diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml
index 8132d0c0f00a1..35e669020296d 100644
--- a/Documentation/devicetree/bindings/sound/simple-card.yaml
+++ b/Documentation/devicetree/bindings/sound/simple-card.yaml
@@ -378,6 +378,8 @@ examples:
- |
sound {
compatible = "simple-audio-card";
+ #address-cells = <1>;
+ #size-cells = <0>;
simple-audio-card,name = "rsnd-ak4643";
simple-audio-card,format = "left_j";
@@ -391,10 +393,12 @@ examples:
"ak4642 Playback", "DAI1 Playback";
dpcmcpu: simple-audio-card,cpu@0 {
+ reg = <0>;
sound-dai = <&rcar_sound 0>;
};
simple-audio-card,cpu@1 {
+ reg = <1>;
sound-dai = <&rcar_sound 1>;
};
@@ -418,6 +422,8 @@ examples:
- |
sound {
compatible = "simple-audio-card";
+ #address-cells = <1>;
+ #size-cells = <0>;
simple-audio-card,routing =
"pcm3168a Playback", "DAI1 Playback",
@@ -426,6 +432,7 @@ examples:
"pcm3168a Playback", "DAI4 Playback";
simple-audio-card,dai-link@0 {
+ reg = <0>;
format = "left_j";
bitclock-master = <&sndcpu0>;
frame-master = <&sndcpu0>;
@@ -439,22 +446,23 @@ examples:
};
simple-audio-card,dai-link@1 {
+ reg = <1>;
format = "i2s";
bitclock-master = <&sndcpu1>;
frame-master = <&sndcpu1>;
convert-channels = <8>; /* TDM Split */
- sndcpu1: cpu@0 {
+ sndcpu1: cpu0 {
sound-dai = <&rcar_sound 1>;
};
- cpu@1 {
+ cpu1 {
sound-dai = <&rcar_sound 2>;
};
- cpu@2 {
+ cpu2 {
sound-dai = <&rcar_sound 3>;
};
- cpu@3 {
+ cpu3 {
sound-dai = <&rcar_sound 4>;
};
codec {
@@ -466,6 +474,7 @@ examples:
};
simple-audio-card,dai-link@2 {
+ reg = <2>;
format = "i2s";
bitclock-master = <&sndcpu2>;
frame-master = <&sndcpu2>;
diff --git a/MAINTAINERS b/MAINTAINERS
index d53db30d1365b..0887816d125e5 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -6956,6 +6956,7 @@ M: Timur Tabi <timur@kernel.org>
M: Nicolin Chen <nicoleotsuka@gmail.com>
M: Xiubo Li <Xiubo.Lee@gmail.com>
R: Fabio Estevam <festevam@gmail.com>
+R: Shengjiu Wang <shengjiu.wang@gmail.com>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
L: linuxppc-dev@lists.ozlabs.org
S: Maintained
@@ -11333,17 +11334,17 @@ F: drivers/iio/adc/at91-sama5d2_adc.c
F: include/dt-bindings/iio/adc/at91-sama5d2_adc.h
MICROCHIP SAMA5D2-COMPATIBLE SHUTDOWN CONTROLLER
-M: Nicolas Ferre <nicolas.ferre@microchip.com>
+M: Claudiu Beznea <claudiu.beznea@microchip.com>
S: Supported
F: drivers/power/reset/at91-sama5d2_shdwc.c
MICROCHIP SPI DRIVER
-M: Nicolas Ferre <nicolas.ferre@microchip.com>
+M: Tudor Ambarus <tudor.ambarus@microchip.com>
S: Supported
F: drivers/spi/spi-atmel.*
MICROCHIP SSC DRIVER
-M: Nicolas Ferre <nicolas.ferre@microchip.com>
+M: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
L: linux-arm-kernel@lists.infradead.org (moderated for non-subscribers)
S: Supported
F: drivers/misc/atmel-ssc.c
diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h
index f9024c7a1600f..02e1d77783549 100644
--- a/include/sound/rt5670.h
+++ b/include/sound/rt5670.h
@@ -12,6 +12,7 @@ struct rt5670_platform_data {
int jd_mode;
bool in2_diff;
bool dev_gpio;
+ bool gpio1_is_ext_spk_en;
bool dmic_en;
unsigned int dmic1_data_pin;
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 212257e84facb..71e178c897932 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -161,6 +161,7 @@ void snd_soc_dai_resume(struct snd_soc_dai *dai);
int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
struct snd_soc_pcm_runtime *rtd, int num);
bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link);
void snd_soc_dai_action(struct snd_soc_dai *dai,
int stream, int action);
static inline void snd_soc_dai_activate(struct snd_soc_dai *dai,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 2756f9bcac3e4..3ce7f0f5aa929 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -444,6 +444,8 @@ int devm_snd_soc_register_component(struct device *dev,
const struct snd_soc_component_driver *component_driver,
struct snd_soc_dai_driver *dai_drv, int num_dai);
void snd_soc_unregister_component(struct device *dev);
+void snd_soc_unregister_component_by_driver(struct device *dev,
+ const struct snd_soc_component_driver *component_driver);
struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev,
const char *driver_name);
struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
diff --git a/sound/core/info.c b/sound/core/info.c
index 8c6bc5241df50..9fec3070f8ba3 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -606,7 +606,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len)
{
int c;
- if (snd_BUG_ON(!buffer || !buffer->buffer))
+ if (snd_BUG_ON(!buffer))
+ return 1;
+ if (!buffer->buffer)
return 1;
if (len <= 0 || buffer->stop || buffer->error)
return 1;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1b06c42612488..1b2d8e56390a5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7587,6 +7587,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
+ SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC),
diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c
index f25ce50f1a901..ebf4388b62621 100644
--- a/sound/soc/amd/raven/pci-acp3x.c
+++ b/sound/soc/amd/raven/pci-acp3x.c
@@ -232,9 +232,7 @@ static int snd_acp3x_probe(struct pci_dev *pci,
}
pm_runtime_set_autosuspend_delay(&pci->dev, 2000);
pm_runtime_use_autosuspend(&pci->dev);
- pm_runtime_set_active(&pci->dev);
pm_runtime_put_noidle(&pci->dev);
- pm_runtime_enable(&pci->dev);
pm_runtime_allow(&pci->dev);
return 0;
@@ -303,7 +301,7 @@ static void snd_acp3x_remove(struct pci_dev *pci)
ret = acp3x_deinit(adata->acp3x_base);
if (ret)
dev_err(&pci->dev, "ACP de-init failed\n");
- pm_runtime_disable(&pci->dev);
+ pm_runtime_forbid(&pci->dev);
pm_runtime_get_noresume(&pci->dev);
pci_disable_msi(pci);
pci_release_regions(pci);
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 96718e3a1ad0e..d87402a86c880 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -779,13 +779,6 @@ static int max98373_probe(struct snd_soc_component *component)
regmap_write(max98373->regmap,
MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2,
0x1);
- /* Set inital volume (0dB) */
- regmap_write(max98373->regmap,
- MAX98373_R203D_AMP_DIG_VOL_CTRL,
- 0x00);
- regmap_write(max98373->regmap,
- MAX98373_R203E_AMP_PATH_GAIN,
- 0x00);
/* Enable DC blocker */
regmap_write(max98373->regmap,
MAX98373_R203F_AMP_DSP_CFG,
@@ -869,7 +862,6 @@ static const struct snd_soc_component_driver soc_codec_dev_max98373 = {
.num_dapm_widgets = ARRAY_SIZE(max98373_dapm_widgets),
.dapm_routes = max98373_audio_map,
.num_dapm_routes = ARRAY_SIZE(max98373_audio_map),
- .idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 9593a9a27bf85..e8d14eefc41bb 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -272,13 +272,13 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf);
*mic = buf & 0x80000000;
}
- if (!*mic) {
+
+ if (!*hp) {
snd_soc_dapm_disable_pin(dapm, "HV");
snd_soc_dapm_disable_pin(dapm, "VREF");
- }
- if (!*hp)
snd_soc_dapm_disable_pin(dapm, "LDO1");
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync(dapm);
+ }
return 0;
}
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 70fee6849ab00..dfbc0ca38ff7f 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -31,18 +31,19 @@
#include "rt5670.h"
#include "rt5670-dsp.h"
-#define RT5670_DEV_GPIO BIT(0)
-#define RT5670_IN2_DIFF BIT(1)
-#define RT5670_DMIC_EN BIT(2)
-#define RT5670_DMIC1_IN2P BIT(3)
-#define RT5670_DMIC1_GPIO6 BIT(4)
-#define RT5670_DMIC1_GPIO7 BIT(5)
-#define RT5670_DMIC2_INR BIT(6)
-#define RT5670_DMIC2_GPIO8 BIT(7)
-#define RT5670_DMIC3_GPIO5 BIT(8)
-#define RT5670_JD_MODE1 BIT(9)
-#define RT5670_JD_MODE2 BIT(10)
-#define RT5670_JD_MODE3 BIT(11)
+#define RT5670_DEV_GPIO BIT(0)
+#define RT5670_IN2_DIFF BIT(1)
+#define RT5670_DMIC_EN BIT(2)
+#define RT5670_DMIC1_IN2P BIT(3)
+#define RT5670_DMIC1_GPIO6 BIT(4)
+#define RT5670_DMIC1_GPIO7 BIT(5)
+#define RT5670_DMIC2_INR BIT(6)
+#define RT5670_DMIC2_GPIO8 BIT(7)
+#define RT5670_DMIC3_GPIO5 BIT(8)
+#define RT5670_JD_MODE1 BIT(9)
+#define RT5670_JD_MODE2 BIT(10)
+#define RT5670_JD_MODE3 BIT(11)
+#define RT5670_GPIO1_IS_EXT_SPK_EN BIT(12)
static unsigned long rt5670_quirk;
static unsigned int quirk_override;
@@ -602,9 +603,9 @@ int rt5670_set_jack_detect(struct snd_soc_component *component,
EXPORT_SYMBOL_GPL(rt5670_set_jack_detect);
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
@@ -1447,6 +1448,33 @@ static int rt5670_hp_event(struct snd_soc_dapm_widget *w,
return 0;
}
+static int rt5670_spk_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 rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+ if (!rt5670->pdata.gpio1_is_ext_spk_en)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_HI);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_LO);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static int rt5670_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -1860,7 +1888,9 @@ static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = {
};
static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = {
- SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ rt5670_spk_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_OUTPUT("SPOLP"),
SND_SOC_DAPM_OUTPUT("SPOLN"),
SND_SOC_DAPM_OUTPUT("SPORP"),
@@ -2857,14 +2887,14 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
},
{
.callback = rt5670_quirk_cb,
- .ident = "Lenovo Thinkpad Tablet 10",
+ .ident = "Lenovo Miix 2 10",
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"),
},
.driver_data = (unsigned long *)(RT5670_DMIC_EN |
RT5670_DMIC1_IN2P |
- RT5670_DEV_GPIO |
+ RT5670_GPIO1_IS_EXT_SPK_EN |
RT5670_JD_MODE2),
},
{
@@ -2924,6 +2954,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
rt5670->pdata.dev_gpio = true;
dev_info(&i2c->dev, "quirk dev_gpio\n");
}
+ if (rt5670_quirk & RT5670_GPIO1_IS_EXT_SPK_EN) {
+ rt5670->pdata.gpio1_is_ext_spk_en = true;
+ dev_info(&i2c->dev, "quirk GPIO1 is external speaker enable\n");
+ }
if (rt5670_quirk & RT5670_IN2_DIFF) {
rt5670->pdata.in2_diff = true;
dev_info(&i2c->dev, "quirk IN2_DIFF\n");
@@ -3023,6 +3057,13 @@ static int rt5670_i2c_probe(struct i2c_client *i2c,
RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
}
+ if (rt5670->pdata.gpio1_is_ext_spk_en) {
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_GPIO1);
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
+ }
+
if (rt5670->pdata.jd_mode) {
regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK,
RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index a8c3e44770b82..de0203369b7cd 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -757,7 +757,7 @@
#define RT5670_PWR_VREF2_BIT 4
#define RT5670_PWR_FV2 (0x1 << 3)
#define RT5670_PWR_FV2_BIT 3
-#define RT5670_LDO_SEL_MASK (0x3)
+#define RT5670_LDO_SEL_MASK (0x7)
#define RT5670_LDO_SEL_SFT 0
/* Power Management for Analog 2 (0x64) */
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 7d6670abdb08e..d503b5bef4ba9 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -967,13 +967,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
rt5682_enable_push_button_irq(component, false);
snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
- if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+ if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
snd_soc_component_update_bits(component,
- RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
- else
+ RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
+ if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
snd_soc_component_update_bits(component,
- RT5682_PWR_ANLG_1,
- RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
+ RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
RT5682_PWR_CBJ, 0);
@@ -992,16 +991,17 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
rt5682->hs_jack = hs_jack;
- if (!rt5682->is_sdw) {
- if (!hs_jack) {
- regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
- RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
- regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
- RT5682_POW_JDH | RT5682_POW_JDL, 0);
- cancel_delayed_work_sync(&rt5682->jack_detect_work);
- return 0;
- }
+ if (!hs_jack) {
+ regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+ RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+ regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+ RT5682_POW_JDH | RT5682_POW_JDL, 0);
+ cancel_delayed_work_sync(&rt5682->jack_detect_work);
+ return 0;
+ }
+
+ if (!rt5682->is_sdw) {
switch (rt5682->pdata.jd_src) {
case RT5682_JD1:
snd_soc_component_update_bits(component,
@@ -1082,7 +1082,8 @@ void rt5682_jack_detect_handler(struct work_struct *work)
/* jack was out, report jack type */
rt5682->jack_type =
rt5682_headset_detect(rt5682->component, 1);
- } else {
+ } else if ((rt5682->jack_type & SND_JACK_HEADSET) ==
+ SND_JACK_HEADSET) {
/* jack is already in, report button event */
rt5682->jack_type = SND_JACK_HEADSET;
btn_type = rt5682_button_detect(rt5682->component);
@@ -1608,8 +1609,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
0, set_filter_clk, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
- SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
- NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0),
/* ASRC */
@@ -2492,6 +2492,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw)
snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
RT5682_PWR_MB, RT5682_PWR_MB);
+
+ snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2");
+ snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+ RT5682_PWR_VREF2 | RT5682_PWR_FV2,
+ RT5682_PWR_VREF2);
+ usleep_range(55000, 60000);
+ snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+ RT5682_PWR_FV2, RT5682_PWR_FV2);
+
snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1");
snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F");
snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B");
@@ -2517,9 +2526,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw)
snd_soc_dapm_mutex_lock(dapm);
snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2");
if (!rt5682->jack_type)
snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+ RT5682_PWR_VREF2 | RT5682_PWR_FV2 |
RT5682_PWR_MB, 0);
+
snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1");
snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F");
snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B");
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 06ba36595ddd6..7cfc89602fc39 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0),
/* Boost mixer */
static const struct snd_kcontrol_new wm8974_boost_mixer[] = {
-SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0),
+SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1),
};
/* Input PGA */
@@ -474,6 +474,10 @@ static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai,
iface |= 0x0008;
break;
case SND_SOC_DAIFMT_DSP_A:
+ if ((fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_IB_IF ||
+ (fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_NB_IF) {
+ return -EINVAL;
+ }
iface |= 0x00018;
break;
default:
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 9ad35d9940fef..97b4f5480a31c 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -317,8 +317,8 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
if (ret < 0)
goto out_put_node;
- dai_link->dpcm_playback = 1;
- dai_link->dpcm_capture = 1;
+ snd_soc_dai_link_set_capabilities(dai_link);
+
dai_link->ops = &graph_ops;
dai_link->init = asoc_simple_dai_init;
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 55e9f8800b3e1..04d4d28ed5112 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -231,8 +231,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
if (ret < 0)
goto out_put_node;
- dai_link->dpcm_playback = 1;
- dai_link->dpcm_capture = 1;
+ snd_soc_dai_link_set_capabilities(dai_link);
+
dai_link->ops = &simple_ops;
dai_link->init = asoc_simple_dai_init;
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index 5f96d7ac0a226..bed4d5f73d9cf 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -354,6 +354,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = {
{
.name = "Codec DSP",
.stream_name = "Wake on Voice",
+ .capture_only = 1,
.ops = &bdw_rt5677_dsp_ops,
SND_SOC_DAILINK_REG(dsp),
},
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 9e5fc9430628c..ecbc58e8a37f5 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -543,8 +543,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
if (cnt) {
ret = device_add_properties(codec_dev, props);
- if (ret)
+ if (ret) {
+ put_device(codec_dev);
return ret;
+ }
}
devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios);
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 7a43c70a1378a..22e432768edb3 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -253,21 +253,20 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
/*
- * Default mode for SSP configuration is TDM 4 slot
+ * Default mode for SSP configuration is TDM 4 slot. One board/design,
+ * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The
+ * second piggy-backed, output-only codec is inside the keyboard-dock
+ * (which has extra speakers). Unlike the main rt5672 codec, we cannot
+ * configure this codec, it is hard coded to use 2 channel 24 bit I2S.
+ * Since we only support 2 channels anyways, there is no need for TDM
+ * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere.
*/
- ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0),
- SND_SOC_DAIFMT_DSP_B |
- SND_SOC_DAIFMT_IB_NF |
+ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0),
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0) {
- dev_err(rtd->dev, "can't set format to TDM %d\n", ret);
- return ret;
- }
-
- /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24);
- if (ret < 0) {
- dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
+ dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
return ret;
}
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index f51b28d1b94d8..92f51d0e9fe2b 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -72,7 +72,7 @@ config SND_SOC_QDSP6_ASM_DAI
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
- depends on QCOM_APR && HAS_DMA
+ depends on QCOM_APR
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_CORE
select SND_SOC_QDSP6_AFE
diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c
index f45e5aaa4b302..9539b0d024fed 100644
--- a/sound/soc/rockchip/rk3399_gru_sound.c
+++ b/sound/soc/rockchip/rk3399_gru_sound.c
@@ -219,19 +219,32 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+static int rockchip_sound_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ return snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
+ 8000, 96000);
+}
+
static const struct snd_soc_ops rockchip_sound_max98357a_ops = {
+ .startup = rockchip_sound_startup,
.hw_params = rockchip_sound_max98357a_hw_params,
};
static const struct snd_soc_ops rockchip_sound_rt5514_ops = {
+ .startup = rockchip_sound_startup,
.hw_params = rockchip_sound_rt5514_hw_params,
};
static const struct snd_soc_ops rockchip_sound_da7219_ops = {
+ .startup = rockchip_sound_startup,
.hw_params = rockchip_sound_da7219_hw_params,
};
static const struct snd_soc_ops rockchip_sound_dmic_ops = {
+ .startup = rockchip_sound_startup,
.hw_params = rockchip_sound_dmic_hw_params,
};
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0f30f5aabaa87..2b8abf88ec603 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -2572,6 +2572,33 @@ int snd_soc_register_component(struct device *dev,
}
EXPORT_SYMBOL_GPL(snd_soc_register_component);
+/**
+ * snd_soc_unregister_component_by_driver - Unregister component using a given driver
+ * from the ASoC core
+ *
+ * @dev: The device to unregister
+ * @component_driver: The component driver to unregister
+ */
+void snd_soc_unregister_component_by_driver(struct device *dev,
+ const struct snd_soc_component_driver *component_driver)
+{
+ struct snd_soc_component *component;
+
+ if (!component_driver)
+ return;
+
+ mutex_lock(&client_mutex);
+ component = snd_soc_lookup_component_nolocked(dev, component_driver->name);
+ if (!component)
+ goto out;
+
+ snd_soc_del_component_unlocked(component);
+
+out:
+ mutex_unlock(&client_mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_component_by_driver);
+
/**
* snd_soc_unregister_component - Unregister all related component
* from the ASoC core
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index b05e18b63a1c3..457159975b01a 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -391,6 +391,44 @@ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir)
return stream->channels_min;
}
+/*
+ * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs
+ */
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
+{
+ struct snd_soc_dai_link_component *cpu;
+ struct snd_soc_dai_link_component *codec;
+ struct snd_soc_dai *dai;
+ bool supported[SNDRV_PCM_STREAM_LAST + 1];
+ int direction;
+ int i;
+
+ for_each_pcm_streams(direction) {
+ supported[direction] = true;
+
+ for_each_link_cpus(dai_link, i, cpu) {
+ dai = snd_soc_find_dai(cpu);
+ if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+ supported[direction] = false;
+ break;
+ }
+ }
+ if (!supported[direction])
+ continue;
+ for_each_link_codecs(dai_link, i, codec) {
+ dai = snd_soc_find_dai(codec);
+ if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+ supported[direction] = false;
+ break;
+ }
+ }
+ }
+
+ dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK];
+ dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE];
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities);
+
void snd_soc_dai_action(struct snd_soc_dai *dai,
int stream, int action)
{
diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c
index 11e5d79623707..4534a1c03e8e5 100644
--- a/sound/soc/soc-devres.c
+++ b/sound/soc/soc-devres.c
@@ -48,7 +48,9 @@ EXPORT_SYMBOL_GPL(devm_snd_soc_register_dai);
static void devm_component_release(struct device *dev, void *res)
{
- snd_soc_unregister_component(*(struct device **)res);
+ const struct snd_soc_component_driver **cmpnt_drv = res;
+
+ snd_soc_unregister_component_by_driver(dev, *cmpnt_drv);
}
/**
@@ -65,7 +67,7 @@ int devm_snd_soc_register_component(struct device *dev,
const struct snd_soc_component_driver *cmpnt_drv,
struct snd_soc_dai_driver *dai_drv, int num_dai)
{
- struct device **ptr;
+ const struct snd_soc_component_driver **ptr;
int ret;
ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL);
@@ -74,7 +76,7 @@ int devm_snd_soc_register_component(struct device *dev,
ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai);
if (ret == 0) {
- *ptr = dev;
+ *ptr = cmpnt_drv;
devres_add(dev, ptr);
} else {
devres_free(ptr);
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 80a4e71f2d95d..61844403f1817 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -478,7 +478,7 @@ void snd_dmaengine_pcm_unregister(struct device *dev)
pcm = soc_component_to_pcm(component);
- snd_soc_unregister_component(dev);
+ snd_soc_unregister_component_by_driver(dev, component->driver);
dmaengine_pcm_release_chan(pcm);
kfree(pcm);
}
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 43e5745b06aa7..6eaa00c210117 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1261,17 +1261,29 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list);
ret = soc_tplg_add_route(tplg, routes[i]);
- if (ret < 0)
+ if (ret < 0) {
+ /*
+ * this route was added to the list, it will
+ * be freed in remove_route() so increment the
+ * counter to skip it in the error handling
+ * below.
+ */
+ i++;
break;
+ }
/* add route, but keep going if some fail */
snd_soc_dapm_add_routes(dapm, routes[i], 1);
}
- /* free memory allocated for all dapm routes in case of error */
- if (ret < 0)
- for (i = 0; i < count ; i++)
- kfree(routes[i]);
+ /*
+ * free memory allocated for all dapm routes not added to the
+ * list in case of error
+ */
+ if (ret < 0) {
+ while (i < count)
+ kfree(routes[i++]);
+ }
/*
* free pointer to array of dapm routes as this is no longer needed.
@@ -1359,7 +1371,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
if (err < 0) {
dev_err(tplg->dev, "ASoC: failed to init %s\n",
mc->hdr.name);
- soc_tplg_free_tlv(tplg, &kc[i]);
goto err_sm;
}
}
@@ -1367,6 +1378,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
err_sm:
for (; i >= 0; i--) {
+ soc_tplg_free_tlv(tplg, &kc[i]);
sm = (struct soc_mixer_control *)kc[i].private_value;
kfree(sm);
kfree(kc[i].name);
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 339c4930b0c0a..adc7c37145d64 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -345,15 +345,15 @@ int snd_sof_device_remove(struct device *dev)
struct snd_sof_pdata *pdata = sdev->pdata;
int ret;
- ret = snd_sof_dsp_power_down_notify(sdev);
- if (ret < 0)
- dev_warn(dev, "error: %d failed to prepare DSP for device removal",
- ret);
-
if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
cancel_work_sync(&sdev->probe_work);
if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) {
+ ret = snd_sof_dsp_power_down_notify(sdev);
+ if (ret < 0)
+ dev_warn(dev, "error: %d failed to prepare DSP for device removal",
+ ret);
+
snd_sof_fw_unload(sdev);
snd_sof_ipc_free(sdev);
snd_sof_free_debug(sdev);
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 63f9c20a1bacf..a4fa8451d8cb3 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -375,6 +375,14 @@ static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev,
static struct snd_soc_dai_driver imx8_dai[] = {
{
.name = "esai-port",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 8,
+ },
},
};
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index fa86a9e2990f8..287114a37688c 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -240,6 +240,14 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev,
static struct snd_soc_dai_driver imx8m_dai[] = {
{
.name = "sai-port",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 32,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 32,
+ },
},
};