd-clsh-v2.o +snd-soc-wcd-mbhc-y := wcd-mbhc-v2.o +snd-soc-wcd9335-y := wcd9335.o +snd-soc-wcd934x-y := wcd934x.o +snd-soc-wcd938x-y := wcd938x.o +snd-soc-wcd938x-sdw-y := wcd938x-sdw.o +snd-soc-wcd939x-y := wcd939x.o +snd-soc-wcd939x-sdw-y := wcd939x-sdw.o +snd-soc-wl1273-y := wl1273.o +snd-soc-wm-adsp-y := wm_adsp.o +snd-soc-wm0010-y := wm0010.o +snd-soc-wm1250-ev1-y := wm1250-ev1.o +snd-soc-wm2000-y := wm2000.o +snd-soc-wm2200-y := wm2200.o +snd-soc-wm5100-y := wm5100.o wm5100-tables.o +snd-soc-wm5102-y := wm5102.o +snd-soc-wm5110-y := wm5110.o +snd-soc-wm8350-y := wm8350.o +snd-soc-wm8400-y := wm8400.o +snd-soc-wm8510-y := wm8510.o +snd-soc-wm8523-y := wm8523.o +snd-soc-wm8524-y := wm8524.o +snd-soc-wm8580-y := wm8580.o +snd-soc-wm8711-y := wm8711.o +snd-soc-wm8727-y := wm8727.o +snd-soc-wm8728-y := wm8728.o +snd-soc-wm8731-y := wm8731.o +snd-soc-wm8731-i2c-y := wm8731-i2c.o +snd-soc-wm8731-spi-y := wm8731-spi.o +snd-soc-wm8737-y := wm8737.o +snd-soc-wm8741-y := wm8741.o +snd-soc-wm8750-y := wm8750.o +snd-soc-wm8753-y := wm8753.o +snd-soc-wm8770-y := wm8770.o +snd-soc-wm8776-y := wm8776.o +snd-soc-wm8782-y := wm8782.o +snd-soc-wm8804-y := wm8804.o +snd-soc-wm8804-i2c-y := wm8804-i2c.o +snd-soc-wm8804-spi-y := wm8804-spi.o +snd-soc-wm8900-y := wm8900.o +snd-soc-wm8903-y := wm8903.o +snd-soc-wm8904-y := wm8904.o +snd-soc-wm8996-y := wm8996.o +snd-soc-wm8940-y := wm8940.o +snd-soc-wm8955-y := wm8955.o +snd-soc-wm8960-y := wm8960.o +snd-soc-wm8961-y := wm8961.o +snd-soc-wm8962-y := wm8962.o +snd-soc-wm8971-y := wm8971.o +snd-soc-wm8974-y := wm8974.o +snd-soc-wm8978-y := wm8978.o +snd-soc-wm8983-y := wm8983.o +snd-soc-wm8985-y := wm8985.o +snd-soc-wm8988-y := wm8988.o +snd-soc-wm8990-y := wm8990.o +snd-soc-wm8991-y := wm8991.o +snd-soc-wm8993-y := wm8993.o +snd-soc-wm8994-y := wm8994.o wm8958-dsp2.o +snd-soc-wm8995-y := wm8995.o +snd-soc-wm8997-y := wm8997.o +snd-soc-wm8998-y := wm8998.o +snd-soc-wm9081-y := wm9081.o +snd-soc-wm9090-y := wm9090.o +snd-soc-wm9705-y := wm9705.o +snd-soc-wm9712-y := wm9712.o +snd-soc-wm9713-y := wm9713.o +snd-soc-wm-hubs-y := wm_hubs.o +snd-soc-wsa881x-y := wsa881x.o +snd-soc-wsa883x-y := wsa883x.o +snd-soc-wsa884x-y := wsa884x.o +snd-soc-zl38060-y := zl38060.o # Amp -snd-soc-max9877-objs := max9877.o -snd-soc-max98504-objs := max98504.o -snd-soc-simple-amplifier-objs := simple-amplifier.o -snd-soc-tpa6130a2-objs := tpa6130a2.o -snd-soc-tas2552-objs := tas2552.o -snd-soc-tas2562-objs := tas2562.o -snd-soc-tas2764-objs := tas2764.o -snd-soc-tas2780-objs := tas2780.o +snd-soc-max9877-y := max9877.o +snd-soc-max98504-y := max98504.o +snd-soc-simple-amplifier-y := simple-amplifier.o +snd-soc-tpa6130a2-y := tpa6130a2.o +snd-soc-tas2552-y := tas2552.o +snd-soc-tas2562-y := tas2562.o +snd-soc-tas2764-y := tas2764.o +snd-soc-tas2780-y := tas2780.o # Mux -snd-soc-simple-mux-objs := simple-mux.o +snd-soc-simple-mux-y := simple-mux.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o From 01e8f66b434f3cca0690f839c417324d7471aa9b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:07 +0200 Subject: [PATCH 490/603] ASoC: adi: Use *-y instead of *-objs in Makefile Signed-off-by: Takashi Iwai *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Link: https://lore.kernel.org/r/20240507155540.24815-4-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/adi/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/adi/Makefile b/sound/soc/adi/Makefile index 125f667b0e089..0d2db8d05806e 100644 --- a/sound/soc/adi/Makefile +++ b/sound/soc/adi/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-adi-axi-i2s-objs := axi-i2s.o -snd-soc-adi-axi-spdif-objs := axi-spdif.o +snd-soc-adi-axi-i2s-y := axi-i2s.o +snd-soc-adi-axi-spdif-y := axi-spdif.o obj-$(CONFIG_SND_SOC_ADI_AXI_I2S) += snd-soc-adi-axi-i2s.o obj-$(CONFIG_SND_SOC_ADI_AXI_SPDIF) += snd-soc-adi-axi-spdif.o From 07214013578b156d0fda288a0311d353ea149674 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:08 +0200 Subject: [PATCH 491/603] ASoC: amd: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-5-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/amd/Makefile | 12 ++++++------ sound/soc/amd/acp/Makefile | 26 +++++++++++++------------- sound/soc/amd/ps/Makefile | 8 ++++---- sound/soc/amd/raven/Makefile | 6 +++--- sound/soc/amd/renoir/Makefile | 6 +++--- sound/soc/amd/rpl/Makefile | 2 +- sound/soc/amd/vangogh/Makefile | 8 ++++---- sound/soc/amd/yc/Makefile | 6 +++--- 8 files changed, 37 insertions(+), 37 deletions(-) diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index ebbe49c2bbff2..4f89d962cce20 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -1,10 +1,10 @@ # SPDX-License-Identifier: GPL-2.0-only -acp_audio_dma-objs := acp-pcm-dma.o -snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o -snd-soc-acp-rt5645-mach-objs := acp-rt5645.o -snd-soc-acp-es8336-mach-objs := acp-es8336.o -snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o -snd-acp-config-objs := acp-config.o +acp_audio_dma-y := acp-pcm-dma.o +snd-soc-acp-da7219mx98357-mach-y := acp-da7219-max98357a.o +snd-soc-acp-rt5645-mach-y := acp-rt5645.o +snd-soc-acp-es8336-mach-y := acp-es8336.o +snd-soc-acp-rt5682-mach-y := acp3x-rt5682-max9836.o +snd-acp-config-y := acp-config.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 1fd581a2aa338..b068bf1f920ed 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -5,23 +5,23 @@ # Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. #common acp driver -snd-acp-pcm-objs := acp-platform.o -snd-acp-i2s-objs := acp-i2s.o -snd-acp-pdm-objs := acp-pdm.o -snd-acp-legacy-common-objs := acp-legacy-common.o -snd-acp-pci-objs := acp-pci.o -snd-amd-sdw-acpi-objs := amd-sdw-acpi.o +snd-acp-pcm-y := acp-platform.o +snd-acp-i2s-y := acp-i2s.o +snd-acp-pdm-y := acp-pdm.o +snd-acp-legacy-common-y := acp-legacy-common.o +snd-acp-pci-y := acp-pci.o +snd-amd-sdw-acpi-y := amd-sdw-acpi.o #platform specific driver -snd-acp-renoir-objs := acp-renoir.o -snd-acp-rembrandt-objs := acp-rembrandt.o -snd-acp63-objs := acp63.o -snd-acp70-objs := acp70.o +snd-acp-renoir-y := acp-renoir.o +snd-acp-rembrandt-y := acp-rembrandt.o +snd-acp63-y := acp63.o +snd-acp70-y := acp70.o #machine specific driver -snd-acp-mach-objs := acp-mach-common.o -snd-acp-legacy-mach-objs := acp-legacy-mach.o acp3x-es83xx/acp3x-es83xx.o -snd-acp-sof-mach-objs := acp-sof-mach.o +snd-acp-mach-y := acp-mach-common.o +snd-acp-legacy-mach-y := acp-legacy-mach.o acp3x-es83xx/acp3x-es83xx.o +snd-acp-sof-mach-y := acp-sof-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o diff --git a/sound/soc/amd/ps/Makefile b/sound/soc/amd/ps/Makefile index b3c254886fd94..b5efb1c5382c6 100644 --- a/sound/soc/amd/ps/Makefile +++ b/sound/soc/amd/ps/Makefile @@ -1,9 +1,9 @@ # SPDX-License-Identifier: GPL-2.0-only # Pink Sardine platform Support -snd-pci-ps-objs := pci-ps.o -snd-ps-pdm-dma-objs := ps-pdm-dma.o -snd-soc-ps-mach-objs := ps-mach.o -snd-ps-sdw-dma-objs := ps-sdw-dma.o +snd-pci-ps-y := pci-ps.o +snd-ps-pdm-dma-y := ps-pdm-dma.o +snd-soc-ps-mach-y := ps-mach.o +snd-ps-sdw-dma-y := ps-sdw-dma.o obj-$(CONFIG_SND_SOC_AMD_PS) += snd-pci-ps.o obj-$(CONFIG_SND_SOC_AMD_PS) += snd-ps-pdm-dma.o diff --git a/sound/soc/amd/raven/Makefile b/sound/soc/amd/raven/Makefile index 62c22b6ed95a2..b2ea030cbf250 100644 --- a/sound/soc/amd/raven/Makefile +++ b/sound/soc/amd/raven/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0+ # Raven Ridge platform Support -snd-pci-acp3x-objs := pci-acp3x.o -snd-acp3x-pcm-dma-objs := acp3x-pcm-dma.o -snd-acp3x-i2s-objs := acp3x-i2s.o +snd-pci-acp3x-y := pci-acp3x.o +snd-acp3x-pcm-dma-y := acp3x-pcm-dma.o +snd-acp3x-i2s-y := acp3x-i2s.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-pci-acp3x.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-pcm-dma.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += snd-acp3x-i2s.o diff --git a/sound/soc/amd/renoir/Makefile b/sound/soc/amd/renoir/Makefile index 4a82690aec16a..76b4a9c3e24fa 100644 --- a/sound/soc/amd/renoir/Makefile +++ b/sound/soc/amd/renoir/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0+ # Renoir platform Support -snd-rn-pci-acp3x-objs := rn-pci-acp3x.o -snd-acp3x-pdm-dma-objs := acp3x-pdm-dma.o -snd-acp3x-rn-objs := acp3x-rn.o +snd-rn-pci-acp3x-y := rn-pci-acp3x.o +snd-acp3x-pdm-dma-y := acp3x-pdm-dma.o +snd-acp3x-rn-y := acp3x-rn.o obj-$(CONFIG_SND_SOC_AMD_RENOIR) += snd-rn-pci-acp3x.o obj-$(CONFIG_SND_SOC_AMD_RENOIR) += snd-acp3x-pdm-dma.o obj-$(CONFIG_SND_SOC_AMD_RENOIR_MACH) += snd-acp3x-rn.o diff --git a/sound/soc/amd/rpl/Makefile b/sound/soc/amd/rpl/Makefile index 11a33a05e94b4..a3825c5be4e7f 100644 --- a/sound/soc/amd/rpl/Makefile +++ b/sound/soc/amd/rpl/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0+ # RPL platform Support -snd-rpl-pci-acp6x-objs := rpl-pci-acp6x.o +snd-rpl-pci-acp6x-y := rpl-pci-acp6x.o obj-$(CONFIG_SND_SOC_AMD_RPL_ACP6x) += snd-rpl-pci-acp6x.o diff --git a/sound/soc/amd/vangogh/Makefile b/sound/soc/amd/vangogh/Makefile index c9e53e04e2470..7eae82faa392a 100644 --- a/sound/soc/amd/vangogh/Makefile +++ b/sound/soc/amd/vangogh/Makefile @@ -1,9 +1,9 @@ # SPDX-License-Identifier: GPL-2.0+ # Vangogh platform Support -snd-pci-acp5x-objs := pci-acp5x.o -snd-acp5x-i2s-objs := acp5x-i2s.o -snd-acp5x-pcm-dma-objs := acp5x-pcm-dma.o -snd-soc-acp5x-mach-objs := acp5x-mach.o +snd-pci-acp5x-y := pci-acp5x.o +snd-acp5x-i2s-y := acp5x-i2s.o +snd-acp5x-pcm-dma-y := acp5x-pcm-dma.o +snd-soc-acp5x-mach-y := acp5x-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP5x) += snd-pci-acp5x.o obj-$(CONFIG_SND_SOC_AMD_ACP5x) += snd-acp5x-i2s.o diff --git a/sound/soc/amd/yc/Makefile b/sound/soc/amd/yc/Makefile index dc29744403884..7a0a3a410b2d6 100644 --- a/sound/soc/amd/yc/Makefile +++ b/sound/soc/amd/yc/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0+ # Yellow Carp platform Support -snd-pci-acp6x-objs := pci-acp6x.o -snd-acp6x-pdm-dma-objs := acp6x-pdm-dma.o -snd-soc-acp6x-mach-objs := acp6x-mach.o +snd-pci-acp6x-y := pci-acp6x.o +snd-acp6x-pdm-dma-y := acp6x-pdm-dma.o +snd-soc-acp6x-mach-y := acp6x-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP6x) += snd-pci-acp6x.o obj-$(CONFIG_SND_SOC_AMD_ACP6x) += snd-acp6x-pdm-dma.o From fa8e54828efee5002b5aed0e315097f7ca4171d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:09 +0200 Subject: [PATCH 492/603] ASoC: apple: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-6-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/apple/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/apple/Makefile b/sound/soc/apple/Makefile index 7a30bf452817e..1eb8fbef60c61 100644 --- a/sound/soc/apple/Makefile +++ b/sound/soc/apple/Makefile @@ -1,3 +1,3 @@ -snd-soc-apple-mca-objs := mca.o +snd-soc-apple-mca-y := mca.o obj-$(CONFIG_SND_SOC_APPLE_MCA) += snd-soc-apple-mca.o From 907cb79d41a7bb54d2d69bbbdd6ebb159a1d1718 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:10 +0200 Subject: [PATCH 493/603] ASoC: atmel: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-7-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/atmel/Makefile | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 043097a08ea81..03d9c419c93fb 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -1,13 +1,13 @@ # SPDX-License-Identifier: GPL-2.0 # AT91 Platform Support -snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o -snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o -snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o -snd-soc-atmel-i2s-objs := atmel-i2s.o -snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o -snd-soc-mchp-spdiftx-objs := mchp-spdiftx.o -snd-soc-mchp-spdifrx-objs := mchp-spdifrx.o -snd-soc-mchp-pdmc-objs := mchp-pdmc.o +snd-soc-atmel-pcm-pdc-y := atmel-pcm-pdc.o +snd-soc-atmel-pcm-dma-y := atmel-pcm-dma.o +snd-soc-atmel_ssc_dai-y := atmel_ssc_dai.o +snd-soc-atmel-i2s-y := atmel-i2s.o +snd-soc-mchp-i2s-mcc-y := mchp-i2s-mcc.o +snd-soc-mchp-spdiftx-y := mchp-spdiftx.o +snd-soc-mchp-spdifrx-y := mchp-spdifrx.o +snd-soc-mchp-pdmc-y := mchp-pdmc.o # pdc and dma need to both be built-in if any user of # ssc is built-in. @@ -25,13 +25,13 @@ obj-$(CONFIG_SND_MCHP_SOC_SPDIFRX) += snd-soc-mchp-spdifrx.o obj-$(CONFIG_SND_MCHP_SOC_PDMC) += snd-soc-mchp-pdmc.o # AT91 Machine Support -snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o -snd-atmel-soc-wm8904-objs := atmel_wm8904.o -snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o -snd-atmel-soc-classd-objs := atmel-classd.o -snd-atmel-soc-pdmic-objs := atmel-pdmic.o -snd-atmel-soc-tse850-pcm5142-objs := tse850-pcm5142.o -snd-soc-mikroe-proto-objs := mikroe-proto.o +snd-soc-sam9g20-wm8731-y := sam9g20_wm8731.o +snd-atmel-soc-wm8904-y := atmel_wm8904.o +snd-soc-sam9x5-wm8731-y := sam9x5_wm8731.o +snd-atmel-soc-classd-y := atmel-classd.o +snd-atmel-soc-pdmic-y := atmel-pdmic.o +snd-atmel-soc-tse850-pcm5142-y := tse850-pcm5142.o +snd-soc-mikroe-proto-y := mikroe-proto.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o From ef993723af434eb91923b39dcef1623bcfa89606 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:11 +0200 Subject: [PATCH 494/603] ASoC: au1x: Use *-y instead of *-objs Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-8-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/au1x/Makefile | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index 33183d7fe0577..9c6f5c38f92d9 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -1,13 +1,13 @@ # SPDX-License-Identifier: GPL-2.0 # Au1200/Au1550 PSC audio -snd-soc-au1xpsc-dbdma-objs := dbdma2.o -snd-soc-au1xpsc-i2s-objs := psc-i2s.o -snd-soc-au1xpsc-ac97-objs := psc-ac97.o +snd-soc-au1xpsc-dbdma-y := dbdma2.o +snd-soc-au1xpsc-i2s-y := psc-i2s.o +snd-soc-au1xpsc-ac97-y := psc-ac97.o # Au1000/1500/1100 Audio units -snd-soc-au1x-dma-objs := dma.o -snd-soc-au1x-ac97c-objs := ac97c.o -snd-soc-au1x-i2sc-objs := i2sc.o +snd-soc-au1x-dma-y := dma.o +snd-soc-au1x-ac97c-y := ac97c.o +snd-soc-au1x-i2sc-y := i2sc.o obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o @@ -17,8 +17,8 @@ obj-$(CONFIG_SND_SOC_AU1XAC97C) += snd-soc-au1x-ac97c.o obj-$(CONFIG_SND_SOC_AU1XI2SC) += snd-soc-au1x-i2sc.o # Boards -snd-soc-db1000-objs := db1000.o -snd-soc-db1200-objs := db1200.o +snd-soc-db1000-y := db1000.o +snd-soc-db1200-y := db1200.o obj-$(CONFIG_SND_SOC_DB1000) += snd-soc-db1000.o obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o From 37f8d0d000b55038938e90d21aa987a6db1dd31a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:12 +0200 Subject: [PATCH 495/603] ASoC: bcm: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-9-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/bcm/Makefile | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile index 7c2d7899603b7..0c1325a97b709 100644 --- a/sound/soc/bcm/Makefile +++ b/sound/soc/bcm/Makefile @@ -1,15 +1,15 @@ # SPDX-License-Identifier: GPL-2.0-only # BCM2835 Platform Support -snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o +snd-soc-bcm2835-i2s-y := bcm2835-i2s.o obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o # CYGNUS Platform Support -snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o +snd-soc-cygnus-y := cygnus-pcm.o cygnus-ssp.o obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o # BCM63XX Platform Support -snd-soc-63xx-objs := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o +snd-soc-63xx-y := bcm63xx-i2s-whistler.o bcm63xx-pcm-whistler.o obj-$(CONFIG_SND_BCM63XX_I2S_WHISTLER) += snd-soc-63xx.o \ No newline at end of file From 591002b84a335f538879b44dff64568875da89b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:13 +0200 Subject: [PATCH 496/603] ASoC: cirrus: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-10-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/cirrus/Makefile | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/cirrus/Makefile b/sound/soc/cirrus/Makefile index 19a86daad6609..ad606b2937159 100644 --- a/sound/soc/cirrus/Makefile +++ b/sound/soc/cirrus/Makefile @@ -1,12 +1,12 @@ # SPDX-License-Identifier: GPL-2.0 # EP93xx Platform Support -snd-soc-ep93xx-objs := ep93xx-pcm.o -snd-soc-ep93xx-i2s-objs := ep93xx-i2s.o +snd-soc-ep93xx-y := ep93xx-pcm.o +snd-soc-ep93xx-i2s-y := ep93xx-i2s.o obj-$(CONFIG_SND_EP93XX_SOC) += snd-soc-ep93xx.o obj-$(CONFIG_SND_EP93XX_SOC_I2S) += snd-soc-ep93xx-i2s.o # EP93XX Machine Support -snd-soc-edb93xx-objs := edb93xx.o +snd-soc-edb93xx-y := edb93xx.o obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o From fe756e1e54c62ee691a3778b3e758808d0a41ab5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:14 +0200 Subject: [PATCH 497/603] ASoC: fsl: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-11-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/fsl/Makefile | 52 +++++++++++++++++++++--------------------- 1 file changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index b45eda80c196d..2fe78eed3a489 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -1,30 +1,30 @@ # SPDX-License-Identifier: GPL-2.0 # P1022 DS Machine Support -snd-soc-p1022-ds-objs := p1022_ds.o +snd-soc-p1022-ds-y := p1022_ds.o obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o # P1022 RDK Machine Support -snd-soc-p1022-rdk-objs := p1022_rdk.o +snd-soc-p1022-rdk-y := p1022_rdk.o obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o # Freescale SSI/DMA/SAI/SPDIF Support -snd-soc-fsl-audmix-objs := fsl_audmix.o -snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o -snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o -snd-soc-fsl-sai-objs := fsl_sai.o +snd-soc-fsl-audmix-y := fsl_audmix.o +snd-soc-fsl-asoc-card-y := fsl-asoc-card.o +snd-soc-fsl-asrc-y := fsl_asrc.o fsl_asrc_dma.o +snd-soc-fsl-sai-y := fsl_sai.o snd-soc-fsl-ssi-y := fsl_ssi.o snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o -snd-soc-fsl-spdif-objs := fsl_spdif.o -snd-soc-fsl-esai-objs := fsl_esai.o -snd-soc-fsl-micfil-objs := fsl_micfil.o -snd-soc-fsl-utils-objs := fsl_utils.o -snd-soc-fsl-dma-objs := fsl_dma.o -snd-soc-fsl-mqs-objs := fsl_mqs.o -snd-soc-fsl-easrc-objs := fsl_easrc.o -snd-soc-fsl-xcvr-objs := fsl_xcvr.o -snd-soc-fsl-aud2htx-objs := fsl_aud2htx.o -snd-soc-fsl-rpmsg-objs := fsl_rpmsg.o -snd-soc-fsl-qmc-audio-objs := fsl_qmc_audio.o +snd-soc-fsl-spdif-y := fsl_spdif.o +snd-soc-fsl-esai-y := fsl_esai.o +snd-soc-fsl-micfil-y := fsl_micfil.o +snd-soc-fsl-utils-y := fsl_utils.o +snd-soc-fsl-dma-y := fsl_dma.o +snd-soc-fsl-mqs-y := fsl_mqs.o +snd-soc-fsl-easrc-y := fsl_easrc.o +snd-soc-fsl-xcvr-y := fsl_xcvr.o +snd-soc-fsl-aud2htx-y := fsl_aud2htx.o +snd-soc-fsl-rpmsg-y := fsl_rpmsg.o +snd-soc-fsl-qmc-audio-y := fsl_qmc_audio.o obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o @@ -53,7 +53,7 @@ obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o # i.MX Platform Support -snd-soc-imx-audmux-objs := imx-audmux.o +snd-soc-imx-audmux-y := imx-audmux.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o @@ -62,14 +62,14 @@ obj-$(CONFIG_SND_SOC_IMX_AUDIO_RPMSG) += imx-audio-rpmsg.o obj-$(CONFIG_SND_SOC_IMX_PCM_RPMSG) += imx-pcm-rpmsg.o # i.MX Machine Support -snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o -snd-soc-imx-es8328-objs := imx-es8328.o -snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o -snd-soc-imx-spdif-objs := imx-spdif.o -snd-soc-imx-audmix-objs := imx-audmix.o -snd-soc-imx-hdmi-objs := imx-hdmi.o -snd-soc-imx-rpmsg-objs := imx-rpmsg.o -snd-soc-imx-card-objs := imx-card.o +snd-soc-eukrea-tlv320-y := eukrea-tlv320.o +snd-soc-imx-es8328-y := imx-es8328.o +snd-soc-imx-sgtl5000-y := imx-sgtl5000.o +snd-soc-imx-spdif-y := imx-spdif.o +snd-soc-imx-audmix-y := imx-audmix.o +snd-soc-imx-hdmi-y := imx-hdmi.o +snd-soc-imx-rpmsg-y := imx-rpmsg.o +snd-soc-imx-card-y := imx-card.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o From 76560b808b41267cd08a538d2ad536f557c510f3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:15 +0200 Subject: [PATCH 498/603] ASoC: generic: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-12-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/generic/Makefile | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 0848621565060..d5abb3eed3df7 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,10 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 -snd-soc-simple-card-utils-objs := simple-card-utils.o -snd-soc-simple-card-objs := simple-card.o -snd-soc-audio-graph-card-objs := audio-graph-card.o -snd-soc-audio-graph-card2-objs := audio-graph-card2.o -snd-soc-audio-graph-card2-custom-sample-objs := audio-graph-card2-custom-sample.o -snd-soc-test-component-objs := test-component.o +snd-soc-simple-card-utils-y := simple-card-utils.o +snd-soc-simple-card-y := simple-card.o +snd-soc-audio-graph-card-y := audio-graph-card.o +snd-soc-audio-graph-card2-y := audio-graph-card2.o +snd-soc-audio-graph-card2-custom-sample-y := audio-graph-card2-custom-sample.o +snd-soc-test-component-y := test-component.o obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o From 590d82e6e6284eba3e99d15264779c4369709291 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:16 +0200 Subject: [PATCH 499/603] ASoC: intel: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-13-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/intel/atom/Makefile | 2 +- sound/soc/intel/atom/sst/Makefile | 6 +-- sound/soc/intel/avs/Makefile | 10 ++-- sound/soc/intel/avs/boards/Makefile | 34 ++++++------ sound/soc/intel/boards/Makefile | 84 ++++++++++++++--------------- sound/soc/intel/catpt/Makefile | 2 +- sound/soc/intel/common/Makefile | 8 +-- sound/soc/intel/keembay/Makefile | 2 +- sound/soc/intel/skylake/Makefile | 6 +-- 9 files changed, 77 insertions(+), 77 deletions(-) diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile index c66f03f5d8d6d..38e4876025c79 100644 --- a/sound/soc/intel/atom/Makefile +++ b/sound/soc/intel/atom/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-sst-atom-hifi2-platform-objs := sst-mfld-platform-pcm.o \ +snd-soc-sst-atom-hifi2-platform-y := sst-mfld-platform-pcm.o \ sst-mfld-platform-compress.o \ sst-atom-controls.o diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile index 5761d30a5f9d1..16be0463424d6 100644 --- a/sound/soc/intel/atom/sst/Makefile +++ b/sound/soc/intel/atom/sst/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o -snd-intel-sst-pci-objs += sst_pci.o -snd-intel-sst-acpi-objs += sst_acpi.o +snd-intel-sst-core-y := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o +snd-intel-sst-pci-y += sst_pci.o +snd-intel-sst-acpi-y += sst_acpi.o obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index 5480500337f85..5139a019a4ada 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -1,17 +1,17 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-avs-objs := dsp.o ipc.o messages.o utils.o core.o loader.o \ +snd-soc-avs-y := dsp.o ipc.o messages.o utils.o core.o loader.o \ topology.o path.o pcm.o board_selection.o control.o \ sysfs.o -snd-soc-avs-objs += cldma.o -snd-soc-avs-objs += skl.o apl.o cnl.o icl.o tgl.o +snd-soc-avs-y += cldma.o +snd-soc-avs-y += skl.o apl.o cnl.o icl.o tgl.o -snd-soc-avs-objs += trace.o +snd-soc-avs-y += trace.o # tell define_trace.h where to find the trace header CFLAGS_trace.o := -I$(src) ifneq ($(CONFIG_DEBUG_FS),) -snd-soc-avs-objs += probes.o debugfs.o +snd-soc-avs-y += probes.o debugfs.o endif obj-$(CONFIG_SND_SOC_INTEL_AVS) += snd-soc-avs.o diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 0ff21d55be242..4fbd936ffb3e0 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -1,22 +1,22 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-avs-da7219-objs := da7219.o -snd-soc-avs-dmic-objs := dmic.o -snd-soc-avs-es8336-objs := es8336.o -snd-soc-avs-hdaudio-objs := hdaudio.o -snd-soc-avs-i2s-test-objs := i2s_test.o -snd-soc-avs-max98927-objs := max98927.o -snd-soc-avs-max98357a-objs := max98357a.o -snd-soc-avs-max98373-objs := max98373.o -snd-soc-avs-nau8825-objs := nau8825.o -snd-soc-avs-probe-objs := probe.o -snd-soc-avs-rt274-objs := rt274.o -snd-soc-avs-rt286-objs := rt286.o -snd-soc-avs-rt298-objs := rt298.o -snd-soc-avs-rt5514-objs := rt5514.o -snd-soc-avs-rt5663-objs := rt5663.o -snd-soc-avs-rt5682-objs := rt5682.o -snd-soc-avs-ssm4567-objs := ssm4567.o +snd-soc-avs-da7219-y := da7219.o +snd-soc-avs-dmic-y := dmic.o +snd-soc-avs-es8336-y := es8336.o +snd-soc-avs-hdaudio-y := hdaudio.o +snd-soc-avs-i2s-test-y := i2s_test.o +snd-soc-avs-max98927-y := max98927.o +snd-soc-avs-max98357a-y := max98357a.o +snd-soc-avs-max98373-y := max98373.o +snd-soc-avs-nau8825-y := nau8825.o +snd-soc-avs-probe-y := probe.o +snd-soc-avs-rt274-y := rt274.o +snd-soc-avs-rt286-y := rt286.o +snd-soc-avs-rt298-y := rt298.o +snd-soc-avs-rt5514-y := rt5514.o +snd-soc-avs-rt5663-y := rt5663.o +snd-soc-avs-rt5682-y := rt5682.o +snd-soc-avs-ssm4567-y := ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DA7219) += snd-soc-avs-da7219.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9630050557304..dc6fe110f2797 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -1,40 +1,40 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-hsw-rt5640-objs := hsw_rt5640.o -snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o -snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o -snd-soc-bdw-rt286-objs := bdw_rt286.o -snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o -snd-soc-sst-bxt-rt298-objs := bxt_rt298.o -snd-soc-sst-sof-pcm512x-objs := sof_pcm512x.o -snd-soc-sst-sof-wm8804-objs := sof_wm8804.o -snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o -snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o -snd-soc-sst-bytcr-wm5102-objs := bytcr_wm5102.o -snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o -snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o -snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o -snd-soc-sst-cht-bsw-nau8824-objs := cht_bsw_nau8824.o -snd-soc-sst-byt-cht-cx2072x-objs := bytcht_cx2072x.o -snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o -snd-soc-sst-byt-cht-es8316-objs := bytcht_es8316.o -snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o -snd-soc-sof_rt5682-objs := sof_rt5682.o -snd-soc-sof_cs42l42-objs := sof_cs42l42.o -snd-soc-sof_es8336-objs := sof_es8336.o -snd-soc-sof_nau8825-objs := sof_nau8825.o -snd-soc-sof_da7219-objs := sof_da7219.o -snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o -snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o -snd-soc-kbl_rt5663_max98927-objs := kbl_rt5663_max98927.o -snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o -snd-soc-kbl_rt5660-objs := kbl_rt5660.o -snd-soc-skl_rt286-objs := skl_rt286.o -snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o -snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o -snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o -snd-soc-ehl-rt5660-objs := ehl_rt5660.o -snd-soc-sof-ssp-amp-objs := sof_ssp_amp.o -snd-soc-sof-sdw-objs += sof_sdw.o \ +snd-soc-hsw-rt5640-y := hsw_rt5640.o +snd-soc-sst-bdw-rt5650-mach-y := bdw-rt5650.o +snd-soc-sst-bdw-rt5677-mach-y := bdw-rt5677.o +snd-soc-bdw-rt286-y := bdw_rt286.o +snd-soc-sst-bxt-da7219_max98357a-y := bxt_da7219_max98357a.o +snd-soc-sst-bxt-rt298-y := bxt_rt298.o +snd-soc-sst-sof-pcm512x-y := sof_pcm512x.o +snd-soc-sst-sof-wm8804-y := sof_wm8804.o +snd-soc-sst-bytcr-rt5640-y := bytcr_rt5640.o +snd-soc-sst-bytcr-rt5651-y := bytcr_rt5651.o +snd-soc-sst-bytcr-wm5102-y := bytcr_wm5102.o +snd-soc-sst-cht-bsw-rt5672-y := cht_bsw_rt5672.o +snd-soc-sst-cht-bsw-rt5645-y := cht_bsw_rt5645.o +snd-soc-sst-cht-bsw-max98090_ti-y := cht_bsw_max98090_ti.o +snd-soc-sst-cht-bsw-nau8824-y := cht_bsw_nau8824.o +snd-soc-sst-byt-cht-cx2072x-y := bytcht_cx2072x.o +snd-soc-sst-byt-cht-da7213-y := bytcht_da7213.o +snd-soc-sst-byt-cht-es8316-y := bytcht_es8316.o +snd-soc-sst-byt-cht-nocodec-y := bytcht_nocodec.o +snd-soc-sof_rt5682-y := sof_rt5682.o +snd-soc-sof_cs42l42-y := sof_cs42l42.o +snd-soc-sof_es8336-y := sof_es8336.o +snd-soc-sof_nau8825-y := sof_nau8825.o +snd-soc-sof_da7219-y := sof_da7219.o +snd-soc-kbl_da7219_max98357a-y := kbl_da7219_max98357a.o +snd-soc-kbl_da7219_max98927-y := kbl_da7219_max98927.o +snd-soc-kbl_rt5663_max98927-y := kbl_rt5663_max98927.o +snd-soc-kbl_rt5663_rt5514_max98927-y := kbl_rt5663_rt5514_max98927.o +snd-soc-kbl_rt5660-y := kbl_rt5660.o +snd-soc-skl_rt286-y := skl_rt286.o +snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o +snd-skl_nau88l25_max98357a-y := skl_nau88l25_max98357a.o +snd-soc-skl_nau88l25_ssm4567-y := skl_nau88l25_ssm4567.o +snd-soc-ehl-rt5660-y := ehl_rt5660.o +snd-soc-sof-ssp-amp-y := sof_ssp_amp.o +snd-soc-sof-sdw-y += sof_sdw.o \ sof_sdw_maxim.o sof_sdw_rt_amp.o \ bridge_cs35l56.o \ sof_sdw_rt5682.o sof_sdw_rt700.o \ @@ -83,20 +83,20 @@ obj-$(CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH) += snd-soc-sof-sdw.o obj-$(CONFIG_SND_SOC_INTEL_SOF_SSP_AMP_MACH) += snd-soc-sof-ssp-amp.o # common modules -snd-soc-intel-hda-dsp-common-objs := hda_dsp_common.o +snd-soc-intel-hda-dsp-common-y := hda_dsp_common.o obj-$(CONFIG_SND_SOC_INTEL_HDA_DSP_COMMON) += snd-soc-intel-hda-dsp-common.o -snd-soc-intel-sof-maxim-common-objs += sof_maxim_common.o +snd-soc-intel-sof-maxim-common-y += sof_maxim_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_MAXIM_COMMON) += snd-soc-intel-sof-maxim-common.o -snd-soc-intel-sof-realtek-common-objs += sof_realtek_common.o +snd-soc-intel-sof-realtek-common-y += sof_realtek_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_REALTEK_COMMON) += snd-soc-intel-sof-realtek-common.o -snd-soc-intel-sof-cirrus-common-objs += sof_cirrus_common.o +snd-soc-intel-sof-cirrus-common-y += sof_cirrus_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CIRRUS_COMMON) += snd-soc-intel-sof-cirrus-common.o -snd-soc-intel-sof-nuvoton-common-objs += sof_nuvoton_common.o +snd-soc-intel-sof-nuvoton-common-y += sof_nuvoton_common.o obj-$(CONFIG_SND_SOC_INTEL_SOF_NUVOTON_COMMON) += snd-soc-intel-sof-nuvoton-common.o -snd-soc-intel-sof-board-helpers-objs += sof_board_helpers.o +snd-soc-intel-sof-board-helpers-y += sof_board_helpers.o obj-$(CONFIG_SND_SOC_INTEL_SOF_BOARD_HELPERS) += snd-soc-intel-sof-board-helpers.o diff --git a/sound/soc/intel/catpt/Makefile b/sound/soc/intel/catpt/Makefile index c393a45795dae..f5f6a7e956cee 100644 --- a/sound/soc/intel/catpt/Makefile +++ b/sound/soc/intel/catpt/Makefile @@ -1,4 +1,4 @@ -snd-soc-catpt-objs := device.o dsp.o loader.o ipc.o messages.o pcm.o sysfs.o +snd-soc-catpt-y := device.o dsp.o loader.o ipc.o messages.o pcm.o sysfs.o # tell define_trace.h where to find the trace header CFLAGS_device.o := -I$(src) diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index a86457674726b..40a74a19c508c 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-sst-dsp-objs := sst-dsp.o -snd-soc-sst-ipc-objs := sst-ipc.o -snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \ +snd-soc-sst-dsp-y := sst-dsp.o +snd-soc-sst-ipc-y := sst-ipc.o +snd-soc-acpi-intel-match-y := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-match.o \ soc-acpi-intel-hsw-bdw-match.o \ soc-acpi-intel-skl-match.o soc-acpi-intel-kbl-match.o \ soc-acpi-intel-bxt-match.o soc-acpi-intel-glk-match.o \ @@ -15,7 +15,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-hda-match.o \ soc-acpi-intel-sdw-mockup-match.o -snd-soc-acpi-intel-match-objs += soc-acpi-intel-ssp-common.o +snd-soc-acpi-intel-match-y += soc-acpi-intel-ssp-common.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o diff --git a/sound/soc/intel/keembay/Makefile b/sound/soc/intel/keembay/Makefile index 9084e8c638545..3da9a6f9ba2a1 100644 --- a/sound/soc/intel/keembay/Makefile +++ b/sound/soc/intel/keembay/Makefile @@ -1,4 +1,4 @@ -snd-soc-kmb_platform-objs := \ +snd-soc-kmb_platform-y := \ kmb_platform.o obj-$(CONFIG_SND_SOC_INTEL_KEEMBAY) += snd-soc-kmb_platform.o diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index 1c4649bccec5a..ad9be6168428e 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -1,15 +1,15 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-skl-objs := skl.o skl-pcm.o skl-nhlt.o skl-messages.o skl-topology.o \ +snd-soc-skl-y := skl.o skl-pcm.o skl-nhlt.o skl-messages.o skl-topology.o \ skl-sst-ipc.o skl-sst-dsp.o cnl-sst-dsp.o skl-sst-cldma.o \ skl-sst.o bxt-sst.o cnl-sst.o skl-sst-utils.o ifdef CONFIG_DEBUG_FS - snd-soc-skl-objs += skl-debug.o + snd-soc-skl-y += skl-debug.o endif obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += snd-soc-skl.o #Skylake Clock device support -snd-soc-skl-ssp-clk-objs := skl-ssp-clk.o +snd-soc-skl-ssp-clk-y := skl-ssp-clk.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_SSP_CLK) += snd-soc-skl-ssp-clk.o From f7d09b7d4d709b34ccb0ca8fd626a499fa672a6d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:17 +0200 Subject: [PATCH 500/603] ASoC: jz4740: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-14-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/jz4740/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile index f8701c9b09fe2..1c7f3f375318c 100644 --- a/sound/soc/jz4740/Makefile +++ b/sound/soc/jz4740/Makefile @@ -2,6 +2,6 @@ # # Jz4740 Platform Support # -snd-soc-jz4740-i2s-objs := jz4740-i2s.o +snd-soc-jz4740-i2s-y := jz4740-i2s.o obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o From 7ca27af84fc3cef3a8769bbef278e460d794c603 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:18 +0200 Subject: [PATCH 501/603] ASoC: kirkwood: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-15-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/kirkwood/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index e2d279f16a46f..9be1eb8203a10 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o +snd-soc-kirkwood-y := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -snd-soc-armada-370-db-objs := armada-370-db.o +snd-soc-armada-370-db-y := armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o From 3299aa652dee0d9e61cf7223a5e0bf131faec63f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:19 +0200 Subject: [PATCH 502/603] ASoC: loongson: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-16-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/loongson/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile index 601a905a48603..578030ad6563c 100644 --- a/sound/soc/loongson/Makefile +++ b/sound/soc/loongson/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0 #Platform Support -snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o loongson_dma.o +snd-soc-loongson-i2s-pci-y := loongson_i2s_pci.o loongson_i2s.o loongson_dma.o obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o #Machine Support -snd-soc-loongson-card-objs := loongson_card.o +snd-soc-loongson-card-y := loongson_card.o obj-$(CONFIG_SND_SOC_LOONGSON_CARD) += snd-soc-loongson-card.o From 9d819c244a478099828f76a92be4b71360073207 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:20 +0200 Subject: [PATCH 503/603] ASoC: mediatek: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-17-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/mediatek/common/Makefile | 4 ++-- sound/soc/mediatek/mt2701/Makefile | 2 +- sound/soc/mediatek/mt6797/Makefile | 2 +- sound/soc/mediatek/mt7986/Makefile | 2 +- sound/soc/mediatek/mt8183/Makefile | 2 +- sound/soc/mediatek/mt8186/Makefile | 2 +- sound/soc/mediatek/mt8188/Makefile | 2 +- sound/soc/mediatek/mt8192/Makefile | 2 +- sound/soc/mediatek/mt8195/Makefile | 2 +- 9 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/mediatek/common/Makefile b/sound/soc/mediatek/common/Makefile index 363cc258a3d55..c90d276cf4ed1 100644 --- a/sound/soc/mediatek/common/Makefile +++ b/sound/soc/mediatek/common/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mtk-common-objs := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o mtk-soundcard-driver.o -snd-soc-mtk-common-objs += mtk-dai-adda-common.o +snd-soc-mtk-common-y := mtk-afe-platform-driver.o mtk-afe-fe-dai.o mtk-dsp-sof-common.o mtk-soundcard-driver.o +snd-soc-mtk-common-y += mtk-dai-adda-common.o obj-$(CONFIG_SND_SOC_MEDIATEK) += snd-soc-mtk-common.o diff --git a/sound/soc/mediatek/mt2701/Makefile b/sound/soc/mediatek/mt2701/Makefile index 21d5e697cfa7b..507fa26c39452 100644 --- a/sound/soc/mediatek/mt2701/Makefile +++ b/sound/soc/mediatek/mt2701/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mt2701-afe-objs := mt2701-afe-pcm.o mt2701-afe-clock-ctrl.o +snd-soc-mt2701-afe-y := mt2701-afe-pcm.o mt2701-afe-clock-ctrl.o obj-$(CONFIG_SND_SOC_MT2701) += snd-soc-mt2701-afe.o # machine driver diff --git a/sound/soc/mediatek/mt6797/Makefile b/sound/soc/mediatek/mt6797/Makefile index bf6e179ea93ff..150021495e94c 100644 --- a/sound/soc/mediatek/mt6797/Makefile +++ b/sound/soc/mediatek/mt6797/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mt6797-afe-objs := \ +snd-soc-mt6797-afe-y := \ mt6797-afe-pcm.o \ mt6797-afe-clk.o \ mt6797-dai-pcm.o \ diff --git a/sound/soc/mediatek/mt7986/Makefile b/sound/soc/mediatek/mt7986/Makefile index fc4c82559b298..4b54bbe88683e 100644 --- a/sound/soc/mediatek/mt7986/Makefile +++ b/sound/soc/mediatek/mt7986/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mt7986-afe-objs := \ +snd-soc-mt7986-afe-y := \ mt7986-afe-pcm.o \ mt7986-dai-etdm.o diff --git a/sound/soc/mediatek/mt8183/Makefile b/sound/soc/mediatek/mt8183/Makefile index c0a3bbc2c1f6c..0d0dcdde00fcc 100644 --- a/sound/soc/mediatek/mt8183/Makefile +++ b/sound/soc/mediatek/mt8183/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mt8183-afe-objs := \ +snd-soc-mt8183-afe-y := \ mt8183-afe-pcm.o \ mt8183-afe-clk.o \ mt8183-dai-i2s.o \ diff --git a/sound/soc/mediatek/mt8186/Makefile b/sound/soc/mediatek/mt8186/Makefile index fa2f0efce8159..ab3f5b763df89 100644 --- a/sound/soc/mediatek/mt8186/Makefile +++ b/sound/soc/mediatek/mt8186/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mt8186-afe-objs := \ +snd-soc-mt8186-afe-y := \ mt8186-afe-pcm.o \ mt8186-audsys-clk.o \ mt8186-afe-clk.o \ diff --git a/sound/soc/mediatek/mt8188/Makefile b/sound/soc/mediatek/mt8188/Makefile index 781e61cbb22b0..1178bce45c50b 100644 --- a/sound/soc/mediatek/mt8188/Makefile +++ b/sound/soc/mediatek/mt8188/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mt8188-afe-objs := \ +snd-soc-mt8188-afe-y := \ mt8188-afe-clk.o \ mt8188-afe-pcm.o \ mt8188-audsys-clk.o \ diff --git a/sound/soc/mediatek/mt8192/Makefile b/sound/soc/mediatek/mt8192/Makefile index 8b27d82626ea4..d60c36bcdccec 100644 --- a/sound/soc/mediatek/mt8192/Makefile +++ b/sound/soc/mediatek/mt8192/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mt8192-afe-objs := \ +snd-soc-mt8192-afe-y := \ mt8192-afe-pcm.o \ mt8192-afe-clk.o \ mt8192-afe-gpio.o \ diff --git a/sound/soc/mediatek/mt8195/Makefile b/sound/soc/mediatek/mt8195/Makefile index aae673ec751b0..014e93dace264 100644 --- a/sound/soc/mediatek/mt8195/Makefile +++ b/sound/soc/mediatek/mt8195/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # platform driver -snd-soc-mt8195-afe-objs := \ +snd-soc-mt8195-afe-y := \ mt8195-audsys-clk.o \ mt8195-afe-clk.o \ mt8195-afe-pcm.o \ From bfc05938ebab6429543fcf1df1279b24953f1a7b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:21 +0200 Subject: [PATCH 504/603] ASoC: meson: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-18-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/meson/Makefile | 50 ++++++++++++++++++++-------------------- 1 file changed, 25 insertions(+), 25 deletions(-) diff --git a/sound/soc/meson/Makefile b/sound/soc/meson/Makefile index e446bc9804818..24078e4396b02 100644 --- a/sound/soc/meson/Makefile +++ b/sound/soc/meson/Makefile @@ -1,30 +1,30 @@ # SPDX-License-Identifier: (GPL-2.0 OR MIT) -snd-soc-meson-aiu-objs := aiu.o -snd-soc-meson-aiu-objs += aiu-acodec-ctrl.o -snd-soc-meson-aiu-objs += aiu-codec-ctrl.o -snd-soc-meson-aiu-objs += aiu-encoder-i2s.o -snd-soc-meson-aiu-objs += aiu-encoder-spdif.o -snd-soc-meson-aiu-objs += aiu-fifo.o -snd-soc-meson-aiu-objs += aiu-fifo-i2s.o -snd-soc-meson-aiu-objs += aiu-fifo-spdif.o -snd-soc-meson-axg-fifo-objs := axg-fifo.o -snd-soc-meson-axg-frddr-objs := axg-frddr.o -snd-soc-meson-axg-toddr-objs := axg-toddr.o -snd-soc-meson-axg-tdm-formatter-objs := axg-tdm-formatter.o -snd-soc-meson-axg-tdm-interface-objs := axg-tdm-interface.o -snd-soc-meson-axg-tdmin-objs := axg-tdmin.o -snd-soc-meson-axg-tdmout-objs := axg-tdmout.o -snd-soc-meson-axg-sound-card-objs := axg-card.o -snd-soc-meson-axg-spdifin-objs := axg-spdifin.o -snd-soc-meson-axg-spdifout-objs := axg-spdifout.o -snd-soc-meson-axg-pdm-objs := axg-pdm.o -snd-soc-meson-card-utils-objs := meson-card-utils.o -snd-soc-meson-codec-glue-objs := meson-codec-glue.o -snd-soc-meson-gx-sound-card-objs := gx-card.o -snd-soc-meson-g12a-toacodec-objs := g12a-toacodec.o -snd-soc-meson-g12a-tohdmitx-objs := g12a-tohdmitx.o -snd-soc-meson-t9015-objs := t9015.o +snd-soc-meson-aiu-y := aiu.o +snd-soc-meson-aiu-y += aiu-acodec-ctrl.o +snd-soc-meson-aiu-y += aiu-codec-ctrl.o +snd-soc-meson-aiu-y += aiu-encoder-i2s.o +snd-soc-meson-aiu-y += aiu-encoder-spdif.o +snd-soc-meson-aiu-y += aiu-fifo.o +snd-soc-meson-aiu-y += aiu-fifo-i2s.o +snd-soc-meson-aiu-y += aiu-fifo-spdif.o +snd-soc-meson-axg-fifo-y := axg-fifo.o +snd-soc-meson-axg-frddr-y := axg-frddr.o +snd-soc-meson-axg-toddr-y := axg-toddr.o +snd-soc-meson-axg-tdm-formatter-y := axg-tdm-formatter.o +snd-soc-meson-axg-tdm-interface-y := axg-tdm-interface.o +snd-soc-meson-axg-tdmin-y := axg-tdmin.o +snd-soc-meson-axg-tdmout-y := axg-tdmout.o +snd-soc-meson-axg-sound-card-y := axg-card.o +snd-soc-meson-axg-spdifin-y := axg-spdifin.o +snd-soc-meson-axg-spdifout-y := axg-spdifout.o +snd-soc-meson-axg-pdm-y := axg-pdm.o +snd-soc-meson-card-utils-y := meson-card-utils.o +snd-soc-meson-codec-glue-y := meson-codec-glue.o +snd-soc-meson-gx-sound-card-y := gx-card.o +snd-soc-meson-g12a-toacodec-y := g12a-toacodec.o +snd-soc-meson-g12a-tohdmitx-y := g12a-tohdmitx.o +snd-soc-meson-t9015-y := t9015.o obj-$(CONFIG_SND_MESON_AIU) += snd-soc-meson-aiu.o obj-$(CONFIG_SND_MESON_AXG_FIFO) += snd-soc-meson-axg-fifo.o From 62c6137ab2d0598603115f5bacff38c440a7d8c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:22 +0200 Subject: [PATCH 505/603] ASoC: mxs: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-19-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/mxs/Makefile | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile index ab0a9a553702b..474bdd75513f9 100644 --- a/sound/soc/mxs/Makefile +++ b/sound/soc/mxs/Makefile @@ -1,11 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # MXS Platform Support -snd-soc-mxs-objs := mxs-saif.o -snd-soc-mxs-pcm-objs := mxs-pcm.o +snd-soc-mxs-y := mxs-saif.o +snd-soc-mxs-pcm-y := mxs-pcm.o obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o snd-soc-mxs-pcm.o # i.MX Machine Support -snd-soc-mxs-sgtl5000-objs := mxs-sgtl5000.o +snd-soc-mxs-sgtl5000-y := mxs-sgtl5000.o obj-$(CONFIG_SND_SOC_MXS_SGTL5000) += snd-soc-mxs-sgtl5000.o From 8a72584d0d7601699f7945ac84899e8a3c7cf617 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:23 +0200 Subject: [PATCH 506/603] ASoC: pxa: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-20-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/pxa/Makefile | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 406605fc7414d..93b4e57eaa5c5 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -1,10 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 # PXA Platform Support -snd-soc-pxa2xx-objs := pxa2xx-pcm.o -snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o -snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o -snd-soc-pxa-ssp-objs := pxa-ssp.o -snd-soc-mmp-sspa-objs := mmp-sspa.o +snd-soc-pxa2xx-y := pxa2xx-pcm.o +snd-soc-pxa2xx-ac97-y := pxa2xx-ac97.o +snd-soc-pxa2xx-i2s-y := pxa2xx-i2s.o +snd-soc-pxa-ssp-y := pxa-ssp.o +snd-soc-mmp-sspa-y := mmp-sspa.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o @@ -13,5 +13,5 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o # PXA Machine Support -snd-soc-spitz-objs := spitz.o +snd-soc-spitz-y := spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o From 078f57bd555b092e85c02510a2cf8bd30773cc3b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:24 +0200 Subject: [PATCH 507/603] ASoC: qcom: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-21-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/qcom/Makefile | 38 +++++++++++++++++------------------ sound/soc/qcom/qdsp6/Makefile | 4 ++-- 2 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 34f3fcb8ee9aa..16db7b53ddac7 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -1,13 +1,13 @@ # SPDX-License-Identifier: GPL-2.0 # Platform -snd-soc-lpass-cpu-objs := lpass-cpu.o -snd-soc-lpass-cdc-dma-objs := lpass-cdc-dma.o -snd-soc-lpass-hdmi-objs := lpass-hdmi.o -snd-soc-lpass-platform-objs := lpass-platform.o -snd-soc-lpass-ipq806x-objs := lpass-ipq806x.o -snd-soc-lpass-apq8016-objs := lpass-apq8016.o -snd-soc-lpass-sc7180-objs := lpass-sc7180.o -snd-soc-lpass-sc7280-objs := lpass-sc7280.o +snd-soc-lpass-cpu-y := lpass-cpu.o +snd-soc-lpass-cdc-dma-y := lpass-cdc-dma.o +snd-soc-lpass-hdmi-y := lpass-hdmi.o +snd-soc-lpass-platform-y := lpass-platform.o +snd-soc-lpass-ipq806x-y := lpass-ipq806x.o +snd-soc-lpass-apq8016-y := lpass-apq8016.o +snd-soc-lpass-sc7180-y := lpass-sc7180.o +snd-soc-lpass-sc7280-y := lpass-sc7280.o obj-$(CONFIG_SND_SOC_LPASS_CPU) += snd-soc-lpass-cpu.o obj-$(CONFIG_SND_SOC_LPASS_CDC_DMA) += snd-soc-lpass-cdc-dma.o @@ -19,17 +19,17 @@ obj-$(CONFIG_SND_SOC_LPASS_SC7180) += snd-soc-lpass-sc7180.o obj-$(CONFIG_SND_SOC_LPASS_SC7280) += snd-soc-lpass-sc7280.o # Machine -snd-soc-storm-objs := storm.o -snd-soc-apq8016-sbc-objs := apq8016_sbc.o -snd-soc-apq8096-objs := apq8096.o -snd-soc-sc7180-objs := sc7180.o -snd-soc-sc7280-objs := sc7280.o -snd-soc-sdm845-objs := sdm845.o -snd-soc-sm8250-objs := sm8250.o -snd-soc-sc8280xp-objs := sc8280xp.o -snd-soc-qcom-common-objs := common.o -snd-soc-qcom-sdw-objs := sdw.o -snd-soc-x1e80100-objs := x1e80100.o +snd-soc-storm-y := storm.o +snd-soc-apq8016-sbc-y := apq8016_sbc.o +snd-soc-apq8096-y := apq8096.o +snd-soc-sc7180-y := sc7180.o +snd-soc-sc7280-y := sc7280.o +snd-soc-sdm845-y := sdm845.o +snd-soc-sm8250-y := sm8250.o +snd-soc-sc8280xp-y := sc8280xp.o +snd-soc-qcom-common-y := common.o +snd-soc-qcom-sdw-y := sdw.o +snd-soc-x1e80100-y := x1e80100.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile index 3963bf2346643..26b7c55c9c11e 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-q6dsp-common-objs := q6dsp-common.o q6dsp-lpass-ports.o q6dsp-lpass-clocks.o -snd-q6apm-objs := q6apm.o audioreach.o topology.o +snd-q6dsp-common-y := q6dsp-common.o q6dsp-lpass-ports.o q6dsp-lpass-clocks.o +snd-q6apm-y := q6apm.o audioreach.o topology.o obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += snd-q6dsp-common.o obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o From 18db1c48515ed358a1c29e5dbda3e39a822b1864 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:25 +0200 Subject: [PATCH 508/603] ASoC: rockchip: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-22-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/rockchip/Makefile | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index 30c57c0d76603..2ee9c08131d1e 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,19 +1,19 @@ # SPDX-License-Identifier: GPL-2.0 # ROCKCHIP Platform Support -snd-soc-rockchip-i2s-objs := rockchip_i2s.o -snd-soc-rockchip-i2s-tdm-objs := rockchip_i2s_tdm.o -snd-soc-rockchip-pdm-objs := rockchip_pdm.o -snd-soc-rockchip-spdif-objs := rockchip_spdif.o +snd-soc-rockchip-i2s-y := rockchip_i2s.o +snd-soc-rockchip-i2s-tdm-y := rockchip_i2s_tdm.o +snd-soc-rockchip-pdm-y := rockchip_pdm.o +snd-soc-rockchip-spdif-y := rockchip_spdif.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-rockchip-i2s.o obj-$(CONFIG_SND_SOC_ROCKCHIP_PDM) += snd-soc-rockchip-pdm.o obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S_TDM) += snd-soc-rockchip-i2s-tdm.o -snd-soc-rockchip-max98090-objs := rockchip_max98090.o -snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o -snd-soc-rk3288-hdmi-analog-objs := rk3288_hdmi_analog.o -snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o +snd-soc-rockchip-max98090-y := rockchip_max98090.o +snd-soc-rockchip-rt5645-y := rockchip_rt5645.o +snd-soc-rk3288-hdmi-analog-y := rk3288_hdmi_analog.o +snd-soc-rk3399-gru-sound-y := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o From e93c3b9606d17344a17e061c51dc4ec0538b3345 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:26 +0200 Subject: [PATCH 509/603] ASoC: samsung: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-23-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/samsung/Makefile | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index f5d327b90a4ec..8d5f091479008 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,10 +1,10 @@ # SPDX-License-Identifier: GPL-2.0 # S3c24XX Platform Support -snd-soc-s3c-dma-objs := dmaengine.o -snd-soc-idma-objs := idma.o -snd-soc-samsung-spdif-objs := spdif.o -snd-soc-pcm-objs := pcm.o -snd-soc-i2s-objs := i2s.o +snd-soc-s3c-dma-y := dmaengine.o +snd-soc-idma-y := idma.o +snd-soc-samsung-spdif-y := spdif.o +snd-soc-pcm-y := pcm.o +snd-soc-i2s-y := i2s.o obj-$(CONFIG_SND_SOC_SAMSUNG) += snd-soc-s3c-dma.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o @@ -13,20 +13,20 @@ obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support -snd-soc-smdk-wm8994-objs := smdk_wm8994.o -snd-soc-snow-objs := snow.o -snd-soc-smdk-spdif-objs := smdk_spdif.o -snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o -snd-soc-speyside-objs := speyside.o -snd-soc-tobermory-objs := tobermory.o -snd-soc-lowland-objs := lowland.o -snd-soc-littlemill-objs := littlemill.o -snd-soc-bells-objs := bells.o -snd-soc-odroid-objs := odroid.o -snd-soc-arndale-objs := arndale.o -snd-soc-tm2-wm5110-objs := tm2_wm5110.o -snd-soc-aries-wm8994-objs := aries_wm8994.o -snd-soc-midas-wm1811-objs := midas_wm1811.o +snd-soc-smdk-wm8994-y := smdk_wm8994.o +snd-soc-snow-y := snow.o +snd-soc-smdk-spdif-y := smdk_spdif.o +snd-soc-smdk-wm8994pcm-y := smdk_wm8994pcm.o +snd-soc-speyside-y := speyside.o +snd-soc-tobermory-y := tobermory.o +snd-soc-lowland-y := lowland.o +snd-soc-littlemill-y := littlemill.o +snd-soc-bells-y := bells.o +snd-soc-odroid-y := odroid.o +snd-soc-arndale-y := arndale.o +snd-soc-tm2-wm5110-y := tm2_wm5110.o +snd-soc-aries-wm8994-y := aries_wm8994.o +snd-soc-midas-wm1811-y := midas_wm1811.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8994) += snd-soc-smdk-wm8994.o obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o From 8c26836f5bade6d8aef415bdbe8d290b8bb8a5d9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:27 +0200 Subject: [PATCH 510/603] ASoC: sh: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-24-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/sh/Makefile | 16 ++++++++-------- sound/soc/sh/rcar/Makefile | 2 +- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index f6fd79948f6ae..f0e19cbd1581b 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -1,13 +1,13 @@ # SPDX-License-Identifier: GPL-2.0 ## DMA engines -snd-soc-dma-sh7760-objs := dma-sh7760.o +snd-soc-dma-sh7760-y := dma-sh7760.o obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o ## audio units found on some SH-4 -snd-soc-hac-objs := hac.o -snd-soc-ssi-objs := ssi.o -snd-soc-fsi-objs := fsi.o -snd-soc-siu-objs := siu_pcm.o siu_dai.o +snd-soc-hac-y := hac.o +snd-soc-ssi-y := ssi.o +snd-soc-fsi-y := fsi.o +snd-soc-siu-y := siu_pcm.o siu_dai.o obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o @@ -17,12 +17,12 @@ obj-$(CONFIG_SND_SOC_SH4_SIU) += snd-soc-siu.o obj-$(CONFIG_SND_SOC_RCAR) += rcar/ ## boards -snd-soc-sh7760-ac97-objs := sh7760-ac97.o -snd-soc-migor-objs := migor.o +snd-soc-sh7760-ac97-y := sh7760-ac97.o +snd-soc-migor-y := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o # RZ/G2L -snd-soc-rz-ssi-objs := rz-ssi.o +snd-soc-rz-ssi-y := rz-ssi.o obj-$(CONFIG_SND_SOC_RZ) += snd-soc-rz-ssi.o diff --git a/sound/soc/sh/rcar/Makefile b/sound/soc/sh/rcar/Makefile index d07eccfa3ac2b..45eb875a912a6 100644 --- a/sound/soc/sh/rcar/Makefile +++ b/sound/soc/sh/rcar/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0 -snd-soc-rcar-objs := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o debugfs.o +snd-soc-rcar-y := core.o gen.o dma.o adg.o ssi.o ssiu.o src.o ctu.o mix.o dvc.o cmd.o debugfs.o obj-$(CONFIG_SND_SOC_RCAR) += snd-soc-rcar.o From 9c2f5b6eb8b7da05e13cde60c32e0a8b1f5873b0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:28 +0200 Subject: [PATCH 511/603] ASoC: SOF: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-25-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/sof/Makefile | 30 +++++++++++++------------- sound/soc/sof/amd/Makefile | 10 ++++----- sound/soc/sof/imx/Makefile | 8 +++---- sound/soc/sof/intel/Makefile | 30 +++++++++++++------------- sound/soc/sof/mediatek/mt8186/Makefile | 2 +- sound/soc/sof/mediatek/mt8195/Makefile | 2 +- sound/soc/sof/xtensa/Makefile | 2 +- 7 files changed, 42 insertions(+), 42 deletions(-) diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 3624124575afd..b0b22e6ebc03f 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -1,44 +1,44 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) -snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ +snd-sof-y := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\ fw-file-profile.o # IPC implementations ifneq ($(CONFIG_SND_SOC_SOF_IPC3),) -snd-sof-objs += ipc3.o ipc3-loader.o ipc3-topology.o ipc3-control.o ipc3-pcm.o\ +snd-sof-y += ipc3.o ipc3-loader.o ipc3-topology.o ipc3-control.o ipc3-pcm.o\ ipc3-dtrace.o endif ifneq ($(CONFIG_SND_SOC_SOF_IPC4),) -snd-sof-objs += ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o\ +snd-sof-y += ipc4.o ipc4-loader.o ipc4-topology.o ipc4-control.o ipc4-pcm.o\ ipc4-mtrace.o ipc4-telemetry.o endif # SOF client support ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) -snd-sof-objs += sof-client.o +snd-sof-y += sof-client.o endif snd-sof-$(CONFIG_SND_SOC_SOF_COMPRESS) += compress.o -snd-sof-pci-objs := sof-pci-dev.o -snd-sof-acpi-objs := sof-acpi-dev.o -snd-sof-of-objs := sof-of-dev.o +snd-sof-pci-y := sof-pci-dev.o +snd-sof-acpi-y := sof-acpi-dev.o +snd-sof-of-y := sof-of-dev.o -snd-sof-ipc-flood-test-objs := sof-client-ipc-flood-test.o -snd-sof-ipc-msg-injector-objs := sof-client-ipc-msg-injector.o -snd-sof-ipc-kernel-injector-objs := sof-client-ipc-kernel-injector.o -snd-sof-probes-objs := sof-client-probes.o +snd-sof-ipc-flood-test-y := sof-client-ipc-flood-test.o +snd-sof-ipc-msg-injector-y := sof-client-ipc-msg-injector.o +snd-sof-ipc-kernel-injector-y := sof-client-ipc-kernel-injector.o +snd-sof-probes-y := sof-client-probes.o ifneq ($(CONFIG_SND_SOC_SOF_IPC3),) -snd-sof-probes-objs += sof-client-probes-ipc3.o +snd-sof-probes-y += sof-client-probes-ipc3.o endif ifneq ($(CONFIG_SND_SOC_SOF_IPC4),) -snd-sof-probes-objs += sof-client-probes-ipc4.o +snd-sof-probes-y += sof-client-probes-ipc4.o endif -snd-sof-nocodec-objs := nocodec.o +snd-sof-nocodec-y := nocodec.o -snd-sof-utils-objs := sof-utils.o +snd-sof-utils-y := sof-utils.o obj-$(CONFIG_SND_SOC_SOF) += snd-sof.o obj-$(CONFIG_SND_SOC_SOF_NOCODEC) += snd-sof-nocodec.o diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile index ad25f42061771..380a9769c3213 100644 --- a/sound/soc/sof/amd/Makefile +++ b/sound/soc/sof/amd/Makefile @@ -4,12 +4,12 @@ # # Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. -snd-sof-amd-acp-objs := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o +snd-sof-amd-acp-y := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) = acp-probes.o -snd-sof-amd-renoir-objs := pci-rn.o renoir.o -snd-sof-amd-rembrandt-objs := pci-rmb.o rembrandt.o -snd-sof-amd-vangogh-objs := pci-vangogh.o vangogh.o -snd-sof-amd-acp63-objs := pci-acp63.o acp63.o +snd-sof-amd-renoir-y := pci-rn.o renoir.o +snd-sof-amd-rembrandt-y := pci-rmb.o rembrandt.o +snd-sof-amd-vangogh-y := pci-vangogh.o vangogh.o +snd-sof-amd-acp63-y := pci-acp63.o acp63.o obj-$(CONFIG_SND_SOC_SOF_AMD_COMMON) += snd-sof-amd-acp.o obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) +=snd-sof-amd-renoir.o diff --git a/sound/soc/sof/imx/Makefile b/sound/soc/sof/imx/Makefile index 798b43a415bf9..be0bf0736dfa3 100644 --- a/sound/soc/sof/imx/Makefile +++ b/sound/soc/sof/imx/Makefile @@ -1,9 +1,9 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) -snd-sof-imx8-objs := imx8.o -snd-sof-imx8m-objs := imx8m.o -snd-sof-imx8ulp-objs := imx8ulp.o +snd-sof-imx8-y := imx8.o +snd-sof-imx8m-y := imx8m.o +snd-sof-imx8ulp-y := imx8ulp.o -snd-sof-imx-common-objs := imx-common.o +snd-sof-imx-common-y := imx-common.o obj-$(CONFIG_SND_SOC_SOF_IMX8) += snd-sof-imx8.o obj-$(CONFIG_SND_SOC_SOF_IMX8M) += snd-sof-imx8m.o diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index 806df08e3fd58..b56fa5530b8b1 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -1,22 +1,22 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) -snd-sof-acpi-intel-byt-objs := byt.o -snd-sof-acpi-intel-bdw-objs := bdw.o +snd-sof-acpi-intel-byt-y := byt.o +snd-sof-acpi-intel-bdw-y := bdw.o -snd-sof-intel-hda-common-objs := hda-loader.o hda-stream.o hda-trace.o \ +snd-sof-intel-hda-common-y := hda-loader.o hda-stream.o hda-trace.o \ hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ hda-dai.o hda-dai-ops.o hda-bus.o \ telemetry.o tracepoints.o -snd-sof-intel-hda-generic-objs := hda.o hda-common-ops.o +snd-sof-intel-hda-generic-y := hda.o hda-common-ops.o -snd-sof-intel-hda-mlink-objs := hda-mlink.o +snd-sof-intel-hda-mlink-y := hda-mlink.o snd-sof-intel-hda-common-$(CONFIG_SND_SOC_SOF_HDA_PROBES) += hda-probes.o -snd-sof-intel-hda-objs := hda-codec.o +snd-sof-intel-hda-y := hda-codec.o -snd-sof-intel-atom-objs := atom.o +snd-sof-intel-atom-y := atom.o obj-$(CONFIG_SND_SOC_SOF_INTEL_ATOM_HIFI_EP) += snd-sof-intel-atom.o obj-$(CONFIG_SND_SOC_SOF_BAYTRAIL) += snd-sof-acpi-intel-byt.o @@ -26,14 +26,14 @@ obj-$(CONFIG_SND_SOC_SOF_HDA_GENERIC) += snd-sof-intel-hda-generic.o obj-$(CONFIG_SND_SOC_SOF_HDA_MLINK) += snd-sof-intel-hda-mlink.o obj-$(CONFIG_SND_SOC_SOF_HDA) += snd-sof-intel-hda.o -snd-sof-pci-intel-tng-objs := pci-tng.o -snd-sof-pci-intel-skl-objs := pci-skl.o skl.o hda-loader-skl.o -snd-sof-pci-intel-apl-objs := pci-apl.o apl.o -snd-sof-pci-intel-cnl-objs := pci-cnl.o cnl.o -snd-sof-pci-intel-icl-objs := pci-icl.o icl.o -snd-sof-pci-intel-tgl-objs := pci-tgl.o tgl.o -snd-sof-pci-intel-mtl-objs := pci-mtl.o mtl.o -snd-sof-pci-intel-lnl-objs := pci-lnl.o lnl.o +snd-sof-pci-intel-tng-y := pci-tng.o +snd-sof-pci-intel-skl-y := pci-skl.o skl.o hda-loader-skl.o +snd-sof-pci-intel-apl-y := pci-apl.o apl.o +snd-sof-pci-intel-cnl-y := pci-cnl.o cnl.o +snd-sof-pci-intel-icl-y := pci-icl.o icl.o +snd-sof-pci-intel-tgl-y := pci-tgl.o tgl.o +snd-sof-pci-intel-mtl-y := pci-mtl.o mtl.o +snd-sof-pci-intel-lnl-y := pci-lnl.o lnl.o obj-$(CONFIG_SND_SOC_SOF_MERRIFIELD) += snd-sof-pci-intel-tng.o obj-$(CONFIG_SND_SOC_SOF_INTEL_SKL) += snd-sof-pci-intel-skl.o diff --git a/sound/soc/sof/mediatek/mt8186/Makefile b/sound/soc/sof/mediatek/mt8186/Makefile index c1f5fc4e24959..022f415afac99 100644 --- a/sound/soc/sof/mediatek/mt8186/Makefile +++ b/sound/soc/sof/mediatek/mt8186/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) -snd-sof-mt8186-objs := mt8186.o mt8186-clk.o mt8186-loader.o +snd-sof-mt8186-y := mt8186.o mt8186-clk.o mt8186-loader.o obj-$(CONFIG_SND_SOC_SOF_MT8186) += snd-sof-mt8186.o diff --git a/sound/soc/sof/mediatek/mt8195/Makefile b/sound/soc/sof/mediatek/mt8195/Makefile index afc4f21fccc50..f5eeda380b500 100644 --- a/sound/soc/sof/mediatek/mt8195/Makefile +++ b/sound/soc/sof/mediatek/mt8195/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) -snd-sof-mt8195-objs := mt8195.o mt8195-clk.o mt8195-loader.o +snd-sof-mt8195-y := mt8195.o mt8195-clk.o mt8195-loader.o obj-$(CONFIG_SND_SOC_SOF_MT8195) += snd-sof-mt8195.o diff --git a/sound/soc/sof/xtensa/Makefile b/sound/soc/sof/xtensa/Makefile index b8376ea04bcfb..b9e6e8f5a7f62 100644 --- a/sound/soc/sof/xtensa/Makefile +++ b/sound/soc/sof/xtensa/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) -snd-sof-xtensa-dsp-objs := core.o +snd-sof-xtensa-dsp-y := core.o obj-$(CONFIG_SND_SOC_SOF_XTENSA) += snd-sof-xtensa-dsp.o From 50537c3fae4114dc71cf8b32c71881ca57fd3ad1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:29 +0200 Subject: [PATCH 512/603] ASoC: spear: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-26-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/spear/Makefile | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile index 31d9dae280e7a..61569d2e76060 100644 --- a/sound/soc/spear/Makefile +++ b/sound/soc/spear/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0 # SPEAR Platform Support -snd-soc-spear-pcm-objs := spear_pcm.o -snd-soc-spear-spdif-in-objs := spdif_in.o -snd-soc-spear-spdif-out-objs := spdif_out.o +snd-soc-spear-pcm-y := spear_pcm.o +snd-soc-spear-spdif-in-y := spdif_in.o +snd-soc-spear-spdif-out-y := spdif_out.o obj-$(CONFIG_SND_SPEAR_SOC) += snd-soc-spear-pcm.o obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += snd-soc-spear-spdif-in.o From c4e93fd13235fdc339d087d4b1f491311f9e6a1d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:30 +0200 Subject: [PATCH 513/603] ASoC: sprd: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-27-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/sprd/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sprd/Makefile b/sound/soc/sprd/Makefile index a95fa56cd000a..c36399414862a 100644 --- a/sound/soc/sprd/Makefile +++ b/sound/soc/sprd/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 # Spreadtrum Audio Support -snd-soc-sprd-platform-objs := sprd-pcm-dma.o sprd-pcm-compress.o +snd-soc-sprd-platform-y := sprd-pcm-dma.o sprd-pcm-compress.o obj-$(CONFIG_SND_SOC_SPRD) += snd-soc-sprd-platform.o From aa4a610309028d38b041cca899b772023178d112 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:31 +0200 Subject: [PATCH 514/603] ASoC: sti: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-28-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/sti/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sti/Makefile b/sound/soc/sti/Makefile index 787ccb5212983..dfd4f4dd36fce 100644 --- a/sound/soc/sti/Makefile +++ b/sound/soc/sti/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only # STI platform support -snd-soc-sti-objs := sti_uniperif.o uniperif_player.o uniperif_reader.o +snd-soc-sti-y := sti_uniperif.o uniperif_player.o uniperif_reader.o obj-$(CONFIG_SND_SOC_STI) += snd-soc-sti.o From f827d7a34bbc1922c36aae80ff30c9ef9fb3647d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:32 +0200 Subject: [PATCH 515/603] ASoC: stm: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-29-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/stm/Makefile | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/stm/Makefile b/sound/soc/stm/Makefile index 3143c0b470426..3372432faa091 100644 --- a/sound/soc/stm/Makefile +++ b/sound/soc/stm/Makefile @@ -1,17 +1,17 @@ # SPDX-License-Identifier: GPL-2.0 # SAI -snd-soc-stm32-sai-sub-objs := stm32_sai_sub.o +snd-soc-stm32-sai-sub-y := stm32_sai_sub.o obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai-sub.o -snd-soc-stm32-sai-objs := stm32_sai.o +snd-soc-stm32-sai-y := stm32_sai.o obj-$(CONFIG_SND_SOC_STM32_SAI) += snd-soc-stm32-sai.o # I2S -snd-soc-stm32-i2s-objs := stm32_i2s.o +snd-soc-stm32-i2s-y := stm32_i2s.o obj-$(CONFIG_SND_SOC_STM32_I2S) += snd-soc-stm32-i2s.o # SPDIFRX -snd-soc-stm32-spdifrx-objs := stm32_spdifrx.o +snd-soc-stm32-spdifrx-y := stm32_spdifrx.o obj-$(CONFIG_SND_SOC_STM32_SPDIFRX) += snd-soc-stm32-spdifrx.o #DFSDM From 51a50d6ad72702dd8de352c5882a18001638dc30 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:33 +0200 Subject: [PATCH 516/603] ASoC: tegra: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-30-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/tegra/Makefile | 46 ++++++++++++++++++++-------------------- 1 file changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index b723c78e665da..cea4b0d543787 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -1,25 +1,25 @@ # SPDX-License-Identifier: GPL-2.0 # Tegra platform Support -snd-soc-tegra-pcm-objs := tegra_pcm.o -snd-soc-tegra-utils-objs += tegra_asoc_utils.o -snd-soc-tegra20-ac97-objs := tegra20_ac97.o -snd-soc-tegra20-das-objs := tegra20_das.o -snd-soc-tegra20-i2s-objs := tegra20_i2s.o -snd-soc-tegra20-spdif-objs := tegra20_spdif.o -snd-soc-tegra30-ahub-objs := tegra30_ahub.o -snd-soc-tegra30-i2s-objs := tegra30_i2s.o -snd-soc-tegra210-ahub-objs := tegra210_ahub.o -snd-soc-tegra210-dmic-objs := tegra210_dmic.o -snd-soc-tegra210-i2s-objs := tegra210_i2s.o -snd-soc-tegra186-asrc-objs := tegra186_asrc.o -snd-soc-tegra186-dspk-objs := tegra186_dspk.o -snd-soc-tegra210-admaif-objs := tegra210_admaif.o -snd-soc-tegra210-mvc-objs := tegra210_mvc.o -snd-soc-tegra210-sfc-objs := tegra210_sfc.o -snd-soc-tegra210-amx-objs := tegra210_amx.o -snd-soc-tegra210-adx-objs := tegra210_adx.o -snd-soc-tegra210-mixer-objs := tegra210_mixer.o -snd-soc-tegra210-ope-objs := tegra210_ope.o tegra210_mbdrc.o tegra210_peq.o +snd-soc-tegra-pcm-y := tegra_pcm.o +snd-soc-tegra-utils-y += tegra_asoc_utils.o +snd-soc-tegra20-ac97-y := tegra20_ac97.o +snd-soc-tegra20-das-y := tegra20_das.o +snd-soc-tegra20-i2s-y := tegra20_i2s.o +snd-soc-tegra20-spdif-y := tegra20_spdif.o +snd-soc-tegra30-ahub-y := tegra30_ahub.o +snd-soc-tegra30-i2s-y := tegra30_i2s.o +snd-soc-tegra210-ahub-y := tegra210_ahub.o +snd-soc-tegra210-dmic-y := tegra210_dmic.o +snd-soc-tegra210-i2s-y := tegra210_i2s.o +snd-soc-tegra186-asrc-y := tegra186_asrc.o +snd-soc-tegra186-dspk-y := tegra186_dspk.o +snd-soc-tegra210-admaif-y := tegra210_admaif.o +snd-soc-tegra210-mvc-y := tegra210_mvc.o +snd-soc-tegra210-sfc-y := tegra210_sfc.o +snd-soc-tegra210-amx-y := tegra210_amx.o +snd-soc-tegra210-adx-y := tegra210_adx.o +snd-soc-tegra210-mixer-y := tegra210_mixer.o +snd-soc-tegra210-ope-y := tegra210_ope.o tegra210_mbdrc.o tegra210_peq.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA20_AC97) += snd-soc-tegra20-ac97.o @@ -42,9 +42,9 @@ obj-$(CONFIG_SND_SOC_TEGRA210_MIXER) += snd-soc-tegra210-mixer.o obj-$(CONFIG_SND_SOC_TEGRA210_OPE) += snd-soc-tegra210-ope.o # Tegra machine Support -snd-soc-tegra-wm8903-objs := tegra_wm8903.o -snd-soc-tegra-machine-objs := tegra_asoc_machine.o -snd-soc-tegra-audio-graph-card-objs := tegra_audio_graph_card.o +snd-soc-tegra-wm8903-y := tegra_wm8903.o +snd-soc-tegra-machine-y := tegra_asoc_machine.o +snd-soc-tegra-audio-graph-card-y := tegra_audio_graph_card.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_MACHINE_DRV) += snd-soc-tegra-machine.o From 72b28ec6de0a4dd27dfd89525c3e1efd75ad327b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:34 +0200 Subject: [PATCH 517/603] ASoC: ti: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-31-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/ti/Makefile | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile index 41cdcaec770dc..421e13bc04dbb 100644 --- a/sound/soc/ti/Makefile +++ b/sound/soc/ti/Makefile @@ -1,20 +1,20 @@ # SPDX-License-Identifier: GPL-2.0 # Platform drivers -snd-soc-ti-edma-objs := edma-pcm.o -snd-soc-ti-sdma-objs := sdma-pcm.o -snd-soc-ti-udma-objs := udma-pcm.o +snd-soc-ti-edma-y := edma-pcm.o +snd-soc-ti-sdma-y := sdma-pcm.o +snd-soc-ti-udma-y := udma-pcm.o obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o # CPU DAI drivers -snd-soc-davinci-asp-objs := davinci-i2s.o -snd-soc-davinci-mcasp-objs := davinci-mcasp.o -snd-soc-omap-dmic-objs := omap-dmic.o -snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o -snd-soc-omap-mcpdm-objs := omap-mcpdm.o +snd-soc-davinci-asp-y := davinci-i2s.o +snd-soc-davinci-mcasp-y := davinci-mcasp.o +snd-soc-omap-dmic-y := omap-dmic.o +snd-soc-omap-mcbsp-y := omap-mcbsp.o omap-mcbsp-st.o +snd-soc-omap-mcpdm-y := omap-mcpdm.o obj-$(CONFIG_SND_SOC_DAVINCI_ASP) += snd-soc-davinci-asp.o obj-$(CONFIG_SND_SOC_DAVINCI_MCASP) += snd-soc-davinci-mcasp.o @@ -23,16 +23,16 @@ obj-$(CONFIG_SND_SOC_OMAP_MCBSP) += snd-soc-omap-mcbsp.o obj-$(CONFIG_SND_SOC_OMAP_MCPDM) += snd-soc-omap-mcpdm.o # Machine drivers -snd-soc-davinci-evm-objs := davinci-evm.o -snd-soc-n810-objs := n810.o -snd-soc-rx51-objs := rx51.o -snd-soc-omap3pandora-objs := omap3pandora.o -snd-soc-omap-twl4030-objs := omap-twl4030.o -snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o -snd-soc-ams-delta-objs := ams-delta.o -snd-soc-omap-hdmi-objs := omap-hdmi.o -snd-soc-osk5912-objs := osk5912.o -snd-soc-j721e-evm-objs := j721e-evm.o +snd-soc-davinci-evm-y := davinci-evm.o +snd-soc-n810-y := n810.o +snd-soc-rx51-y := rx51.o +snd-soc-omap3pandora-y := omap3pandora.o +snd-soc-omap-twl4030-y := omap-twl4030.o +snd-soc-omap-abe-twl6040-y := omap-abe-twl6040.o +snd-soc-ams-delta-y := ams-delta.o +snd-soc-omap-hdmi-y := omap-hdmi.o +snd-soc-osk5912-y := osk5912.o +snd-soc-j721e-evm-y := j721e-evm.o obj-$(CONFIG_SND_SOC_DAVINCI_EVM) += snd-soc-davinci-evm.o obj-$(CONFIG_SND_SOC_NOKIA_N810) += snd-soc-n810.o From d5c337c5e3d54a98a1458942a4c1eb7a307b180c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:35 +0200 Subject: [PATCH 518/603] ASoC: uniphier: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-32-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/uniphier/Makefile | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/uniphier/Makefile b/sound/soc/uniphier/Makefile index 88169395f68a3..5fee21b6074cb 100644 --- a/sound/soc/uniphier/Makefile +++ b/sound/soc/uniphier/Makefile @@ -1,11 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 -snd-soc-uniphier-aio-cpu-objs := aio-core.o aio-dma.o aio-cpu.o aio-compress.o -snd-soc-uniphier-aio-ld11-objs := aio-ld11.o -snd-soc-uniphier-aio-pxs2-objs := aio-pxs2.o +snd-soc-uniphier-aio-cpu-y := aio-core.o aio-dma.o aio-cpu.o aio-compress.o +snd-soc-uniphier-aio-ld11-y := aio-ld11.o +snd-soc-uniphier-aio-pxs2-y := aio-pxs2.o obj-$(CONFIG_SND_SOC_UNIPHIER_AIO) += snd-soc-uniphier-aio-cpu.o obj-$(CONFIG_SND_SOC_UNIPHIER_LD11) += snd-soc-uniphier-aio-ld11.o obj-$(CONFIG_SND_SOC_UNIPHIER_PXS2) += snd-soc-uniphier-aio-pxs2.o -snd-soc-uniphier-evea-objs := evea.o +snd-soc-uniphier-evea-y := evea.o obj-$(CONFIG_SND_SOC_UNIPHIER_EVEA_CODEC) += snd-soc-uniphier-evea.o From c4850e856f9db799be4e5a211dc85bdfabbe210b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:36 +0200 Subject: [PATCH 519/603] ASoC: ux500: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-33-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/ux500/Makefile | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index e7d6de51b32bb..a63787d9d6643 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -1,11 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 # Ux500 Platform Support -snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o +snd-soc-ux500-plat-msp-i2s-y := ux500_msp_dai.o ux500_msp_i2s.o obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o -snd-soc-ux500-plat-dma-objs := ux500_pcm.o +snd-soc-ux500-plat-dma-y := ux500_pcm.o obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o -snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +snd-soc-ux500-mach-mop500-y := mop500.o mop500_ab8500.o obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o From a9732bf75785d5baa3259a59ed4cac80da138ed4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:37 +0200 Subject: [PATCH 520/603] ASoC: xilinx: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-34-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/xilinx/Makefile | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/xilinx/Makefile b/sound/soc/xilinx/Makefile index be7652ce7c13d..61cb47cf74dc7 100644 --- a/sound/soc/xilinx/Makefile +++ b/sound/soc/xilinx/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-xlnx-i2s-objs := xlnx_i2s.o +snd-soc-xlnx-i2s-y := xlnx_i2s.o obj-$(CONFIG_SND_SOC_XILINX_I2S) += snd-soc-xlnx-i2s.o -snd-soc-xlnx-formatter-pcm-objs := xlnx_formatter_pcm.o +snd-soc-xlnx-formatter-pcm-y := xlnx_formatter_pcm.o obj-$(CONFIG_SND_SOC_XILINX_AUDIO_FORMATTER) += snd-soc-xlnx-formatter-pcm.o -snd-soc-xlnx-spdif-objs := xlnx_spdif.o +snd-soc-xlnx-spdif-y := xlnx_spdif.o obj-$(CONFIG_SND_SOC_XILINX_SPDIF) += snd-soc-xlnx-spdif.o From 476d0a22bcc4d305080e3fc2d759010db23ccec1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 17:55:38 +0200 Subject: [PATCH 521/603] ASoC: xtensa: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240507155540.24815-35-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/xtensa/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile index b8707f63c4ae3..be4e847d1b20b 100644 --- a/sound/soc/xtensa/Makefile +++ b/sound/soc/xtensa/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o +snd-soc-xtfpga-i2s-y := xtfpga-i2s.o obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o From d45ae45f328b6824966ab7dbf8b1c712f7d0e877 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Tue, 7 May 2024 23:56:59 +0300 Subject: [PATCH 522/603] ASoC: atmel: tse850-pcm5142: Remove unused of_gpio.h of_gpio.h is deprecated and subject to remove. The driver doesn't use it, simply remove the unused header. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20240507205659.690270-1-andriy.shevchenko@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/atmel/tse850-pcm5142.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 611da23325d32..5d208e0b4b905 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -35,10 +35,9 @@ // of the (filtered) output from the PCM5142 codec. #include -#include +#include #include #include -#include #include #include From 1dec672e8686b824aa8742cf9cc00748289de95b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:19:00 +0200 Subject: [PATCH 523/603] ASoC: ab8500: Add missing module description Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Suggested-by: Andy Shevchenko Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091909.27062-2-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/ab8500-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 68342917419e4..04b5e1d5a6530 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2571,4 +2571,5 @@ static struct platform_driver ab8500_codec_platform_driver = { }; module_platform_driver(ab8500_codec_platform_driver); +MODULE_DESCRIPTION("ASoC AB8500 codec driver"); MODULE_LICENSE("GPL v2"); From 1a7dc1cf9e46d2e4c732679775c3a06fba9d94a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:19:01 +0200 Subject: [PATCH 524/603] ASoC: sigmadsp: Add missing module description Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Suggested-by: Andy Shevchenko Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091909.27062-3-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/codecs/sigmadsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 56546e2394aba..201f74e3a7aec 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -805,4 +805,5 @@ int sigmadsp_restrict_params(struct sigmadsp *sigmadsp, } EXPORT_SYMBOL_GPL(sigmadsp_restrict_params); +MODULE_DESCRIPTION("Analog Devices SigmaStudio firmware helpers"); MODULE_LICENSE("GPL"); From 6d21f0a2be88d34e53fcbf66fb8e298787837c01 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:19:02 +0200 Subject: [PATCH 525/603] ASoC: qcom: Add missing module descriptions Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Suggested-by: Andy Shevchenko Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091909.27062-4-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/qcom/common.c | 2 ++ sound/soc/qcom/qdsp6/q6dsp-common.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 747041fa78665..3d02aa3844f29 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -239,4 +239,6 @@ int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd, return 0; } EXPORT_SYMBOL_GPL(qcom_snd_wcd_jack_setup); + +MODULE_DESCRIPTION("ASoC Qualcomm helper functions"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/qcom/qdsp6/q6dsp-common.c b/sound/soc/qcom/qdsp6/q6dsp-common.c index 95585dea2b36d..f74585d88bd65 100644 --- a/sound/soc/qcom/qdsp6/q6dsp-common.c +++ b/sound/soc/qcom/qdsp6/q6dsp-common.c @@ -98,4 +98,6 @@ int q6dsp_get_channel_allocation(int channels) return channel_allocation; } EXPORT_SYMBOL_GPL(q6dsp_get_channel_allocation); + +MODULE_DESCRIPTION("ASoC MSM QDSP6 helper functions"); MODULE_LICENSE("GPL v2"); From 2c846d7cc4ee3382aed6dfd322fcaaf309eb2c9a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:19:03 +0200 Subject: [PATCH 526/603] ASoC: dmaengine: Add missing module description Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Suggested-by: Andy Shevchenko Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091909.27062-5-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 83db1a83d8ba8..ea3bc9318412a 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -494,4 +494,5 @@ void snd_dmaengine_pcm_unregister(struct device *dev) } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); +MODULE_DESCRIPTION("ASoC helpers for generic PCM dmaengine API"); MODULE_LICENSE("GPL"); From ea762d10b92a423b6e4980241a7bfa809c7cffd1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:19:04 +0200 Subject: [PATCH 527/603] ASoC: topology-test: Add missing module description Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Suggested-by: Andy Shevchenko Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091909.27062-6-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-topology-test.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index 02aa2448f0e67..d62a02ec58963 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -819,4 +819,5 @@ static struct kunit_suite snd_soc_tplg_test_suite = { kunit_test_suites(&snd_soc_tplg_test_suite); +MODULE_DESCRIPTION("ASoC Topology Kernel Unit Tests"); MODULE_LICENSE("GPL"); From e85d8aeef23376ef10569c555819d301f3364f28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:19:05 +0200 Subject: [PATCH 528/603] ASoC: ux500: Add missing module description Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Suggested-by: Andy Shevchenko Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091909.27062-7-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index cde0dd8e25691..3fd13e8dd1107 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -820,4 +820,5 @@ static struct platform_driver msp_i2s_driver = { }; module_platform_driver(msp_i2s_driver); +MODULE_DESCRIPTION("ASoC Ux500 I2S driver"); MODULE_LICENSE("GPL v2"); From 3423ad19ea912694a15ebbc48cbc8e1fcd8439da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:19:06 +0200 Subject: [PATCH 529/603] ASoC: xilinx: Add missing module descriptions Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Suggested-by: Andy Shevchenko Link: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091909.27062-8-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 ++ sound/soc/xilinx/xlnx_i2s.c | 1 + 2 files changed, 3 insertions(+) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 299cfb5e20224..158fc21a86c10 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -721,5 +721,7 @@ static struct platform_driver xlnx_formatter_pcm_driver = { }; module_platform_driver(xlnx_formatter_pcm_driver); + +MODULE_DESCRIPTION("ASoC driver for Xilinx audio formatter"); MODULE_AUTHOR("Maruthi Srinivas Bayyavarapu "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/xilinx/xlnx_i2s.c b/sound/soc/xilinx/xlnx_i2s.c index 9de92d35e30ee..ca915a001ad5f 100644 --- a/sound/soc/xilinx/xlnx_i2s.c +++ b/sound/soc/xilinx/xlnx_i2s.c @@ -253,6 +253,7 @@ static struct platform_driver xlnx_i2s_aud_driver = { module_platform_driver(xlnx_i2s_aud_driver); +MODULE_DESCRIPTION("ASoC driver for Xilinx I2S audio"); MODULE_LICENSE("GPL v2"); MODULE_AUTHOR("Praveen Vuppala "); MODULE_AUTHOR("Maruthi Srinivas Bayyavarapu "); From 32d7c6cdc98f7131f13c41251e87837df81e0b31 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:44 +0200 Subject: [PATCH 530/603] ALSA: hda: intel: Reduce CONFIG_PM dependencies snd-hda-intel contains lots of CONFIG_PM dependent code although CONFIG_PM is almost mandatory nowadays, and it makes the code unnecessarily complex. Let's reduce the dependencies of CONFIG_PM in snd-hda-intel driver code. I left a few module options to be dependent on CONFIG_PM (which are visible to users), but other places are either enabled or optimized by compiler automatically. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-2-tiwai@suse.de --- sound/pci/hda/hda_intel.c | 46 ++++++++++----------------------- sound/pci/hda/hda_intel_trace.h | 2 -- 2 files changed, 14 insertions(+), 34 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b550c42db092..31b0c954b0c8d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -186,8 +186,10 @@ MODULE_PARM_DESC(pm_blacklist, "Enable power-management denylist"); static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); -#else +#else /* CONFIG_PM */ #define power_save 0 +#define pm_blacklist false +#define power_save_controller false #endif /* CONFIG_PM */ static int align_buffer_size = -1; @@ -890,7 +892,6 @@ static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) display_power(chip, false); } -#ifdef CONFIG_PM static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); @@ -916,7 +917,7 @@ static void azx_del_card_list(struct azx *chip) } /* trigger power-save check at writing parameter */ -static int param_set_xint(const char *val, const struct kernel_param *kp) +static int __maybe_unused param_set_xint(const char *val, const struct kernel_param *kp) { struct hda_intel *hda; struct azx *chip; @@ -987,7 +988,6 @@ static void __azx_runtime_resume(struct azx *chip) display_power(chip, false); } -#ifdef CONFIG_PM_SLEEP static int azx_prepare(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -1046,7 +1046,7 @@ static int azx_suspend(struct device *dev) return 0; } -static int azx_resume(struct device *dev) +static int __maybe_unused azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; @@ -1097,9 +1097,8 @@ static int azx_thaw_noirq(struct device *dev) return 0; } -#endif /* CONFIG_PM_SLEEP */ -static int azx_runtime_suspend(struct device *dev) +static int __maybe_unused azx_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; @@ -1116,7 +1115,7 @@ static int azx_runtime_suspend(struct device *dev) return 0; } -static int azx_runtime_resume(struct device *dev) +static int __maybe_unused azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; @@ -1133,7 +1132,7 @@ static int azx_runtime_resume(struct device *dev) return 0; } -static int azx_runtime_idle(struct device *dev) +static int __maybe_unused azx_runtime_idle(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; @@ -1159,23 +1158,14 @@ static int azx_runtime_idle(struct device *dev) } static const struct dev_pm_ops azx_pm = { - SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) -#ifdef CONFIG_PM_SLEEP - .prepare = azx_prepare, - .complete = azx_complete, - .freeze_noirq = azx_freeze_noirq, - .thaw_noirq = azx_thaw_noirq, -#endif + SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + .prepare = pm_sleep_ptr(azx_prepare), + .complete = pm_sleep_ptr(azx_complete), + .freeze_noirq = pm_sleep_ptr(azx_freeze_noirq), + .thaw_noirq = pm_sleep_ptr(azx_thaw_noirq), SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, azx_runtime_idle) }; -#define AZX_PM_OPS &azx_pm -#else -#define azx_add_card_list(chip) /* NOP */ -#define azx_del_card_list(chip) /* NOP */ -#define AZX_PM_OPS NULL -#endif /* CONFIG_PM */ - static int azx_probe_continue(struct azx *chip); @@ -2206,7 +2196,6 @@ static int azx_probe(struct pci_dev *pci, return err; } -#ifdef CONFIG_PM /* On some boards setting power_save to a non 0 value leads to clicking / * popping sounds when ever we enter/leave powersaving mode. Ideally we would * figure out how to avoid these sounds, but that is not always feasible. @@ -2248,13 +2237,11 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x1734, 0x1232, "KONTRON SinglePC", 0), {} }; -#endif /* CONFIG_PM */ static void set_default_power_save(struct azx *chip) { int val = power_save; -#ifdef CONFIG_PM if (pm_blacklist) { const struct snd_pci_quirk *q; @@ -2265,7 +2252,6 @@ static void set_default_power_save(struct azx *chip) val = 0; } } -#endif /* CONFIG_PM */ snd_hda_set_power_save(&chip->bus, val * 1000); } @@ -2321,10 +2307,6 @@ static int azx_probe_continue(struct azx *chip) chip->fw->data); if (err < 0) goto out_free; -#ifndef CONFIG_PM - release_firmware(chip->fw); /* no longer needed */ - chip->fw = NULL; -#endif } #endif @@ -2765,7 +2747,7 @@ static struct pci_driver azx_driver = { .remove = azx_remove, .shutdown = azx_shutdown, .driver = { - .pm = AZX_PM_OPS, + .pm = &azx_pm, }, }; diff --git a/sound/pci/hda/hda_intel_trace.h b/sound/pci/hda/hda_intel_trace.h index 73a7adfa192d0..2775fa81a5007 100644 --- a/sound/pci/hda/hda_intel_trace.h +++ b/sound/pci/hda/hda_intel_trace.h @@ -34,7 +34,6 @@ DEFINE_EVENT(hda_pm, azx_resume, TP_ARGS(chip) ); -#ifdef CONFIG_PM DEFINE_EVENT(hda_pm, azx_runtime_suspend, TP_PROTO(struct azx *chip), TP_ARGS(chip) @@ -44,7 +43,6 @@ DEFINE_EVENT(hda_pm, azx_runtime_resume, TP_PROTO(struct azx *chip), TP_ARGS(chip) ); -#endif #endif /* _TRACE_HDA_INTEL_H */ From 9e993b3d722fb452e274e1f8694d8940db183323 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:45 +0200 Subject: [PATCH 531/603] ALSA: hda: codec: Reduce CONFIG_PM dependencies CONFIG_PM is almost mandatory nowadays for real systems, but we have lots of CONFIG_PM dependent code in snd-hda-codec helper code. Let's reduce the dependencies of CONFIG_PM now. The only visible drawback would be a couple of superfluous trace entries for runtime PM, but we can live with that. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-3-tiwai@suse.de --- include/sound/hda_codec.h | 11 ---------- sound/pci/hda/hda_codec.c | 37 +++++++++------------------------- sound/pci/hda/hda_controller.c | 2 -- sound/pci/hda/hda_sysfs.c | 4 ---- 4 files changed, 10 insertions(+), 44 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 9c94ba7c183df..575e55aa08ca9 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -109,11 +109,9 @@ struct hda_codec_ops { void (*unsol_event)(struct hda_codec *codec, unsigned int res); void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); -#ifdef CONFIG_PM int (*suspend)(struct hda_codec *codec); int (*resume)(struct hda_codec *codec); int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); -#endif void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on); }; @@ -259,11 +257,9 @@ struct hda_codec { unsigned int no_stream_clean_at_suspend:1; /* do not clean streams at suspend */ unsigned int ctl_dev_id:1; /* old control element id build behaviour */ -#ifdef CONFIG_PM unsigned long power_on_acct; unsigned long power_off_acct; unsigned long power_jiffies; -#endif /* filter the requested power state per nid */ unsigned int (*power_filter)(struct hda_codec *codec, hda_nid_t nid, @@ -481,10 +477,8 @@ extern const struct dev_pm_ops hda_codec_driver_pm; static inline int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) { -#ifdef CONFIG_PM if (codec->patch_ops.check_power_status) return codec->patch_ops.check_power_status(codec, nid); -#endif return 0; } @@ -495,14 +489,9 @@ int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) #define snd_hda_power_up_pm(codec) snd_hdac_power_up_pm(&(codec)->core) #define snd_hda_power_down(codec) snd_hdac_power_down(&(codec)->core) #define snd_hda_power_down_pm(codec) snd_hdac_power_down_pm(&(codec)->core) -#ifdef CONFIG_PM void snd_hda_codec_set_power_save(struct hda_codec *codec, int delay); void snd_hda_set_power_save(struct hda_bus *bus, int delay); void snd_hda_update_power_acct(struct hda_codec *codec); -#else -static inline void snd_hda_codec_set_power_save(struct hda_codec *codec, int delay) {} -static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} -#endif static inline bool hda_codec_need_resume(struct hda_codec *codec) { diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2cac337f52632..325e8f0b99a82 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -616,7 +616,6 @@ void snd_hda_shutup_pins(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_shutup_pins); -#ifdef CONFIG_PM /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ static void restore_shutup_pins(struct hda_codec *codec) { @@ -634,7 +633,6 @@ static void restore_shutup_pins(struct hda_codec *codec) } codec->pins_shutup = 0; } -#endif static void hda_jackpoll_work(struct work_struct *work) { @@ -1001,9 +999,7 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, codec->card = card; codec->addr = codec_addr; -#ifdef CONFIG_PM codec->power_jiffies = jiffies; -#endif snd_hda_sysfs_init(codec); @@ -1238,7 +1234,6 @@ static void purify_inactive_streams(struct hda_codec *codec) } } -#ifdef CONFIG_PM /* clean up all streams; called from suspend */ static void hda_cleanup_all_streams(struct hda_codec *codec) { @@ -1250,7 +1245,6 @@ static void hda_cleanup_all_streams(struct hda_codec *codec) really_cleanup_stream(codec, p); } } -#endif /* * amp access functions @@ -2858,7 +2852,6 @@ static void hda_exec_init_verbs(struct hda_codec *codec) static inline void hda_exec_init_verbs(struct hda_codec *codec) {} #endif -#ifdef CONFIG_PM /* update the power on/off account with the current jiffies */ static void update_power_acct(struct hda_codec *codec, bool on) { @@ -2966,9 +2959,6 @@ static int hda_codec_runtime_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM */ - -#ifdef CONFIG_PM_SLEEP static int hda_codec_pm_prepare(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); @@ -3023,22 +3013,19 @@ static int hda_codec_pm_restore(struct device *dev) dev->power.power_state = PMSG_RESTORE; return pm_runtime_force_resume(dev); } -#endif /* CONFIG_PM_SLEEP */ /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { -#ifdef CONFIG_PM_SLEEP - .prepare = hda_codec_pm_prepare, - .complete = hda_codec_pm_complete, - .suspend = hda_codec_pm_suspend, - .resume = hda_codec_pm_resume, - .freeze = hda_codec_pm_freeze, - .thaw = hda_codec_pm_thaw, - .poweroff = hda_codec_pm_suspend, - .restore = hda_codec_pm_restore, -#endif /* CONFIG_PM_SLEEP */ - SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume, - NULL) + .prepare = pm_sleep_ptr(hda_codec_pm_prepare), + .complete = pm_sleep_ptr(hda_codec_pm_complete), + .suspend = pm_sleep_ptr(hda_codec_pm_suspend), + .resume = pm_sleep_ptr(hda_codec_pm_resume), + .freeze = pm_sleep_ptr(hda_codec_pm_freeze), + .thaw = pm_sleep_ptr(hda_codec_pm_thaw), + .poweroff = pm_sleep_ptr(hda_codec_pm_suspend), + .restore = pm_sleep_ptr(hda_codec_pm_restore), + .runtime_suspend = pm_ptr(hda_codec_runtime_suspend), + .runtime_resume = pm_ptr(hda_codec_runtime_resume), }; /* suspend the codec at shutdown; called from driver's shutdown callback */ @@ -3425,7 +3412,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls); -#ifdef CONFIG_PM /** * snd_hda_codec_set_power_save - Configure codec's runtime PM * @codec: codec device to configure @@ -3516,7 +3502,6 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); -#endif /* * input MUX helper @@ -4060,12 +4045,10 @@ void snd_hda_bus_reset_codecs(struct hda_bus *bus) /* FIXME: maybe a better way needed for forced reset */ if (current_work() != &codec->jackpoll_work.work) cancel_delayed_work_sync(&codec->jackpoll_work); -#ifdef CONFIG_PM if (hda_codec_is_power_on(codec)) { hda_call_codec_suspend(codec); hda_call_codec_resume(codec); } -#endif } } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 206306a0eb827..c7142eee9f353 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1075,11 +1075,9 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) bool active, handled = false; int repeat = 0; /* count for avoiding endless loop */ -#ifdef CONFIG_PM if (azx_has_pm_runtime(chip)) if (!pm_runtime_active(chip->card->dev)) return IRQ_NONE; -#endif spin_lock(&bus->reg_lock); diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 69ebc37a4d6f3..265fd47378931 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -26,7 +26,6 @@ struct hda_hint { const char *val; /* contained in the same alloc as key */ }; -#ifdef CONFIG_PM static ssize_t power_on_acct_show(struct device *dev, struct device_attribute *attr, char *buf) @@ -47,7 +46,6 @@ static ssize_t power_off_acct_show(struct device *dev, static DEVICE_ATTR_RO(power_on_acct); static DEVICE_ATTR_RO(power_off_acct); -#endif /* CONFIG_PM */ #define CODEC_INFO_SHOW(type, field) \ static ssize_t type##_show(struct device *dev, \ @@ -745,10 +743,8 @@ static struct attribute *hda_dev_attrs[] = { &dev_attr_modelname.attr, &dev_attr_init_pin_configs.attr, &dev_attr_driver_pin_configs.attr, -#ifdef CONFIG_PM &dev_attr_power_on_acct.attr, &dev_attr_power_off_acct.attr, -#endif #ifdef CONFIG_SND_HDA_RECONFIG &dev_attr_init_verbs.attr, &dev_attr_hints.attr, From 6c8fd3499423fc3ebb735f32d4a52bc5825f6301 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:46 +0200 Subject: [PATCH 532/603] ALSA: hda: generic: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio generic codec driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-4-tiwai@suse.de --- sound/pci/hda/hda_generic.c | 4 ---- sound/pci/hda/hda_generic.h | 2 -- 2 files changed, 6 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index de2a3d08c73c1..f64d9dc197a31 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -6021,7 +6021,6 @@ void snd_hda_gen_free(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_gen_free); -#ifdef CONFIG_PM /** * snd_hda_gen_check_power_status - check the loopback power save state * @codec: the HDA codec @@ -6035,7 +6034,6 @@ int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid) return snd_hda_check_amp_list_power(codec, &spec->loopback, nid); } EXPORT_SYMBOL_GPL(snd_hda_gen_check_power_status); -#endif /* @@ -6048,9 +6046,7 @@ static const struct hda_codec_ops generic_patch_ops = { .init = snd_hda_gen_init, .free = snd_hda_gen_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, -#endif }; /* diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index a8eea83676299..8f5ecf740c491 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -340,9 +340,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_callback *jack); void snd_hda_gen_update_outputs(struct hda_codec *codec); -#ifdef CONFIG_PM int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid); -#endif unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state); From b61f98b2c33570211a83d5d2355c3c7130469f3d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:47 +0200 Subject: [PATCH 533/603] ALSA: hda: analog: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio analog codec driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-5-tiwai@suse.de --- sound/pci/hda/patch_analog.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 8afe6000f7dad..1e9dadcdc51be 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -72,7 +72,6 @@ static int create_beep_ctls(struct hda_codec *codec) #define create_beep_ctls(codec) 0 #endif -#ifdef CONFIG_PM static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front, hda_nid_t hp) { @@ -118,7 +117,6 @@ static int ad198x_suspend(struct hda_codec *codec) ad198x_power_eapd(codec); return 0; } -#endif /* follow EAPD via vmaster hook */ static void ad_vmaster_eapd_hook(void *private_data, int enabled) @@ -158,10 +156,8 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = { .init = snd_hda_gen_init, .free = snd_hda_gen_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, .suspend = ad198x_suspend, -#endif }; From 9fff3e28f54d8eeac707f82d3f0aeaf1beb9192f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:48 +0200 Subject: [PATCH 534/603] ALSA: hda: ca0132: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio ca0132 codec driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-6-tiwai@suse.de --- sound/pci/hda/patch_ca0132.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index aa312441604f8..e4673a71551a3 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -9682,7 +9682,6 @@ static void dbpro_free(struct hda_codec *codec) kfree(codec->spec); } -#ifdef CONFIG_PM static int ca0132_suspend(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; @@ -9690,7 +9689,6 @@ static int ca0132_suspend(struct hda_codec *codec) cancel_delayed_work_sync(&spec->unsol_hp_work); return 0; } -#endif static const struct hda_codec_ops ca0132_patch_ops = { .build_controls = ca0132_build_controls, @@ -9698,9 +9696,7 @@ static const struct hda_codec_ops ca0132_patch_ops = { .init = ca0132_init, .free = ca0132_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM .suspend = ca0132_suspend, -#endif }; static const struct hda_codec_ops dbpro_patch_ops = { From 2bf6ae888a5aa7c8af723f038f96666eef9a463e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:49 +0200 Subject: [PATCH 535/603] ALSA: hda: cirrus: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio cirrus codec driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-7-tiwai@suse.de --- sound/pci/hda/patch_cirrus.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 6807b4708a176..654724559355e 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1128,7 +1128,6 @@ static int cs421x_parse_auto_config(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM /* * Manage PDREF, when transitioning to D3hot * (DAC,ADC) -> D3, PDREF=1, AFG->D3 @@ -1153,7 +1152,6 @@ static int cs421x_suspend(struct hda_codec *codec) return 0; } -#endif static const struct hda_codec_ops cs421x_patch_ops = { .build_controls = snd_hda_gen_build_controls, @@ -1161,9 +1159,7 @@ static const struct hda_codec_ops cs421x_patch_ops = { .init = cs421x_init, .free = cs_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM .suspend = cs421x_suspend, -#endif }; static int patch_cs4210(struct hda_codec *codec) From 29d57f6dc62485ee0752767debdfa2783d162beb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:50 +0200 Subject: [PATCH 536/603] ALSA: hda: conexant: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio conexant codec driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-8-tiwai@suse.de --- sound/pci/hda/patch_conexant.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e8209178d87bb..17389a3801bd1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -294,13 +294,11 @@ static void cx_jack_unsol_event(struct hda_codec *codec, unsigned int res) snd_hda_jack_unsol_event(codec, res); } -#ifdef CONFIG_PM static int cx_auto_suspend(struct hda_codec *codec) { cx_auto_shutdown(codec); return 0; } -#endif static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = snd_hda_gen_build_controls, @@ -308,10 +306,8 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .init = cx_auto_init, .free = cx_auto_free, .unsol_event = cx_jack_unsol_event, -#ifdef CONFIG_PM .suspend = cx_auto_suspend, .check_power_status = snd_hda_gen_check_power_status, -#endif }; /* From 4e0549ff81cd9c82712ee87adcd5a278879b4c05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:51 +0200 Subject: [PATCH 537/603] ALSA: hda: cs4809: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio generic cs8409 driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-9-tiwai@suse.de --- sound/pci/hda/patch_cs8409.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index e41316e2e9833..26f3c31600d7b 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -909,7 +909,6 @@ static void cs42l42_resume(struct sub_codec *cs42l42) cs42l42_enable_jack_detect(cs42l42); } -#ifdef CONFIG_PM static void cs42l42_suspend(struct sub_codec *cs42l42) { struct hda_codec *codec = cs42l42->codec; @@ -948,7 +947,6 @@ static void cs42l42_suspend(struct sub_codec *cs42l42) spec->gpio_data &= ~cs42l42->reset_gpio; snd_hda_codec_write(codec, CS8409_PIN_AFG, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_data); } -#endif static void cs8409_free(struct hda_codec *codec) { @@ -1003,7 +1001,6 @@ static void cs8409_cs42l42_jack_unsol_event(struct hda_codec *codec, unsigned in } } -#ifdef CONFIG_PM /* Manage PDREF, when transition to D3hot */ static int cs8409_cs42l42_suspend(struct hda_codec *codec) { @@ -1025,7 +1022,6 @@ static int cs8409_cs42l42_suspend(struct hda_codec *codec) return 0; } -#endif /* Vendor specific HW configuration * PLL, ASP, I2C, SPI, GPIOs, DMIC etc... @@ -1080,9 +1076,7 @@ static const struct hda_codec_ops cs8409_cs42l42_patch_ops = { .init = cs8409_init, .free = cs8409_free, .unsol_event = cs8409_cs42l42_jack_unsol_event, -#ifdef CONFIG_PM .suspend = cs8409_cs42l42_suspend, -#endif }; static int cs8409_cs42l42_exec_verb(struct hdac_device *dev, unsigned int cmd, unsigned int flags, @@ -1310,9 +1304,7 @@ static const struct hda_codec_ops cs8409_dolphin_patch_ops = { .init = cs8409_init, .free = cs8409_free, .unsol_event = dolphin_jack_unsol_event, -#ifdef CONFIG_PM .suspend = cs8409_cs42l42_suspend, -#endif }; static int dolphin_exec_verb(struct hdac_device *dev, unsigned int cmd, unsigned int flags, From aa423b7d526469620521a1f871918f5435c8541c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:52 +0200 Subject: [PATCH 538/603] ALSA: hda: hdmi: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio HDMI codec driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-10-tiwai@suse.de --- sound/pci/hda/patch_hdmi.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 495d63101186f..b2c42cd543514 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2513,7 +2513,6 @@ static void generic_hdmi_free(struct hda_codec *codec) generic_spec_free(codec); } -#ifdef CONFIG_PM static int generic_hdmi_suspend(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2540,7 +2539,6 @@ static int generic_hdmi_resume(struct hda_codec *codec) } return 0; } -#endif static const struct hda_codec_ops generic_hdmi_patch_ops = { .init = generic_hdmi_init, @@ -2548,10 +2546,8 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .build_pcms = generic_hdmi_build_pcms, .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, -#ifdef CONFIG_PM .suspend = generic_hdmi_suspend, .resume = generic_hdmi_resume, -#endif }; static const struct hdmi_ops generic_standard_hdmi_ops = { @@ -2952,7 +2948,6 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, } } -#ifdef CONFIG_PM static int i915_adlp_hdmi_suspend(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -3032,7 +3027,6 @@ static int i915_adlp_hdmi_resume(struct hda_codec *codec) return res; } -#endif /* precondition and allocation for Intel codecs */ static int alloc_intel_hdmi(struct hda_codec *codec) @@ -3167,10 +3161,8 @@ static int patch_i915_adlp_hdmi(struct hda_codec *codec) if (spec->silent_stream_type) { spec->silent_stream_type = SILENT_STREAM_KAE; -#ifdef CONFIG_PM codec->patch_ops.resume = i915_adlp_hdmi_resume; codec->patch_ops.suspend = i915_adlp_hdmi_suspend; -#endif } } From 03002d182d6f0bfd2fbaebe069b7038c0cae8e40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:53 +0200 Subject: [PATCH 539/603] ALSA: hda: realtek: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio realtek codec driver, too. Simply drop CONFIG_PM ifdefs. A superfluous __maybe_unused attribute was dropped as well. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-11-tiwai@suse.de --- sound/pci/hda/patch_realtek.c | 16 +--------------- 1 file changed, 1 insertion(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b29739bd330b1..944f36f94a5a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -109,9 +109,7 @@ struct alc_spec { /* hooks */ void (*init_hook)(struct hda_codec *codec); -#ifdef CONFIG_PM void (*power_hook)(struct hda_codec *codec); -#endif void (*shutup)(struct hda_codec *codec); int init_amp; @@ -947,7 +945,6 @@ static int alc_init(struct hda_codec *codec) #define alc_free snd_hda_gen_free -#ifdef CONFIG_PM static inline void alc_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -986,7 +983,6 @@ static int alc_resume(struct hda_codec *codec) hda_call_check_power_status(codec, 0x01); return 0; } -#endif /* */ @@ -996,11 +992,9 @@ static const struct hda_codec_ops alc_patch_ops = { .init = alc_init, .free = alc_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM .resume = alc_resume, .suspend = alc_suspend, .check_power_status = snd_hda_gen_check_power_status, -#endif }; @@ -4041,7 +4035,6 @@ static void alc5505_dsp_init(struct hda_codec *codec) #define alc5505_dsp_resume(codec) alc5505_dsp_back_from_halt(codec) #endif -#ifdef CONFIG_PM static int alc269_suspend(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4087,7 +4080,6 @@ static int alc269_resume(struct hda_codec *codec) return 0; } -#endif /* CONFIG_PM */ static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -7204,7 +7196,7 @@ static void alc287_alc1318_playback_pcm_hook(struct hda_pcm_stream *hinfo, } } -static void __maybe_unused alc287_s4_power_gpio3_default(struct hda_codec *codec) +static void alc287_s4_power_gpio3_default(struct hda_codec *codec) { if (is_s4_suspend(codec)) { alc_write_coef_idx(codec, 0x10, 0x8806); /* Change MLK to GPIO3 */ @@ -7219,9 +7211,7 @@ static void alc287_fixup_lenovo_thinkpad_with_alc1318(struct hda_codec *codec, if (action != HDA_FIXUP_ACT_PRE_PROBE) return; -#ifdef CONFIG_PM spec->power_hook = alc287_s4_power_gpio3_default; -#endif spec->gen.pcm_playback_hook = alc287_alc1318_playback_pcm_hook; } @@ -11287,10 +11277,8 @@ static int patch_alc269(struct hda_codec *codec) codec->power_save_node = 0; spec->en_3kpull_low = true; -#ifdef CONFIG_PM codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; -#endif spec->shutup = alc_default_shutup; spec->init_hook = alc_default_init; @@ -11588,9 +11576,7 @@ static int patch_alc861(struct hda_codec *codec) if (has_cdefine_beep(codec)) spec->gen.beep_nid = 0x23; -#ifdef CONFIG_PM spec->power_hook = alc_power_eapd; -#endif alc_pre_init(codec); From 9c9d7fd5b792a100de633621bf0b541775872fa5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:54 +0200 Subject: [PATCH 540/603] ALSA: hda: sigmantel: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio sigmatel codec driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-12-tiwai@suse.de --- sound/pci/hda/patch_sigmatel.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 61258b0aac8d6..ae1a34c68c616 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2154,10 +2154,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ -#ifdef CONFIG_PM /* resetting controller clears GPIO, so we need to keep on */ codec->core.power_caps &= ~AC_PWRST_CLKSTOP; -#endif } } @@ -4442,7 +4440,6 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #define stac927x_proc_hook NULL #endif -#ifdef CONFIG_PM static int stac_suspend(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -4456,9 +4453,6 @@ static int stac_suspend(struct hda_codec *codec) return 0; } -#else -#define stac_suspend NULL -#endif /* CONFIG_PM */ static const struct hda_codec_ops stac_patch_ops = { .build_controls = snd_hda_gen_build_controls, @@ -4466,9 +4460,7 @@ static const struct hda_codec_ops stac_patch_ops = { .init = stac_init, .free = stac_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM .suspend = stac_suspend, -#endif }; static int alloc_stac_spec(struct hda_codec *codec) From 2ff85dc64df5bc0ff12e2f4e23fae7bbadbf1d5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 May 2024 18:13:55 +0200 Subject: [PATCH 541/603] ALSA: hda: via: Reduce CONFIG_PM dependencies CONFIG_PM dependencies got reduced in HD-audio codec core driver, and now it's time to reduce in HD-audio via codec driver, too. Simply drop CONFIG_PM ifdefs. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506161359.6960-13-tiwai@suse.de --- sound/pci/hda/patch_via.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 2994f85bc1b9a..a8ef4bb70dd05 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -379,7 +379,6 @@ static void via_free(struct hda_codec *codec) snd_hda_gen_free(codec); } -#ifdef CONFIG_PM static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -400,9 +399,7 @@ static int via_resume(struct hda_codec *codec) snd_hda_regmap_sync(codec); return 0; } -#endif -#ifdef CONFIG_PM static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct via_spec *spec = codec->spec; @@ -410,7 +407,6 @@ static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) vt1708_update_hp_work(codec); return snd_hda_check_amp_list_power(codec, &spec->gen.loopback, nid); } -#endif /* */ @@ -423,11 +419,9 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM .suspend = via_suspend, .resume = via_resume, .check_power_status = via_check_power_status, -#endif }; From 762eba7096e3d4d81faefffcc57074a82b53613d Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 29 Apr 2024 16:48:52 +0100 Subject: [PATCH 542/603] ALSA: hda: cs35l41: Ignore errors when configuring IRQs IRQs used for CS35L41 HDA are used to detect and attempt to recover from errors. Without these interrupts, the driver should behave as normal. For laptops which contain a bad configuration for the interrupt in the BIOS, the current behaviour of failing when trying to configure the interrupt means the probe fails, and audio is broken. It is better for the user experience if the driver instead warns that no interrupt is configured rather than simply failing. The drawback is that if an error occurs without the interrupt, we firstly would not be able to trace the issue, and secondly would not be able to attempt to recover from the issue, but this is better than failing immediately. Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240429154853.9393-2-sbinding@opensource.cirrus.com --- sound/pci/hda/cs35l41_hda.c | 69 +++++++++++++++++++++++++------------ 1 file changed, 47 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 7fde8a7a923bc..2b0404f6e2716 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1593,13 +1593,56 @@ static struct regmap_irq_chip cs35l41_regmap_irq_chip = { .runtime_pm = true, }; +static void cs35l41_configure_interrupt(struct cs35l41_hda *cs35l41, int irq_pol) +{ + int irq; + int ret; + int i; + + if (!cs35l41->irq) { + dev_warn(cs35l41->dev, "No Interrupt Found"); + goto err; + } + + ret = devm_regmap_add_irq_chip(cs35l41->dev, cs35l41->regmap, cs35l41->irq, + IRQF_ONESHOT | IRQF_SHARED | irq_pol, + 0, &cs35l41_regmap_irq_chip, &cs35l41->irq_data); + if (ret) { + dev_dbg(cs35l41->dev, "Unable to add IRQ Chip: %d.", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(cs35l41_irqs); i++) { + irq = regmap_irq_get_virq(cs35l41->irq_data, cs35l41_irqs[i].irq); + if (irq < 0) { + ret = irq; + dev_dbg(cs35l41->dev, "Unable to map IRQ %s: %d.", cs35l41_irqs[i].name, + ret); + goto err; + } + + ret = devm_request_threaded_irq(cs35l41->dev, irq, NULL, + cs35l41_irqs[i].handler, + IRQF_ONESHOT | IRQF_SHARED | irq_pol, + cs35l41_irqs[i].name, cs35l41); + if (ret) { + dev_dbg(cs35l41->dev, "Unable to allocate IRQ %s:: %d.", + cs35l41_irqs[i].name, ret); + goto err; + } + } + return; +err: + dev_warn(cs35l41->dev, + "IRQ Config Failed. Amp errors may not be recoverable without reboot."); +} + static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41) { struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; bool using_irq = false; - int irq, irq_pol; + int irq_pol; int ret; - int i; if (!cs35l41->hw_cfg.valid) return -EINVAL; @@ -1642,26 +1685,8 @@ static int cs35l41_hda_apply_properties(struct cs35l41_hda *cs35l41) irq_pol = cs35l41_gpio_config(cs35l41->regmap, hw_cfg); - if (cs35l41->irq && using_irq) { - ret = devm_regmap_add_irq_chip(cs35l41->dev, cs35l41->regmap, cs35l41->irq, - IRQF_ONESHOT | IRQF_SHARED | irq_pol, - 0, &cs35l41_regmap_irq_chip, &cs35l41->irq_data); - if (ret) - return ret; - - for (i = 0; i < ARRAY_SIZE(cs35l41_irqs); i++) { - irq = regmap_irq_get_virq(cs35l41->irq_data, cs35l41_irqs[i].irq); - if (irq < 0) - return irq; - - ret = devm_request_threaded_irq(cs35l41->dev, irq, NULL, - cs35l41_irqs[i].handler, - IRQF_ONESHOT | IRQF_SHARED | irq_pol, - cs35l41_irqs[i].name, cs35l41); - if (ret) - return ret; - } - } + if (using_irq) + cs35l41_configure_interrupt(cs35l41, irq_pol); return cs35l41_hda_channel_map(cs35l41->dev, 0, NULL, 1, &hw_cfg->spk_pos); } From 4bfea1dcd5cbf2d92f999e86d95cfcb86ac8345f Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Mon, 29 Apr 2024 16:48:53 +0100 Subject: [PATCH 543/603] ALSA: hda: cs35l41: Add support for ASUS ROG 2024 Laptops All of these laptops do not have _DSD, so need to be added to the configuration table. Signed-off-by: Stefan Binding Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240429154853.9393-3-sbinding@opensource.cirrus.com --- sound/pci/hda/cs35l41_hda_property.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index be97e973accb6..ad9b6abdf269f 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -97,6 +97,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431863", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "104318D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "10431A83", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431B93", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431C9F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431CAF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431CCF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, @@ -110,6 +111,10 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10433A20", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10433A30", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10433A40", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10433A50", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA3865", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3866", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, @@ -492,6 +497,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431863", generic_dsd_config }, { "CSC3551", "104318D3", generic_dsd_config }, { "CSC3551", "10431A83", generic_dsd_config }, + { "CSC3551", "10431B93", generic_dsd_config }, { "CSC3551", "10431C9F", generic_dsd_config }, { "CSC3551", "10431CAF", generic_dsd_config }, { "CSC3551", "10431CCF", generic_dsd_config }, @@ -505,6 +511,10 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, + { "CSC3551", "10433A20", generic_dsd_config }, + { "CSC3551", "10433A30", generic_dsd_config }, + { "CSC3551", "10433A40", generic_dsd_config }, + { "CSC3551", "10433A50", generic_dsd_config }, { "CSC3551", "10433A60", generic_dsd_config }, { "CSC3551", "17AA3865", generic_dsd_config }, { "CSC3551", "17AA3866", generic_dsd_config }, From 44f69ddccb66bcdf969c44d8bb5d4dea4d6b2933 Mon Sep 17 00:00:00 2001 From: Manuel Barrio Linares Date: Tue, 30 Apr 2024 14:10:18 -0300 Subject: [PATCH 544/603] ALSA: usb-audio: Add sampling rates support for Mbox3 This adds support for all sample rates supported by the hardware,Digidesign Mbox 3 supports: {44100, 48000, 88200, 96000} Fixes syncing clock issues that presented as pops. To test this, without this patch playing 440hz tone produces pops. Clock is now synced between playback and capture interfaces so no more latency drift issue when using pipewire pro-profile. (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3900) Signed-off-by: Manuel Barrio Linares Link: https://lore.kernel.org/r/20240430171020.192285-1-mbarriolinares@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 38 +++++++++++++-------- sound/usb/quirks.c | 74 +++++++++++++++++++++++++++++++--------- 2 files changed, 81 insertions(+), 31 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5d72dc8441cbb..73abc38a54006 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3013,21 +3013,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = 0x00, - .endpoint = 0x01, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_OUT, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .sync_ep = USB_RECIP_INTERFACE | USB_DIR_IN, + .sync_iface = 3, + .sync_altsetting = 1, + .sync_ep_idx = 1, + .implicit_fb = 1, } }, { @@ -3035,22 +3042,25 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 3, .altsetting = 1, .altset_idx = 1, - .endpoint = 0x81, .attributes = 0x00, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_IN, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x009c, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .implicit_fb = 0, } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e61c606e..2f961f0e93782 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -984,21 +984,13 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } -static void mbox3_setup_48_24_magic(struct usb_device *dev) +static void mbox3_setup_defaults(struct usb_device *dev) { /* The Mbox 3 is "little endian" */ /* max volume is: 0x0000. */ /* min volume is: 0x0080 (shown in little endian form) */ - - /* Load 48000Hz rate into buffer */ - u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00}; - - /* Set 48000Hz sample rate */ - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed? - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4); + u8 com_buff[2]; /* Deactivate Tuner */ /* on = 0x01*/ @@ -1008,6 +1000,8 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 0x01, 0x21, 0x0003, 0x2001, &com_buff, 1); /* Set clock source to Internal (as opposed to S/PDIF) */ + /* Internal = 0x01*/ + /* S/PDIF = 0x02*/ com_buff[0] = 0x01; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, 0x21, 0x0100, 0x8001, &com_buff, 1); @@ -1113,9 +1107,11 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 1, 0x21, 0x0107, 0x4201, &com_buff, 2); /* Toggle allowing host control */ + /* Not needed com_buff[0] = 0x02; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + */ /* Do not dim fx returns */ com_buff[0] = 0x00; @@ -1259,26 +1255,27 @@ static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { - dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + dev_err(&dev->dev, "MBOX3: Invalid descriptor size=%d.\n", descriptor_size); return -ENODEV; } - dev_dbg(&dev->dev, "device initialised!\n"); + dev_dbg(&dev->dev, "MBOX3: device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) - dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + dev_dbg(&dev->dev, "MBOX3: error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) - dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); - dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + dev_dbg(&dev->dev, "MBOX3: error usb_reset_configuration: %d\n", err); + + dev_dbg(&dev->dev, "MBOX3: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); - mbox3_setup_48_24_magic(dev); - dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + mbox3_setup_defaults(dev); + dev_info(&dev->dev, "MBOX3: Initialized."); return 0; /* Successful boot */ } @@ -1734,6 +1731,46 @@ static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, return 0; } +static void mbox3_set_format_quirk(struct snd_usb_substream *subs, + const struct audioformat *fmt) +{ + __le32 buff4 = 0; + u8 buff1 = 0x01; + u32 new_rate = subs->data_endpoint->cur_rate; + u32 current_rate; + + // Get current rate from card and check if changing it is needed + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4); + current_rate = le32_to_cpu(buff4); + dev_dbg(&subs->dev->dev, + "MBOX3: Current configured sample rate: %d", current_rate); + if (current_rate == new_rate) { + dev_dbg(&subs->dev->dev, + "MBOX3: No change needed (current rate:%d == new rate:%d)", + current_rate, new_rate); + return; + } + + // Set new rate + dev_info(&subs->dev->dev, + "MBOX3: Changing sample rate to: %d", new_rate); + buff4 = cpu_to_le32(new_rate); + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &buff4, 4); + + // Set clock source to Internal + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8001, &buff1, 1); + + // Check whether the change was successful + buff4 = 0; + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4); + if (new_rate != le32_to_cpu(buff4)) + dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate"); +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1755,6 +1792,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */ pioneer_djm_set_format_quirk(subs, 0x0086); break; + case USB_ID(0x0dba, 0x5000): + mbox3_set_format_quirk(subs, fmt); /* Digidesign Mbox 3 */ + break; } } From 222a596640295e5f2deb7bd0a805f8f6a3068069 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Tue, 30 Apr 2024 14:10:27 +0200 Subject: [PATCH 545/603] ALSA: aoa: soundbus: i2sbus: pcm: use 'time_left' variable with wait_for_completion_timeout() There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_completion_timeout() causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code self explaining. Fix to the proper variable type 'unsigned long' while here. Signed-off-by: Wolfram Sang Link: https://lore.kernel.org/r/20240430121028.30443-1-wsa+renesas@sang-engineering.com Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 07df5cc0f2d7c..98b812ffbde68 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -255,24 +255,24 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev, { unsigned long flags; DECLARE_COMPLETION_ONSTACK(done); - long timeout; + unsigned long time_left; spin_lock_irqsave(&i2sdev->low_lock, flags); if (pi->dbdma_ring.stopping) { pi->stop_completion = &done; spin_unlock_irqrestore(&i2sdev->low_lock, flags); - timeout = wait_for_completion_timeout(&done, HZ); + time_left = wait_for_completion_timeout(&done, HZ); spin_lock_irqsave(&i2sdev->low_lock, flags); pi->stop_completion = NULL; - if (timeout == 0) { + if (time_left == 0) { /* timeout expired, stop dbdma forcefully */ printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n"); /* make sure RUN, PAUSE and S0 bits are cleared */ out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16); pi->dbdma_ring.stopping = 0; - timeout = 10; + time_left = 10; while (in_le32(&pi->dbdma->status) & ACTIVE) { - if (--timeout <= 0) + if (--time_left <= 0) break; udelay(1); } From e8336a63de03eab16927abe1fc686c3ea10b27c6 Mon Sep 17 00:00:00 2001 From: Chaitanya Kumar Borah Date: Mon, 6 May 2024 10:55:31 +0530 Subject: [PATCH 546/603] ALSA: hda: Add Intel BMG PCI ID and HDMI codec vid Add HD Audio PCI ID and HDMI codec vendor ID for Intel Battlemage. Signed-off-by: Chaitanya Kumar Borah Link: https://lore.kernel.org/r/20240506052531.1150062-1-chaitanya.kumar.borah@intel.com Signed-off-by: Takashi Iwai --- include/linux/pci_ids.h | 1 + include/sound/hdaudio.h | 1 + sound/pci/hda/hda_intel.c | 2 ++ sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 5 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index a0c75e467df36..ae8e662429274 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -3106,6 +3106,7 @@ #define PCI_DEVICE_ID_INTEL_HDA_CML_S 0xa3f0 #define PCI_DEVICE_ID_INTEL_HDA_LNL_P 0xa828 #define PCI_DEVICE_ID_INTEL_S21152BB 0xb152 +#define PCI_DEVICE_ID_INTEL_HDA_BMG 0xe2f7 #define PCI_DEVICE_ID_INTEL_HDA_CML_R 0xf0c8 #define PCI_DEVICE_ID_INTEL_HDA_RKL_S 0xf1c8 diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 85d2c3d00a271..1d10939e40afc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -732,6 +732,7 @@ static inline unsigned int snd_array_index(struct snd_array *array, void *ptr) { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_0) }, \ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_1) }, \ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_DG2_2) }, \ + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_HDA_BMG) }, \ { } \ }, pci) || HDA_CONTROLLER_IS_HSW(pci)) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6a3022a87245a..a03f17f1328fc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2505,6 +2505,8 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_RPL_M, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, { PCI_DEVICE_DATA(INTEL, HDA_RPL_PX, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, { PCI_DEVICE_DATA(INTEL, HDA_MTL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, + /* Battlemage */ + { PCI_DEVICE_DATA(INTEL, HDA_BMG, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, /* Lunarlake-P */ { PCI_DEVICE_DATA(INTEL, HDA_LNL_P, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_LNL) }, /* Arrow Lake-S */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 495d63101186f..0a00c5ea42e05 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4642,6 +4642,7 @@ HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281d, "Meteor Lake HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281e, "Battlemage HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281f, "Raptor Lake P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862820, "Lunar Lake HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), From 5e7488cdeb26bd5bbc1beb265dba5b33d7b60e68 Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Mon, 6 May 2024 11:52:19 +0300 Subject: [PATCH 547/603] ALSA: docs: Correct the kernel object suffix of target The correct suffix is 'y' for the kernel code and 'objs' for the user space. Update documentation. Signed-off-by: Andy Shevchenko Link: https://lore.kernel.org/r/20240506085219.3403731-1-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- Documentation/sound/kernel-api/writing-an-alsa-driver.rst | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 2d2998faff62b..801b0bb57e974 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -3976,7 +3976,7 @@ Driver with A Single Source File Suppose you have a file xyz.c. Add the following two lines:: - snd-xyz-objs := xyz.o + snd-xyz-y := xyz.o obj-$(CONFIG_SND_XYZ) += snd-xyz.o 2. Create the Kconfig entry @@ -4019,7 +4019,7 @@ located in the new subdirectory, sound/pci/xyz. 2. Under the directory ``sound/pci/xyz``, create a Makefile:: - snd-xyz-objs := xyz.o abc.o def.o + snd-xyz-y := xyz.o abc.o def.o obj-$(CONFIG_SND_XYZ) += snd-xyz.o 3. Create the Kconfig entry From b9112b17950c955071abfd4331d4daa162d6ec4d Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 6 May 2024 09:54:19 +0200 Subject: [PATCH 548/603] selftests/alsa: make dump_config_tree() as void function dump_config_tree() is declared to return an int, but the compiler cannot prove that it always returns any value at all. This leads to a clang warning, when building via: make LLVM=1 -C tools/testing/selftests Suggested-by: John Hubbard Cc: Mark Brown Signed-off-by: Jaroslav Kysela Reviewed-by: Mark Brown Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240506075419.301780-1-perex@perex.cz --- tools/testing/selftests/alsa/conf.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/conf.c b/tools/testing/selftests/alsa/conf.c index 89e3656a042d7..e2b3a5810f474 100644 --- a/tools/testing/selftests/alsa/conf.c +++ b/tools/testing/selftests/alsa/conf.c @@ -105,7 +105,7 @@ static struct card_cfg_data *conf_data_by_card(int card, bool msg) return NULL; } -static int dump_config_tree(snd_config_t *top) +static void dump_config_tree(snd_config_t *top) { snd_output_t *out; int err; From 318555454100fe64ae8b82866c904f2880829e19 Mon Sep 17 00:00:00 2001 From: Junhao Pei Date: Mon, 6 May 2024 16:10:39 +0800 Subject: [PATCH 549/603] ALSA: hda/realtek: Fix internal speakers for Legion Y9000X 2022 IAH7 This fixes the sound not working from internal speakers on Lenovo Legion Y9000X 2022 IAH7 models. Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218744 Signed-off-by: Junhao Pei Link: https://lore.kernel.org/r/TYCP286MB2535ACD5188A91C000B7CD13C41C2@TYCP286MB2535.JPNP286.PROD.OUTLOOK.COM Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 3 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index ad9b6abdf269f..6a7a6d486916a 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -118,6 +118,7 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10433A60", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "17AA3865", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3866", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "17AA386E", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 2, -1, 0, 0, 0 }, { "17AA386F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, @@ -518,6 +519,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10433A60", generic_dsd_config }, { "CSC3551", "17AA3865", generic_dsd_config }, { "CSC3551", "17AA3866", generic_dsd_config }, + { "CSC3551", "17AA386E", generic_dsd_config }, { "CSC3551", "17AA386F", generic_dsd_config }, { "CSC3551", "17AA3877", generic_dsd_config }, { "CSC3551", "17AA3878", generic_dsd_config }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2eb419c814f6a..8502b30d5d5d4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10492,6 +10492,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3865, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3866, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7/7i", ALC287_FIXUP_LENOVO_LEGION_7), SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), From 21a522f935e476e5c0119f5a1f6d2278cff57daa Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 7 May 2024 17:01:31 +0100 Subject: [PATCH 550/603] ALSA: hda/realtek: Add support for ASUS Zenbook 2024 HN7306W This laptop uses CS35L41 HDA with 2 amps using I2C and Internal Boost. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20240507160131.356114-1-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8502b30d5d5d4..8c157158f1cb3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10276,6 +10276,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), From 3983f7b9e3b932daa8e510c8ca80cc7f18674d71 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:01 +0200 Subject: [PATCH 551/603] ALSA: core: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-2-tiwai@suse.de --- sound/Makefile | 2 +- sound/core/Makefile | 18 +++++++++--------- sound/core/oss/Makefile | 2 +- sound/core/seq/Makefile | 14 +++++++------- sound/core/seq/oss/Makefile | 2 +- 5 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/Makefile b/sound/Makefile index 04ef04b1168f3..5942311a4232c 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -17,4 +17,4 @@ ifeq ($(CONFIG_SND),y) obj-y += last.o endif -soundcore-objs := sound_core.o +soundcore-y := sound_core.o diff --git a/sound/core/Makefile b/sound/core/Makefile index b8aa886198ab0..31a0623cc89d1 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -24,18 +24,18 @@ snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o CFLAGS_pcm_lib.o := -I$(src) CFLAGS_pcm_native.o := -I$(src) -snd-pcm-dmaengine-objs := pcm_dmaengine.o +snd-pcm-dmaengine-y := pcm_dmaengine.o -snd-ctl-led-objs := control_led.o -snd-rawmidi-objs := rawmidi.o -snd-ump-objs := ump.o +snd-ctl-led-y := control_led.o +snd-rawmidi-y := rawmidi.o +snd-ump-y := ump.o snd-ump-$(CONFIG_SND_UMP_LEGACY_RAWMIDI) += ump_convert.o -snd-timer-objs := timer.o -snd-hrtimer-objs := hrtimer.o -snd-hwdep-objs := hwdep.o -snd-seq-device-objs := seq_device.o +snd-timer-y := timer.o +snd-hrtimer-y := hrtimer.o +snd-hwdep-y := hwdep.o +snd-seq-device-y := seq_device.o -snd-compress-objs := compress_offload.o +snd-compress-y := compress_offload.o obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_CTL_LED) += snd-ctl-led.o diff --git a/sound/core/oss/Makefile b/sound/core/oss/Makefile index ae25edcc3b42e..d5f48ae6ba966 100644 --- a/sound/core/oss/Makefile +++ b/sound/core/oss/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 1999 by Jaroslav Kysela # -snd-mixer-oss-objs := mixer_oss.o +snd-mixer-oss-y := mixer_oss.o snd-pcm-oss-y := pcm_oss.o snd-pcm-oss-$(CONFIG_SND_PCM_OSS_PLUGINS) += pcm_plugin.o \ diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index 990eec7c83ad1..0904aa48d88b6 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -4,17 +4,17 @@ # Copyright (c) 1999 by Jaroslav Kysela # -snd-seq-objs := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \ +snd-seq-y := seq.o seq_lock.o seq_clientmgr.o seq_memory.o seq_queue.o \ seq_fifo.o seq_prioq.o seq_timer.o \ seq_system.o seq_ports.o snd-seq-$(CONFIG_SND_PROC_FS) += seq_info.o snd-seq-$(CONFIG_SND_SEQ_UMP) += seq_ump_convert.o -snd-seq-midi-objs := seq_midi.o -snd-seq-midi-emul-objs := seq_midi_emul.o -snd-seq-midi-event-objs := seq_midi_event.o -snd-seq-dummy-objs := seq_dummy.o -snd-seq-virmidi-objs := seq_virmidi.o -snd-seq-ump-client-objs := seq_ump_client.o +snd-seq-midi-y := seq_midi.o +snd-seq-midi-emul-y := seq_midi_emul.o +snd-seq-midi-event-y := seq_midi_event.o +snd-seq-dummy-y := seq_dummy.o +snd-seq-virmidi-y := seq_virmidi.o +snd-seq-ump-client-y := seq_ump_client.o obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o obj-$(CONFIG_SND_SEQUENCER_OSS) += oss/ diff --git a/sound/core/seq/oss/Makefile b/sound/core/seq/oss/Makefile index f1a60878549a9..4e47418342087 100644 --- a/sound/core/seq/oss/Makefile +++ b/sound/core/seq/oss/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 1999 by Jaroslav Kysela # -snd-seq-oss-objs := seq_oss.o seq_oss_init.o seq_oss_timer.o seq_oss_ioctl.o \ +snd-seq-oss-y := seq_oss.o seq_oss_init.o seq_oss_timer.o seq_oss_ioctl.o \ seq_oss_event.o seq_oss_rw.o seq_oss_synth.o \ seq_oss_midi.o seq_oss_readq.o seq_oss_writeq.o From 1f75c498d615d8f6b94fff27a14229c2b52c3572 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:02 +0200 Subject: [PATCH 552/603] ALSA: pci: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-3-tiwai@suse.de --- sound/pci/Makefile | 48 ++++++++++++++++----------------- sound/pci/ali5451/Makefile | 2 +- sound/pci/asihpi/Makefile | 2 +- sound/pci/au88x0/Makefile | 6 ++--- sound/pci/aw2/Makefile | 2 +- sound/pci/ca0106/Makefile | 2 +- sound/pci/ctxfi/Makefile | 2 +- sound/pci/echoaudio/Makefile | 28 +++++++++---------- sound/pci/emu10k1/Makefile | 6 ++--- sound/pci/hda/Makefile | 52 ++++++++++++++++++------------------ sound/pci/ice1712/Makefile | 6 ++--- sound/pci/korg1212/Makefile | 2 +- sound/pci/lx6464es/Makefile | 2 +- sound/pci/mixart/Makefile | 2 +- sound/pci/nm256/Makefile | 2 +- sound/pci/oxygen/Makefile | 8 +++--- sound/pci/pcxhr/Makefile | 2 +- sound/pci/riptide/Makefile | 2 +- sound/pci/rme9652/Makefile | 6 ++--- sound/pci/trident/Makefile | 2 +- sound/pci/vx222/Makefile | 2 +- sound/pci/ymfpci/Makefile | 2 +- 22 files changed, 94 insertions(+), 94 deletions(-) diff --git a/sound/pci/Makefile b/sound/pci/Makefile index 04cac74691398..18b673018dfd6 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -4,30 +4,30 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-ad1889-objs := ad1889.o -snd-als300-objs := als300.o -snd-als4000-objs := als4000.o -snd-atiixp-objs := atiixp.o -snd-atiixp-modem-objs := atiixp_modem.o -snd-azt3328-objs := azt3328.o -snd-bt87x-objs := bt87x.o -snd-cmipci-objs := cmipci.o -snd-cs4281-objs := cs4281.o -snd-cs5530-objs := cs5530.o -snd-ens1370-objs := ens1370.o ak4531_codec.o -snd-ens1371-objs := ens1371.o -snd-es1938-objs := es1938.o -snd-es1968-objs := es1968.o -snd-fm801-objs := fm801.o -snd-intel8x0-objs := intel8x0.o -snd-intel8x0m-objs := intel8x0m.o -snd-maestro3-objs := maestro3.o -snd-rme32-objs := rme32.o -snd-rme96-objs := rme96.o -snd-sis7019-objs := sis7019.o -snd-sonicvibes-objs := sonicvibes.o -snd-via82xx-objs := via82xx.o -snd-via82xx-modem-objs := via82xx_modem.o +snd-ad1889-y := ad1889.o +snd-als300-y := als300.o +snd-als4000-y := als4000.o +snd-atiixp-y := atiixp.o +snd-atiixp-modem-y := atiixp_modem.o +snd-azt3328-y := azt3328.o +snd-bt87x-y := bt87x.o +snd-cmipci-y := cmipci.o +snd-cs4281-y := cs4281.o +snd-cs5530-y := cs5530.o +snd-ens1370-y := ens1370.o ak4531_codec.o +snd-ens1371-y := ens1371.o +snd-es1938-y := es1938.o +snd-es1968-y := es1968.o +snd-fm801-y := fm801.o +snd-intel8x0-y := intel8x0.o +snd-intel8x0m-y := intel8x0m.o +snd-maestro3-y := maestro3.o +snd-rme32-y := rme32.o +snd-rme96-y := rme96.o +snd-sis7019-y := sis7019.o +snd-sonicvibes-y := sonicvibes.o +snd-via82xx-y := via82xx.o +snd-via82xx-modem-y := via82xx_modem.o # Toplevel Module Dependency obj-$(CONFIG_SND_AD1889) += snd-ad1889.o diff --git a/sound/pci/ali5451/Makefile b/sound/pci/ali5451/Makefile index 8156198fbaebf..e319a4c1d6b27 100644 --- a/sound/pci/ali5451/Makefile +++ b/sound/pci/ali5451/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-ali5451-objs := ali5451.o +snd-ali5451-y := ali5451.o # Toplevel Module Dependency obj-$(CONFIG_SND_ALI5451) += snd-ali5451.o diff --git a/sound/pci/asihpi/Makefile b/sound/pci/asihpi/Makefile index 8351f8f5b5239..d558a974fa7e1 100644 --- a/sound/pci/asihpi/Makefile +++ b/sound/pci/asihpi/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-asihpi-objs := asihpi.o hpioctl.o hpimsginit.o\ +snd-asihpi-y := asihpi.o hpioctl.o hpimsginit.o\ hpicmn.o hpifunc.o hpidebug.o hpidspcd.o\ hpios.o hpi6000.o hpi6205.o hpimsgx.o diff --git a/sound/pci/au88x0/Makefile b/sound/pci/au88x0/Makefile index 78ab11562f4de..5ec5abdee28d2 100644 --- a/sound/pci/au88x0/Makefile +++ b/sound/pci/au88x0/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 -snd-au8810-objs := au8810.o -snd-au8820-objs := au8820.o -snd-au8830-objs := au8830.o +snd-au8810-y := au8810.o +snd-au8820-y := au8820.o +snd-au8830-y := au8830.o obj-$(CONFIG_SND_AU8810) += snd-au8810.o obj-$(CONFIG_SND_AU8820) += snd-au8820.o diff --git a/sound/pci/aw2/Makefile b/sound/pci/aw2/Makefile index f9045afb4cdaf..c246f7c7f2bf3 100644 --- a/sound/pci/aw2/Makefile +++ b/sound/pci/aw2/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-aw2-objs := aw2-alsa.o aw2-saa7146.o +snd-aw2-y := aw2-alsa.o aw2-saa7146.o obj-$(CONFIG_SND_AW2) += snd-aw2.o diff --git a/sound/pci/ca0106/Makefile b/sound/pci/ca0106/Makefile index 9e51d3df3ee8c..693dc4d809250 100644 --- a/sound/pci/ca0106/Makefile +++ b/sound/pci/ca0106/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-ca0106-objs := ca0106_main.o ca0106_mixer.o ca_midi.o +snd-ca0106-y := ca0106_main.o ca0106_mixer.o ca_midi.o snd-ca0106-$(CONFIG_SND_PROC_FS) += ca0106_proc.o obj-$(CONFIG_SND_CA0106) += snd-ca0106.o diff --git a/sound/pci/ctxfi/Makefile b/sound/pci/ctxfi/Makefile index 70888706a0afe..ff2b1cba3a3ca 100644 --- a/sound/pci/ctxfi/Makefile +++ b/sound/pci/ctxfi/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-ctxfi-objs := xfi.o ctatc.o ctvmem.o ctpcm.o ctmixer.o ctresource.o \ +snd-ctxfi-y := xfi.o ctatc.o ctvmem.o ctpcm.o ctmixer.o ctresource.o \ ctsrc.o ctamixer.o ctdaio.o ctimap.o cthardware.o cttimer.o \ cthw20k2.o cthw20k1.o diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile index 4865b8fe74343..96667641c7cf5 100644 --- a/sound/pci/echoaudio/Makefile +++ b/sound/pci/echoaudio/Makefile @@ -4,20 +4,20 @@ # Copyright (c) 2003 by Giuliano Pochini # -snd-darla20-objs := darla20.o -snd-gina20-objs := gina20.o -snd-layla20-objs := layla20.o -snd-darla24-objs := darla24.o -snd-gina24-objs := gina24.o -snd-layla24-objs := layla24.o -snd-mona-objs := mona.o -snd-mia-objs := mia.o -snd-echo3g-objs := echo3g.o -snd-indigo-objs := indigo.o -snd-indigoio-objs := indigoio.o -snd-indigodj-objs := indigodj.o -snd-indigoiox-objs := indigoiox.o -snd-indigodjx-objs := indigodjx.o +snd-darla20-y := darla20.o +snd-gina20-y := gina20.o +snd-layla20-y := layla20.o +snd-darla24-y := darla24.o +snd-gina24-y := gina24.o +snd-layla24-y := layla24.o +snd-mona-y := mona.o +snd-mia-y := mia.o +snd-echo3g-y := echo3g.o +snd-indigo-y := indigo.o +snd-indigoio-y := indigoio.o +snd-indigodj-y := indigodj.o +snd-indigoiox-y := indigoiox.o +snd-indigodjx-y := indigodjx.o obj-$(CONFIG_SND_DARLA20) += snd-darla20.o obj-$(CONFIG_SND_GINA20) += snd-gina20.o diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile index 17d5527be3194..1f325abcb3ef4 100644 --- a/sound/pci/emu10k1/Makefile +++ b/sound/pci/emu10k1/Makefile @@ -4,12 +4,12 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-emu10k1-objs := emu10k1.o emu10k1_main.o \ +snd-emu10k1-y := emu10k1.o emu10k1_main.o \ irq.o memory.o voice.o emumpu401.o emupcm.o io.o \ emumixer.o emufx.o timer.o p16v.o snd-emu10k1-$(CONFIG_SND_PROC_FS) += emuproc.o -snd-emu10k1-synth-objs := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o -snd-emu10k1x-objs := emu10k1x.o +snd-emu10k1-synth-y := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o +snd-emu10k1x-y := emu10k1x.o # Toplevel Module Dependency obj-$(CONFIG_SND_EMU10K1) += snd-emu10k1.o diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 13e04e1f65de2..058ca0a289e4d 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 -snd-hda-intel-objs := hda_intel.o -snd-hda-tegra-objs := hda_tegra.o +snd-hda-intel-y := hda_intel.o +snd-hda-tegra-y := hda_tegra.o snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o snd-hda-codec-y += hda_controller.o @@ -13,32 +13,32 @@ snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o CFLAGS_hda_controller.o := -I$(src) CFLAGS_hda_intel.o := -I$(src) -snd-hda-codec-generic-objs := hda_generic.o -snd-hda-codec-realtek-objs := patch_realtek.o -snd-hda-codec-cmedia-objs := patch_cmedia.o -snd-hda-codec-analog-objs := patch_analog.o -snd-hda-codec-idt-objs := patch_sigmatel.o -snd-hda-codec-si3054-objs := patch_si3054.o -snd-hda-codec-cirrus-objs := patch_cirrus.o -snd-hda-codec-cs8409-objs := patch_cs8409.o patch_cs8409-tables.o -snd-hda-codec-ca0110-objs := patch_ca0110.o -snd-hda-codec-ca0132-objs := patch_ca0132.o -snd-hda-codec-conexant-objs := patch_conexant.o -snd-hda-codec-via-objs := patch_via.o -snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o +snd-hda-codec-generic-y := hda_generic.o +snd-hda-codec-realtek-y := patch_realtek.o +snd-hda-codec-cmedia-y := patch_cmedia.o +snd-hda-codec-analog-y := patch_analog.o +snd-hda-codec-idt-y := patch_sigmatel.o +snd-hda-codec-si3054-y := patch_si3054.o +snd-hda-codec-cirrus-y := patch_cirrus.o +snd-hda-codec-cs8409-y := patch_cs8409.o patch_cs8409-tables.o +snd-hda-codec-ca0110-y := patch_ca0110.o +snd-hda-codec-ca0132-y := patch_ca0132.o +snd-hda-codec-conexant-y := patch_conexant.o +snd-hda-codec-via-y := patch_via.o +snd-hda-codec-hdmi-y := patch_hdmi.o hda_eld.o # side codecs -snd-hda-cirrus-scodec-objs := cirrus_scodec.o -snd-hda-cirrus-scodec-test-objs := cirrus_scodec_test.o -snd-hda-scodec-cs35l41-objs := cs35l41_hda.o cs35l41_hda_property.o -snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o -snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o -snd-hda-scodec-cs35l56-objs := cs35l56_hda.o -snd-hda-scodec-cs35l56-i2c-objs := cs35l56_hda_i2c.o -snd-hda-scodec-cs35l56-spi-objs := cs35l56_hda_spi.o -snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o -snd-hda-scodec-component-objs := hda_component.o -snd-hda-scodec-tas2781-i2c-objs := tas2781_hda_i2c.o +snd-hda-cirrus-scodec-y := cirrus_scodec.o +snd-hda-cirrus-scodec-test-y := cirrus_scodec_test.o +snd-hda-scodec-cs35l41-y := cs35l41_hda.o cs35l41_hda_property.o +snd-hda-scodec-cs35l41-i2c-y := cs35l41_hda_i2c.o +snd-hda-scodec-cs35l41-spi-y := cs35l41_hda_spi.o +snd-hda-scodec-cs35l56-y := cs35l56_hda.o +snd-hda-scodec-cs35l56-i2c-y := cs35l56_hda_i2c.o +snd-hda-scodec-cs35l56-spi-y := cs35l56_hda_spi.o +snd-hda-cs-dsp-ctls-y := hda_cs_dsp_ctl.o +snd-hda-scodec-component-y := hda_component.o +snd-hda-scodec-tas2781-i2c-y := tas2781_hda_i2c.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile index 1196f22a9b452..f406a048374ce 100644 --- a/sound/pci/ice1712/Makefile +++ b/sound/pci/ice1712/Makefile @@ -4,9 +4,9 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-ice17xx-ak4xxx-objs := ak4xxx.o -snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o -snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o quartet.o psc724.o wm8766.o wm8776.o +snd-ice17xx-ak4xxx-y := ak4xxx.o +snd-ice1712-y := ice1712.o delta.o hoontech.o ews.o +snd-ice1724-y := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o quartet.o psc724.o wm8766.o wm8776.o # Toplevel Module Dependency obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o diff --git a/sound/pci/korg1212/Makefile b/sound/pci/korg1212/Makefile index 42eb287c77af8..ab0186ffbd589 100644 --- a/sound/pci/korg1212/Makefile +++ b/sound/pci/korg1212/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-korg1212-objs := korg1212.o +snd-korg1212-y := korg1212.o # Toplevel Module Dependency obj-$(CONFIG_SND_KORG1212) += snd-korg1212.o diff --git a/sound/pci/lx6464es/Makefile b/sound/pci/lx6464es/Makefile index c295f68bac68f..2b3047c7a388a 100644 --- a/sound/pci/lx6464es/Makefile +++ b/sound/pci/lx6464es/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-lx6464es-objs := lx6464es.o lx_core.o +snd-lx6464es-y := lx6464es.o lx_core.o obj-$(CONFIG_SND_LX6464ES) += snd-lx6464es.o diff --git a/sound/pci/mixart/Makefile b/sound/pci/mixart/Makefile index 16cfeb78a0b6c..b803e5e72791e 100644 --- a/sound/pci/mixart/Makefile +++ b/sound/pci/mixart/Makefile @@ -4,6 +4,6 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-mixart-objs := mixart.o mixart_core.o mixart_hwdep.o mixart_mixer.o +snd-mixart-y := mixart.o mixart_core.o mixart_hwdep.o mixart_mixer.o obj-$(CONFIG_SND_MIXART) += snd-mixart.o diff --git a/sound/pci/nm256/Makefile b/sound/pci/nm256/Makefile index 3063766ac56bb..7d55fe774d20b 100644 --- a/sound/pci/nm256/Makefile +++ b/sound/pci/nm256/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-nm256-objs := nm256.o +snd-nm256-y := nm256.o # Toplevel Module Dependency obj-$(CONFIG_SND_NM256) += snd-nm256.o diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index 0dfc4f8409922..cc0c246947500 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0 -snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o -snd-oxygen-objs := oxygen.o xonar_dg_mixer.o xonar_dg.o -snd-se6x-objs := se6x.o -snd-virtuoso-objs := virtuoso.o xonar_lib.o \ +snd-oxygen-lib-y := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o +snd-oxygen-y := oxygen.o xonar_dg_mixer.o xonar_dg.o +snd-se6x-y := se6x.o +snd-virtuoso-y := virtuoso.o xonar_lib.o \ xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o diff --git a/sound/pci/pcxhr/Makefile b/sound/pci/pcxhr/Makefile index 5993d86cfb5d9..0ea1e5ccb56ff 100644 --- a/sound/pci/pcxhr/Makefile +++ b/sound/pci/pcxhr/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o pcxhr_mix22.o +snd-pcxhr-y := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o pcxhr_mix22.o obj-$(CONFIG_SND_PCXHR) += snd-pcxhr.o diff --git a/sound/pci/riptide/Makefile b/sound/pci/riptide/Makefile index 9a505bae243eb..9b4e9595859ac 100644 --- a/sound/pci/riptide/Makefile +++ b/sound/pci/riptide/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-riptide-objs := riptide.o +snd-riptide-y := riptide.o obj-$(CONFIG_SND_RIPTIDE) += snd-riptide.o diff --git a/sound/pci/rme9652/Makefile b/sound/pci/rme9652/Makefile index a3351447ddc00..cc99ae8922118 100644 --- a/sound/pci/rme9652/Makefile +++ b/sound/pci/rme9652/Makefile @@ -4,9 +4,9 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-rme9652-objs := rme9652.o -snd-hdsp-objs := hdsp.o -snd-hdspm-objs := hdspm.o +snd-rme9652-y := rme9652.o +snd-hdsp-y := hdsp.o +snd-hdspm-y := hdspm.o # Toplevel Module Dependency obj-$(CONFIG_SND_RME9652) += snd-rme9652.o diff --git a/sound/pci/trident/Makefile b/sound/pci/trident/Makefile index e8975bc37fcbf..476d16abcfc96 100644 --- a/sound/pci/trident/Makefile +++ b/sound/pci/trident/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-trident-objs := trident.o trident_main.o trident_memory.o +snd-trident-y := trident.o trident_main.o trident_memory.o # Toplevel Module Dependency obj-$(CONFIG_SND_TRIDENT) += snd-trident.o diff --git a/sound/pci/vx222/Makefile b/sound/pci/vx222/Makefile index dda900e453856..6889137eb438f 100644 --- a/sound/pci/vx222/Makefile +++ b/sound/pci/vx222/Makefile @@ -4,6 +4,6 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-vx222-objs := vx222.o vx222_ops.o +snd-vx222-y := vx222.o vx222_ops.o obj-$(CONFIG_SND_VX222) += snd-vx222.o diff --git a/sound/pci/ymfpci/Makefile b/sound/pci/ymfpci/Makefile index 40a1d83e1a9e7..2d78564033712 100644 --- a/sound/pci/ymfpci/Makefile +++ b/sound/pci/ymfpci/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-ymfpci-objs := ymfpci.o ymfpci_main.o +snd-ymfpci-y := ymfpci.o ymfpci_main.o # Toplevel Module Dependency obj-$(CONFIG_SND_YMFPCI) += snd-ymfpci.o From b26693926c6a9caf4fc405d26157cdd12c7d2d2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:03 +0200 Subject: [PATCH 553/603] ALSA: hda: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-4-tiwai@suse.de --- sound/hda/Makefile | 8 ++++---- sound/hda/ext/Makefile | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 78f487a635f88..83cceafe0d4c3 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0 -snd-hda-core-objs := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \ +snd-hda-core-y := hda_bus_type.o hdac_bus.o hdac_device.o hdac_sysfs.o \ hdac_regmap.o hdac_controller.o hdac_stream.o array.o hdmi_chmap.o -snd-hda-core-objs += trace.o +snd-hda-core-y += trace.o CFLAGS_trace.o := -I$(src) # for sync with i915 gfx driver @@ -14,9 +14,9 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-dspcfg-objs := intel-dsp-config.o +snd-intel-dspcfg-y := intel-dsp-config.o snd-intel-dspcfg-$(CONFIG_SND_INTEL_NHLT) += intel-nhlt.o obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o -snd-intel-sdw-acpi-objs := intel-sdw-acpi.o +snd-intel-sdw-acpi-y := intel-sdw-acpi.o obj-$(CONFIG_SND_INTEL_SOUNDWIRE_ACPI) += snd-intel-sdw-acpi.o diff --git a/sound/hda/ext/Makefile b/sound/hda/ext/Makefile index 154779bdc0baa..05883fb28d289 100644 --- a/sound/hda/ext/Makefile +++ b/sound/hda/ext/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-hda-ext-core-objs := hdac_ext_bus.o hdac_ext_controller.o hdac_ext_stream.o +snd-hda-ext-core-y := hdac_ext_bus.o hdac_ext_controller.o hdac_ext_stream.o obj-$(CONFIG_SND_HDA_EXT_CORE) += snd-hda-ext-core.o From 2e7b8fd6d9bccaa54b7504c3e0a729f0c47765c1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:04 +0200 Subject: [PATCH 554/603] ALSA: isa: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-5-tiwai@suse.de --- sound/isa/Makefile | 18 +++++++++--------- sound/isa/ad1816a/Makefile | 2 +- sound/isa/ad1848/Makefile | 2 +- sound/isa/cs423x/Makefile | 4 ++-- sound/isa/es1688/Makefile | 4 ++-- sound/isa/galaxy/Makefile | 4 ++-- sound/isa/gus/Makefile | 12 ++++++------ sound/isa/msnd/Makefile | 6 +++--- sound/isa/opti9xx/Makefile | 8 ++++---- sound/isa/sb/Makefile | 18 +++++++++--------- sound/isa/wavefront/Makefile | 2 +- sound/isa/wss/Makefile | 2 +- 12 files changed, 41 insertions(+), 41 deletions(-) diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 5eaddbf4a7123..2135d68a15ac8 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -4,15 +4,15 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-adlib-objs := adlib.o -snd-als100-objs := als100.o -snd-azt2320-objs := azt2320.o -snd-cmi8328-objs := cmi8328.o -snd-cmi8330-objs := cmi8330.o -snd-es18xx-objs := es18xx.o -snd-opl3sa2-objs := opl3sa2.o -snd-sc6000-objs := sc6000.o -snd-sscape-objs := sscape.o +snd-adlib-y := adlib.o +snd-als100-y := als100.o +snd-azt2320-y := azt2320.o +snd-cmi8328-y := cmi8328.o +snd-cmi8330-y := cmi8330.o +snd-es18xx-y := es18xx.o +snd-opl3sa2-y := opl3sa2.o +snd-sc6000-y := sc6000.o +snd-sscape-y := sscape.o # Toplevel Module Dependency obj-$(CONFIG_SND_ADLIB) += snd-adlib.o diff --git a/sound/isa/ad1816a/Makefile b/sound/isa/ad1816a/Makefile index 93def7f169332..5733252285346 100644 --- a/sound/isa/ad1816a/Makefile +++ b/sound/isa/ad1816a/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-ad1816a-objs := ad1816a.o ad1816a_lib.o +snd-ad1816a-y := ad1816a.o ad1816a_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_AD1816A) += snd-ad1816a.o diff --git a/sound/isa/ad1848/Makefile b/sound/isa/ad1848/Makefile index 4eab89bbc8458..5fdfc1c9f0598 100644 --- a/sound/isa/ad1848/Makefile +++ b/sound/isa/ad1848/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-ad1848-objs := ad1848.o +snd-ad1848-y := ad1848.o # Toplevel Module Dependency obj-$(CONFIG_SND_AD1848) += snd-ad1848.o diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile index 91c6b8d644244..013a777d23fab 100644 --- a/sound/isa/cs423x/Makefile +++ b/sound/isa/cs423x/Makefile @@ -4,8 +4,8 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-cs4231-objs := cs4231.o -snd-cs4236-objs := cs4236.o cs4236_lib.o +snd-cs4231-y := cs4231.o +snd-cs4236-y := cs4236.o cs4236_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS4231) += snd-cs4231.o diff --git a/sound/isa/es1688/Makefile b/sound/isa/es1688/Makefile index c683ac36c50ee..7d6c44a8eaad0 100644 --- a/sound/isa/es1688/Makefile +++ b/sound/isa/es1688/Makefile @@ -4,8 +4,8 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-es1688-lib-objs := es1688_lib.o -snd-es1688-objs := es1688.o +snd-es1688-lib-y := es1688_lib.o +snd-es1688-y := es1688.o # Toplevel Module Dependency obj-$(CONFIG_SND_ES1688) += snd-es1688.o snd-es1688-lib.o diff --git a/sound/isa/galaxy/Makefile b/sound/isa/galaxy/Makefile index ff861f2380939..2dbd519860a62 100644 --- a/sound/isa/galaxy/Makefile +++ b/sound/isa/galaxy/Makefile @@ -4,8 +4,8 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-azt1605-objs := azt1605.o -snd-azt2316-objs := azt2316.o +snd-azt1605-y := azt1605.o +snd-azt2316-y := azt2316.o obj-$(CONFIG_SND_AZT1605) += snd-azt1605.o obj-$(CONFIG_SND_AZT2316) += snd-azt2316.o diff --git a/sound/isa/gus/Makefile b/sound/isa/gus/Makefile index c6f32ffd34205..4924c1904fa45 100644 --- a/sound/isa/gus/Makefile +++ b/sound/isa/gus/Makefile @@ -4,18 +4,18 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-gus-lib-objs := gus_main.o \ +snd-gus-lib-y := gus_main.o \ gus_io.o gus_irq.o gus_timer.o \ gus_mem.o gus_mem_proc.o gus_dram.o gus_dma.o gus_volume.o \ gus_pcm.o gus_mixer.o \ gus_uart.o \ gus_reset.o -snd-gusclassic-objs := gusclassic.o -snd-gusextreme-objs := gusextreme.o -snd-gusmax-objs := gusmax.o -snd-interwave-objs := interwave.o -snd-interwave-stb-objs := interwave-stb.o +snd-gusclassic-y := gusclassic.o +snd-gusextreme-y := gusextreme.o +snd-gusmax-y := gusmax.o +snd-interwave-y := interwave.o +snd-interwave-stb-y := interwave-stb.o # Toplevel Module Dependency obj-$(CONFIG_SND_GUSCLASSIC) += snd-gusclassic.o snd-gus-lib.o diff --git a/sound/isa/msnd/Makefile b/sound/isa/msnd/Makefile index ec231a7b1d5e7..5f8d6b472722b 100644 --- a/sound/isa/msnd/Makefile +++ b/sound/isa/msnd/Makefile @@ -1,8 +1,8 @@ # SPDX-License-Identifier: GPL-2.0 -snd-msnd-lib-objs := msnd.o msnd_midi.o msnd_pinnacle_mixer.o -snd-msnd-pinnacle-objs := msnd_pinnacle.o -snd-msnd-classic-objs := msnd_classic.o +snd-msnd-lib-y := msnd.o msnd_midi.o msnd_pinnacle_mixer.o +snd-msnd-pinnacle-y := msnd_pinnacle.o +snd-msnd-classic-y := msnd_classic.o # Toplevel Module Dependency obj-$(CONFIG_SND_MSND_PINNACLE) += snd-msnd-pinnacle.o snd-msnd-lib.o diff --git a/sound/isa/opti9xx/Makefile b/sound/isa/opti9xx/Makefile index a9dcdeb502bd9..44a2fb220456d 100644 --- a/sound/isa/opti9xx/Makefile +++ b/sound/isa/opti9xx/Makefile @@ -4,10 +4,10 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-opti92x-ad1848-objs := opti92x-ad1848.o -snd-opti92x-cs4231-objs := opti92x-cs4231.o -snd-opti93x-objs := opti93x.o -snd-miro-objs := miro.o +snd-opti92x-ad1848-y := opti92x-ad1848.o +snd-opti92x-cs4231-y := opti92x-cs4231.o +snd-opti93x-y := opti93x.o +snd-miro-y := miro.o # Toplevel Module Dependency obj-$(CONFIG_SND_OPTI92X_AD1848) += snd-opti92x-ad1848.o diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index f174a5b3c8e4a..96a926feb17a5 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -4,15 +4,15 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-sb-common-objs := sb_common.o sb_mixer.o -snd-sb8-dsp-objs := sb8_main.o sb8_midi.o -snd-sb16-dsp-objs := sb16_main.o -snd-sb16-csp-objs := sb16_csp.o -snd-sb8-objs := sb8.o -snd-sb16-objs := sb16.o -snd-sbawe-objs := sbawe.o emu8000.o -snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o -snd-jazz16-objs := jazz16.o +snd-sb-common-y := sb_common.o sb_mixer.o +snd-sb8-dsp-y := sb8_main.o sb8_midi.o +snd-sb16-dsp-y := sb16_main.o +snd-sb16-csp-y := sb16_csp.o +snd-sb8-y := sb8.o +snd-sb16-y := sb16.o +snd-sbawe-y := sbawe.o emu8000.o +snd-emu8000-synth-y := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o +snd-jazz16-y := jazz16.o # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o diff --git a/sound/isa/wavefront/Makefile b/sound/isa/wavefront/Makefile index b8406dce81f54..3ba85fb2e6cd5 100644 --- a/sound/isa/wavefront/Makefile +++ b/sound/isa/wavefront/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-wavefront-objs := wavefront.o wavefront_fx.o wavefront_synth.o wavefront_midi.o +snd-wavefront-y := wavefront.o wavefront_fx.o wavefront_synth.o wavefront_midi.o # Toplevel Module Dependency obj-$(CONFIG_SND_WAVEFRONT) += snd-wavefront.o diff --git a/sound/isa/wss/Makefile b/sound/isa/wss/Makefile index 34d0636b3dc37..f23e71d0d5d43 100644 --- a/sound/isa/wss/Makefile +++ b/sound/isa/wss/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2008 by Jaroslav Kysela # -snd-wss-lib-objs := wss_lib.o +snd-wss-lib-y := wss_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_WSS_LIB) += snd-wss-lib.o From a355e0786ce8c9e33f3f5653c398e9ee2db5345f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:05 +0200 Subject: [PATCH 555/603] ALSA: usb: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-6-tiwai@suse.de --- sound/usb/6fire/Makefile | 2 +- sound/usb/Makefile | 4 ++-- sound/usb/hiface/Makefile | 2 +- sound/usb/misc/Makefile | 2 +- sound/usb/usx2y/Makefile | 4 ++-- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/usb/6fire/Makefile b/sound/usb/6fire/Makefile index 7d353bbf74931..587f25c64e56a 100644 --- a/sound/usb/6fire/Makefile +++ b/sound/usb/6fire/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-usb-6fire-objs += chip.o comm.o midi.o control.o firmware.o pcm.o +snd-usb-6fire-y += chip.o comm.o midi.o control.o firmware.o pcm.o obj-$(CONFIG_SND_USB_6FIRE) += snd-usb-6fire.o diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 8c657c2753c84..0532499dbc6d6 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -3,7 +3,7 @@ # Makefile for ALSA # -snd-usb-audio-objs := card.o \ +snd-usb-audio-y := card.o \ clock.o \ endpoint.o \ format.o \ @@ -25,7 +25,7 @@ snd-usb-audio-objs := card.o \ snd-usb-audio-$(CONFIG_SND_USB_AUDIO_MIDI_V2) += midi2.o snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o -snd-usbmidi-lib-objs := midi.o +snd-usbmidi-lib-y := midi.o # Toplevel Module Dependency obj-$(CONFIG_SND_USB_AUDIO) += snd-usb-audio.o snd-usbmidi-lib.o diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile index 8f3b24e7d6c21..997c1558d0cb7 100644 --- a/sound/usb/hiface/Makefile +++ b/sound/usb/hiface/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-usb-hiface-objs := chip.o pcm.o +snd-usb-hiface-y := chip.o pcm.o obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o diff --git a/sound/usb/misc/Makefile b/sound/usb/misc/Makefile index 068ecd7bc0434..3e9f4adc28de4 100644 --- a/sound/usb/misc/Makefile +++ b/sound/usb/misc/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-ua101-objs := ua101.o +snd-ua101-y := ua101.o obj-$(CONFIG_SND_USB_UA101) += snd-ua101.o diff --git a/sound/usb/usx2y/Makefile b/sound/usb/usx2y/Makefile index cc4c2f1efab2c..fc033aba03a4c 100644 --- a/sound/usb/usx2y/Makefile +++ b/sound/usb/usx2y/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 -snd-usb-usx2y-objs := usbusx2y.o usX2Yhwdep.o usx2yhwdeppcm.o -snd-usb-us122l-objs := us122l.o +snd-usb-usx2y-y := usbusx2y.o usX2Yhwdep.o usx2yhwdeppcm.o +snd-usb-us122l-y := us122l.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usb-usx2y.o obj-$(CONFIG_SND_USB_US122L) += snd-usb-us122l.o From 4d256122945ec26623260babdeda90ab0796ecea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:06 +0200 Subject: [PATCH 556/603] ALSA: drivers: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-7-tiwai@suse.de --- sound/drivers/Makefile | 18 +++++++++--------- sound/drivers/mpu401/Makefile | 4 ++-- sound/drivers/opl3/Makefile | 2 +- sound/drivers/opl4/Makefile | 4 ++-- sound/drivers/pcsp/Makefile | 2 +- sound/drivers/vx/Makefile | 2 +- 6 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index 2c0c7092d396f..a08bdd70ec9c2 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -4,15 +4,15 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-dummy-objs := dummy.o -snd-aloop-objs := aloop.o -snd-mtpav-objs := mtpav.o -snd-mts64-objs := mts64.o -snd-pcmtest-objs := pcmtest.o -snd-portman2x4-objs := portman2x4.o -snd-serial-u16550-objs := serial-u16550.o -snd-serial-generic-objs := serial-generic.o -snd-virmidi-objs := virmidi.o +snd-dummy-y := dummy.o +snd-aloop-y := aloop.o +snd-mtpav-y := mtpav.o +snd-mts64-y := mts64.o +snd-pcmtest-y := pcmtest.o +snd-portman2x4-y := portman2x4.o +snd-serial-u16550-y := serial-u16550.o +snd-serial-generic-y := serial-generic.o +snd-virmidi-y := virmidi.o # Toplevel Module Dependency obj-$(CONFIG_SND_DUMMY) += snd-dummy.o diff --git a/sound/drivers/mpu401/Makefile b/sound/drivers/mpu401/Makefile index 3dfd5b374c4f8..0a96e238ee92a 100644 --- a/sound/drivers/mpu401/Makefile +++ b/sound/drivers/mpu401/Makefile @@ -4,8 +4,8 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-mpu401-objs := mpu401.o -snd-mpu401-uart-objs := mpu401_uart.o +snd-mpu401-y := mpu401.o +snd-mpu401-uart-y := mpu401_uart.o obj-$(CONFIG_SND_MPU401_UART) += snd-mpu401-uart.o diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile index 83bca9f1fbdf3..cf48263083652 100644 --- a/sound/drivers/opl3/Makefile +++ b/sound/drivers/opl3/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-opl3-lib-objs := opl3_lib.o opl3_synth.o +snd-opl3-lib-y := opl3_lib.o opl3_synth.o snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o ifneq ($(CONFIG_SND_SEQUENCER_OSS),) snd-opl3-synth-y += opl3_oss.o diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile index 6e86a4092b4c1..a841630b45c22 100644 --- a/sound/drivers/opl4/Makefile +++ b/sound/drivers/opl4/Makefile @@ -4,9 +4,9 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o +snd-opl4-lib-y := opl4_lib.o opl4_mixer.o snd-opl4-lib-$(CONFIG_SND_PROC_FS) += opl4_proc.o -snd-opl4-synth-objs := opl4_seq.o opl4_synth.o yrw801.o +snd-opl4-synth-y := opl4_seq.o opl4_synth.o yrw801.o obj-$(CONFIG_SND_OPL4_LIB) += snd-opl4-lib.o obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-opl4-synth.o diff --git a/sound/drivers/pcsp/Makefile b/sound/drivers/pcsp/Makefile index 77dc0ee1b5984..309c094972612 100644 --- a/sound/drivers/pcsp/Makefile +++ b/sound/drivers/pcsp/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-pcsp-objs := pcsp.o pcsp_lib.o pcsp_mixer.o pcsp_input.o +snd-pcsp-y := pcsp.o pcsp_lib.o pcsp_mixer.o pcsp_input.o obj-$(CONFIG_SND_PCSP) += snd-pcsp.o diff --git a/sound/drivers/vx/Makefile b/sound/drivers/vx/Makefile index d9f9ac6703787..ae1b3e09283f9 100644 --- a/sound/drivers/vx/Makefile +++ b/sound/drivers/vx/Makefile @@ -4,6 +4,6 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-vx-lib-objs := vx_core.o vx_hwdep.o vx_pcm.o vx_mixer.o vx_cmd.o vx_uer.o +snd-vx-lib-y := vx_core.o vx_hwdep.o vx_pcm.o vx_mixer.o vx_cmd.o vx_uer.o obj-$(CONFIG_SND_VX_LIB) += snd-vx-lib.o From fdd51b3e73e906aac056f2c337710185607d43d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:07 +0200 Subject: [PATCH 557/603] ALSA: firewire: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Cc: Takashi Sakamoto Signed-off-by: Takashi Iwai Reviewed-by: Takashi Sakamoto Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-8-tiwai@suse.de --- sound/firewire/Makefile | 4 ++-- sound/firewire/bebob/Makefile | 2 +- sound/firewire/dice/Makefile | 2 +- sound/firewire/digi00x/Makefile | 2 +- sound/firewire/fireface/Makefile | 2 +- sound/firewire/fireworks/Makefile | 2 +- sound/firewire/motu/Makefile | 2 +- sound/firewire/oxfw/Makefile | 2 +- sound/firewire/tascam/Makefile | 2 +- 9 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 44a7b510b75bf..45018a5c224fe 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -2,9 +2,9 @@ # To find a header included by define_trace.h. CFLAGS_amdtp-stream.o := -I$(src) -snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ +snd-firewire-lib-y := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp-stream.o amdtp-am824.o -snd-isight-objs := isight.o +snd-isight-y := isight.o obj-$(CONFIG_SND_FIREWIRE_LIB) += snd-firewire-lib.o obj-$(CONFIG_SND_DICE) += dice/ diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile index 14bc84c51ef58..b913e805bd7a0 100644 --- a/sound/firewire/bebob/Makefile +++ b/sound/firewire/bebob/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0 -snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \ +snd-bebob-y := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \ bebob_pcm.o bebob_hwdep.o bebob_terratec.o \ bebob_yamaha_terratec.o bebob_focusrite.o bebob_maudio.o \ bebob.o diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index bac8712f9014d..36e25a3cf3c63 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ +snd-dice-y := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o \ dice-harman.o dice-focusrite.o dice-weiss.o diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile index 8add0cd9af3a5..6dc18bd2e1861 100644 --- a/sound/firewire/digi00x/Makefile +++ b/sound/firewire/digi00x/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \ +snd-firewire-digi00x-y := amdtp-dot.o digi00x-stream.o digi00x-proc.o \ digi00x-pcm.o digi00x-hwdep.o \ digi00x-transaction.o digi00x-midi.o digi00x.o obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o diff --git a/sound/firewire/fireface/Makefile b/sound/firewire/fireface/Makefile index 3aef221ce4b02..b397d95877a0d 100644 --- a/sound/firewire/fireface/Makefile +++ b/sound/firewire/fireface/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-fireface-objs := ff.o ff-transaction.o ff-midi.o ff-proc.o amdtp-ff.o \ +snd-fireface-y := ff.o ff-transaction.o ff-midi.o ff-proc.o amdtp-ff.o \ ff-stream.o ff-pcm.o ff-hwdep.o ff-protocol-former.o \ ff-protocol-latter.o obj-$(CONFIG_SND_FIREFACE) += snd-fireface.o diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile index 3386121b2a04f..baaf3066c9b15 100644 --- a/sound/firewire/fireworks/Makefile +++ b/sound/firewire/fireworks/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \ +snd-fireworks-y := fireworks_transaction.o fireworks_command.o \ fireworks_stream.o fireworks_proc.o fireworks_midi.o \ fireworks_pcm.o fireworks_hwdep.o fireworks.o obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o diff --git a/sound/firewire/motu/Makefile b/sound/firewire/motu/Makefile index 3bef2a0b1e2ee..df0fe886dbc06 100644 --- a/sound/firewire/motu/Makefile +++ b/sound/firewire/motu/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 CFLAGS_amdtp-motu.o := -I$(src) -snd-firewire-motu-objs := motu.o amdtp-motu.o motu-transaction.o motu-stream.o \ +snd-firewire-motu-y := motu.o amdtp-motu.o motu-transaction.o motu-stream.o \ motu-proc.o motu-pcm.o motu-midi.o motu-hwdep.o \ motu-protocol-v2.o motu-protocol-v3.o \ motu-protocol-v1.o motu-register-dsp-message-parser.o \ diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile index 669d1e8238df9..9ac8893a926f7 100644 --- a/sound/firewire/oxfw/Makefile +++ b/sound/firewire/oxfw/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-pcm.o oxfw-proc.o \ +snd-oxfw-y := oxfw-command.o oxfw-stream.o oxfw-pcm.o oxfw-proc.o \ oxfw-midi.o oxfw-hwdep.o oxfw-spkr.o oxfw-scs1x.o oxfw.o obj-$(CONFIG_SND_OXFW) += snd-oxfw.o diff --git a/sound/firewire/tascam/Makefile b/sound/firewire/tascam/Makefile index a1d21f244d64f..43fed14cf172c 100644 --- a/sound/firewire/tascam/Makefile +++ b/sound/firewire/tascam/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-firewire-tascam-objs := tascam-proc.o amdtp-tascam.o tascam-stream.o \ +snd-firewire-tascam-y := tascam-proc.o amdtp-tascam.o tascam-stream.o \ tascam-pcm.o tascam-hwdep.o tascam-transaction.o \ tascam-midi.o tascam.o obj-$(CONFIG_SND_FIREWIRE_TASCAM) += snd-firewire-tascam.o From c02ce1735b150cf7c3b43790b48e23dcd17c0d46 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:08 +0200 Subject: [PATCH 558/603] ALSA: aoa: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-9-tiwai@suse.de --- sound/aoa/codecs/Makefile | 6 +++--- sound/aoa/core/Makefile | 2 +- sound/aoa/fabrics/Makefile | 2 +- sound/aoa/soundbus/Makefile | 2 +- sound/aoa/soundbus/i2sbus/Makefile | 2 +- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile index 95f4c3849d550..8feedc771bd9f 100644 --- a/sound/aoa/codecs/Makefile +++ b/sound/aoa/codecs/Makefile @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0 -snd-aoa-codec-onyx-objs := onyx.o -snd-aoa-codec-tas-objs := tas.o -snd-aoa-codec-toonie-objs := toonie.o +snd-aoa-codec-onyx-y := onyx.o +snd-aoa-codec-tas-y := tas.o +snd-aoa-codec-toonie-y := toonie.o obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o diff --git a/sound/aoa/core/Makefile b/sound/aoa/core/Makefile index 056d69683b1e9..f586c340fe127 100644 --- a/sound/aoa/core/Makefile +++ b/sound/aoa/core/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 obj-$(CONFIG_SND_AOA) += snd-aoa.o -snd-aoa-objs := core.o \ +snd-aoa-y := core.o \ alsa.o \ gpio-pmf.o \ gpio-feature.o diff --git a/sound/aoa/fabrics/Makefile b/sound/aoa/fabrics/Makefile index 3f1d55f3f1fc5..2c3bee6cfa2c2 100644 --- a/sound/aoa/fabrics/Makefile +++ b/sound/aoa/fabrics/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-aoa-fabric-layout-objs += layout.o +snd-aoa-fabric-layout-y += layout.o obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o diff --git a/sound/aoa/soundbus/Makefile b/sound/aoa/soundbus/Makefile index e0b61cf5518e3..a10b102daf81b 100644 --- a/sound/aoa/soundbus/Makefile +++ b/sound/aoa/soundbus/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only obj-$(CONFIG_SND_AOA_SOUNDBUS) += snd-aoa-soundbus.o -snd-aoa-soundbus-objs := core.o sysfs.o +snd-aoa-soundbus-y := core.o sysfs.o obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += i2sbus/ diff --git a/sound/aoa/soundbus/i2sbus/Makefile b/sound/aoa/soundbus/i2sbus/Makefile index 1b38c87fef094..1ddaa0e17d676 100644 --- a/sound/aoa/soundbus/i2sbus/Makefile +++ b/sound/aoa/soundbus/i2sbus/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0-only obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o -snd-aoa-i2sbus-objs := core.o pcm.o control.o +snd-aoa-i2sbus-y := core.o pcm.o control.o From b7df4cc3a088a8ce6973c96731bc792dbf54ce28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 May 2024 15:55:09 +0200 Subject: [PATCH 559/603] ALSA: misc: Use *-y instead of *-objs in Makefile *-objs suffix is reserved rather for (user-space) host programs while usually *-y suffix is used for kernel drivers (although *-objs works for that purpose for now). Let's correct the old usages of *-objs in Makefiles. Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240507135513.14919-10-tiwai@suse.de --- sound/arm/Makefile | 4 ++-- sound/atmel/Makefile | 2 +- sound/i2c/Makefile | 6 +++--- sound/i2c/other/Makefile | 10 +++++----- sound/mips/Makefile | 4 ++-- sound/parisc/Makefile | 2 +- sound/pcmcia/pdaudiocf/Makefile | 2 +- sound/pcmcia/vx/Makefile | 2 +- sound/ppc/Makefile | 2 +- sound/sh/Makefile | 4 ++-- sound/sparc/Makefile | 6 +++--- sound/spi/Makefile | 2 +- sound/synth/Makefile | 2 +- sound/synth/emux/Makefile | 2 +- sound/virtio/Makefile | 2 +- sound/x86/Makefile | 2 +- sound/xen/Makefile | 2 +- 17 files changed, 28 insertions(+), 28 deletions(-) diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 34c7694898779..899edb4bb278d 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -4,11 +4,11 @@ # obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o -snd-aaci-objs := aaci.o +snd-aaci-y := aaci.o obj-$(CONFIG_SND_PXA2XX_LIB) += snd-pxa2xx-lib.o snd-pxa2xx-lib-y := pxa2xx-pcm-lib.o snd-pxa2xx-lib-$(CONFIG_SND_PXA2XX_LIB_AC97) += pxa2xx-ac97-lib.o obj-$(CONFIG_SND_PXA2XX_AC97) += snd-pxa2xx-ac97.o -snd-pxa2xx-ac97-objs := pxa2xx-ac97.o +snd-pxa2xx-ac97-y := pxa2xx-ac97.o diff --git a/sound/atmel/Makefile b/sound/atmel/Makefile index 57bc6f65be193..a8917d1854c7d 100644 --- a/sound/atmel/Makefile +++ b/sound/atmel/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-atmel-ac97c-objs := ac97c.o +snd-atmel-ac97c-y := ac97c.o obj-$(CONFIG_SND_ATMEL_AC97C) += snd-atmel-ac97c.o diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile index 09978855e08e0..c827f9f70a339 100644 --- a/sound/i2c/Makefile +++ b/sound/i2c/Makefile @@ -4,9 +4,9 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-i2c-objs := i2c.o -snd-cs8427-objs := cs8427.o -snd-tea6330t-objs := tea6330t.o +snd-i2c-y := i2c.o +snd-cs8427-y := cs8427.o +snd-tea6330t-y := tea6330t.o obj-$(CONFIG_SND) += other/ diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile index 1a4ce12361466..0a2c0d147ab88 100644 --- a/sound/i2c/other/Makefile +++ b/sound/i2c/other/Makefile @@ -4,11 +4,11 @@ # Copyright (c) 2003 by Jaroslav Kysela # -snd-ak4114-objs := ak4114.o -snd-ak4117-objs := ak4117.o -snd-ak4113-objs := ak4113.o -snd-ak4xxx-adda-objs := ak4xxx-adda.o -snd-pt2258-objs := pt2258.o +snd-ak4114-y := ak4114.o +snd-ak4117-y := ak4117.o +snd-ak4113-y := ak4113.o +snd-ak4xxx-adda-y := ak4xxx-adda.o +snd-pt2258-y := pt2258.o # Module Dependency obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 7c86268b2bf3f..bfbf3bda487ba 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -3,8 +3,8 @@ # Makefile for ALSA # -snd-sgi-o2-objs := sgio2audio.o ad1843.o -snd-sgi-hal2-objs := hal2.o +snd-sgi-o2-y := sgio2audio.o ad1843.o +snd-sgi-hal2-y := hal2.o # Toplevel Module Dependency obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o diff --git a/sound/parisc/Makefile b/sound/parisc/Makefile index 10891c3b7d914..84c71490fb72c 100644 --- a/sound/parisc/Makefile +++ b/sound/parisc/Makefile @@ -3,7 +3,7 @@ # Makefile for ALSA # -snd-harmony-objs := harmony.o +snd-harmony-y := harmony.o # Toplevel Module Dependency obj-$(CONFIG_SND_HARMONY) += snd-harmony.o diff --git a/sound/pcmcia/pdaudiocf/Makefile b/sound/pcmcia/pdaudiocf/Makefile index ea0d67576df9e..34a288c1eebdb 100644 --- a/sound/pcmcia/pdaudiocf/Makefile +++ b/sound/pcmcia/pdaudiocf/Makefile @@ -4,6 +4,6 @@ # Copyright (c) 2004 by Jaroslav Kysela # -snd-pdaudiocf-objs := pdaudiocf.o pdaudiocf_core.o pdaudiocf_irq.o pdaudiocf_pcm.o +snd-pdaudiocf-y := pdaudiocf.o pdaudiocf_core.o pdaudiocf_irq.o pdaudiocf_pcm.o obj-$(CONFIG_SND_PDAUDIOCF) += snd-pdaudiocf.o diff --git a/sound/pcmcia/vx/Makefile b/sound/pcmcia/vx/Makefile index b25006e4d25a8..abd1875449461 100644 --- a/sound/pcmcia/vx/Makefile +++ b/sound/pcmcia/vx/Makefile @@ -4,6 +4,6 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-vxpocket-objs := vxpocket.o vxp_ops.o vxp_mixer.o +snd-vxpocket-y := vxpocket.o vxp_ops.o vxp_mixer.o obj-$(CONFIG_SND_VXPOCKET) += snd-vxpocket.o diff --git a/sound/ppc/Makefile b/sound/ppc/Makefile index 0188ce3e30b8f..655bcffba843c 100644 --- a/sound/ppc/Makefile +++ b/sound/ppc/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-powermac-objs := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o +snd-powermac-y := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o # Toplevel Module Dependency obj-$(CONFIG_SND_POWERMAC) += snd-powermac.o diff --git a/sound/sh/Makefile b/sound/sh/Makefile index c0bbc500c17c7..6871dece28a55 100644 --- a/sound/sh/Makefile +++ b/sound/sh/Makefile @@ -3,8 +3,8 @@ # Makefile for ALSA # -snd-aica-objs := aica.o -snd-sh_dac_audio-objs := sh_dac_audio.o +snd-aica-y := aica.o +snd-sh_dac_audio-y := sh_dac_audio.o # Toplevel Module Dependency obj-$(CONFIG_SND_AICA) += snd-aica.o diff --git a/sound/sparc/Makefile b/sound/sparc/Makefile index e1f596571d7ff..0a03123933c66 100644 --- a/sound/sparc/Makefile +++ b/sound/sparc/Makefile @@ -4,9 +4,9 @@ # Copyright (c) 2002 by David S. Miller # -snd-sun-amd7930-objs := amd7930.o -snd-sun-cs4231-objs := cs4231.o -snd-sun-dbri-objs := dbri.o +snd-sun-amd7930-y := amd7930.o +snd-sun-cs4231-y := cs4231.o +snd-sun-dbri-y := dbri.o obj-$(CONFIG_SND_SUN_AMD7930) += snd-sun-amd7930.o obj-$(CONFIG_SND_SUN_CS4231) += snd-sun-cs4231.o diff --git a/sound/spi/Makefile b/sound/spi/Makefile index a3834919b0f6e..d6a198a449174 100644 --- a/sound/spi/Makefile +++ b/sound/spi/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 # Makefile for SPI drivers -snd-at73c213-objs := at73c213.o +snd-at73c213-y := at73c213.o obj-$(CONFIG_SND_AT73C213) += snd-at73c213.o diff --git a/sound/synth/Makefile b/sound/synth/Makefile index b9f71d5dbc8c0..369f3be2cd176 100644 --- a/sound/synth/Makefile +++ b/sound/synth/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-util-mem-objs := util_mem.o +snd-util-mem-y := util_mem.o # Toplevel Module Dependency obj-$(CONFIG_SND_EMU10K1) += snd-util-mem.o diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile index ed28c81ac12e1..59357da6d1ef8 100644 --- a/sound/synth/emux/Makefile +++ b/sound/synth/emux/Makefile @@ -4,7 +4,7 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-emux-synth-objs := emux.o emux_synth.o emux_seq.o emux_nrpn.o \ +snd-emux-synth-y := emux.o emux_synth.o emux_seq.o emux_nrpn.o \ emux_effect.o emux_hwdep.o soundfont.o snd-emux-synth-$(CONFIG_SND_PROC_FS) += emux_proc.o ifneq ($(CONFIG_SND_SEQUENCER_OSS),) diff --git a/sound/virtio/Makefile b/sound/virtio/Makefile index a839f8c8b5e64..3ca41b3c8b95b 100644 --- a/sound/virtio/Makefile +++ b/sound/virtio/Makefile @@ -2,7 +2,7 @@ obj-$(CONFIG_SND_VIRTIO) += virtio_snd.o -virtio_snd-objs := \ +virtio_snd-y := \ virtio_card.o \ virtio_chmap.o \ virtio_ctl_msg.o \ diff --git a/sound/x86/Makefile b/sound/x86/Makefile index 6b5ffb329d472..44d2a339615d9 100644 --- a/sound/x86/Makefile +++ b/sound/x86/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-hdmi-lpe-audio-objs += \ +snd-hdmi-lpe-audio-y += \ intel_hdmi_audio.o obj-$(CONFIG_HDMI_LPE_AUDIO) += snd-hdmi-lpe-audio.o diff --git a/sound/xen/Makefile b/sound/xen/Makefile index 24031775b7153..5dab1616eac09 100644 --- a/sound/xen/Makefile +++ b/sound/xen/Makefile @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 OR MIT -snd_xen_front-objs := xen_snd_front.o \ +snd_xen_front-y := xen_snd_front.o \ xen_snd_front_cfg.o \ xen_snd_front_evtchnl.o \ xen_snd_front_alsa.o From 172811e3a557d8681a5e2d0f871dc04a2d17eb13 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 8 May 2024 10:56:27 +0100 Subject: [PATCH 560/603] ALSA: hda/cs_dsp_ctl: Use private_free for control cleanup Use the control private_free callback to free the associated data block. This ensures that the memory won't leak, whatever way the control gets destroyed. The original implementation didn't actually remove the ALSA controls in hda_cs_dsp_control_remove(). It only freed the internal tracking structure. This meant it was possible to remove/unload the amp driver while leaving its ALSA controls still present in the soundcard. Obviously attempting to access them could cause segfaults or at least dereferencing stale pointers. Signed-off-by: Richard Fitzgerald Fixes: 3233b978af23 ("ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls") Link: https://lore.kernel.org/r/20240508095627.44476-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_cs_dsp_ctl.c | 47 ++++++++++++++++++++++------------ 1 file changed, 31 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 463ca06036bfe..9db45d7c17e5f 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -8,6 +8,7 @@ #include #include +#include #include #include #include "hda_cs_dsp_ctl.h" @@ -97,11 +98,23 @@ static unsigned int wmfw_convert_flags(unsigned int in) return out; } -static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char *name) +static void hda_cs_dsp_free_kcontrol(struct snd_kcontrol *kctl) { + struct hda_cs_dsp_coeff_ctl *ctl = (struct hda_cs_dsp_coeff_ctl *)snd_kcontrol_chip(kctl); struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; + + /* NULL priv to prevent a double-free in hda_cs_dsp_control_remove() */ + cs_ctl->priv = NULL; + kfree(ctl); +} + +static void hda_cs_dsp_add_kcontrol(struct cs_dsp_coeff_ctl *cs_ctl, + const struct hda_cs_dsp_ctl_info *info, + const char *name) +{ struct snd_kcontrol_new kcontrol = {0}; struct snd_kcontrol *kctl; + struct hda_cs_dsp_coeff_ctl *ctl __free(kfree) = NULL; int ret = 0; if (cs_ctl->len > ADSP_MAX_STD_CTRL_SIZE) { @@ -110,6 +123,13 @@ static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char return; } + ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); + if (!ctl) + return; + + ctl->cs_ctl = cs_ctl; + ctl->card = info->card; + kcontrol.name = name; kcontrol.info = hda_cs_dsp_coeff_info; kcontrol.iface = SNDRV_CTL_ELEM_IFACE_MIXER; @@ -117,20 +137,22 @@ static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char kcontrol.get = hda_cs_dsp_coeff_get; kcontrol.put = hda_cs_dsp_coeff_put; - /* Save ctl inside private_data, ctl is owned by cs_dsp, - * and will be freed when cs_dsp removes the control */ kctl = snd_ctl_new1(&kcontrol, (void *)ctl); if (!kctl) return; - ret = snd_ctl_add(ctl->card, kctl); + kctl->private_free = hda_cs_dsp_free_kcontrol; + ctl->kctl = kctl; + + /* snd_ctl_add() calls our private_free on error, which will kfree(ctl) */ + cs_ctl->priv = no_free_ptr(ctl); + ret = snd_ctl_add(info->card, kctl); if (ret) { dev_err(cs_ctl->dsp->dev, "Failed to add KControl %s = %d\n", kcontrol.name, ret); return; } dev_dbg(cs_ctl->dsp->dev, "Added KControl: %s\n", kcontrol.name); - ctl->kctl = kctl; } static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, @@ -138,7 +160,6 @@ static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, { struct cs_dsp *cs_dsp = cs_ctl->dsp; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - struct hda_cs_dsp_coeff_ctl *ctl; const char *region_name; int ret; @@ -163,15 +184,7 @@ static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, " %.*s", cs_ctl->subname_len - skip, cs_ctl->subname + skip); } - ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); - if (!ctl) - return; - - ctl->cs_ctl = cs_ctl; - ctl->card = info->card; - cs_ctl->priv = ctl; - - hda_cs_dsp_add_kcontrol(ctl, name); + hda_cs_dsp_add_kcontrol(cs_ctl, info, name); } void hda_cs_dsp_add_controls(struct cs_dsp *dsp, const struct hda_cs_dsp_ctl_info *info) @@ -203,7 +216,9 @@ void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) { struct hda_cs_dsp_coeff_ctl *ctl = cs_ctl->priv; - kfree(ctl); + /* ctl and kctl may already have been removed by ALSA private_free */ + if (ctl && ctl->kctl) + snd_ctl_remove(ctl->card, ctl->kctl); } EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS); From ec6f32bc924d1c00cbcd5672510758f7088f2513 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 8 May 2024 11:03:47 +0100 Subject: [PATCH 561/603] ALSA: hda: hda_component: Initialize shared data during bind callback Move the initialization of the shared struct hda_component array into hda_component_manager_bind(). The purpose of the manager bind() callback is to allow it to perform initialization before binding in the component drivers. This is the correct place to initialize the shared data. The original implementation initialized the shared data in hda_component_manager_init(). This is only done once during probe() of the manager driver. So if the component binding was unbound and then rebound, the shared data would not be re-initialized. Signed-off-by: Richard Fitzgerald Fixes: fd895a74dc1d ("ALSA: hda: realtek: Move hda_component implementation to module") Link: https://lore.kernel.org/r/20240508100347.47283-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_component.c | 16 +++++++++++++++- sound/pci/hda/hda_component.h | 7 ++----- sound/pci/hda/patch_realtek.c | 2 +- 3 files changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_component.c b/sound/pci/hda/hda_component.c index cd299d7d84baf..d02589014a3fa 100644 --- a/sound/pci/hda/hda_component.c +++ b/sound/pci/hda/hda_component.c @@ -123,6 +123,21 @@ static int hda_comp_match_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +int hda_component_manager_bind(struct hda_codec *cdc, + struct hda_component *comps, int count) +{ + int i; + + /* Init shared data */ + for (i = 0; i < count; ++i) { + memset(&comps[i], 0, sizeof(comps[i])); + comps[i].codec = cdc; + } + + return component_bind_all(hda_codec_dev(cdc), comps); +} +EXPORT_SYMBOL_NS_GPL(hda_component_manager_bind, SND_HDA_SCODEC_COMPONENT); + int hda_component_manager_init(struct hda_codec *cdc, struct hda_component *comps, int count, const char *bus, const char *hid, @@ -143,7 +158,6 @@ int hda_component_manager_init(struct hda_codec *cdc, sm->hid = hid; sm->match_str = match_str; sm->index = i; - comps[i].codec = cdc; component_match_add(dev, &match, hda_comp_match_dev_name, sm); } diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index c80a66691b5d8..c70b3de68ab20 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -75,11 +75,8 @@ int hda_component_manager_init(struct hda_codec *cdc, void hda_component_manager_free(struct hda_codec *cdc, const struct component_master_ops *ops); -static inline int hda_component_manager_bind(struct hda_codec *cdc, - struct hda_component *comps) -{ - return component_bind_all(hda_codec_dev(cdc), comps); -} +int hda_component_manager_bind(struct hda_codec *cdc, + struct hda_component *comps, int count); static inline void hda_component_manager_unbind(struct hda_codec *cdc, struct hda_component *comps) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c157158f1cb3..8c39bbdc6c5e2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6785,7 +6785,7 @@ static int comp_bind(struct device *dev) struct alc_spec *spec = cdc->spec; int ret; - ret = hda_component_manager_bind(cdc, spec->comps); + ret = hda_component_manager_bind(cdc, spec->comps, ARRAY_SIZE(spec->comps)); if (ret) return ret; From d344873c4cbde249b7152d36a273bcc45864001e Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 8 May 2024 11:08:11 +0100 Subject: [PATCH 562/603] ALSA: hda: cs35l56: Fix lifetime of cs_dsp instance The cs_dsp instance is initialized in the driver probe() so it should be freed in the driver remove(). Also fix a missing call to cs_dsp_remove() in the error path of cs35l56_hda_common_probe(). The call to cs_dsp_remove() was being done in the component unbind callback cs35l56_hda_unbind(). This meant that if the driver was unbound and then re-bound it would be using an uninitialized cs_dsp instance. It is best to initialize the cs_dsp instance in probe() so that it can return an error if it fails. The component binding API doesn't have any error handling so there's no way to handle a failure if cs_dsp was initialized in the bind. Signed-off-by: Richard Fitzgerald Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Link: https://lore.kernel.org/r/20240508100811.49514-1-rf@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l56_hda.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 558c1f38fe971..11b0570ff56d4 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -732,8 +732,6 @@ static void cs35l56_hda_unbind(struct device *dev, struct device *master, void * if (cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); - cs_dsp_remove(&cs35l56->cs_dsp); - if (comps[cs35l56->index].dev == dev) memset(&comps[cs35l56->index], 0, sizeof(*comps)); @@ -1035,7 +1033,7 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) ARRAY_SIZE(cs35l56_hda_dai_config)); ret = cs35l56_force_sync_asp1_registers_from_cache(&cs35l56->base); if (ret) - goto err; + goto dsp_err; /* * By default only enable one ASP1TXn, where n=amplifier index, @@ -1061,6 +1059,8 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) pm_err: pm_runtime_disable(cs35l56->base.dev); +dsp_err: + cs_dsp_remove(&cs35l56->cs_dsp); err: gpiod_set_value_cansleep(cs35l56->base.reset_gpio, 0); @@ -1078,6 +1078,8 @@ void cs35l56_hda_remove(struct device *dev) component_del(cs35l56->base.dev, &cs35l56_hda_comp_ops); + cs_dsp_remove(&cs35l56->cs_dsp); + kfree(cs35l56->system_name); pm_runtime_put_noidle(cs35l56->base.dev); From c6f597bc598a8c3456fa017eb0c55e7a41bfce2c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 9 May 2024 00:33:11 +0000 Subject: [PATCH 563/603] ASoC: audio-graph-card2: remove unneeded of_node_get() "lnk" is used as "port0", and it will be used to get "port1" by of_get_next_child(ports, lnk). It will call of_node_put() inside. This function is calling of_node_get() to make up for it, but it doesn't call paired of_node_put(port0) when it quit function. This of_node_get() itself is not needed, let's remove it. Reported-by: Dan Carpenter Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87jzk3akoo.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 62606e20be9a3..c4d52fadebb62 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1203,8 +1203,6 @@ static int graph_count_c2c(struct simple_util_priv *priv, struct device_node *codec0 = of_graph_get_remote_port(ep0); struct device_node *codec1 = of_graph_get_remote_port(ep1); - of_node_get(lnk); - /* * codec2codec { * ports { From 2d95e8d48499fbb14ee7c1b93c73938bb0751275 Mon Sep 17 00:00:00 2001 From: Steven 'Steve' Kendall Date: Wed, 8 May 2024 22:29:25 +0000 Subject: [PATCH 564/603] ALSA: usb-audio: Add name for HP Engage Go dock This dock previously presented as 'generic USB Audio'. UCM may now be applied specific to this dock. Signed-off-by: Steven 'Steve' Kendall Link: https://lore.kernel.org/r/20240508-usb-audio-hp-engage-go-dock-v1-1-9e2984d49e63@chromium.org Signed-off-by: Takashi Iwai --- sound/usb/card.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 1b2edc0fd2e99..bdb04fa37a71d 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -425,6 +425,10 @@ static const struct usb_audio_device_name usb_audio_names[] = { DEVICE_NAME(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), + /* Dock/Stand for HP Engage Go */ + PROFILE_NAME(0x103c, 0x830a, "HP", "HP Engage Go Dock", + "HP-Engage-Go-Dock"), + /* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */ DEVICE_NAME(0x103d, 0x0100, "Stanton", "ScratchAmp"), DEVICE_NAME(0x103d, 0x0101, "Stanton", "ScratchAmp"), From 92641cc5b80499e2af21c965b38b64b4dcabe959 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:11:20 +0200 Subject: [PATCH 565/603] ALSA: ac97: legacy: Add missing module description Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Reported-by: Andy Shevchenko Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091128.25274-2-tiwai@suse.de --- sound/ac97_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 7ea274c559005..1484fc178fa45 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -95,4 +95,5 @@ module_exit(ac97_bus_exit); EXPORT_SYMBOL(ac97_bus_type); +MODULE_DESCRIPTION("Legacy AC97 bus interface"); MODULE_LICENSE("GPL"); From ddefb24b3effa6dbca148d176cdd5e24e8e3cf3b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:11:21 +0200 Subject: [PATCH 566/603] ALSA: ac97: bus: Add missing module description Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Reported-by: Andy Shevchenko Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091128.25274-3-tiwai@suse.de --- sound/ac97/bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 5e46b972a3daf..40e88d79c483b 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -551,5 +551,6 @@ static void __exit ac97_bus_exit(void) } module_exit(ac97_bus_exit); +MODULE_DESCRIPTION("AC97 bus interface"); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Robert Jarzmik "); From b58a6b1ee60a49249030bb1fb6d0d139bfc5bf0a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:11:22 +0200 Subject: [PATCH 567/603] ALSA: pcm_dmaengine: Add missing module description Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Reported-by: Andy Shevchenko Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091128.25274-4-tiwai@suse.de --- sound/core/pcm_dmaengine.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 494ec0c207fad..12aa1cef11a13 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -470,4 +470,5 @@ int snd_dmaengine_pcm_refine_runtime_hwparams( } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_refine_runtime_hwparams); +MODULE_DESCRIPTION("PCM dmaengine helper APIs"); MODULE_LICENSE("GPL"); From 568d0ae72a281d5a08ac1a092e3adb85eff7aee3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:11:23 +0200 Subject: [PATCH 568/603] ALSA: kunit: Add missing module descriptions Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Reported-by: Andy Shevchenko Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091128.25274-5-tiwai@suse.de --- sound/core/sound_kunit.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c index 165bd0f5338af..bfed1a25fc8f7 100644 --- a/sound/core/sound_kunit.c +++ b/sound/core/sound_kunit.c @@ -308,5 +308,6 @@ static struct kunit_suite sound_utils_suite = { }; kunit_test_suite(sound_utils_suite); +MODULE_DESCRIPTION("Sound core KUnit test"); MODULE_AUTHOR("Ivan Orlov"); MODULE_LICENSE("GPL"); From e3e21cefa23b95093577903c39838cf49b1c429a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:11:24 +0200 Subject: [PATCH 569/603] ALSA: pcmtest: Add missing module descriptions Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Reported-by: Andy Shevchenko Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091128.25274-6-tiwai@suse.de --- sound/drivers/pcmtest.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index b8bff5522bce2..21cefaf5419aa 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -772,6 +772,7 @@ static void __exit mod_exit(void) platform_device_unregister(&pcmtst_pdev); } +MODULE_DESCRIPTION("Virtual ALSA driver for PCM testing/fuzzing"); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Ivan Orlov"); module_init(mod_init); From cdd08e4639a0290da9ace88a969781c694238e9f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:11:25 +0200 Subject: [PATCH 570/603] ALSA: hda: cirrus_scodec_test: Add missing module descriptions Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Reported-by: Andy Shevchenko Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091128.25274-7-tiwai@suse.de --- sound/pci/hda/cirrus_scodec_test.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/cirrus_scodec_test.c b/sound/pci/hda/cirrus_scodec_test.c index 8ae373676bd1a..e925ebe21ccba 100644 --- a/sound/pci/hda/cirrus_scodec_test.c +++ b/sound/pci/hda/cirrus_scodec_test.c @@ -366,5 +366,6 @@ static struct kunit_suite cirrus_scodec_test_suite = { kunit_test_suite(cirrus_scodec_test_suite); MODULE_IMPORT_NS(SND_HDA_CIRRUS_SCODEC); +MODULE_DESCRIPTION("KUnit test for the Cirrus side-codec library"); MODULE_AUTHOR("Richard Fitzgerald "); MODULE_LICENSE("GPL"); From 914728fbf2af4d96a5ba896256e696d78c474a4f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 May 2024 11:11:26 +0200 Subject: [PATCH 571/603] sound: oss: dmasound: Add missing module descriptions Now that make W=1 starts complaining the lack of MODULE_DESCRIPTION(), let's add the missing information. Reported-by: Andy Shevchenko Closes: https://lore.kernel.org/r/ZjpQm-hxLQtpgkUx@smile.fi.intel.com Signed-off-by: Takashi Iwai Reviewed-by: Andy Shevchenko Reviewed-by: Dragan Simic Link: https://lore.kernel.org/r/20240508091128.25274-8-tiwai@suse.de --- sound/oss/dmasound/dmasound_atari.c | 2 ++ sound/oss/dmasound/dmasound_paula.c | 1 + 2 files changed, 3 insertions(+) diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 81c6a98307273..6188469de8af1 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -1618,4 +1618,6 @@ static void __exit dmasound_atari_cleanup(void) module_init(dmasound_atari_init); module_exit(dmasound_atari_cleanup); + +MODULE_DESCRIPTION("Atari TT and Falcon DMA Sound Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 3a593da09280d..b8fad12f9e5f5 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -740,5 +740,6 @@ static struct platform_driver amiga_audio_driver __refdata = { module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe); +MODULE_DESCRIPTION("Amiga Paula DMA Sound Driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:amiga-audio"); From a85ed162f0efcfdd664954414a05d1d560cc95dc Mon Sep 17 00:00:00 2001 From: Hsin-Te Yuan Date: Thu, 9 May 2024 07:31:29 +0000 Subject: [PATCH 572/603] ASoC: mediatek: mt8192: fix register configuration for tdm For DSP_A, data is a BCK cycle behind LRCK trigger edge. For DSP_B, this delay doesn't exist. Fix the delay configuration to match the standard. Fixes: 52fcd65414abfc ("ASoC: mediatek: mt8192: support tdm in platform driver") Signed-off-by: Hsin-Te Yuan Reviewed-by: AngeloGioacchino Del Regno Reviewed-by: Chen-Yu Tsai Link: https://lore.kernel.org/r/20240509-8192-tdm-v1-1-530b54645763@chromium.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c index 9ce06821c7d0f..49440db370af0 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-tdm.c @@ -566,10 +566,10 @@ static int mtk_dai_tdm_hw_params(struct snd_pcm_substream *substream, tdm_con |= 1 << DELAY_DATA_SFT; tdm_con |= get_tdm_lrck_width(format) << LRCK_TDM_WIDTH_SFT; } else if (tdm_priv->tdm_out_mode == TDM_OUT_DSP_A) { - tdm_con |= 0 << DELAY_DATA_SFT; + tdm_con |= 1 << DELAY_DATA_SFT; tdm_con |= 0 << LRCK_TDM_WIDTH_SFT; } else if (tdm_priv->tdm_out_mode == TDM_OUT_DSP_B) { - tdm_con |= 1 << DELAY_DATA_SFT; + tdm_con |= 0 << DELAY_DATA_SFT; tdm_con |= 0 << LRCK_TDM_WIDTH_SFT; } From 51bebf3460cb7c380ac625e8ca0a0b31d4c6428c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 19 Apr 2024 10:43:06 +0200 Subject: [PATCH 573/603] ALSA: hda: intel-dsp-config: Switch to ACPI NHLT MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Now that NHLT support in ACPI framework was introduced, migrate intel-dsp-config driver to new API. Signed-off-by: Amadeusz Sławiński Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20240419084307.2718881-2-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/hda/Kconfig | 1 + sound/hda/intel-dsp-config.c | 16 +++++++++------- 2 files changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 741179ccbd4e4..e2ac247fc1d40 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -42,6 +42,7 @@ config SND_INTEL_NHLT config SND_INTEL_DSP_CONFIG tristate + select ACPI_NHLT if ACPI select SND_INTEL_NHLT if ACPI select SND_INTEL_SOUNDWIRE_ACPI if ACPI # this config should be selected only for Intel DSP platforms. diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 6a384b922e4fa..ea050805c20f5 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -13,6 +13,8 @@ #include #include +#include + static int dsp_driver; module_param(dsp_driver, int, 0444); @@ -570,15 +572,15 @@ static const struct config_entry *snd_intel_dsp_find_config static int snd_intel_dsp_check_dmic(struct pci_dev *pci) { - struct nhlt_acpi_table *nhlt; int ret = 0; - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_has_endpoint_type(nhlt, NHLT_LINK_DMIC)) - ret = 1; - intel_nhlt_free(nhlt); - } + acpi_nhlt_get_gbl_table(); + + if (acpi_nhlt_find_endpoint(ACPI_NHLT_LINKTYPE_PDM, -1, -1, -1)) + ret = 1; + + acpi_nhlt_put_gbl_table(); + return ret; } From e09f9f529caf43ba94245f61564b966cab912935 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:58 -0500 Subject: [PATCH 574/603] ALSA: hda: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Reviewed-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/patch_hdmi.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 1030a8bfb3cda..766734dc5be27 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -3,7 +3,7 @@ * * Implementation of primary alsa driver code base for Intel HD Audio. * - * Copyright(c) 2004 Intel Corporation. All rights reserved. + * Copyright(c) 2004 Intel Corporation * * Copyright (c) 2004 Takashi Iwai * PeiSen Hou diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2bd652dcb5b34..3500108f6ba37 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -4,7 +4,7 @@ * hda_intel.c - Implementation of primary alsa driver code base * for Intel HD Audio. * - * Copyright(c) 2004 Intel Corporation. All rights reserved. + * Copyright(c) 2004 Intel Corporation * * Copyright (c) 2004 Takashi Iwai * PeiSen Hou diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index db24c8fc7cd0f..707d203ba6527 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3,7 +3,7 @@ * * patch_hdmi.c - routines for HDMI/DisplayPort codecs * - * Copyright(c) 2008-2010 Intel Corporation. All rights reserved. + * Copyright(c) 2008-2010 Intel Corporation * Copyright (c) 2006 ATI Technologies Inc. * Copyright (c) 2008 NVIDIA Corp. All rights reserved. * Copyright (c) 2008 Wei Ni From ea89a742daf4317038fbab6776d36726dd7a1e2a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 3 May 2024 09:03:59 -0500 Subject: [PATCH 575/603] ALSA/ASoC: include: clarify Copyright information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For some reason a number of files included the "All rights reserved" statement. Good old copy-paste made sure this mistake proliferated. Remove the "All rights reserved" in all Intel-copyright to align with internal guidance. Reviewed-by: Cezary Rojewski Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20240503140359.259762-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hda-mlink.h | 2 +- include/sound/soc-acpi-intel-match.h | 2 +- include/sound/soc-acpi.h | 2 +- include/sound/sof.h | 2 +- include/sound/sof/channel_map.h | 2 +- include/sound/sof/control.h | 2 +- include/sound/sof/dai-intel.h | 2 +- include/sound/sof/dai.h | 2 +- include/sound/sof/debug.h | 2 +- include/sound/sof/ext_manifest.h | 2 +- include/sound/sof/ext_manifest4.h | 2 +- include/sound/sof/header.h | 2 +- include/sound/sof/info.h | 2 +- include/sound/sof/ipc4/header.h | 2 +- include/sound/sof/pm.h | 2 +- include/sound/sof/stream.h | 2 +- include/sound/sof/topology.h | 2 +- include/sound/sof/trace.h | 2 +- include/sound/sof/xtensa.h | 2 +- include/trace/events/sof.h | 2 +- include/trace/events/sof_intel.h | 2 +- include/uapi/sound/intel/avs/tokens.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/fw.h | 2 +- include/uapi/sound/sof/header.h | 2 +- include/uapi/sound/sof/tokens.h | 2 +- 26 files changed, 26 insertions(+), 26 deletions(-) diff --git a/include/sound/hda-mlink.h b/include/sound/hda-mlink.h index d849d9b24f13b..9ced94686ce3f 100644 --- a/include/sound/hda-mlink.h +++ b/include/sound/hda-mlink.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2022-2023 Intel Corporation. All rights reserved. + * Copyright(c) 2022-2023 Intel Corporation */ struct hdac_bus; diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 845e7608ac375..4843b57798f69 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 * - * Copyright (C) 2017, Intel Corporation. All rights reserved. + * Copyright (C) 2017, Intel Corporation */ #ifndef __LINUX_SND_SOC_ACPI_INTEL_MATCH_H diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 23d6d6bfb0736..8967d07a27361 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only * - * Copyright (C) 2013-15, Intel Corporation. All rights reserved. + * Copyright (C) 2013-15, Intel Corporation */ #ifndef __LINUX_SND_SOC_ACPI_H diff --git a/include/sound/sof.h b/include/sound/sof.h index 05213bb515a38..cde0587778325 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation * * Author: Liam Girdwood */ diff --git a/include/sound/sof/channel_map.h b/include/sound/sof/channel_map.h index d363f0ca6979a..2a177a1938e93 100644 --- a/include/sound/sof/channel_map.h +++ b/include/sound/sof/channel_map.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2019 Intel Corporation. All rights reserved. + * Copyright(c) 2019 Intel Corporation */ #ifndef __IPC_CHANNEL_MAP_H__ diff --git a/include/sound/sof/control.h b/include/sound/sof/control.h index 983d374fe511c..8d3300dd9d8cc 100644 --- a/include/sound/sof/control.h +++ b/include/sound/sof/control.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_CONTROL_H__ diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h index 5b93b7292f5e1..e4e710b43d2d9 100644 --- a/include/sound/sof/dai-intel.h +++ b/include/sound/sof/dai-intel.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_DAI_INTEL_H__ diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 0764a80c17a9b..36809f7127236 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_DAI_H__ diff --git a/include/sound/sof/debug.h b/include/sound/sof/debug.h index 38693e3fb5149..8b308d7e5eeed 100644 --- a/include/sound/sof/debug.h +++ b/include/sound/sof/debug.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2020 Intel Corporation. All rights reserved. + * Copyright(c) 2020 Intel Corporation * * Author: Karol Trzcinski */ diff --git a/include/sound/sof/ext_manifest.h b/include/sound/sof/ext_manifest.h index 2a7e055584f9f..fc0231d04a941 100644 --- a/include/sound/sof/ext_manifest.h +++ b/include/sound/sof/ext_manifest.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2020 Intel Corporation. All rights reserved. + * Copyright(c) 2020 Intel Corporation */ /* diff --git a/include/sound/sof/ext_manifest4.h b/include/sound/sof/ext_manifest4.h index ec97edcbbfc39..4e1d379d96ad0 100644 --- a/include/sound/sof/ext_manifest4.h +++ b/include/sound/sof/ext_manifest4.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2022 Intel Corporation. All rights reserved. + * Copyright(c) 2022 Intel Corporation */ /* diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index b22e925c70e27..4e406dc22f31c 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_HEADER_H__ diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h index 75193850ead0c..08400fbe5402a 100644 --- a/include/sound/sof/info.h +++ b/include/sound/sof/info.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_INFO_H__ diff --git a/include/sound/sof/ipc4/header.h b/include/sound/sof/ipc4/header.h index 1eb538e18236a..0c0cf47946b1d 100644 --- a/include/sound/sof/ipc4/header.h +++ b/include/sound/sof/ipc4/header.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2022 Intel Corporation. All rights reserved. + * Copyright(c) 2022 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_IPC4_HEADER_H__ diff --git a/include/sound/sof/pm.h b/include/sound/sof/pm.h index 366aa6ec442b1..df55bfe583042 100644 --- a/include/sound/sof/pm.h +++ b/include/sound/sof/pm.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_PM_H__ diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h index 9377113f13e49..1bb25487d120c 100644 --- a/include/sound/sof/stream.h +++ b/include/sound/sof/stream.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_STREAM_H__ diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h index b3ca886fa28f5..3ba086f619831 100644 --- a/include/sound/sof/topology.h +++ b/include/sound/sof/topology.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_TOPOLOGY_H__ diff --git a/include/sound/sof/trace.h b/include/sound/sof/trace.h index 25ea99f62d37d..ac2ebb6fb38ff 100644 --- a/include/sound/sof/trace.h +++ b/include/sound/sof/trace.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_TRACE_H__ diff --git a/include/sound/sof/xtensa.h b/include/sound/sof/xtensa.h index 87a07e5204156..ef70f8e266a20 100644 --- a/include/sound/sof/xtensa.h +++ b/include/sound/sof/xtensa.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_SOUND_SOF_XTENSA_H__ diff --git a/include/trace/events/sof.h b/include/trace/events/sof.h index 21c2a1efb9f6e..cd8e6844cca2e 100644 --- a/include/trace/events/sof.h +++ b/include/trace/events/sof.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2022 Intel Corporation. All rights reserved. + * Copyright(c) 2022 Intel Corporation * * Author: Noah Klayman */ diff --git a/include/trace/events/sof_intel.h b/include/trace/events/sof_intel.h index 2a77f9d26c0bf..4cac5fef99a27 100644 --- a/include/trace/events/sof_intel.h +++ b/include/trace/events/sof_intel.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0-only */ /* - * Copyright(c) 2022 Intel Corporation. All rights reserved. + * Copyright(c) 2022 Intel Corporation * * Author: Noah Klayman */ diff --git a/include/uapi/sound/intel/avs/tokens.h b/include/uapi/sound/intel/avs/tokens.h index 4beca03405c07..3e3fb258dd543 100644 --- a/include/uapi/sound/intel/avs/tokens.h +++ b/include/uapi/sound/intel/avs/tokens.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* - * Copyright(c) 2021 Intel Corporation. All rights reserved. + * Copyright(c) 2021 Intel Corporation * * Authors: Cezary Rojewski * Amadeusz Slawinski diff --git a/include/uapi/sound/sof/abi.h b/include/uapi/sound/sof/abi.h index 45c657c3919e6..937ed9408c239 100644 --- a/include/uapi/sound/sof/abi.h +++ b/include/uapi/sound/sof/abi.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ /** diff --git a/include/uapi/sound/sof/fw.h b/include/uapi/sound/sof/fw.h index e9f697467a861..fcfa71faf239e 100644 --- a/include/uapi/sound/sof/fw.h +++ b/include/uapi/sound/sof/fw.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ /* diff --git a/include/uapi/sound/sof/header.h b/include/uapi/sound/sof/header.h index cb3c1ace69e31..228d4c3eb225c 100644 --- a/include/uapi/sound/sof/header.h +++ b/include/uapi/sound/sof/header.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation */ #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__ diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index 6bf00c09d30db..0a246bc218d32 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2018 Intel Corporation. All rights reserved. + * Copyright(c) 2018 Intel Corporation * Author: Liam Girdwood * Keyon Jie */ From 9a9d31b149f3a71ad0835ea295743482601dd322 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:01 -0500 Subject: [PATCH 576/603] ASoC: Intel: sof_sdw_rt_sdca_jack_common: remove -sdca for new codecs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Realtek new SoundWire codecs are all -sdca version. No need to add -sdca to distinguish the non-sdca version. To be consistent with "spk:" and "mic:" components string, remove "-sdca" suffix from "hs:" components string. Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../boards/sof_sdw_rt_sdca_jack_common.c | 23 ++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 85c09513bc35c..69161ea5b290a 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -88,6 +88,15 @@ static const char * const jack_codecs[] = { "rt711", "rt712", "rt713", "rt722" }; +/* + * The sdca suffix is required for rt711 since there are two generations of the same chip. + * RT713 is an SDCA device but the sdca suffix is required for backwards-compatibility with + * previous UCM definitions. + */ +static const char * const need_sdca_suffix[] = { + "rt711", "rt713" +}; + int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -96,6 +105,7 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; + int i; codec_dai = get_codec_dai_by_name(rtd, jack_codecs, ARRAY_SIZE(jack_codecs)); if (!codec_dai) @@ -103,11 +113,22 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) component = codec_dai->component; card->components = devm_kasprintf(card->dev, GFP_KERNEL, - "%s hs:%s-sdca", + "%s hs:%s", card->components, component->name_prefix); if (!card->components) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(need_sdca_suffix); i++) { + if (strstr(codec_dai->name, need_sdca_suffix[i])) { + /* Add -sdca suffix for existing UCMs */ + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s-sdca", card->components); + if (!card->components) + return -ENOMEM; + break; + } + } + ret = snd_soc_add_card_controls(card, rt_sdca_jack_controls, ARRAY_SIZE(rt_sdca_jack_controls)); if (ret) { From 426c43ae9549559f5f5fd405bf464f9fa175e418 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:02 -0500 Subject: [PATCH 577/603] ASoC: Intel: sof-rt5682: remove DMI quirk for hatch The quirk for cml chromebook hatch could be removed since we have cml_rt5682_def board id now. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index e3a2ec6b4c7cd..5a189c165e2da 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -67,15 +67,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { .driver_data = (void *)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(1)), }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Hatch"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_SSP_PORT_CODEC(0) | - SOF_SSP_PORT_AMP(1)), - }, { .callback = sof_rt5682_quirk_cb, .matches = { From f1a5cb6a3bd2fc518dc21561dfc784530dab60b2 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:03 -0500 Subject: [PATCH 578/603] ASoC: Intel: sof_rt5682: board id cleanup for mtl boards Since mtl chromebook is using DMI quirk for board config, changing the SSP port config of mtl_rt5682_def to RVP to simplify mtl board id and enumeration table. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 16 +------------ .../intel/common/soc-acpi-intel-mtl-match.c | 24 ------------------- 2 files changed, 1 insertion(+), 39 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5a189c165e2da..3dc373855a58a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -866,27 +866,13 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_MASK_HDMI_CAPTURE(0x5)), }, { - .name = "mtl_mx98357_rt5682", + .name = "mtl_rt5682_def", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_SSP_PORT_CODEC(0) | SOF_SSP_PORT_AMP(1) | SOF_SSP_PORT_BT_OFFLOAD(2) | SOF_BT_OFFLOAD_PRESENT), }, - { - .name = "mtl_mx98360_rt5682", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_SSP_PORT_CODEC(0) | - SOF_SSP_PORT_AMP(1)), - }, - { - .name = "mtl_rt5682_def", - .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | - SOF_SSP_PORT_CODEC(2) | - SOF_SSP_PORT_AMP(0) | - SOF_SSP_PORT_BT_OFFLOAD(1) | - SOF_BT_OFFLOAD_PRESENT), - }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 4eeec0bc92dc4..0bff913177687 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -11,16 +11,6 @@ #include #include "soc-acpi-intel-sdw-mockup-match.h" -static const struct snd_soc_acpi_codecs mtl_max98357a_amp = { - .num_codecs = 1, - .codecs = {"MX98357A"} -}; - -static const struct snd_soc_acpi_codecs mtl_max98360a_amp = { - .num_codecs = 1, - .codecs = {"MX98360A"} -}; - static const struct snd_soc_acpi_codecs mtl_rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID}, @@ -37,20 +27,6 @@ static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = { }; struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { - { - .comp_ids = &mtl_rt5682_rt5682s_hp, - .drv_name = "mtl_mx98357_rt5682", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &mtl_max98357a_amp, - .sof_tplg_filename = "sof-mtl-max98357a-rt5682.tplg", - }, - { - .comp_ids = &mtl_rt5682_rt5682s_hp, - .drv_name = "mtl_mx98360_rt5682", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &mtl_max98360a_amp, - .sof_tplg_filename = "sof-mtl-max98360a-rt5682.tplg", - }, { .comp_ids = &mtl_essx_83x6, .drv_name = "mtl_es83x6_c1_h02", From eed867325e4b825a16946539275364699eed83d9 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:04 -0500 Subject: [PATCH 579/603] ASoC: Intel: realtek-common: remove 2-spk rt1015p config The original code for rt1015p could support 1 or 2 rt1015p device instances in ACPI. However, all designs implement only 1 instance in ACPI table so we remove the code segment which is for 2 instances. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_realtek_common.c | 57 ++------------------- sound/soc/intel/boards/sof_realtek_common.h | 1 - 2 files changed, 5 insertions(+), 53 deletions(-) diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index 6915ff8b8716b..dda346e0f7374 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -314,59 +314,21 @@ EXPORT_SYMBOL_NS(sof_rt1011_codec_conf, SND_SOC_INTEL_SOF_REALTEK_COMMON); * * For stereo output, there are always two amplifiers on the board. * However, the ACPI implements only one device instance (UID=0) if they - * are sharing the same enable pin. The code will detect the number of - * device instance and use corresponding DAPM structures for - * initialization. + * are sharing the same enable pin. This is the case of rt1015p. */ -static const struct snd_soc_dapm_route rt1015p_1dev_dapm_routes[] = { +static const struct snd_soc_dapm_route rt1015p_dapm_routes[] = { /* speaker */ { "Left Spk", NULL, "Speaker" }, { "Right Spk", NULL, "Speaker" }, }; -static const struct snd_soc_dapm_route rt1015p_2dev_dapm_routes[] = { - /* speaker */ - { "Left Spk", NULL, "Left Speaker" }, - { "Right Spk", NULL, "Right Speaker" }, -}; - -static struct snd_soc_codec_conf rt1015p_codec_confs[] = { - { - .dlc = COMP_CODEC_CONF(RT1015P_DEV0_NAME), - .name_prefix = "Left", - }, - { - .dlc = COMP_CODEC_CONF(RT1015P_DEV1_NAME), - .name_prefix = "Right", - }, -}; - static struct snd_soc_dai_link_component rt1015p_dai_link_components[] = { { .name = RT1015P_DEV0_NAME, .dai_name = RT1015P_CODEC_DAI, }, - { - .name = RT1015P_DEV1_NAME, - .dai_name = RT1015P_CODEC_DAI, - }, }; -static int rt1015p_get_num_codecs(void) -{ - static int dev_num; - - if (dev_num) - return dev_num; - - if (!acpi_dev_present("RTL1015", "1", -1)) - dev_num = 1; - else - dev_num = 2; - - return dev_num; -} - static int rt1015p_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -398,12 +360,8 @@ static int rt1015p_init(struct snd_soc_pcm_runtime *rtd) return ret; } - if (rt1015p_get_num_codecs() == 1) - ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_1dev_dapm_routes, - ARRAY_SIZE(rt1015p_1dev_dapm_routes)); - else - ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_2dev_dapm_routes, - ARRAY_SIZE(rt1015p_2dev_dapm_routes)); + ret = snd_soc_dapm_add_routes(&card->dapm, rt1015p_dapm_routes, + ARRAY_SIZE(rt1015p_dapm_routes)); if (ret) dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); return ret; @@ -412,7 +370,7 @@ static int rt1015p_init(struct snd_soc_pcm_runtime *rtd) void sof_rt1015p_dai_link(struct snd_soc_dai_link *link) { link->codecs = rt1015p_dai_link_components; - link->num_codecs = rt1015p_get_num_codecs(); + link->num_codecs = ARRAY_SIZE(rt1015p_dai_link_components); link->init = rt1015p_init; link->ops = &rt1015p_ops; } @@ -420,11 +378,6 @@ EXPORT_SYMBOL_NS(sof_rt1015p_dai_link, SND_SOC_INTEL_SOF_REALTEK_COMMON); void sof_rt1015p_codec_conf(struct snd_soc_card *card) { - if (rt1015p_get_num_codecs() == 1) - return; - - card->codec_conf = rt1015p_codec_confs; - card->num_configs = ARRAY_SIZE(rt1015p_codec_confs); } EXPORT_SYMBOL_NS(sof_rt1015p_codec_conf, SND_SOC_INTEL_SOF_REALTEK_COMMON); diff --git a/sound/soc/intel/boards/sof_realtek_common.h b/sound/soc/intel/boards/sof_realtek_common.h index d0e331530b398..876290555c220 100644 --- a/sound/soc/intel/boards/sof_realtek_common.h +++ b/sound/soc/intel/boards/sof_realtek_common.h @@ -31,7 +31,6 @@ void sof_rt1011_codec_conf(struct device *dev, struct snd_soc_card *card); */ #define RT1015P_CODEC_DAI "HiFi" #define RT1015P_DEV0_NAME RT1015P_ACPI_HID ":00" -#define RT1015P_DEV1_NAME RT1015P_ACPI_HID ":01" void sof_rt1015p_dai_link(struct snd_soc_dai_link *link); void sof_rt1015p_codec_conf(struct snd_soc_card *card); From f77ae7fcdc47630eb7653983f3c57ac44103aebc Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:05 -0500 Subject: [PATCH 580/603] ASoC: Intel: soc-acpi-intel-mtl-match: add cs42l43 only support cs42l43 is on link 0. No amp in this configuration, will use cs42l43 speaker or bridge. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 0bff913177687..48252fa9e39e0 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -668,6 +668,14 @@ static const struct snd_soc_acpi_link_adr cs42l42_link0_max98363_link2[] = { {} }; +static const struct snd_soc_acpi_link_adr mtl_cs42l43_l0[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, +}; + static const struct snd_soc_acpi_link_adr mtl_cs42l43_cs35l56[] = { { .mask = BIT(0), @@ -782,6 +790,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-cs42l43-l0-cs35l56-l12.tplg", }, + { + .link_mask = BIT(0), + .links = mtl_cs42l43_l0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-cs42l43-l0.tplg", + }, { .link_mask = GENMASK(3, 0), .links = mtl_3_in_1_sdca, From dd3bd9dc47084195fcb3c1b371cb03046abb13ab Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:06 -0500 Subject: [PATCH 581/603] ASoC: Intel: soc-acpi-intel-lnl-match: add cs42l43 only support cs42l43 is on link 0. No amp in this configuration, will use cs42l43 speaker or bridge. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-lnl-match.c | 50 +++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 0c08859c4773c..e6ffcd5be6c5a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -76,6 +76,42 @@ static const struct snd_soc_acpi_endpoint rt722_endpoints[] = { }, }; +static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { + { /* Jack Playback Endpoint */ + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* DMIC Capture Endpoint */ + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Jack Capture Endpoint */ + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Speaker Playback Endpoint */ + .num = 3, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + +static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = { + { + .adr = 0x00003001FA424301ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + } +}; + static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { { .adr = 0x000030025D071101ull, @@ -166,6 +202,14 @@ static const struct snd_soc_acpi_adr_device rt714_1_adr[] = { } }; +static const struct snd_soc_acpi_link_adr lnl_cs42l43_l0[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, +}; + static const struct snd_soc_acpi_link_adr lnl_rvp[] = { { .mask = BIT(0), @@ -268,6 +312,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt711-l0-rt1316-l23-rt714-l1.tplg", }, + { + .link_mask = BIT(0), + .links = lnl_cs42l43_l0, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-cs42l43-l0.tplg", + }, { .link_mask = BIT(0), .links = lnl_rvp, From 3895aa82b665d43b24ba1ac307e20df3396a6b65 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:07 -0500 Subject: [PATCH 582/603] ASoC: Intel: maxim-common: add max_98373_dai_link function Add a helper function, max_98373_dai_link(), for machine driver to initialize dai link of maxim max98373 speaker amplifier. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_maxim_common.c | 9 +++++++++ sound/soc/intel/boards/sof_maxim_common.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index 7e1f485dce7ab..c38c5598d87fa 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -188,6 +188,15 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd) } EXPORT_SYMBOL_NS(max_98373_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); +void max_98373_dai_link(struct device *dev, struct snd_soc_dai_link *link) +{ + link->codecs = max_98373_components; + link->num_codecs = ARRAY_SIZE(max_98373_components); + link->init = max_98373_spk_codec_init; + link->ops = &max_98373_ops; +} +EXPORT_SYMBOL_NS(max_98373_dai_link, SND_SOC_INTEL_SOF_MAXIM_COMMON); + void max_98373_set_codec_conf(struct snd_soc_card *card) { card->codec_conf = max_98373_codec_conf; diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 8886f985de68b..2d8257cc52294 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -25,6 +25,7 @@ extern struct snd_soc_ops max_98373_ops; extern const struct snd_soc_dapm_route max_98373_dapm_routes[]; int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); +void max_98373_dai_link(struct device *dev, struct snd_soc_dai_link *link); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); From b533ed0d85cb64f9323b4221fcaad41259b08556 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:08 -0500 Subject: [PATCH 583/603] ASoC: Intel: sof_da7219: use max_98373_dai_link function Use max_98373_dai_link() function to initialize dai link of maxim max98373 speaker amplifier. No functional change. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-9-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_da7219.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_da7219.c b/sound/soc/intel/boards/sof_da7219.c index d7b605896f895..886771e9b9d69 100644 --- a/sound/soc/intel/boards/sof_da7219.c +++ b/sound/soc/intel/boards/sof_da7219.c @@ -275,9 +275,8 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, max_98360a_dai_link(ctx->amp_link); break; case CODEC_MAX98373: - ctx->amp_link->codecs = max_98373_components; - ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components); - ctx->amp_link->init = max_98373_spk_codec_init; + max_98373_dai_link(dev, ctx->amp_link); + if (ctx->da7219.is_jsl_board) { ctx->amp_link->ops = &max98373_ops; /* use local ops */ } else { From d304ab5ecb758f76bd4eafa6d7e0fac4eba48e88 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:09 -0500 Subject: [PATCH 584/603] ASoC: Intel: sof_nau8825: use max_98373_dai_link function Use max_98373_dai_link() function to initialize dai link of maxim max98373 speaker amplifier. No functional change. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-10-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index dd4dc4ec42556..c08b4eef0bcbe 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -213,10 +213,7 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, max_98360a_dai_link(ctx->amp_link); break; case CODEC_MAX98373: - ctx->amp_link->codecs = max_98373_components; - ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components); - ctx->amp_link->init = max_98373_spk_codec_init; - ctx->amp_link->ops = &max_98373_ops; + max_98373_dai_link(dev, ctx->amp_link); break; case CODEC_NAU8318: nau8318_set_dai_link(ctx->amp_link); From 834c4f95673fdb4394c6cf26242d8763ee6ccf28 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:10 -0500 Subject: [PATCH 585/603] ASoC: Intel: sof_rt5682: use max_98373_dai_link function Use max_98373_dai_link() function to initialize dai link of maxim max98373 speaker amplifier. No functional change. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 3dc373855a58a..6fc6eb0c5172c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -591,10 +591,7 @@ sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, max_98360a_dai_link(ctx->amp_link); break; case CODEC_MAX98373: - ctx->amp_link->codecs = max_98373_components; - ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components); - ctx->amp_link->init = max_98373_spk_codec_init; - ctx->amp_link->ops = &max_98373_ops; + max_98373_dai_link(dev, ctx->amp_link); break; case CODEC_MAX98390: max_98390_dai_link(dev, ctx->amp_link); From 15ce635f396bdb416a41840bfb5e911128585e4d Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:11 -0500 Subject: [PATCH 586/603] ASoC: Intel: sof_sdw: add max98373 dapm routes Add dapm routes of max98373 to sdw driver and remove dependency of maxim-common module. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 - sound/soc/intel/boards/sof_sdw.c | 1 - sound/soc/intel/boards/sof_sdw_maxim.c | 6 +++++- 3 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 17bd0f883b9e2..3ed81ab649c53 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -681,7 +681,6 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_CS35L56_SDW select SND_SOC_DMIC select SND_SOC_INTEL_HDA_DSP_COMMON - select SND_SOC_INTEL_SOF_MAXIM_COMMON imply SND_SOC_SDW_MOCKUP help Add support for Intel SoundWire-based platforms connected to diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b1595fdb500df..2b97972f85ef8 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -2067,4 +2067,3 @@ MODULE_AUTHOR("Rander Wang "); MODULE_AUTHOR("Pierre-Louis Bossart "); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index 432e5112415ab..57c1a5fb17a98 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -11,7 +11,6 @@ #include #include #include "sof_sdw_common.h" -#include "sof_maxim_common.h" static int maxim_part_id; #define SOF_SDW_PART_ID_MAX98363 0x8363 @@ -27,6 +26,11 @@ static const struct snd_kcontrol_new maxim_controls[] = { SOC_DAPM_PIN_SWITCH("Right Spk"), }; +static const struct snd_soc_dapm_route max_98373_dapm_routes[] = { + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; From 2bb765f053910339126626d49ae851b937d06206 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 9 May 2024 11:34:12 -0500 Subject: [PATCH 587/603] ASoC: Intel: maxim-common: change max98373 data to static Since there is dai link helper max_98373_dai_link, we could change all functions and data of max98373 to static. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-13-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_maxim_common.c | 18 ++++++++---------- sound/soc/intel/boards/sof_maxim_common.h | 6 ------ 2 files changed, 8 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index c38c5598d87fa..6c40ecc04723c 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -39,14 +39,16 @@ static unsigned int get_num_codecs(const char *hid) return dev_num; } +/* + * Maxim MAX98373 + */ #define MAX_98373_PIN_NAME 16 -const struct snd_soc_dapm_route max_98373_dapm_routes[] = { +static const struct snd_soc_dapm_route max_98373_dapm_routes[] = { /* speaker */ { "Left Spk", NULL, "Left BE_OUT" }, { "Right Spk", NULL, "Right BE_OUT" }, }; -EXPORT_SYMBOL_NS(max_98373_dapm_routes, SND_SOC_INTEL_SOF_MAXIM_COMMON); static struct snd_soc_codec_conf max_98373_codec_conf[] = { { @@ -59,7 +61,7 @@ static struct snd_soc_codec_conf max_98373_codec_conf[] = { }, }; -struct snd_soc_dai_link_component max_98373_components[] = { +static struct snd_soc_dai_link_component max_98373_components[] = { { /* For Right */ .name = MAX_98373_DEV0_NAME, .dai_name = MAX_98373_CODEC_DAI, @@ -69,7 +71,6 @@ struct snd_soc_dai_link_component max_98373_components[] = { .dai_name = MAX_98373_CODEC_DAI, }, }; -EXPORT_SYMBOL_NS(max_98373_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); static int max_98373_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -96,7 +97,7 @@ static int max_98373_hw_params(struct snd_pcm_substream *substream, return 0; } -int max_98373_trigger(struct snd_pcm_substream *substream, int cmd) +static int max_98373_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai; @@ -139,15 +140,13 @@ int max_98373_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -EXPORT_SYMBOL_NS(max_98373_trigger, SND_SOC_INTEL_SOF_MAXIM_COMMON); -struct snd_soc_ops max_98373_ops = { +static const struct snd_soc_ops max_98373_ops = { .hw_params = max_98373_hw_params, .trigger = max_98373_trigger, }; -EXPORT_SYMBOL_NS(max_98373_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); -int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +static int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; unsigned int num_codecs = get_num_codecs(MAX_98373_ACPI_HID); @@ -186,7 +185,6 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } -EXPORT_SYMBOL_NS(max_98373_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); void max_98373_dai_link(struct device *dev, struct snd_soc_dai_link *link) { diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 2d8257cc52294..3d34c7dae6f57 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -20,14 +20,8 @@ #define MAX_98373_DEV0_NAME "i2c-" MAX_98373_ACPI_HID ":00" #define MAX_98373_DEV1_NAME "i2c-" MAX_98373_ACPI_HID ":01" -extern struct snd_soc_dai_link_component max_98373_components[2]; -extern struct snd_soc_ops max_98373_ops; -extern const struct snd_soc_dapm_route max_98373_dapm_routes[]; - -int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_dai_link(struct device *dev, struct snd_soc_dai_link *link); void max_98373_set_codec_conf(struct snd_soc_card *card); -int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); /* * Maxim MAX98390 From 1628e1c8f6f1446460fc33a06f52e5ce52ac587a Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:13 -0500 Subject: [PATCH 588/603] ASoC: Intel: sof_sdw_cs_amp: rename Speakers to Speaker MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit To be consistent to other amps that used in the machine driver. Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_cs_amp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index e29a586ce7c0e..3538d3c752b78 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -15,7 +15,7 @@ #define CODEC_NAME_SIZE 8 static const struct snd_soc_dapm_widget sof_widgets[] = { - SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), }; int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) @@ -24,7 +24,7 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; char codec_name[CODEC_NAME_SIZE]; char widget_name[16]; - struct snd_soc_dapm_route route = { "Speakers", NULL, widget_name }; + struct snd_soc_dapm_route route = { "Speaker", NULL, widget_name }; struct snd_soc_dai *codec_dai; int i, ret; From 01c266af92f4d24d44939e7d21c36e898caaa18f Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:14 -0500 Subject: [PATCH 589/603] ASoC: Intel: sof_sdw: use generic name for controls/widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Some controls and widgets have exactly the same items. Rename them to be generic. This is a preparation for further cleanup. No function change. Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-15-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_cs42l42.c | 12 ++++++------ sound/soc/intel/boards/sof_sdw_cs42l43.c | 18 +++++++++--------- sound/soc/intel/boards/sof_sdw_cs_amp.c | 6 +++--- sound/soc/intel/boards/sof_sdw_rt5682.c | 12 ++++++------ sound/soc/intel/boards/sof_sdw_rt711.c | 12 ++++++------ sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 12 ++++++------ sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 12 ++++++------ sound/soc/intel/boards/sof_sdw_rt_amp.c | 16 ++++++++-------- .../intel/boards/sof_sdw_rt_sdca_jack_common.c | 12 ++++++------ 9 files changed, 56 insertions(+), 56 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index b999f4e7901a5..82555c53a9c9e 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -17,7 +17,7 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget cs42l42_widgets[] = { +static const struct snd_soc_dapm_widget generic_jack_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; @@ -30,7 +30,7 @@ static const struct snd_soc_dapm_route cs42l42_map[] = { {"cs42l42 HS", NULL, "Headset Mic"}, }; -static const struct snd_kcontrol_new cs42l42_controls[] = { +static const struct snd_kcontrol_new generic_jack_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), }; @@ -70,15 +70,15 @@ int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, cs42l42_controls, - ARRAY_SIZE(cs42l42_controls)); + ret = snd_soc_add_card_controls(card, generic_jack_controls, + ARRAY_SIZE(generic_jack_controls)); if (ret) { dev_err(card->dev, "cs42l42 control addition failed: %d\n", ret); return ret; } - ret = snd_soc_dapm_new_controls(&card->dapm, cs42l42_widgets, - ARRAY_SIZE(cs42l42_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, + ARRAY_SIZE(generic_jack_widgets)); if (ret) { dev_err(card->dev, "cs42l42 widgets addition failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index 0fd5e099bb1a7..f76a16795fda6 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -18,7 +18,7 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget cs42l43_hs_widgets[] = { +static const struct snd_soc_dapm_widget generic_jack_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; @@ -30,7 +30,7 @@ static const struct snd_soc_dapm_route cs42l43_hs_map[] = { { "cs42l43 ADC1_IN1_N", NULL, "Headset Mic" }, }; -static const struct snd_soc_dapm_widget cs42l43_spk_widgets[] = { +static const struct snd_soc_dapm_widget generic_spk_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), }; @@ -41,7 +41,7 @@ static const struct snd_soc_dapm_route cs42l43_spk_map[] = { { "Speaker", NULL, "cs42l43 AMP2_OUT_N", }, }; -static const struct snd_soc_dapm_widget cs42l43_dmic_widgets[] = { +static const struct snd_soc_dapm_widget generic_dmic_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; @@ -74,8 +74,8 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_hs_widgets, - ARRAY_SIZE(cs42l43_hs_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, + ARRAY_SIZE(generic_jack_widgets)); if (ret) { dev_err(card->dev, "cs42l43 hs widgets addition failed: %d\n", ret); return ret; @@ -133,8 +133,8 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) return -ENOMEM; } - ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_spk_widgets, - ARRAY_SIZE(cs42l43_spk_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, + ARRAY_SIZE(generic_spk_widgets)); if (ret) { dev_err(card->dev, "cs42l43 speaker widgets addition failed: %d\n", ret); return ret; @@ -172,8 +172,8 @@ int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_dapm_new_controls(&card->dapm, cs42l43_dmic_widgets, - ARRAY_SIZE(cs42l43_dmic_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_dmic_widgets, + ARRAY_SIZE(generic_dmic_widgets)); if (ret) { dev_err(card->dev, "cs42l43 dmic widgets addition failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index 3538d3c752b78..e15a4753f103e 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -14,7 +14,7 @@ #define CODEC_NAME_SIZE 8 -static const struct snd_soc_dapm_widget sof_widgets[] = { +static const struct snd_soc_dapm_widget generic_spk_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), }; @@ -35,8 +35,8 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_dapm_new_controls(&card->dapm, sof_widgets, - ARRAY_SIZE(sof_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, + ARRAY_SIZE(generic_spk_widgets)); if (ret) { dev_err(card->dev, "widgets addition failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index f812aea64322d..9fbe36322a835 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -17,7 +17,7 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget rt5682_widgets[] = { +static const struct snd_soc_dapm_widget generic_jack_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; @@ -29,7 +29,7 @@ static const struct snd_soc_dapm_route rt5682_map[] = { { "rt5682 IN1P", NULL, "Headset Mic" }, }; -static const struct snd_kcontrol_new rt5682_controls[] = { +static const struct snd_kcontrol_new generic_jack_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), }; @@ -69,15 +69,15 @@ int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, rt5682_controls, - ARRAY_SIZE(rt5682_controls)); + ret = snd_soc_add_card_controls(card, generic_jack_controls, + ARRAY_SIZE(generic_jack_controls)); if (ret) { dev_err(card->dev, "rt5682 control addition failed: %d\n", ret); return ret; } - ret = snd_soc_dapm_new_controls(&card->dapm, rt5682_widgets, - ARRAY_SIZE(rt5682_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, + ARRAY_SIZE(generic_jack_widgets)); if (ret) { dev_err(card->dev, "rt5682 widgets addition failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 7e54fc5cbe097..1a0f6692034ab 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -42,7 +42,7 @@ static int rt711_add_codec_device_props(struct device *sdw_dev) return ret; } -static const struct snd_soc_dapm_widget rt711_widgets[] = { +static const struct snd_soc_dapm_widget generic_jack_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; @@ -53,7 +53,7 @@ static const struct snd_soc_dapm_route rt711_map[] = { { "rt711 MIC2", NULL, "Headset Mic" }, }; -static const struct snd_kcontrol_new rt711_controls[] = { +static const struct snd_kcontrol_new generic_jack_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), }; @@ -93,15 +93,15 @@ int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, rt711_controls, - ARRAY_SIZE(rt711_controls)); + ret = snd_soc_add_card_controls(card, generic_jack_controls, + ARRAY_SIZE(generic_jack_controls)); if (ret) { dev_err(card->dev, "rt711 controls addition failed: %d\n", ret); return ret; } - ret = snd_soc_dapm_new_controls(&card->dapm, rt711_widgets, - ARRAY_SIZE(rt711_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, + ARRAY_SIZE(generic_jack_widgets)); if (ret) { dev_err(card->dev, "rt711 widgets addition failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index 0c4cd4cdbd45b..42308228c634b 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -15,7 +15,7 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget rt712_spk_widgets[] = { +static const struct snd_soc_dapm_widget generic_spk_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), }; @@ -30,7 +30,7 @@ static const struct snd_soc_dapm_route rt712_spk_map[] = { { "Speaker", NULL, "rt712 SPOR" }, }; -static const struct snd_kcontrol_new rt712_spk_controls[] = { +static const struct snd_kcontrol_new generic_spk_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), }; @@ -45,15 +45,15 @@ int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, rt712_spk_controls, - ARRAY_SIZE(rt712_spk_controls)); + ret = snd_soc_add_card_controls(card, generic_spk_controls, + ARRAY_SIZE(generic_spk_controls)); if (ret) { dev_err(card->dev, "rt712 spk controls addition failed: %d\n", ret); return ret; } - ret = snd_soc_dapm_new_controls(&card->dapm, rt712_spk_widgets, - ARRAY_SIZE(rt712_spk_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, + ARRAY_SIZE(generic_spk_widgets)); if (ret) { dev_err(card->dev, "rt712 spk widgets addition failed: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c index e5c2a36e400b8..1653e45a0f7d6 100644 --- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c @@ -15,7 +15,7 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget rt722_spk_widgets[] = { +static const struct snd_soc_dapm_widget generic_spk_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), }; @@ -23,7 +23,7 @@ static const struct snd_soc_dapm_route rt722_spk_map[] = { { "Speaker", NULL, "rt722 SPK" }, }; -static const struct snd_kcontrol_new rt722_spk_controls[] = { +static const struct snd_kcontrol_new generic_spk_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), }; @@ -38,15 +38,15 @@ int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, rt722_spk_controls, - ARRAY_SIZE(rt722_spk_controls)); + ret = snd_soc_add_card_controls(card, generic_spk_controls, + ARRAY_SIZE(generic_spk_controls)); if (ret) { dev_err(card->dev, "failed to add rt722 spk controls: %d\n", ret); return ret; } - ret = snd_soc_dapm_new_controls(&card->dapm, rt722_spk_widgets, - ARRAY_SIZE(rt722_spk_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, + ARRAY_SIZE(generic_spk_widgets)); if (ret) { dev_err(card->dev, "failed to add rt722 spk widgets: %d\n", ret); return ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index 1b415708500e3..03e0978e6a88e 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -131,12 +131,12 @@ static int rt_amp_add_device_props(struct device *sdw_dev) return ret; } -static const struct snd_kcontrol_new rt_amp_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), +static const struct snd_soc_dapm_widget generic_spk_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), }; -static const struct snd_soc_dapm_widget rt_amp_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), +static const struct snd_kcontrol_new generic_spk_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), }; /* @@ -202,15 +202,15 @@ int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, rt_amp_controls, - ARRAY_SIZE(rt_amp_controls)); + ret = snd_soc_add_card_controls(card, generic_spk_controls, + ARRAY_SIZE(generic_spk_controls)); if (ret) { dev_err(card->dev, "%s controls addition failed: %d\n", codec_name, ret); return ret; } - ret = snd_soc_dapm_new_controls(&card->dapm, rt_amp_widgets, - ARRAY_SIZE(rt_amp_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, + ARRAY_SIZE(generic_spk_widgets)); if (ret) { dev_err(card->dev, "%s widgets addition failed: %d\n", codec_name, ret); return ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 69161ea5b290a..932fba8ccf9a0 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -43,7 +43,7 @@ static int rt_sdca_jack_add_codec_device_props(struct device *sdw_dev) return ret; } -static const struct snd_soc_dapm_widget rt_sdca_jack_widgets[] = { +static const struct snd_soc_dapm_widget generic_jack_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; @@ -68,7 +68,7 @@ static const struct snd_soc_dapm_route rt722_sdca_map[] = { { "rt722 MIC2", NULL, "Headset Mic" }, }; -static const struct snd_kcontrol_new rt_sdca_jack_controls[] = { +static const struct snd_kcontrol_new generic_jack_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), }; @@ -129,15 +129,15 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) } } - ret = snd_soc_add_card_controls(card, rt_sdca_jack_controls, - ARRAY_SIZE(rt_sdca_jack_controls)); + ret = snd_soc_add_card_controls(card, generic_jack_controls, + ARRAY_SIZE(generic_jack_controls)); if (ret) { dev_err(card->dev, "rt sdca jack controls addition failed: %d\n", ret); return ret; } - ret = snd_soc_dapm_new_controls(&card->dapm, rt_sdca_jack_widgets, - ARRAY_SIZE(rt_sdca_jack_widgets)); + ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, + ARRAY_SIZE(generic_jack_widgets)); if (ret) { dev_err(card->dev, "rt sdca jack widgets addition failed: %d\n", ret); return ret; From 70d470f05f0b5bb8dea67915cac6ed6308120a89 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:15 -0500 Subject: [PATCH 590/603] ASoC: Intel: sof_sdw: add controls and dapm widgets in codec_info MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Currently, we add card controls and dapm widgets one by one in the codec_info->dais->rtd_init callback. Duplicated controls and dapm widgets will be added if there are more than one types of amps in the dai link. Moving it to sof_sdw_rtd_init() and only add the controls/widgets of the first codec dai can avoid the duplications. Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-16-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 31 +++++++++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 4 ++++ 2 files changed, 35 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 2b97972f85ef8..3acc2db61b358 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1248,6 +1248,7 @@ static const char *get_codec_name(struct device *dev, static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_card *card = rtd->card; struct sof_sdw_codec_info *codec_info; struct snd_soc_dai *dai; int dai_index; @@ -1267,6 +1268,36 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) */ if (codec_info->dais[dai_index].rtd_init_done) continue; + + /* + * Add card controls and dapm widgets for the first codec dai. + * The controls and widgets will be used for all codec dais. + */ + + if (i > 0) + goto skip_add_controls_widgets; + + if (codec_info->dais[dai_index].controls) { + ret = snd_soc_add_card_controls(card, codec_info->dais[dai_index].controls, + codec_info->dais[dai_index].num_controls); + if (ret) { + dev_err(card->dev, "%#x controls addition failed: %d\n", + codec_info->part_id, ret); + return ret; + } + } + if (codec_info->dais[dai_index].widgets) { + ret = snd_soc_dapm_new_controls(&card->dapm, + codec_info->dais[dai_index].widgets, + codec_info->dais[dai_index].num_widgets); + if (ret) { + dev_err(card->dev, "%#x widgets addition failed: %d\n", + codec_info->part_id, ret); + return ret; + } + } + +skip_add_controls_widgets: if (codec_info->dais[dai_index].rtd_init) { ret = codec_info->dais[dai_index].rtd_init(rtd); if (ret) diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 94657dd210f56..a8ba39bd5fd8e 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -86,6 +86,10 @@ struct sof_sdw_dai_info { const char *dai_name; const int dai_type; const int dailink[2]; /* dailink id for each direction */ + const struct snd_kcontrol_new *controls; + const int num_controls; + const struct snd_soc_dapm_widget *widgets; + const int num_widgets; int (*init)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, From 2e3bc94796776e2695054183e7324d1a6ac27837 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:16 -0500 Subject: [PATCH 591/603] ASoC: Intel: sof_sdw: use .controls/.widgets to add controls/widgets MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sof_sdw_rtd_init() will add the controls and widgets if we set them in the codec_info. Move the additions from .rtd_init callback to sof_sdw_rtd_init(). Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-17-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 112 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_cs42l42.c | 24 ---- sound/soc/intel/boards/sof_sdw_cs42l43.c | 34 ------ sound/soc/intel/boards/sof_sdw_cs_amp.c | 11 -- sound/soc/intel/boards/sof_sdw_maxim.c | 24 ---- sound/soc/intel/boards/sof_sdw_rt5682.c | 24 ---- sound/soc/intel/boards/sof_sdw_rt700.c | 26 ---- sound/soc/intel/boards/sof_sdw_rt711.c | 24 ---- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 22 ---- sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 22 ---- sound/soc/intel/boards/sof_sdw_rt_amp.c | 22 ---- .../boards/sof_sdw_rt_sdca_jack_common.c | 24 ---- 12 files changed, 112 insertions(+), 257 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3acc2db61b358..8c1497bab2efd 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -515,6 +515,50 @@ static struct snd_soc_dai_link_component platform_component[] = { } }; +static const struct snd_soc_dapm_widget generic_dmic_widgets[] = { + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static const struct snd_soc_dapm_widget generic_jack_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_kcontrol_new generic_jack_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget generic_spk_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_kcontrol_new generic_spk_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_widget maxim_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_kcontrol_new maxim_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget rt700_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_kcontrol_new rt700_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("AMIC"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + struct snd_soc_dai *get_codec_dai_by_name(struct snd_soc_pcm_runtime *rtd, const char * const dai_name[], int num_dais) @@ -681,6 +725,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .rtd_init = rt700_rtd_init, + .controls = rt700_controls, + .num_controls = ARRAY_SIZE(rt700_controls), + .widgets = rt700_widgets, + .num_widgets = ARRAY_SIZE(rt700_widgets), }, }, .dai_num = 1, @@ -697,6 +745,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, .rtd_init = rt_sdca_jack_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), }, }, .dai_num = 1, @@ -713,6 +765,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt711_init, .exit = sof_sdw_rt711_exit, .rtd_init = rt711_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), }, }, .dai_num = 1, @@ -729,6 +785,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, .rtd_init = rt_sdca_jack_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), }, { .direction = {true, false}, @@ -738,6 +798,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt712_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), }, }, .dai_num = 2, @@ -768,6 +832,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, .rtd_init = rt_sdca_jack_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), }, }, .dai_num = 1, @@ -798,6 +866,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt_amp_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), }, }, .dai_num = 1, @@ -814,6 +886,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt_amp_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), }, }, .dai_num = 1, @@ -829,6 +905,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt_amp_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), }, }, .dai_num = 1, @@ -905,6 +985,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, .rtd_init = rt_sdca_jack_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), }, { .direction = {true, false}, @@ -915,6 +999,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt722_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), }, { .direction = {false, true}, @@ -936,6 +1024,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, .init = sof_sdw_maxim_init, .rtd_init = maxim_spk_rtd_init, + .controls = maxim_controls, + .num_controls = ARRAY_SIZE(maxim_controls), + .widgets = maxim_widgets, + .num_widgets = ARRAY_SIZE(maxim_widgets), }, }, .dai_num = 1, @@ -950,6 +1042,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, .init = sof_sdw_maxim_init, .rtd_init = maxim_spk_rtd_init, + .controls = maxim_controls, + .num_controls = ARRAY_SIZE(maxim_controls), + .widgets = maxim_widgets, + .num_widgets = ARRAY_SIZE(maxim_widgets), }, }, .dai_num = 1, @@ -963,6 +1059,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .rtd_init = rt5682_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), }, }, .dai_num = 1, @@ -977,6 +1077,8 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, .init = sof_sdw_cs_amp_init, .rtd_init = cs_spk_rtd_init, + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), }, }, .dai_num = 1, @@ -990,6 +1092,10 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, .rtd_init = cs42l42_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), }, }, .dai_num = 1, @@ -1006,6 +1112,8 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_JACK, .dailink = {SDW_JACK_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, .rtd_init = cs42l43_hs_rtd_init, + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), }, { .direction = {false, true}, @@ -1013,6 +1121,8 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_type = SOF_SDW_DAI_TYPE_MIC, .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, .rtd_init = cs42l43_dmic_rtd_init, + .widgets = generic_dmic_widgets, + .num_widgets = ARRAY_SIZE(generic_dmic_widgets), }, { .direction = {false, true}, @@ -1027,6 +1137,8 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, .init = sof_sdw_cs42l43_spk_init, .rtd_init = cs42l43_spk_rtd_init, + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), .quirk = SOF_CODEC_SPKR | SOF_SIDECAR_AMPS, }, }, diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index 82555c53a9c9e..9783d5fb2d91c 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -17,11 +17,6 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget generic_jack_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), -}; - static const struct snd_soc_dapm_route cs42l42_map[] = { /* HP jack connectors - unknown if we have jack detection */ {"Headphone", NULL, "cs42l42 HP"}, @@ -30,11 +25,6 @@ static const struct snd_soc_dapm_route cs42l42_map[] = { {"cs42l42 HS", NULL, "Headset Mic"}, }; -static const struct snd_kcontrol_new generic_jack_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), -}; - static struct snd_soc_jack_pin cs42l42_jack_pins[] = { { .pin = "Headphone", @@ -70,20 +60,6 @@ int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, generic_jack_controls, - ARRAY_SIZE(generic_jack_controls)); - if (ret) { - dev_err(card->dev, "cs42l42 control addition failed: %d\n", ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, - ARRAY_SIZE(generic_jack_widgets)); - if (ret) { - dev_err(card->dev, "cs42l42 widgets addition failed: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, cs42l42_map, ARRAY_SIZE(cs42l42_map)); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index f76a16795fda6..2dd0f085fe60b 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -18,11 +18,6 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget generic_jack_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), -}; - static const struct snd_soc_dapm_route cs42l43_hs_map[] = { { "Headphone", NULL, "cs42l43 AMP3_OUT" }, { "Headphone", NULL, "cs42l43 AMP4_OUT" }, @@ -30,10 +25,6 @@ static const struct snd_soc_dapm_route cs42l43_hs_map[] = { { "cs42l43 ADC1_IN1_N", NULL, "Headset Mic" }, }; -static const struct snd_soc_dapm_widget generic_spk_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - static const struct snd_soc_dapm_route cs42l43_spk_map[] = { { "Speaker", NULL, "cs42l43 AMP1_OUT_P", }, { "Speaker", NULL, "cs42l43 AMP1_OUT_N", }, @@ -41,10 +32,6 @@ static const struct snd_soc_dapm_route cs42l43_spk_map[] = { { "Speaker", NULL, "cs42l43 AMP2_OUT_N", }, }; -static const struct snd_soc_dapm_widget generic_dmic_widgets[] = { - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - static const struct snd_soc_dapm_route cs42l43_dmic_map[] = { { "cs42l43 PDM1_DIN", NULL, "DMIC" }, { "cs42l43 PDM2_DIN", NULL, "DMIC" }, @@ -74,13 +61,6 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, - ARRAY_SIZE(generic_jack_widgets)); - if (ret) { - dev_err(card->dev, "cs42l43 hs widgets addition failed: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_hs_map, ARRAY_SIZE(cs42l43_hs_map)); if (ret) { @@ -133,13 +113,6 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) return -ENOMEM; } - ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, - ARRAY_SIZE(generic_spk_widgets)); - if (ret) { - dev_err(card->dev, "cs42l43 speaker widgets addition failed: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_spk_map, ARRAY_SIZE(cs42l43_spk_map)); if (ret) @@ -172,13 +145,6 @@ int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_dapm_new_controls(&card->dapm, generic_dmic_widgets, - ARRAY_SIZE(generic_dmic_widgets)); - if (ret) { - dev_err(card->dev, "cs42l43 dmic widgets addition failed: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, cs42l43_dmic_map, ARRAY_SIZE(cs42l43_dmic_map)); if (ret) diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index e15a4753f103e..663c0e1d2c563 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -14,10 +14,6 @@ #define CODEC_NAME_SIZE 8 -static const struct snd_soc_dapm_widget generic_spk_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { const char *dai_name = rtd->dai_link->codecs->dai_name; @@ -35,13 +31,6 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, - ARRAY_SIZE(generic_spk_widgets)); - if (ret) { - dev_err(card->dev, "widgets addition failed: %d\n", ret); - return ret; - } - for_each_rtd_codec_dais(rtd, i, codec_dai) { if (!strstr(codec_dai->name, "cs35l56")) continue; diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index 57c1a5fb17a98..a7a8db624818b 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -16,16 +16,6 @@ static int maxim_part_id; #define SOF_SDW_PART_ID_MAX98363 0x8363 #define SOF_SDW_PART_ID_MAX98373 0x8373 -static const struct snd_soc_dapm_widget maxim_widgets[] = { - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), -}; - -static const struct snd_kcontrol_new maxim_controls[] = { - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), -}; - static const struct snd_soc_dapm_route max_98373_dapm_routes[] = { { "Left Spk", NULL, "Left BE_OUT" }, { "Right Spk", NULL, "Right BE_OUT" }, @@ -45,20 +35,6 @@ int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) dev_dbg(card->dev, "soundwire maxim card components assigned : %s\n", card->components); - ret = snd_soc_add_card_controls(card, maxim_controls, - ARRAY_SIZE(maxim_controls)); - if (ret) { - dev_err(card->dev, "mx%04x ctrls addition failed: %d\n", maxim_part_id, ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, maxim_widgets, - ARRAY_SIZE(maxim_widgets)); - if (ret) { - dev_err(card->dev, "mx%04x widgets addition failed: %d\n", maxim_part_id, ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, max_98373_dapm_routes, 2); if (ret) dev_err(rtd->dev, "failed to add first SPK map: %d\n", ret); diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index 9fbe36322a835..1741226fe052f 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -17,11 +17,6 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget generic_jack_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), -}; - static const struct snd_soc_dapm_route rt5682_map[] = { /*Headphones*/ { "Headphone", NULL, "rt5682 HPOL" }, @@ -29,11 +24,6 @@ static const struct snd_soc_dapm_route rt5682_map[] = { { "rt5682 IN1P", NULL, "Headset Mic" }, }; -static const struct snd_kcontrol_new generic_jack_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), -}; - static struct snd_soc_jack_pin rt5682_jack_pins[] = { { .pin = "Headphone", @@ -69,20 +59,6 @@ int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, generic_jack_controls, - ARRAY_SIZE(generic_jack_controls)); - if (ret) { - dev_err(card->dev, "rt5682 control addition failed: %d\n", ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, - ARRAY_SIZE(generic_jack_widgets)); - if (ret) { - dev_err(card->dev, "rt5682 widgets addition failed: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, rt5682_map, ARRAY_SIZE(rt5682_map)); diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c index a2648c900e747..58677a8968abe 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -15,12 +15,6 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget rt700_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - static const struct snd_soc_dapm_route rt700_map[] = { /* Headphones */ { "Headphones", NULL, "rt700 HP" }, @@ -28,12 +22,6 @@ static const struct snd_soc_dapm_route rt700_map[] = { { "rt700 MIC2", NULL, "AMIC" }, }; -static const struct snd_kcontrol_new rt700_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphones"), - SOC_DAPM_PIN_SWITCH("AMIC"), - SOC_DAPM_PIN_SWITCH("Speaker"), -}; - static struct snd_soc_jack_pin rt700_jack_pins[] = { { .pin = "Headphones", @@ -69,20 +57,6 @@ int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, rt700_controls, - ARRAY_SIZE(rt700_controls)); - if (ret) { - dev_err(card->dev, "rt700 controls addition failed: %d\n", ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, rt700_widgets, - ARRAY_SIZE(rt700_widgets)); - if (ret) { - dev_err(card->dev, "rt700 widgets addition failed: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, rt700_map, ARRAY_SIZE(rt700_map)); diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 1a0f6692034ab..4cff48a37e3c1 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -42,22 +42,12 @@ static int rt711_add_codec_device_props(struct device *sdw_dev) return ret; } -static const struct snd_soc_dapm_widget generic_jack_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), -}; - static const struct snd_soc_dapm_route rt711_map[] = { /* Headphones */ { "Headphone", NULL, "rt711 HP" }, { "rt711 MIC2", NULL, "Headset Mic" }, }; -static const struct snd_kcontrol_new generic_jack_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), -}; - static struct snd_soc_jack_pin rt711_jack_pins[] = { { .pin = "Headphone", @@ -93,20 +83,6 @@ int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, generic_jack_controls, - ARRAY_SIZE(generic_jack_controls)); - if (ret) { - dev_err(card->dev, "rt711 controls addition failed: %d\n", ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, - ARRAY_SIZE(generic_jack_widgets)); - if (ret) { - dev_err(card->dev, "rt711 widgets addition failed: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, rt711_map, ARRAY_SIZE(rt711_map)); diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index 42308228c634b..ee2c57a50f0d4 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -15,10 +15,6 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget generic_spk_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - /* * dapm routes for rt712 spk will be registered dynamically according * to the number of rt712 spk used. The first two entries will be registered @@ -30,10 +26,6 @@ static const struct snd_soc_dapm_route rt712_spk_map[] = { { "Speaker", NULL, "rt712 SPOR" }, }; -static const struct snd_kcontrol_new generic_spk_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), -}; - int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -45,20 +37,6 @@ int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, generic_spk_controls, - ARRAY_SIZE(generic_spk_controls)); - if (ret) { - dev_err(card->dev, "rt712 spk controls addition failed: %d\n", ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, - ARRAY_SIZE(generic_spk_widgets)); - if (ret) { - dev_err(card->dev, "rt712 spk widgets addition failed: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, rt712_spk_map, ARRAY_SIZE(rt712_spk_map)); if (ret) dev_err(rtd->dev, "failed to add SPK map: %d\n", ret); diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c index 1653e45a0f7d6..02ee8e33a99ae 100644 --- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c @@ -15,18 +15,10 @@ #include #include "sof_sdw_common.h" -static const struct snd_soc_dapm_widget generic_spk_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - static const struct snd_soc_dapm_route rt722_spk_map[] = { { "Speaker", NULL, "rt722 SPK" }, }; -static const struct snd_kcontrol_new generic_spk_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), -}; - int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -38,20 +30,6 @@ int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, generic_spk_controls, - ARRAY_SIZE(generic_spk_controls)); - if (ret) { - dev_err(card->dev, "failed to add rt722 spk controls: %d\n", ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, - ARRAY_SIZE(generic_spk_widgets)); - if (ret) { - dev_err(card->dev, "failed to add rt722 spk widgets: %d\n", ret); - return ret; - } - ret = snd_soc_dapm_add_routes(&card->dapm, rt722_spk_map, ARRAY_SIZE(rt722_spk_map)); if (ret) dev_err(rtd->dev, "failed to add rt722 spk map: %d\n", ret); diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index 03e0978e6a88e..218f6c4efe383 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -131,14 +131,6 @@ static int rt_amp_add_device_props(struct device *sdw_dev) return ret; } -static const struct snd_soc_dapm_widget generic_spk_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - -static const struct snd_kcontrol_new generic_spk_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), -}; - /* * dapm routes for rt1308/rt1316/rt1318 will be registered dynamically * according to the number of rt1308/rt1316/rt1318 used. The first two @@ -202,20 +194,6 @@ int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - ret = snd_soc_add_card_controls(card, generic_spk_controls, - ARRAY_SIZE(generic_spk_controls)); - if (ret) { - dev_err(card->dev, "%s controls addition failed: %d\n", codec_name, ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, generic_spk_widgets, - ARRAY_SIZE(generic_spk_widgets)); - if (ret) { - dev_err(card->dev, "%s widgets addition failed: %d\n", codec_name, ret); - return ret; - } - for_each_rtd_codec_dais(rtd, i, dai) { if (strstr(dai->component->name_prefix, "-1")) ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map, 2); diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 932fba8ccf9a0..c24a5707fb6c3 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -43,11 +43,6 @@ static int rt_sdca_jack_add_codec_device_props(struct device *sdw_dev) return ret; } -static const struct snd_soc_dapm_widget generic_jack_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), -}; - static const struct snd_soc_dapm_route rt711_sdca_map[] = { { "Headphone", NULL, "rt711 HP" }, { "rt711 MIC2", NULL, "Headset Mic" }, @@ -68,11 +63,6 @@ static const struct snd_soc_dapm_route rt722_sdca_map[] = { { "rt722 MIC2", NULL, "Headset Mic" }, }; -static const struct snd_kcontrol_new generic_jack_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), -}; - static struct snd_soc_jack_pin rt_sdca_jack_pins[] = { { .pin = "Headphone", @@ -129,20 +119,6 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) } } - ret = snd_soc_add_card_controls(card, generic_jack_controls, - ARRAY_SIZE(generic_jack_controls)); - if (ret) { - dev_err(card->dev, "rt sdca jack controls addition failed: %d\n", ret); - return ret; - } - - ret = snd_soc_dapm_new_controls(&card->dapm, generic_jack_widgets, - ARRAY_SIZE(generic_jack_widgets)); - if (ret) { - dev_err(card->dev, "rt sdca jack widgets addition failed: %d\n", ret); - return ret; - } - if (strstr(component->name_prefix, "rt711")) { ret = snd_soc_dapm_add_routes(&card->dapm, rt711_sdca_map, ARRAY_SIZE(rt711_sdca_map)); From 84e0a19adb73d7cec5a43e02f9e2d2aafc5c7176 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:17 -0500 Subject: [PATCH 592/603] ASoC: Intel: sof_sdw: add dai parameter to rtd_init callback MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit A generic .rtd_init() callback could be used by different dais. It is useful to pass dai parameter to the callback. The dai parameter will be used in the follow up commit. No functional change here. Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-18-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_sdw_common.h | 30 +++++++++---------- sound/soc/intel/boards/sof_sdw_cs42l42.c | 2 +- sound/soc/intel/boards/sof_sdw_cs42l43.c | 6 ++-- sound/soc/intel/boards/sof_sdw_cs_amp.c | 2 +- sound/soc/intel/boards/sof_sdw_maxim.c | 2 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 2 +- sound/soc/intel/boards/sof_sdw_rt700.c | 2 +- sound/soc/intel/boards/sof_sdw_rt711.c | 2 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 2 +- sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 2 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 10 +++---- sound/soc/intel/boards/sof_sdw_rt_dmic.c | 2 +- .../boards/sof_sdw_rt_sdca_jack_common.c | 2 +- 14 files changed, 34 insertions(+), 34 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 8c1497bab2efd..e41b0d95e0ff7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1411,7 +1411,7 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) skip_add_controls_widgets: if (codec_info->dais[dai_index].rtd_init) { - ret = codec_info->dais[dai_index].rtd_init(rtd); + ret = codec_info->dais[dai_index].rtd_init(rtd, dai); if (ret) return ret; } diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index a8ba39bd5fd8e..3dfba6f6b95dc 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -95,7 +95,7 @@ struct sof_sdw_dai_info { struct sof_sdw_codec_info *info, bool playback); int (*exit)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); - int (*rtd_init)(struct snd_soc_pcm_runtime *rtd); + int (*rtd_init)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); bool rtd_init_done; /* Indicate that the rtd_init callback is done */ unsigned long quirk; }; @@ -208,19 +208,19 @@ int sof_sdw_cs_amp_init(struct snd_soc_card *card, /* dai_link init callbacks */ -int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd); -int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd); -int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); -int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); -int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); -int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd); -int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd); +int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index 9783d5fb2d91c..fdb75fc71c264 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -40,7 +40,7 @@ static const char * const jack_codecs[] = { "cs42l42" }; -int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd) +int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index 2dd0f085fe60b..b7e2750c10745 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -48,7 +48,7 @@ static struct snd_soc_jack_pin sof_jack_pins[] = { }, }; -int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd) +int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -99,7 +99,7 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd) return ret; } -int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) +int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; @@ -135,7 +135,7 @@ int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, return bridge_cs35l56_spk_init(card, dai_links, info, playback); } -int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) +int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index 663c0e1d2c563..10e08207619a9 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -14,7 +14,7 @@ #define CODEC_NAME_SIZE 8 -int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) +int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { const char *dai_name = rtd->dai_link->codecs->dai_name; struct snd_soc_card *card = rtd->card; diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index a7a8db624818b..b7f73177867f4 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -21,7 +21,7 @@ static const struct snd_soc_dapm_route max_98373_dapm_routes[] = { { "Right Spk", NULL, "Right BE_OUT" }, }; -int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) +int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index 1741226fe052f..96f1937985407 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -39,7 +39,7 @@ static const char * const jack_codecs[] = { "rt5682" }; -int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c index 58677a8968abe..f9575db9d99ca 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -37,7 +37,7 @@ static const char * const jack_codecs[] = { "rt700" }; -int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 4cff48a37e3c1..d49e5aa786c3b 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -63,7 +63,7 @@ static const char * const jack_codecs[] = { "rt711" }; -int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index ee2c57a50f0d4..7887964618858 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -26,7 +26,7 @@ static const struct snd_soc_dapm_route rt712_spk_map[] = { { "Speaker", NULL, "rt712 SPOR" }, }; -int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c index 02ee8e33a99ae..083d281bd0525 100644 --- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c @@ -19,7 +19,7 @@ static const struct snd_soc_dapm_route rt722_spk_map[] = { { "Speaker", NULL, "rt722 SPK" }, }; -int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index 218f6c4efe383..df03441ab3647 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -177,12 +177,12 @@ static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_ return rt1318_map; } -int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; const struct snd_soc_dapm_route *rt_amp_map; char codec_name[CODEC_NAME_SIZE]; - struct snd_soc_dai *dai; + struct snd_soc_dai *codec_dai; int ret; int i; @@ -194,10 +194,10 @@ int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; - for_each_rtd_codec_dais(rtd, i, dai) { - if (strstr(dai->component->name_prefix, "-1")) + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (strstr(codec_dai->component->name_prefix, "-1")) ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map, 2); - else if (strstr(dai->component->name_prefix, "-2")) + else if (strstr(codec_dai->component->name_prefix, "-2")) ret = snd_soc_dapm_add_routes(&card->dapm, rt_amp_map + 2, 2); } diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c index 2f7ed9b31e793..b8b493d5c6ec6 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_dmic.c +++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c @@ -19,7 +19,7 @@ static const char * const dmics[] = { "rt722-sdca", }; -int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct snd_soc_component *component; diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index c24a5707fb6c3..701b0372f59e3 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -87,7 +87,7 @@ static const char * const need_sdca_suffix[] = { "rt711", "rt713" }; -int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) +int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); From 4c11132a886ea93865e205e2d92d810722237b48 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:34:18 -0500 Subject: [PATCH 593/603] ASoC: Intel: sof_sdw_rt_amp: use dai parameter MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit get_codec_name_and_route() get dai name from the first codec dai in a dai link. However, the dai may not always be the first codec dai. Use the dai parameter to make sure get_codec_name_and_route() refers to the correct dai. Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163418.67746-19-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt_amp.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index df03441ab3647..797ea9ffa77af 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -158,15 +158,11 @@ static const struct snd_soc_dapm_route rt1318_map[] = { { "Speaker", NULL, "rt1318-2 SPOR" }, }; -static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_pcm_runtime *rtd, +static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_dai *dai, char *codec_name) { - const char *dai_name; - - dai_name = rtd->dai_link->codecs->dai_name; - /* get the codec name */ - snprintf(codec_name, CODEC_NAME_SIZE, "%s", dai_name); + snprintf(codec_name, CODEC_NAME_SIZE, "%s", dai->name); /* choose the right codec's map */ if (strcmp(codec_name, "rt1308") == 0) @@ -186,7 +182,7 @@ int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai int ret; int i; - rt_amp_map = get_codec_name_and_route(rtd, codec_name); + rt_amp_map = get_codec_name_and_route(dai, codec_name); card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s spk:%s", From c8bdf9e727acb6e1b37febf422ef1751e5a2c7d1 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 9 May 2024 11:36:58 -0500 Subject: [PATCH 594/603] ASoC: rt715-sdca-sdw: Fix wrong complete waiting in rt715_dev_resume() enumeration_complete will be completed when a peripheral is attached. And initialization_complete will be completed when a peripheral is initialized. rt715_dev_resume() should wait for initialization_complete instead of enumeration_complete. the issue exists since commit 20d17057f0a8 ("ASoC: rt715-sdca: Add RT715 sdca vendor-specific driver"), but the commit can only apply to commit f892e66fcabc ("ASoC: rt-sdw*: add __func__ to all error logs"). Fixes: f892e66fcabc ("ASoC: rt-sdw*: add __func__ to all error logs") Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20240509163658.68062-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt715-sdca-sdw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index d3fb02e8f31a4..c8dabb9b16b58 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -234,10 +234,10 @@ static int __maybe_unused rt715_dev_resume(struct device *dev) if (!slave->unattach_request) goto regmap_sync; - time = wait_for_completion_timeout(&slave->enumeration_complete, + time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT715_PROBE_TIMEOUT)); if (!time) { - dev_err(&slave->dev, "%s: Enumeration not complete, timed out\n", __func__); + dev_err(&slave->dev, "%s: Initialization not complete, timed out\n", __func__); sdw_show_ping_status(slave->bus, true); return -ETIMEDOUT; From acfff66ee405060576ca88fab5eb569d10adfb45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 May 2024 09:37:37 +0200 Subject: [PATCH 595/603] ASoC: SOF: amd: Correct spaces in Makefile A space should be put around "+=" in each line. Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20240510073739.23541-1-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/sof/amd/Makefile | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile index ad25f42061771..5f2eca283690b 100644 --- a/sound/soc/sof/amd/Makefile +++ b/sound/soc/sof/amd/Makefile @@ -12,7 +12,7 @@ snd-sof-amd-vangogh-objs := pci-vangogh.o vangogh.o snd-sof-amd-acp63-objs := pci-acp63.o acp63.o obj-$(CONFIG_SND_SOC_SOF_AMD_COMMON) += snd-sof-amd-acp.o -obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) +=snd-sof-amd-renoir.o -obj-$(CONFIG_SND_SOC_SOF_AMD_REMBRANDT) +=snd-sof-amd-rembrandt.o -obj-$(CONFIG_SND_SOC_SOF_AMD_VANGOGH) +=snd-sof-amd-vangogh.o -obj-$(CONFIG_SND_SOC_SOF_AMD_ACP63) +=snd-sof-amd-acp63.o +obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) += snd-sof-amd-renoir.o +obj-$(CONFIG_SND_SOC_SOF_AMD_REMBRANDT) += snd-sof-amd-rembrandt.o +obj-$(CONFIG_SND_SOC_SOF_AMD_VANGOGH) += snd-sof-amd-vangogh.o +obj-$(CONFIG_SND_SOC_SOF_AMD_ACP63) += snd-sof-amd-acp63.o From e54f128b0c2fe543816941342e085e21f49c5b6c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 10 May 2024 00:24:55 +0000 Subject: [PATCH 596/603] ASoC: audio-graph-card2: call of_node_get() before of_get_next_child() commit c6f597bc598a8 ("ASoC: audio-graph-card2: remove unneeded of_node_get()") removed of_node_get(), but it need to keep original reference, so of_node_get() itself is needed. Because of_get_next_child() will call of_node_put() inside, if the reference count dropped to zero, then of_node_get() after that will be use afer free. Need to call of_node_get() *before* that. Fixes: c6f597bc598a ("ASoC: audio-graph-card2: remove unneeded of_node_get()") Link: https://lore.kernel.org/r/f930862e-9d30-4ea3-b3e7-b4b4f411f6d1@moroto.mountain Reported-by: Dan Carpenter Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/875xvmqzs8.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 62606e20be9a3..0148688b7152b 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1197,7 +1197,7 @@ static int graph_count_c2c(struct simple_util_priv *priv, { struct device_node *ports = of_get_parent(lnk); struct device_node *port0 = lnk; - struct device_node *port1 = of_get_next_child(ports, lnk); + struct device_node *port1 = of_get_next_child(ports, of_node_get(lnk)); struct device_node *ep0 = port_to_endpoint(port0); struct device_node *ep1 = port_to_endpoint(port1); struct device_node *codec0 = of_graph_get_remote_port(ep0); From 87988a534d8e12f2e6fc01fe63e6c1925dc5307c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 May 2024 12:14:23 +0200 Subject: [PATCH 597/603] ALSA: Fix deadlocks with kctl removals at disconnection In snd_card_disconnect(), we set card->shutdown flag at the beginning, call callbacks and do sync for card->power_ref_sleep waiters at the end. The callback may delete a kctl element, and this can lead to a deadlock when the device was in the suspended state. Namely: * A process waits for the power up at snd_power_ref_and_wait() in snd_ctl_info() or read/write() inside card->controls_rwsem. * The system gets disconnected meanwhile, and the driver tries to delete a kctl via snd_ctl_remove*(); it tries to take card->controls_rwsem again, but this is already locked by the above. Since the sleeper isn't woken up, this deadlocks. An easy fix is to wake up sleepers before processing the driver disconnect callbacks but right after setting the card->shutdown flag. Then all sleepers will abort immediately, and the code flows again. So, basically this patch moves the wait_event() call at the right timing. While we're at it, just to be sure, call wait_event_all() instead of wait_event(), although we don't use exclusive events on this queue for now. Link: https://bugzilla.kernel.org/show_bug.cgi?id=218816 Cc: Reviewed-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20240510101424.6279-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/init.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/core/init.c b/sound/core/init.c index 4ed5037d8693b..89c8354862c4b 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -516,6 +516,14 @@ void snd_card_disconnect(struct snd_card *card) } } +#ifdef CONFIG_PM + /* wake up sleepers here before other callbacks for avoiding potential + * deadlocks with other locks (e.g. in kctls); + * then this notifies the shutdown and sleepers would abort immediately + */ + wake_up_all(&card->power_sleep); +#endif + /* notify all connected devices about disconnection */ /* at this point, they cannot respond to any calls except release() */ @@ -543,7 +551,6 @@ void snd_card_disconnect(struct snd_card *card) } #ifdef CONFIG_PM - wake_up(&card->power_sleep); snd_power_sync_ref(card); #endif } From da0713fff528112890aac02fea08937b65d5c8ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 May 2024 14:51:27 +0200 Subject: [PATCH 598/603] ALSA: core: Remove superfluous CONFIG_PM Since the recent code change, the conditional build with CONFIG_PM is calling only snd_power_sync_ref(). As a dummy function is provided for this function, we can get rid of CONFIG_PM gracefully now. Link: https://lore.kernel.org/r/20240510125128.6058-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/init.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 89c8354862c4b..6b127864a1a32 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -550,9 +550,7 @@ void snd_card_disconnect(struct snd_card *card) clear_bit(card->number, snd_cards_lock); } -#ifdef CONFIG_PM snd_power_sync_ref(card); -#endif } EXPORT_SYMBOL(snd_card_disconnect); From d5ca9ad58e15e4169d75c54b2b4dfebaaee0931f Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Sat, 11 May 2024 18:27:45 +0930 Subject: [PATCH 599/603] ALSA: scarlett2: Add S/PDIF source selection controls Add S/PDIF Source/Digital I/O Mode selection controls for the Scarlett 3rd Gen 18i8/18i20 and Clarett 4Pre/8Pre interfaces. These models have both coax S/PDIF and optical inputs, and the optical inputs are switchable between being used as S/PDIF and ADAT inputs. The Scarlett 3rd Gen 18i20 also has a "Dual ADAT" mode for 8-channel audio at 88.2/96kHz. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/Zj8zCTjzPsTDENN+@m.b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 179 ++++++++++++++++++++++++++++++++++++ 1 file changed, 179 insertions(+) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index b251d25f2a858..2dd03b46964ed 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -541,6 +541,7 @@ enum { SCARLETT2_CONFIG_PCM_INPUT_SWITCH, SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN, SCARLETT2_CONFIG_BLUETOOTH_VOLUME, + SCARLETT2_CONFIG_SPDIF_MODE, SCARLETT2_CONFIG_COUNT }; @@ -754,6 +755,9 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3c = { [SCARLETT2_CONFIG_TALKBACK_MAP] = { .offset = 0xb0, .size = 16, .activate = 10 }, + + [SCARLETT2_CONFIG_SPDIF_MODE] = { + .offset = 0x94, .size = 8, .activate = 6 }, } }; @@ -977,6 +981,9 @@ static const struct scarlett2_config_set scarlett2_config_set_clarett = { [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { .offset = 0x8d, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_SPDIF_MODE] = { + .offset = 0x9e, .size = 8, .activate = 4 }, } }; @@ -1147,6 +1154,11 @@ struct scarlett2_device_info { /* has a Bluetooth module with volume control */ u8 has_bluetooth; + /* S/PDIF Source/Digital I/O mode control */ + const char * const spdif_mode_control_name; + const u8 *spdif_mode_values; + const char * const *spdif_mode_texts; + /* remap analogue outputs; 18i8 Gen 3 has "line 3/4" connected * internally to the analogue 7/8 outputs */ @@ -1255,6 +1267,7 @@ struct scarlett2_data { u8 standalone_switch; u8 power_status; u8 bluetooth_volume; + u8 spdif_mode; u8 meter_level_map[SCARLETT2_MAX_METERS]; struct snd_kcontrol *sync_ctl; struct snd_kcontrol *master_vol_ctl; @@ -1582,6 +1595,14 @@ static const struct scarlett2_device_info s8i6_gen3_info = { } }; +static const u8 scarlett2_spdif_s18i8_gen3_values[] = { 0, 2, 0xff }; + +static const char * const scarlett2_spdif_s18i8_gen3_texts[] = { + "RCA", + "Optical", + NULL +}; + static const struct scarlett2_device_info s18i8_gen3_info = { .config_set = &scarlett2_config_set_gen3c, .has_speaker_switching = 1, @@ -1591,6 +1612,10 @@ static const struct scarlett2_device_info s18i8_gen3_info = { .phantom_count = 2, .inputs_per_phantom = 2, + .spdif_mode_control_name = "S/PDIF Mode Capture Enum", + .spdif_mode_values = scarlett2_spdif_s18i8_gen3_values, + .spdif_mode_texts = scarlett2_spdif_s18i8_gen3_texts, + .line_out_remap_enable = 1, .line_out_remap = { 0, 1, 6, 7, 2, 3, 4, 5 }, .line_out_unmap = { 0, 1, 4, 5, 6, 7, 2, 3 }, @@ -1661,6 +1686,15 @@ static const struct scarlett2_device_info s18i8_gen3_info = { } }; +static const u8 scarlett2_spdif_s18i20_gen3_values[] = { 0, 6, 1, 0xff }; + +static const char * const scarlett2_spdif_s18i20_gen3_texts[] = { + "S/PDIF RCA", + "S/PDIF Optical", + "Dual ADAT", + NULL +}; + static const struct scarlett2_device_info s18i20_gen3_info = { .config_set = &scarlett2_config_set_gen3c, .has_speaker_switching = 1, @@ -1671,6 +1705,10 @@ static const struct scarlett2_device_info s18i20_gen3_info = { .phantom_count = 2, .inputs_per_phantom = 4, + .spdif_mode_control_name = "Digital I/O Mode Capture Enum", + .spdif_mode_values = scarlett2_spdif_s18i20_gen3_values, + .spdif_mode_texts = scarlett2_spdif_s18i20_gen3_texts, + .line_out_descrs = { "Monitor 1 L", "Monitor 1 R", @@ -2019,11 +2057,24 @@ static const struct scarlett2_device_info clarett_2pre_info = { } }; +static const u8 scarlett2_spdif_clarett_values[] = { 0, 1, 2, 0xff }; + +static const char * const scarlett2_spdif_clarett_texts[] = { + "None", + "Optical", + "RCA", + NULL +}; + static const struct scarlett2_device_info clarett_4pre_info = { .config_set = &scarlett2_config_set_clarett, .level_input_count = 2, .air_input_count = 4, + .spdif_mode_control_name = "S/PDIF Source Capture Enum", + .spdif_mode_values = scarlett2_spdif_clarett_values, + .spdif_mode_texts = scarlett2_spdif_clarett_texts, + .line_out_descrs = { "Monitor L", "Monitor R", @@ -2076,6 +2127,10 @@ static const struct scarlett2_device_info clarett_8pre_info = { .level_input_count = 2, .air_input_count = 8, + .spdif_mode_control_name = "S/PDIF Source Capture Enum", + .spdif_mode_values = scarlett2_spdif_clarett_values, + .spdif_mode_texts = scarlett2_spdif_clarett_texts, + .line_out_descrs = { "Monitor L", "Monitor R", @@ -7885,6 +7940,121 @@ static int scarlett2_add_bluetooth_volume_ctl( &private->bluetooth_volume_ctl); } +/*** S/PDIF Mode Controls ***/ + +static int scarlett2_update_spdif_mode(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err, i; + u8 mode; + const u8 *mode_values = private->info->spdif_mode_values; + + if (!private->info->spdif_mode_control_name) + return 0; + + err = scarlett2_usb_get_config(mixer, SCARLETT2_CONFIG_SPDIF_MODE, + 1, &mode); + if (err < 0) + return err; + + private->spdif_mode = 0; + + for (i = 0; *mode_values != 0xff; i++, mode_values++) + if (*mode_values == mode) { + private->spdif_mode = i; + break; + } + + return 0; +} + +static int scarlett2_spdif_mode_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; + const char * const *mode_texts = private->info->spdif_mode_texts; + int count = 0; + + while (*mode_texts++) + count++; + + return snd_ctl_enum_info(uinfo, 1, count, + private->info->spdif_mode_texts); +} + +static int scarlett2_spdif_mode_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; + + ucontrol->value.enumerated.item[0] = private->spdif_mode; + return 0; +} + +static int scarlett2_spdif_mode_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int oval, val, err = 0; + int i; + + mutex_lock(&private->data_mutex); + + oval = private->spdif_mode; + val = ucontrol->value.enumerated.item[0]; + + if (val < 0) { + err = -EINVAL; + goto unlock; + } + + for (i = 0; i <= val; i++) + if (private->info->spdif_mode_values[i] == 0xff) { + err = -EINVAL; + goto unlock; + } + + if (oval == val) + goto unlock; + + private->spdif_mode = val; + + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_SPDIF_MODE, 0, + private->info->spdif_mode_values[val]); + if (!err) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_spdif_mode_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_spdif_mode_ctl_info, + .get = scarlett2_spdif_mode_ctl_get, + .put = scarlett2_spdif_mode_ctl_put, +}; + +static int scarlett2_add_spdif_mode_ctl(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + + if (!private->info->spdif_mode_control_name) + return 0; + + return scarlett2_add_new_ctl(mixer, &scarlett2_spdif_mode_ctl, + 0, 1, + private->info->spdif_mode_control_name, + NULL); +} + /*** Notification Handlers ***/ /* Notify on sync change */ @@ -8797,6 +8967,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; + err = scarlett2_update_spdif_mode(mixer); + if (err < 0) + return err; + err = scarlett2_update_mix(mixer); if (err < 0) return err; @@ -8929,6 +9103,11 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; + /* Create the S/PDIF mode control */ + err = scarlett2_add_spdif_mode_ctl(mixer); + if (err < 0) + return err; + /* Set the access mode of controls disabled during * autogain/phantom power switching. */ From 762e6af39883a816290286d5d46a20bbd0f3a135 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Sun, 12 May 2024 02:14:17 +0930 Subject: [PATCH 600/603] ALSA: scarlett2: Increase mixer range to +12dB The values loaded into the mixer are 16-bit values, with 8192 representing 0dB, going up to a current maximum of 16345 (+6dB). All supported interfaces have no problem going up to 32612 (+12dB), so update SCARLETT2_MIXER_MAX_DB and scarlett2_mixer_values[] to allow for this. Tested with: - Scarlett 2nd Gen 6i6, 18i8, 18i20 - Scarlett 3rd Gen 4i4, 8i6, 18i8, 18i20 - Scarlett 4th Gen Solo, 2i2, 4i4 - Clarett+ 2Pre, 4Pre, 8Pre - Vocaster One and Two Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/Zj+gYT4F2XeKTD93@m.b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 2dd03b46964ed..1150cf104985c 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -184,16 +184,16 @@ /* maximum Bluetooth volume value */ #define SCARLETT2_MAX_BLUETOOTH_VOLUME 30 -/* mixer range from -80dB to +6dB in 0.5dB steps */ +/* mixer range from -80dB to +12dB in 0.5dB steps */ #define SCARLETT2_MIXER_MIN_DB -80 #define SCARLETT2_MIXER_BIAS (-SCARLETT2_MIXER_MIN_DB * 2) -#define SCARLETT2_MIXER_MAX_DB 6 +#define SCARLETT2_MIXER_MAX_DB 12 #define SCARLETT2_MIXER_MAX_VALUE \ ((SCARLETT2_MIXER_MAX_DB - SCARLETT2_MIXER_MIN_DB) * 2) #define SCARLETT2_MIXER_VALUE_COUNT (SCARLETT2_MIXER_MAX_VALUE + 1) /* map from (dB + 80) * 2 to mixer value - * for dB in 0 .. 172: int(8192 * pow(10, ((dB - 160) / 2 / 20))) + * for dB in 0 .. 184: int(8192 * pow(10, ((dB - 160) / 2 / 20))) */ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, @@ -209,7 +209,8 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { 3078, 3261, 3454, 3659, 3876, 4105, 4349, 4606, 4879, 5168, 5475, 5799, 6143, 6507, 6892, 7301, 7733, 8192, 8677, 9191, 9736, 10313, 10924, 11571, 12257, 12983, 13752, 14567, 15430, - 16345 + 16345, 17313, 18339, 19426, 20577, 21796, 23088, 24456, 25905, + 27440, 29066, 30788, 32612 }; /* Maximum number of analogue outputs */ From 9b4843e877c86538eaf0bc6fdb3d0f2ffc0ffd2c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 May 2024 09:36:54 +0200 Subject: [PATCH 601/603] ASoC: SOF: amd: Fix build error with built-in config Makefile in AMD ACP driver has a line substitution with "=" instead of "+="; this overrides the preexisting item, hence it broke the build after the recent change to replace *-objs with *-y. This patch corrects the line. Fixes: 1a74b21ce59f ("ASoC: SOF: amd: Add Probe functionality support for amd platforms.") Fixes: 9c2f5b6eb8b7 ("ASoC: SOF: Use *-y instead of *-objs in Makefile") Reported-by: Stephen Rothwell Closes: https://lore.kernel.org/r/20240510170305.03b67d9f@canb.auug.org.au Link: https://lore.kernel.org/r/20240510073656.23491-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/soc/sof/amd/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile index 701d578d63b21..63fe0d55fd0eb 100644 --- a/sound/soc/sof/amd/Makefile +++ b/sound/soc/sof/amd/Makefile @@ -5,7 +5,7 @@ # Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. snd-sof-amd-acp-y := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o -snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) = acp-probes.o +snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) += acp-probes.o snd-sof-amd-renoir-y := pci-rn.o renoir.o snd-sof-amd-rembrandt-y := pci-rmb.o rembrandt.o snd-sof-amd-vangogh-y := pci-vangogh.o vangogh.o From 0df2b9ed489a06643eaaa150a57ab01fa9ad3005 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Mon, 13 May 2024 16:38:11 +0800 Subject: [PATCH 602/603] ALSA: hda/realtek - fixed headset Mic not show ALC256 run on SOF mode. Boot with plugged headset, the Headset Mic will be gone. Plugged headset after boot. It had partial fail with Headset Mic detect. Add spec->en_3kpull_low = false will solve all issues. Signed-off-by: Kailang Yang Link: https://lore.kernel.org/r/c8b638590c5f45a6a5c6aeb20c31fd5b@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c39bbdc6c5e2..cd49b3e217283 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6527,6 +6527,20 @@ static void alc295_fixup_chromebook(struct hda_codec *codec, } } +static void alc256_fixup_chromebook(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->gen.suppress_auto_mute = 1; + spec->gen.suppress_auto_mic = 1; + spec->en_3kpull_low = false; + break; + } +} + static void alc_fixup_disable_mic_vref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -7505,6 +7519,7 @@ enum { ALC285_FIXUP_ASUS_GU605_SPI_2_HEADSET_MIC, ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1, ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318, + ALC256_FIXUP_CHROME_BOOK, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9787,6 +9802,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, + [ALC256_FIXUP_CHROME_BOOK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_chromebook, + .chained = true, + .chain_id = ALC225_FIXUP_HEADSET_JACK + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10753,6 +10774,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"}, {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"}, {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"}, + {.id = ALC256_FIXUP_CHROME_BOOK, .name = "alc-2024y-chromebook"}, {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, From d731b1ed15052580b7b2f40559021012d280f1d9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 May 2024 08:40:08 +0200 Subject: [PATCH 603/603] ALSA: hda/realtek: Drop doubly quirk entry for 103c:8a2e There are two quirk entries for SSID 103c:8a2e. Drop the latter one that isn't applied in anyway. As both point to the same quirk action, there is no actual behavior change. Fixes: aa8e3ef4fe53 ("ALSA: hda/realtek: Add quirks for various HP ENVY models") Link: https://lore.kernel.org/r/20240513064010.17546-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cd49b3e217283..a696943aec0da 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10114,7 +10114,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8a2c, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a2d, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4),