From e958b5884725dac86d36c1e7afe5a55f31feb0b2 Mon Sep 17 00:00:00 2001 From: Robert Hancock Date: Fri, 7 Jan 2022 15:47:06 -0600 Subject: [PATCH 01/52] ASoC: xilinx: xlnx_formatter_pcm: Make buffer bytes multiple of period bytes This patch is based on one in the Xilinx kernel tree, "ASoc: xlnx: Make buffer bytes multiple of period bytes" by Devarsh Thakkar. The same issue exists in the mainline version of the driver. The original patch description is as follows: "The Xilinx Audio Formatter IP has a constraint on period bytes to be multiple of 64. This leads to driver changing the period size to suitable frames such that period bytes are multiple of 64. Now since period bytes and period size are updated but not the buffer bytes, this may make the buffer bytes unaligned and not multiple of period bytes. When this happens we hear popping noise as while DMA is being done the buffer bytes are not enough to complete DMA access for last period of frame within the application buffer boundary. To avoid this, align buffer bytes too as multiple of 64, and set another constraint to always enforce number of periods as integer. Now since, there is already a rule in alsa core to enforce Buffer size = Number of Periods * Period Size this automatically aligns buffer bytes as multiple of period bytes." Fixes: 6f6c3c36f091 ("ASoC: xlnx: add pcm formatter platform driver") Cc: Devarsh Thakkar Signed-off-by: Robert Hancock Link: https://lore.kernel.org/r/20220107214711.1100162-2-robert.hancock@calian.com Signed-off-by: Mark Brown --- sound/soc/xilinx/xlnx_formatter_pcm.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 91afea9d5de67..ce19a6058b279 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -37,6 +37,7 @@ #define XLNX_AUD_XFER_COUNT 0x28 #define XLNX_AUD_CH_STS_START 0x2C #define XLNX_BYTES_PER_CH 0x44 +#define XLNX_AUD_ALIGN_BYTES 64 #define AUD_STS_IOC_IRQ_MASK BIT(31) #define AUD_STS_CH_STS_MASK BIT(29) @@ -368,12 +369,32 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component, snd_soc_set_runtime_hwparams(substream, &xlnx_pcm_hardware); runtime->private_data = stream_data; - /* Resize the period size divisible by 64 */ + /* Resize the period bytes as divisible by 64 */ err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + XLNX_AUD_ALIGN_BYTES); if (err) { dev_err(component->dev, - "unable to set constraint on period bytes\n"); + "Unable to set constraint on period bytes\n"); + return err; + } + + /* Resize the buffer bytes as divisible by 64 */ + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + XLNX_AUD_ALIGN_BYTES); + if (err) { + dev_err(component->dev, + "Unable to set constraint on buffer bytes\n"); + return err; + } + + /* Set periods as integer multiple */ + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + dev_err(component->dev, + "Unable to set constraint on periods to be integer\n"); return err; } From a64067f4cecaaa4deed8e33d3266bc0bcc189142 Mon Sep 17 00:00:00 2001 From: Robert Hancock Date: Fri, 7 Jan 2022 15:47:10 -0600 Subject: [PATCH 02/52] ASoC: simple-card: fix probe failure on platform component A previous change to simple-card resulted in asoc_simple_parse_dai attempting to retrieve the dai_name for platform components, which are unlikely to have a valid DAI name. This caused simple-card to fail to probe when using the xlnx_formatter_pcm as the platform component, since it does not register any DAI components. Since the dai_name is not used for platform components, just skip trying to retrieve it for those. Fixes: f107294c6422 ("ASoC: simple-card: support snd_soc_dai_link_component style for cpu") Signed-off-by: Robert Hancock Link: https://lore.kernel.org/r/20220107214711.1100162-6-robert.hancock@calian.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index a89d1cfdda327..78419e18717d6 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -28,6 +28,30 @@ static const struct snd_soc_ops simple_ops = { .hw_params = asoc_simple_hw_params, }; +static int asoc_simple_parse_platform(struct device_node *node, + struct snd_soc_dai_link_component *dlc) +{ + struct of_phandle_args args; + int ret; + + if (!node) + return 0; + + /* + * Get node via "sound-dai = <&phandle port>" + * it will be used as xxx_of_node on soc_bind_dai_link() + */ + ret = of_parse_phandle_with_args(node, DAI, CELL, 0, &args); + if (ret) + return ret; + + /* dai_name is not required and may not exist for plat component */ + + dlc->of_node = args.np; + + return 0; +} + static int asoc_simple_parse_dai(struct device_node *node, struct snd_soc_dai_link_component *dlc, int *is_single_link) @@ -289,7 +313,7 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_parse_dai(plat, platforms, NULL); + ret = asoc_simple_parse_platform(plat, platforms); if (ret < 0) goto dai_link_of_err; From f7a6021aaf02088870559f82fc13c58cda7fea1a Mon Sep 17 00:00:00 2001 From: Jiasheng Jiang Date: Tue, 11 Jan 2022 10:50:48 +0800 Subject: [PATCH 03/52] ASoC: cpcap: Check for NULL pointer after calling of_get_child_by_name If the device does not exist, of_get_child_by_name() will return NULL pointer. And devm_snd_soc_register_component() does not check it. Also, I have noticed that cpcap_codec_driver has not been used yet. Therefore, it should be better to check it in order to avoid the future dereference of the NULL pointer. Fixes: f6cdf2d3445d ("ASoC: cpcap: new codec") Signed-off-by: Jiasheng Jiang Link: https://lore.kernel.org/r/20220111025048.524134-1-jiasheng@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/codecs/cpcap.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index 598e09024e239..ffdf8b615efaa 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1667,6 +1667,8 @@ static int cpcap_codec_probe(struct platform_device *pdev) { struct device_node *codec_node = of_get_child_by_name(pdev->dev.parent->of_node, "audio-codec"); + if (!codec_node) + return -ENODEV; pdev->dev.of_node = codec_node; From 4c907bcd9dcd233da6707059d777ab389dcbd964 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 19 Jan 2022 15:31:01 +0300 Subject: [PATCH 04/52] ASoC: max9759: fix underflow in speaker_gain_control_put() Check for negative values of "priv->gain" to prevent an out of bounds access. The concern is that these might come from the user via: -> snd_ctl_elem_write_user() -> snd_ctl_elem_write() -> kctl->put() Fixes: fa8d915172b8 ("ASoC: max9759: Add Amplifier Driver") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/20220119123101.GA9509@kili Signed-off-by: Mark Brown --- sound/soc/codecs/max9759.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max9759.c b/sound/soc/codecs/max9759.c index d75fd61b90321..bc57d7687f16a 100644 --- a/sound/soc/codecs/max9759.c +++ b/sound/soc/codecs/max9759.c @@ -64,7 +64,8 @@ static int speaker_gain_control_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); struct max9759 *priv = snd_soc_component_get_drvdata(c); - if (ucontrol->value.integer.value[0] > 3) + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 3) return -EINVAL; priv->gain = ucontrol->value.integer.value[0]; From 579b2c8f72d974f27d85bbd53846f34675ee3b01 Mon Sep 17 00:00:00 2001 From: Julian Braha Date: Mon, 17 Jan 2022 00:03:24 -0500 Subject: [PATCH 05/52] ASoC: mediatek: fix unmet dependency on GPIOLIB for SND_SOC_DMIC When SND_SOC_MT8195_MT6359_RT1011_RT5682 is selected, and GPIOLIB is not selected, Kbuild gives the following warning: WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8195_MT6359_RT1011_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8195 [=y] && MTK_PMIC_WRAP [=y] This is because SND_SOC_MT8195_MT6359_RT1011_RT5682 selects SND_SOC_DMIC without selecting or depending on GPIOLIB, depsite SND_SOC_DMIC depending on GPIOLIB. This unmet dependency bug was detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha Reviewed-by: Tzung-Bi Shih Link: https://lore.kernel.org/r/20220117050324.68371-1-julianbraha@gmail.com Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 9306b7ca26442..0d154350f180e 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -216,7 +216,7 @@ config SND_SOC_MT8195_MT6359_RT1019_RT5682 config SND_SOC_MT8195_MT6359_RT1011_RT5682 tristate "ASoC Audio driver for MT8195 with MT6359 RT1011 RT5682 codec" - depends on I2C + depends on I2C && GPIOLIB depends on SND_SOC_MT8195 && MTK_PMIC_WRAP select SND_SOC_MT6359 select SND_SOC_RT1011 From 248be352bbae1a0f14d0d3511a5b0bb9665097f5 Mon Sep 17 00:00:00 2001 From: Ajit Kumar Pandey Date: Thu, 20 Jan 2022 19:06:01 +0530 Subject: [PATCH 06/52] ASoC: amd: acp-mach: Fix Left and Right rt1019 amp devices We're setting wrong card codec conf for rt1019 amp devices in our machine driver. Due to this left and right amp channels data are reversed in our machines as wrong device prefix results in wrong value for "Mono LR Select" rt1019 mixer control. Reverse dev ids in codec conf with Left and Right name_prefix to fix such issue. Signed-off-by: Ajit Kumar Pandey Link: https://lore.kernel.org/r/20220120133605.476138-1-AjitKumar.Pandey@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index c9caade5cb746..cd05ee2802c9e 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -303,11 +303,11 @@ static const struct snd_soc_dapm_route rt1019_map_lr[] = { static struct snd_soc_codec_conf rt1019_conf[] = { { - .dlc = COMP_CODEC_CONF("i2c-10EC1019:00"), + .dlc = COMP_CODEC_CONF("i2c-10EC1019:01"), .name_prefix = "Left", }, { - .dlc = COMP_CODEC_CONF("i2c-10EC1019:01"), + .dlc = COMP_CODEC_CONF("i2c-10EC1019:00"), .name_prefix = "Right", }, }; From 817f7c9335ec01e0f5e8caffc4f1dcd5e458a4c0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Jan 2022 15:32:51 +0000 Subject: [PATCH 07/52] ASoC: ops: Reject out of bounds values in snd_soc_put_volsw() We don't currently validate that the values being set are within the range we advertised to userspace as being valid, do so and reject any values that are out of range. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220124153253.3548853-2-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 08eaa9ddf191e..fbe5d326b0f2d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -316,13 +316,27 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, if (sign_bit) mask = BIT(sign_bit + 1) - 1; - val = ((ucontrol->value.integer.value[0] + min) & mask); + val = ucontrol->value.integer.value[0]; + if (mc->platform_max && val > mc->platform_max) + return -EINVAL; + if (val > max - min) + return -EINVAL; + if (val < 0) + return -EINVAL; + val = (val + min) & mask; if (invert) val = max - val; val_mask = mask << shift; val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { - val2 = ((ucontrol->value.integer.value[1] + min) & mask); + val2 = ucontrol->value.integer.value[1]; + if (mc->platform_max && val2 > mc->platform_max) + return -EINVAL; + if (val2 > max - min) + return -EINVAL; + if (val2 < 0) + return -EINVAL; + val2 = (val2 + min) & mask; if (invert) val2 = max - val2; if (reg == reg2) { From 4f1e50d6a9cf9c1b8c859d449b5031cacfa8404e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Jan 2022 15:32:52 +0000 Subject: [PATCH 08/52] ASoC: ops: Reject out of bounds values in snd_soc_put_volsw_sx() We don't currently validate that the values being set are within the range we advertised to userspace as being valid, do so and reject any values that are out of range. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220124153253.3548853-3-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index fbe5d326b0f2d..c31e63b27193c 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -423,8 +423,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int err = 0; unsigned int val, val_mask; + val = ucontrol->value.integer.value[0]; + if (mc->platform_max && val > mc->platform_max) + return -EINVAL; + if (val > max - min) + return -EINVAL; + if (val < 0) + return -EINVAL; val_mask = mask << shift; - val = (ucontrol->value.integer.value[0] + min) & mask; + val = (val + min) & mask; val = val << shift; err = snd_soc_component_update_bits(component, reg, val_mask, val); From 4cf28e9ae6e2e11a044be1bcbcfa1b0d8675fe4d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Jan 2022 15:32:53 +0000 Subject: [PATCH 09/52] ASoC: ops: Reject out of bounds values in snd_soc_put_xr_sx() We don't currently validate that the values being set are within the range we advertised to userspace as being valid, do so and reject any values that are out of range. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220124153253.3548853-4-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index c31e63b27193c..dc0e7c8d31f37 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -879,6 +879,8 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, long val = ucontrol->value.integer.value[0]; unsigned int i; + if (val < mc->min || val > mc->max) + return -EINVAL; if (invert) val = max - val; val &= mask; From c5c1546a654f613e291a7c5d6f3660fc1eb6d0c7 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 26 Jan 2022 11:35:46 +0000 Subject: [PATCH 10/52] ASoC: codecs: wcd938x: fix incorrect used of portid Mixer controls have the channel id in mixer->reg, which is not same as port id. port id should be derived from chan_info array. So fix this. Without this, its possible that we could corrupt struct wcd938x_sdw_priv by accessing port_map array out of range with channel id instead of port id. Fixes: e8ba1e05bdc0 ("ASoC: codecs: wcd938x: add basic controls") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220126113549.8853-2-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 25 ++++++++++++++----------- 1 file changed, 14 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index eff200a07d9f4..5994644c8702f 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -1432,14 +1432,10 @@ static int wcd938x_sdw_connect_port(struct wcd938x_sdw_ch_info *ch_info, return 0; } -static int wcd938x_connect_port(struct wcd938x_sdw_priv *wcd, u8 ch_id, u8 enable) +static int wcd938x_connect_port(struct wcd938x_sdw_priv *wcd, u8 port_num, u8 ch_id, u8 enable) { - u8 port_num; - - port_num = wcd->ch_info[ch_id].port_num; - return wcd938x_sdw_connect_port(&wcd->ch_info[ch_id], - &wcd->port_config[port_num], + &wcd->port_config[port_num - 1], enable); } @@ -2593,6 +2589,7 @@ static int wcd938x_set_compander(struct snd_kcontrol *kcontrol, struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); struct wcd938x_sdw_priv *wcd; int value = ucontrol->value.integer.value[0]; + int portidx; struct soc_mixer_control *mc; bool hphr; @@ -2606,10 +2603,12 @@ static int wcd938x_set_compander(struct snd_kcontrol *kcontrol, else wcd938x->comp1_enable = value; + portidx = wcd->ch_info[mc->reg].port_num; + if (value) - wcd938x_connect_port(wcd, mc->reg, true); + wcd938x_connect_port(wcd, portidx, mc->reg, true); else - wcd938x_connect_port(wcd, mc->reg, false); + wcd938x_connect_port(wcd, portidx, mc->reg, false); return 0; } @@ -2882,9 +2881,11 @@ static int wcd938x_get_swr_port(struct snd_kcontrol *kcontrol, struct wcd938x_sdw_priv *wcd; struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value; int dai_id = mixer->shift; - int portidx = mixer->reg; + int portidx, ch_idx = mixer->reg; + wcd = wcd938x->sdw_priv[dai_id]; + portidx = wcd->ch_info[ch_idx].port_num; ucontrol->value.integer.value[0] = wcd->port_enable[portidx]; @@ -2899,12 +2900,14 @@ static int wcd938x_set_swr_port(struct snd_kcontrol *kcontrol, struct wcd938x_sdw_priv *wcd; struct soc_mixer_control *mixer = (struct soc_mixer_control *)kcontrol->private_value; - int portidx = mixer->reg; + int ch_idx = mixer->reg; + int portidx; int dai_id = mixer->shift; bool enable; wcd = wcd938x->sdw_priv[dai_id]; + portidx = wcd->ch_info[ch_idx].port_num; if (ucontrol->value.integer.value[0]) enable = true; else @@ -2912,7 +2915,7 @@ static int wcd938x_set_swr_port(struct snd_kcontrol *kcontrol, wcd->port_enable[portidx] = enable; - wcd938x_connect_port(wcd, portidx, enable); + wcd938x_connect_port(wcd, portidx, ch_idx, enable); return 0; From fca041a3ab70a099a6d5519ecb689b6279bd04f3 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 26 Jan 2022 11:35:47 +0000 Subject: [PATCH 11/52] ASoC: codecs: lpass-rx-macro: fix sidetone register offsets For some reason we ended up with incorrect register offfset calcuations for sidetone. regmap clearly throw errors when accessing these incorrect registers as these do not belong to any read/write ranges. so fix them to point to correct register offsets. Fixes: f3ce6f3c9a99 ("ASoC: codecs: lpass-rx-macro: add iir widgets") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220126113549.8853-3-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-rx-macro.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index aec5127260fd4..6ffe88345de5f 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2688,8 +2688,8 @@ static uint32_t get_iir_band_coeff(struct snd_soc_component *component, int reg, b2_reg; /* Address does not automatically update if reading */ - reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B1_CTL + 16 * iir_idx; - b2_reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B2_CTL + 16 * iir_idx; + reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B1_CTL + 0x80 * iir_idx; + b2_reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B2_CTL + 0x80 * iir_idx; snd_soc_component_write(component, reg, ((band_idx * BAND_MAX + coeff_idx) * @@ -2718,7 +2718,7 @@ static uint32_t get_iir_band_coeff(struct snd_soc_component *component, static void set_iir_band_coeff(struct snd_soc_component *component, int iir_idx, int band_idx, uint32_t value) { - int reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B2_CTL + 16 * iir_idx; + int reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B2_CTL + 0x80 * iir_idx; snd_soc_component_write(component, reg, (value & 0xFF)); snd_soc_component_write(component, reg, (value >> 8) & 0xFF); @@ -2739,7 +2739,7 @@ static int rx_macro_put_iir_band_audio_mixer( int iir_idx = ctl->iir_idx; int band_idx = ctl->band_idx; u32 coeff[BAND_MAX]; - int reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B1_CTL + 16 * iir_idx; + int reg = CDC_RX_SIDETONE_IIR0_IIR_COEF_B1_CTL + 0x80 * iir_idx; memcpy(&coeff[0], ucontrol->value.bytes.data, params->max); From bd2347fd67d8da0fa76296507cc556da0a233bcb Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 26 Jan 2022 11:35:48 +0000 Subject: [PATCH 12/52] ASoC: codecs: wcd938x: fix return value of mixer put function wcd938x_ear_pa_put_gain, wcd938x_set_swr_port and wcd938x_set_compander currently returns zero eventhough it changes the value. Fix this, so that change notifications are sent correctly. Fixes: e8ba1e05bdc01 ("ASoC: codecs: wcd938x: add basic controls") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220126113549.8853-4-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 5994644c8702f..36cbc66914f90 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -2559,7 +2559,7 @@ static int wcd938x_ear_pa_put_gain(struct snd_kcontrol *kcontrol, WCD938X_EAR_GAIN_MASK, ucontrol->value.integer.value[0]); - return 0; + return 1; } static int wcd938x_get_compander(struct snd_kcontrol *kcontrol, @@ -2610,7 +2610,7 @@ static int wcd938x_set_compander(struct snd_kcontrol *kcontrol, else wcd938x_connect_port(wcd, portidx, mc->reg, false); - return 0; + return 1; } static int wcd938x_ldoh_get(struct snd_kcontrol *kcontrol, @@ -2917,7 +2917,7 @@ static int wcd938x_set_swr_port(struct snd_kcontrol *kcontrol, wcd938x_connect_port(wcd, portidx, ch_idx, enable); - return 0; + return 1; } From 8f2e5c65ec7534cce6d315fccf2c3aef023f68f0 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 26 Jan 2022 11:35:49 +0000 Subject: [PATCH 13/52] ASoC: qdsp6: q6apm-dai: only stop graphs that are started Its possible that the sound card is just opened and closed without actually playing stream, ex: if the audio file itself is missing. Even in such cases we do call stop on graphs that are not yet started. DSP can throw errors in such cases, so add a check to see if the graph was started before stopping it. Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220126113549.8853-5-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6apm-dai.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index eb1c3aec479ba..19c4a90ec1ea9 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -308,8 +308,11 @@ static int q6apm_dai_close(struct snd_soc_component *component, struct snd_pcm_runtime *runtime = substream->runtime; struct q6apm_dai_rtd *prtd = runtime->private_data; - q6apm_graph_stop(prtd->graph); - q6apm_unmap_memory_regions(prtd->graph, substream->stream); + if (prtd->state) { /* only stop graph that is started */ + q6apm_graph_stop(prtd->graph); + q6apm_unmap_memory_regions(prtd->graph, substream->stream); + } + q6apm_graph_close(prtd->graph); prtd->graph = NULL; kfree(prtd); From 549f8ffc7b2f7561bea7f90930b6c5104318e87b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 26 Jan 2022 15:50:11 +0100 Subject: [PATCH 14/52] ALSA: hda: Fix UAF of leds class devs at unbinding The LED class devices that are created by HD-audio codec drivers are registered via devm_led_classdev_register() and associated with the HD-audio codec device. Unfortunately, it turned out that the devres release doesn't work for this case; namely, since the codec resource release happens before the devm call chain, it triggers a NULL dereference or a UAF for a stale set_brightness_delay callback. For fixing the bug, this patch changes the LED class device register and unregister in a manual manner without devres, keeping the instances in hda_gen_spec. Reported-by: Alexander Sergeyev Cc: Link: https://lore.kernel.org/r/20220111195229.a77wrpjclqwrx4bx@localhost.localdomain Link: https://lore.kernel.org/r/20220126145011.16728-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 17 +++++++++++++++-- sound/pci/hda/hda_generic.h | 3 +++ 2 files changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3bf5e34107038..fc114e5224806 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -91,6 +91,12 @@ static void snd_hda_gen_spec_free(struct hda_gen_spec *spec) free_kctls(spec); snd_array_free(&spec->paths); snd_array_free(&spec->loopback_list); +#ifdef CONFIG_SND_HDA_GENERIC_LEDS + if (spec->led_cdevs[LED_AUDIO_MUTE]) + led_classdev_unregister(spec->led_cdevs[LED_AUDIO_MUTE]); + if (spec->led_cdevs[LED_AUDIO_MICMUTE]) + led_classdev_unregister(spec->led_cdevs[LED_AUDIO_MICMUTE]); +#endif } /* @@ -3922,7 +3928,10 @@ static int create_mute_led_cdev(struct hda_codec *codec, enum led_brightness), bool micmute) { + struct hda_gen_spec *spec = codec->spec; struct led_classdev *cdev; + int idx = micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE; + int err; cdev = devm_kzalloc(&codec->core.dev, sizeof(*cdev), GFP_KERNEL); if (!cdev) @@ -3932,10 +3941,14 @@ static int create_mute_led_cdev(struct hda_codec *codec, cdev->max_brightness = 1; cdev->default_trigger = micmute ? "audio-micmute" : "audio-mute"; cdev->brightness_set_blocking = callback; - cdev->brightness = ledtrig_audio_get(micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE); + cdev->brightness = ledtrig_audio_get(idx); cdev->flags = LED_CORE_SUSPENDRESUME; - return devm_led_classdev_register(&codec->core.dev, cdev); + err = led_classdev_register(&codec->core.dev, cdev); + if (err < 0) + return err; + spec->led_cdevs[idx] = cdev; + return 0; } /** diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 8e1bc8ea74fc3..34eba40cc6e67 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -294,6 +294,9 @@ struct hda_gen_spec { struct hda_jack_callback *cb); void (*mic_autoswitch_hook)(struct hda_codec *codec, struct hda_jack_callback *cb); + + /* leds */ + struct led_classdev *led_cdevs[NUM_AUDIO_LEDS]; }; /* values for add_stereo_mix_input flag */ From 3da4b7403db87d39bc2613cfd790de1de99a70ab Mon Sep 17 00:00:00 2001 From: Tom Rix Date: Wed, 26 Jan 2022 10:21:42 -0800 Subject: [PATCH 15/52] ALSA: usb-audio: initialize variables that could ignore errors clang static analysis reports this representative issue mixer.c:1548:35: warning: Assigned value is garbage or undefined ucontrol->value.integer.value[0] = val; ^ ~~~ The filter_error() macro allows errors to be ignored. If errors can be ignored, initialize variables so garbage will not be used. Fixes: 48cc42973509 ("ALSA: usb-audio: Filter error from connector kctl ops, too") Signed-off-by: Tom Rix Link: https://lore.kernel.org/r/20220126182142.1184819-1-trix@redhat.com Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e8f3f8d622ec5..630766ba259fd 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1527,6 +1527,10 @@ static int get_connector_value(struct usb_mixer_elem_info *cval, usb_audio_err(chip, "cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", UAC_GET_CUR, validx, idx, cval->val_type); + + if (val) + *val = 0; + return filter_error(cval, ret); } From 0444f82766f0b5b9c8302ad802dafa5dd0e722d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 27 Jan 2022 14:57:17 +0100 Subject: [PATCH 16/52] ALSA: hda: Fix signedness of sscanf() arguments The %x format of sscanf() takes an unsigned int pointer, while we pass a signed int pointer. Practically it's OK, but this may result in a compile warning. Let's fix it. Fixes: a235d5b8e550 ("ALSA: hda: Allow model option to specify PCI SSID alias") Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220127135717.31751-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 82c492b056671..cd1db943b7e07 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -981,7 +981,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, int id = HDA_FIXUP_ID_NOT_SET; const char *name = NULL; const char *type = NULL; - int vendor, device; + unsigned int vendor, device; if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) return; From fb25621da5702c104ce0a48de5b174ced09e5b4e Mon Sep 17 00:00:00 2001 From: Miaoqian Lin Date: Thu, 27 Jan 2022 13:13:34 +0000 Subject: [PATCH 17/52] ASoC: fsl: Add missing error handling in pcm030_fabric_probe Add the missing platform_device_put() and platform_device_del() before return from pcm030_fabric_probe in the error handling case. Fixes: c912fa913446 ("ASoC: fsl: register the wm9712-codec") Signed-off-by: Miaoqian Lin Link: https://lore.kernel.org/r/20220127131336.30214-1-linmq006@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index af3c3b90c0aca..83b4a22bf15ac 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -93,16 +93,21 @@ static int pcm030_fabric_probe(struct platform_device *op) dev_err(&op->dev, "platform_device_alloc() failed\n"); ret = platform_device_add(pdata->codec_device); - if (ret) + if (ret) { dev_err(&op->dev, "platform_device_add() failed: %d\n", ret); + platform_device_put(pdata->codec_device); + } ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); + platform_device_del(pdata->codec_device); + platform_device_put(pdata->codec_device); + } platform_set_drvdata(op, pdata); - return ret; + } static int pcm030_fabric_remove(struct platform_device *op) From 3c75c0ea5da749bd1efebd1387f2e5011b8c7d78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 19 Jan 2022 16:52:48 +0100 Subject: [PATCH 18/52] ASoC: soc-pcm: Fix DPCM lockdep warning due to nested stream locks The recent change for DPCM locking caused spurious lockdep warnings. Actually the warnings are false-positive, as those are triggered due to the nested stream locks for FE and BE. Since both locks belong to the same lock class, lockdep sees it as if a deadlock. For fixing this, we need to take PCM stream locks for BE with the nested lock primitives. Since currently snd_pcm_stream_lock*() helper assumes only the top-level single locking, a new helper function snd_pcm_stream_lock_irqsave_nested() is defined for a single-depth nested lock, which is now used in the BE DAI trigger that is always performed inside a FE stream lock. Fixes: b2ae80663008 ("ASoC: soc-pcm: serialize BE triggers") Reported-and-tested-by: Hans de Goede Reported-and-tested-by: Marek Szyprowski Link: https://lore.kernel.org/r/73018f3c-9769-72ea-0325-b3f8e2381e30@redhat.com Link: https://lore.kernel.org/alsa-devel/9a0abddd-49e9-872d-2f00-a1697340f786@samsung.com Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220119155249.26754-2-tiwai@suse.de Signed-off-by: Mark Brown --- include/sound/pcm.h | 15 +++++++++++++++ sound/core/pcm_native.c | 13 +++++++++++++ sound/soc/soc-pcm.c | 6 +++--- 3 files changed, 31 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 33451f8ff755b..524220fe1af6c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -614,6 +614,7 @@ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream); void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream); void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream); unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream); +unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream); /** * snd_pcm_stream_lock_irqsave - Lock the PCM stream @@ -632,6 +633,20 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream); void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, unsigned long flags); +/** + * snd_pcm_stream_lock_irqsave_nested - Single-nested PCM stream locking + * @substream: PCM substream + * @flags: irq flags + * + * This locks the PCM stream like snd_pcm_stream_lock_irqsave() but with + * the single-depth lockdep subclass. + */ +#define snd_pcm_stream_lock_irqsave_nested(substream, flags) \ + do { \ + typecheck(unsigned long, flags); \ + flags = _snd_pcm_stream_lock_irqsave_nested(substream); \ + } while (0) + /** * snd_pcm_group_for_each_entry - iterate over the linked substreams * @s: the iterator diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 621883e711949..a056b3ef3c843 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -172,6 +172,19 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); +unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream) +{ + unsigned long flags = 0; + if (substream->pcm->nonatomic) + mutex_lock_nested(&substream->self_group.mutex, + SINGLE_DEPTH_NESTING); + else + spin_lock_irqsave_nested(&substream->self_group.lock, flags, + SINGLE_DEPTH_NESTING); + return flags; +} +EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave_nested); + /** * snd_pcm_stream_unlock_irqrestore - Unlock the PCM stream * @substream: PCM substream diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7abfc48b26ca5..e8876e65c6496 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -46,8 +46,8 @@ static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd, snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream)); } -#define snd_soc_dpcm_stream_lock_irqsave(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave(snd_soc_dpcm_get_substream(rtd, stream), flags) +#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ + snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd, int stream) @@ -2094,7 +2094,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave(be, stream, flags); + snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) From 9f620684c1ef5a002b6622ecc7b5818e81252f48 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 19 Jan 2022 16:52:49 +0100 Subject: [PATCH 19/52] ASoC: soc-pcm: Move debugfs removal out of spinlock The recent fix for DPCM locking also covered the loop in dpcm_be_disconnect() with the FE stream lock. This caused an unexpected side effect, thought: calling debugfs_remove_recursive() in the spinlock may lead to lockdep splats as the code there assumes the SOFTIRQ-safe context. For avoiding the problem, this patch changes the disconnection procedure to two phases: at first, the matching entries are removed from the linked list, then the resources are freed outside the lock. Fixes: b7898396f4bb ("ASoC: soc-pcm: Fix and cleanup DPCM locking") Reported-and-tested-by: Marek Szyprowski Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/20220119155249.26754-3-tiwai@suse.de Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e8876e65c6496..9a954680d4928 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1268,6 +1268,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; + LIST_HEAD(deleted_dpcms); snd_soc_dpcm_mutex_assert_held(fe); @@ -1287,13 +1288,18 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) /* BEs still alive need new FE */ dpcm_be_reparent(fe, dpcm->be, stream); - dpcm_remove_debugfs_state(dpcm); - list_del(&dpcm->list_be); + list_move(&dpcm->list_fe, &deleted_dpcms); + } + snd_soc_dpcm_stream_unlock_irq(fe, stream); + + while (!list_empty(&deleted_dpcms)) { + dpcm = list_first_entry(&deleted_dpcms, struct snd_soc_dpcm, + list_fe); list_del(&dpcm->list_fe); + dpcm_remove_debugfs_state(dpcm); kfree(dpcm); } - snd_soc_dpcm_stream_unlock_irq(fe, stream); } /* get BE for DAI widget and stream */ From 06feec6005c9d9500cd286ec440aabf8b2ddd94d Mon Sep 17 00:00:00 2001 From: Dmitry Osipenko Date: Wed, 12 Jan 2022 22:50:39 +0300 Subject: [PATCH 20/52] ASoC: hdmi-codec: Fix OOB memory accesses Correct size of iec_status array by changing it to the size of status array of the struct snd_aes_iec958. This fixes out-of-bounds slab read accesses made by memcpy() of the hdmi-codec driver. This problem is reported by KASAN. Cc: stable@vger.kernel.org Signed-off-by: Dmitry Osipenko Link: https://lore.kernel.org/r/20220112195039.1329-1-digetx@gmail.com Signed-off-by: Mark Brown --- include/uapi/sound/asound.h | 4 +++- sound/soc/codecs/hdmi-codec.c | 2 +- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index ff7e638221c53..228279ea0670c 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -56,8 +56,10 @@ * * ****************************************************************************/ +#define AES_IEC958_STATUS_SIZE 24 + struct snd_aes_iec958 { - unsigned char status[24]; /* AES/IEC958 channel status bits */ + unsigned char status[AES_IEC958_STATUS_SIZE]; /* AES/IEC958 channel status bits */ unsigned char subcode[147]; /* AES/IEC958 subcode bits */ unsigned char pad; /* nothing */ unsigned char dig_subframe[4]; /* AES/IEC958 subframe bits */ diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b61f980cabdc0..b07607a9ecea4 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -277,7 +277,7 @@ struct hdmi_codec_priv { bool busy; struct snd_soc_jack *jack; unsigned int jack_status; - u8 iec_status[5]; + u8 iec_status[AES_IEC958_STATUS_SIZE]; }; static const struct snd_soc_dapm_widget hdmi_widgets[] = { From 4045daf0fa87846a27f56329fddad2deeb5ca354 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 26 Jan 2022 12:03:25 +0200 Subject: [PATCH 21/52] ASoC: rt5682: Fix deadlock on resume On resume from suspend the following chain of events can happen: A rt5682_resume() -> mod_delayed_work() for jack_detect_work B DAPM sequence starts ( DAPM is locked now) A1. rt5682_jack_detect_handler() scheduled - Takes both jdet_mutex and calibrate_mutex - Calls in to rt5682_headset_detect() which tries to take DAPM lock, it starts to wait for it as B path took it already. B1. DAPM sequence reaches the "HP Amp", rt5682_hp_event() tries to take the jdet_mutex, but it is locked in A1, so it waits. Deadlock. To solve the deadlock, drop the jdet_mutex, use the jack_detect_work to do the jack removal handling, move the dapm lock up one level to protect the most of the rt5682_jack_detect_handler(), but not the jack reporting as it might trigger a DAPM sequence. The rt5682_headset_detect() can be changed to static as well. Fixes: 8deb34a90f063 ("ASoC: rt5682: fix the wrong jack type detected") Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20220126100325.16513-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682-i2c.c | 15 ++++----------- sound/soc/codecs/rt5682.c | 24 ++++++++---------------- sound/soc/codecs/rt5682.h | 2 -- 3 files changed, 12 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 20e0f90ea4986..20fc0f3766ded 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -59,18 +59,12 @@ static void rt5682_jd_check_handler(struct work_struct *work) struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, jd_check_work.work); - if (snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL) - & RT5682_JDH_RS_MASK) { + if (snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL) & RT5682_JDH_RS_MASK) /* jack out */ - rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); - - snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, - SND_JACK_HEADSET | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3); - } else { + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, 0); + else schedule_delayed_work(&rt5682->jd_check_work, 500); - } } static irqreturn_t rt5682_irq(int irq, void *data) @@ -198,7 +192,6 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } mutex_init(&rt5682->calibrate_mutex); - mutex_init(&rt5682->jdet_mutex); rt5682_calibrate(rt5682); rt5682_apply_patch_list(rt5682, &i2c->dev); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 415ec564c82e2..0a0ec4a021e11 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -922,15 +922,13 @@ static void rt5682_enable_push_button_irq(struct snd_soc_component *component, * * Returns detect status. */ -int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) +static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = &component->dapm; unsigned int val, count; if (jack_insert) { - snd_soc_dapm_mutex_lock(dapm); - snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2 | RT5682_PWR_MB, RT5682_PWR_VREF2 | RT5682_PWR_MB); @@ -981,8 +979,6 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) snd_soc_component_update_bits(component, RT5682_MICBIAS_2, RT5682_PWR_CLK25M_MASK | RT5682_PWR_CLK1M_MASK, RT5682_PWR_CLK25M_PU | RT5682_PWR_CLK1M_PU); - - snd_soc_dapm_mutex_unlock(dapm); } else { rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, @@ -1011,7 +1007,6 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type); return rt5682->jack_type; } -EXPORT_SYMBOL_GPL(rt5682_headset_detect); static int rt5682_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) @@ -1094,6 +1089,7 @@ void rt5682_jack_detect_handler(struct work_struct *work) { struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, jack_detect_work.work); + struct snd_soc_dapm_context *dapm; int val, btn_type; while (!rt5682->component) @@ -1102,7 +1098,9 @@ void rt5682_jack_detect_handler(struct work_struct *work) while (!rt5682->component->card->instantiated) usleep_range(10000, 15000); - mutex_lock(&rt5682->jdet_mutex); + dapm = snd_soc_component_get_dapm(rt5682->component); + + snd_soc_dapm_mutex_lock(dapm); mutex_lock(&rt5682->calibrate_mutex); val = snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL) @@ -1162,6 +1160,9 @@ void rt5682_jack_detect_handler(struct work_struct *work) rt5682->irq_work_delay_time = 50; } + mutex_unlock(&rt5682->calibrate_mutex); + snd_soc_dapm_mutex_unlock(dapm); + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | @@ -1174,9 +1175,6 @@ void rt5682_jack_detect_handler(struct work_struct *work) else cancel_delayed_work_sync(&rt5682->jd_check_work); } - - mutex_unlock(&rt5682->calibrate_mutex); - mutex_unlock(&rt5682->jdet_mutex); } EXPORT_SYMBOL_GPL(rt5682_jack_detect_handler); @@ -1526,7 +1524,6 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -1538,17 +1535,12 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_DEPOP_1, 0x60, 0x60); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); - - mutex_lock(&rt5682->jdet_mutex); - snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN); usleep_range(5000, 10000); snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_L); - - mutex_unlock(&rt5682->jdet_mutex); break; case SND_SOC_DAPM_POST_PMD: diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index c917c76200ea2..52ff0d9c36c58 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1463,7 +1463,6 @@ struct rt5682_priv { int jack_type; int irq_work_delay_time; - struct mutex jdet_mutex; }; extern const char *rt5682_supply_names[RT5682_NUM_SUPPLIES]; @@ -1473,7 +1472,6 @@ int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev); -int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert); void rt5682_jack_detect_handler(struct work_struct *work); bool rt5682_volatile_register(struct device *dev, unsigned int reg); From 1601033da2dd2052e0489137f7788a46a8fcd82f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 28 Jan 2022 19:24:43 +0000 Subject: [PATCH 22/52] ASoC: ops: Check for negative values before reading them The controls allow inputs to be specified as negative but our manipulating them into register fields need to be done on unsigned variables so the checks for negative numbers weren't taking effect properly. Do the checks for negative values on the variable in the ABI struct rather than on our local unsigned copy. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220128192443.3504823-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index dc0e7c8d31f37..9833611b83d1f 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -316,26 +316,26 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, if (sign_bit) mask = BIT(sign_bit + 1) - 1; + if (ucontrol->value.integer.value[0] < 0) + return -EINVAL; val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; - if (val < 0) - return -EINVAL; val = (val + min) & mask; if (invert) val = max - val; val_mask = mask << shift; val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { + if (ucontrol->value.integer.value[1] < 0) + return -EINVAL; val2 = ucontrol->value.integer.value[1]; if (mc->platform_max && val2 > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; - if (val2 < 0) - return -EINVAL; val2 = (val2 + min) & mask; if (invert) val2 = max - val2; @@ -423,13 +423,13 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int err = 0; unsigned int val, val_mask; + if (ucontrol->value.integer.value[0] < 0) + return -EINVAL; val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; - if (val < 0) - return -EINVAL; val_mask = mask << shift; val = (val + min) & mask; val = val << shift; From b837a9f5ab3bdfab9233c9f98a6bef717673a3e5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 31 Jan 2022 08:57:38 +0100 Subject: [PATCH 23/52] ALSA: hda: realtek: Fix race at concurrent COEF updates The COEF access is done with two steps: setting the index then read or write the data. When multiple COEF accesses are performed concurrently, the index and data might be paired unexpectedly. In most cases, this isn't a big problem as the COEF setup is done at the initialization, but some dynamic changes like the mute LED may hit such a race. For avoiding the racy COEF accesses, this patch introduces a new mutex coef_mutex to alc_spec, and wrap the COEF accessing functions with it. Reported-by: Alexander Sergeyev Cc: Link: https://lore.kernel.org/r/20220111195229.a77wrpjclqwrx4bx@localhost.localdomain Link: https://lore.kernel.org/r/20220131075738.24323-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 61 ++++++++++++++++++++++++++++------- 1 file changed, 50 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 668274e526745..a5677be0a4055 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -98,6 +98,7 @@ struct alc_spec { unsigned int gpio_mic_led_mask; struct alc_coef_led mute_led_coef; struct alc_coef_led mic_led_coef; + struct mutex coef_mutex; hda_nid_t headset_mic_pin; hda_nid_t headphone_mic_pin; @@ -137,8 +138,8 @@ struct alc_spec { * COEF access helper functions */ -static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, - unsigned int coef_idx) +static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx) { unsigned int val; @@ -147,28 +148,61 @@ static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, return val; } +static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + + mutex_lock(&spec->coef_mutex); + val = __alc_read_coefex_idx(codec, nid, coef_idx); + mutex_unlock(&spec->coef_mutex); + return val; +} + #define alc_read_coef_idx(codec, coef_idx) \ alc_read_coefex_idx(codec, 0x20, coef_idx) -static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, - unsigned int coef_idx, unsigned int coef_val) +static void __alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int coef_val) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, coef_idx); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PROC_COEF, coef_val); } +static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int coef_val) +{ + struct alc_spec *spec = codec->spec; + + mutex_lock(&spec->coef_mutex); + __alc_write_coefex_idx(codec, nid, coef_idx, coef_val); + mutex_unlock(&spec->coef_mutex); +} + #define alc_write_coef_idx(codec, coef_idx, coef_val) \ alc_write_coefex_idx(codec, 0x20, coef_idx, coef_val) +static void __alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int mask, + unsigned int bits_set) +{ + unsigned int val = __alc_read_coefex_idx(codec, nid, coef_idx); + + if (val != -1) + __alc_write_coefex_idx(codec, nid, coef_idx, + (val & ~mask) | bits_set); +} + static void alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx, unsigned int mask, unsigned int bits_set) { - unsigned int val = alc_read_coefex_idx(codec, nid, coef_idx); + struct alc_spec *spec = codec->spec; - if (val != -1) - alc_write_coefex_idx(codec, nid, coef_idx, - (val & ~mask) | bits_set); + mutex_lock(&spec->coef_mutex); + __alc_update_coefex_idx(codec, nid, coef_idx, mask, bits_set); + mutex_unlock(&spec->coef_mutex); } #define alc_update_coef_idx(codec, coef_idx, mask, bits_set) \ @@ -201,13 +235,17 @@ struct coef_fw { static void alc_process_coef_fw(struct hda_codec *codec, const struct coef_fw *fw) { + struct alc_spec *spec = codec->spec; + + mutex_lock(&spec->coef_mutex); for (; fw->nid; fw++) { if (fw->mask == (unsigned short)-1) - alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val); + __alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val); else - alc_update_coefex_idx(codec, fw->nid, fw->idx, - fw->mask, fw->val); + __alc_update_coefex_idx(codec, fw->nid, fw->idx, + fw->mask, fw->val); } + mutex_unlock(&spec->coef_mutex); } /* @@ -1153,6 +1191,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) codec->spdif_status_reset = 1; codec->forced_resume = 1; codec->patch_ops = alc_patch_ops; + mutex_init(&spec->coef_mutex); err = alc_codec_rename_from_preset(codec); if (err < 0) { From 63394a16086fc2152869d7902621e2525e14bc40 Mon Sep 17 00:00:00 2001 From: Christian Lachner Date: Sat, 29 Jan 2022 12:32:41 +0100 Subject: [PATCH 24/52] ALSA: hda/realtek: Add missing fixup-model entry for Gigabyte X570 ALC1220 quirks The initial commit of the new Gigabyte X570 ALC1220 quirks lacked the fixup-model entry in alc882_fixup_models[]. It seemed not to cause any ill effects but for completeness sake this commit makes up for that. Signed-off-by: Christian Lachner Cc: Link: https://lore.kernel.org/r/20220129113243.93068-2-gladiac@gmail.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 a5677be0a4055..d662ad8059604 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2665,6 +2665,7 @@ static const struct hda_model_fixup alc882_fixup_models[] = { {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, {.id = ALC887_FIXUP_ASUS_BASS, .name = "asus-bass"}, {.id = ALC1220_FIXUP_GB_DUAL_CODECS, .name = "dual-codecs"}, + {.id = ALC1220_FIXUP_GB_X570, .name = "gb-x570"}, {.id = ALC1220_FIXUP_CLEVO_P950, .name = "clevo-p950"}, {} }; From 41a8601302ecbe704ac970552c33dc942300fc37 Mon Sep 17 00:00:00 2001 From: Christian Lachner Date: Sat, 29 Jan 2022 12:32:42 +0100 Subject: [PATCH 25/52] ALSA: hda/realtek: Fix silent output on Gigabyte X570S Aorus Master (newer chipset) Newer versions of the X570 Master come with a newer revision of the mainboard chipset - the X570S. These boards have the same ALC1220 codec but seem to initialize the codec with a different parameter in Coef 0x7 which causes the output audio to be very low. We therefore write a known-good value to Coef 0x7 to fix that. As the value is the exact same as on the other X570(non-S) boards the same quirk-function can be shared between both generations. This commit adds the Gigabyte X570S Aorus Master to the list of boards using the ALC1220_FIXUP_GB_X570 quirk. This fixes both, the silent output and the no-audio after reboot from windows problems. This work has been tested by the folks over at the level1techs forum here: https://forum.level1techs.com/t/has-anybody-gotten-audio-working-in-linux-on-aorus-x570-master/154072 Signed-off-by: Christian Lachner Cc: Link: https://lore.kernel.org/r/20220129113243.93068-3-gladiac@gmail.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 d662ad8059604..54301d208d2bb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2164,6 +2164,7 @@ static void alc1220_fixup_gb_x570(struct hda_codec *codec, { static const hda_nid_t conn1[] = { 0x0c }; static const struct coef_fw gb_x570_coefs[] = { + WRITE_COEF(0x07, 0x03c0), WRITE_COEF(0x1a, 0x01c1), WRITE_COEF(0x1b, 0x0202), WRITE_COEF(0x43, 0x3005), @@ -2591,6 +2592,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_GB_X570), SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1458, 0xa0d5, "Gigabyte X570S Aorus Master", ALC1220_FIXUP_GB_X570), SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1229, "MSI-GP73", ALC1220_FIXUP_CLEVO_P950), From ea3541961376f733373839cc90493aafa8a7f733 Mon Sep 17 00:00:00 2001 From: Christian Lachner Date: Sat, 29 Jan 2022 12:32:43 +0100 Subject: [PATCH 26/52] ALSA: hda/realtek: Fix silent output on Gigabyte X570 Aorus Xtreme after reboot from Windows This commit switches the Gigabyte X570 Aorus Xtreme from using the ALC1220_FIXUP_CLEVO_P950 to the ALC1220_FIXUP_GB_X570 quirk. This fixes the no-audio after reboot from windows problem. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 Signed-off-by: Christian Lachner Cc: Link: https://lore.kernel.org/r/20220129113243.93068-4-gladiac@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 54301d208d2bb..0d52eca57bbc0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2591,7 +2591,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_GB_X570), - SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_GB_X570), SND_PCI_QUIRK(0x1458, 0xa0d5, "Gigabyte X570S Aorus Master", ALC1220_FIXUP_GB_X570), SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), From 94db9cc8f8fa2d5426ce79ec4ca16028f7084224 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Albert=20Geant=C4=83?= Date: Mon, 31 Jan 2022 03:05:23 +0200 Subject: [PATCH 27/52] ALSA: hda/realtek: Add quirk for ASUS GU603 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The ASUS GU603 (Zephyrus M16 - SSID 1043:16b2) requires a quirk similar to other ASUS devices for correctly routing the 4 integrated speakers. This fixes it by adding a corresponding quirk entry, which connects the bass speakers to the proper DAC. Signed-off-by: Albert Geantă Cc: Link: https://lore.kernel.org/r/20220131010523.546386-1-albertgeanta@gmail.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 0d52eca57bbc0..8315bf7d4c381 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9011,6 +9011,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), From a4f399a1416f645ac701064a55b0cb5203707ac9 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sun, 30 Jan 2022 09:06:36 +0100 Subject: [PATCH 28/52] Input: wm97xx: Simplify resource management Since the commit in the Fixes tag below, 'wm->input_dev' is a managed resource that doesn't need to be explicitly unregistered or freed (see devm_input_allocate_device() documentation) So, remove some unless line of code to slightly simplify it. Fixes: c72f61e74073 ("Input: wm97xx: split out touchscreen registering") Signed-off-by: Christophe JAILLET Acked-by: Charles Keepax Link: https://lore.kernel.org/r/87dce7e80ea9b191843fa22415ca3aef5f3cc2e6.1643529968.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- drivers/input/touchscreen/wm97xx-core.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/drivers/input/touchscreen/wm97xx-core.c b/drivers/input/touchscreen/wm97xx-core.c index 78d2ee99f37ac..1b58611c80840 100644 --- a/drivers/input/touchscreen/wm97xx-core.c +++ b/drivers/input/touchscreen/wm97xx-core.c @@ -615,10 +615,9 @@ static int wm97xx_register_touch(struct wm97xx *wm) * extensions) */ wm->touch_dev = platform_device_alloc("wm97xx-touch", -1); - if (!wm->touch_dev) { - ret = -ENOMEM; - goto touch_err; - } + if (!wm->touch_dev) + return -ENOMEM; + platform_set_drvdata(wm->touch_dev, wm); wm->touch_dev->dev.parent = wm->dev; wm->touch_dev->dev.platform_data = pdata; @@ -629,9 +628,6 @@ static int wm97xx_register_touch(struct wm97xx *wm) return 0; touch_reg_err: platform_device_put(wm->touch_dev); -touch_err: - input_unregister_device(wm->input_dev); - wm->input_dev = NULL; return ret; } @@ -639,8 +635,6 @@ static int wm97xx_register_touch(struct wm97xx *wm) static void wm97xx_unregister_touch(struct wm97xx *wm) { platform_device_unregister(wm->touch_dev); - input_unregister_device(wm->input_dev); - wm->input_dev = NULL; } static int _wm97xx_probe(struct wm97xx *wm) From ff4865b3c8cd746ef72f59bdd485848b4cebd43d Mon Sep 17 00:00:00 2001 From: "Rafael J. Wysocki" Date: Wed, 26 Jan 2022 20:48:49 +0100 Subject: [PATCH 29/52] ALSA: Replace acpi_bus_get_device() Replace acpi_bus_get_device() that is going to be dropped with acpi_fetch_acpi_dev(). No intentional functional impact. Signed-off-by: Rafael J. Wysocki Link: https://lore.kernel.org/r/2828205.e9J7NaK4W3@kreacher Signed-off-by: Takashi Iwai --- sound/hda/intel-sdw-acpi.c | 7 +++---- sound/soc/soc-acpi.c | 7 ++----- 2 files changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index b7758dbe23714..5cb92f7ccbcac 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -50,11 +50,11 @@ static bool is_link_enabled(struct fwnode_handle *fw_node, int i) static int sdw_intel_scan_controller(struct sdw_intel_acpi_info *info) { - struct acpi_device *adev; + struct acpi_device *adev = acpi_fetch_acpi_dev(info->handle); int ret, i; u8 count; - if (acpi_bus_get_device(info->handle, &adev)) + if (!adev) return -EINVAL; /* Found controller, find links supported */ @@ -119,7 +119,6 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, void *cdata, void **return_value) { struct sdw_intel_acpi_info *info = cdata; - struct acpi_device *adev; acpi_status status; u64 adr; @@ -127,7 +126,7 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, if (ACPI_FAILURE(status)) return AE_OK; /* keep going */ - if (acpi_bus_get_device(handle, &adev)) { + if (!acpi_fetch_acpi_dev(handle)) { pr_err("%s: Couldn't find ACPI handle\n", __func__); return AE_NOT_FOUND; } diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index cbd7ea48837b2..142476f1396ff 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -55,16 +55,13 @@ EXPORT_SYMBOL_GPL(snd_soc_acpi_find_machine); static acpi_status snd_soc_acpi_find_package(acpi_handle handle, u32 level, void *context, void **ret) { - struct acpi_device *adev; + struct acpi_device *adev = acpi_fetch_acpi_dev(handle); acpi_status status; struct snd_soc_acpi_package_context *pkg_ctx = context; pkg_ctx->data_valid = false; - if (acpi_bus_get_device(handle, &adev)) - return AE_OK; - - if (adev->status.present && adev->status.functional) { + if (adev && adev->status.present && adev->status.functional) { struct acpi_buffer buffer = {ACPI_ALLOCATE_BUFFER, NULL}; union acpi_object *myobj = NULL; From 4ee02e20893d2f9e951c7888f2284fa608ddaa35 Mon Sep 17 00:00:00 2001 From: Jonas Hahnfeld Date: Mon, 31 Jan 2022 19:35:16 +0100 Subject: [PATCH 30/52] ALSA: usb-audio: Correct quirk for VF0770 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This device provides both audio and video. The original quirk added in commit 48827e1d6af5 ("ALSA: usb-audio: Add quirk for VF0770") used USB_DEVICE to match the vendor and product ID. Depending on module order, if snd-usb-audio was asked first, it would match the entire device and uvcvideo wouldn't get to see it. Change the matching to USB_AUDIO_DEVICE to restore uvcvideo matching in all cases. Fixes: 48827e1d6af5 ("ALSA: usb-audio: Add quirk for VF0770") Reported-by: Jukka Heikintalo Tested-by: Jukka Heikintalo Reported-by: Paweł Susicki Tested-by: Paweł Susicki Cc: # 5.4, 5.10, 5.14, 5.15 Signed-off-by: Jonas Hahnfeld Link: https://lore.kernel.org/r/20220131183516.61191-1-hahnjo@hahnjo.de Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b1522e43173e1..0ea39565e6232 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -84,7 +84,7 @@ * combination. */ { - USB_DEVICE(0x041e, 0x4095), + USB_AUDIO_DEVICE(0x041e, 0x4095), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, From 1c7f0e349aa5f8f80b1cac3d4917405332e14cdf Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 1 Feb 2022 13:21:44 +0200 Subject: [PATCH 31/52] ALSA: hda: Skip codec shutdown in case the codec is not registered If the codec->registered is not set then it means that pm_runtime is not yet enabled and the codec->pcm_list_head has not been initialized. The access to the not initialized pcm_list_head will lead a kernel crash during shutdown. Reported-by: Guennadi Liakhovetski Signed-off-by: Peter Ujfalusi Tested-by: Guennadi Liakhovetski Fixes: b98444ed597d ("ALSA: hda: Suspend codec at shutdown") Link: https://lore.kernel.org/r/20220201112144.29411-1-peter.ujfalusi@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7016b48227bf2..f552785d301e0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3000,6 +3000,10 @@ void snd_hda_codec_shutdown(struct hda_codec *codec) { struct hda_pcm *cpcm; + /* Skip the shutdown if codec is not registered */ + if (!codec->registered) + return; + list_for_each_entry(cpcm, &codec->pcm_list_head, list) snd_pcm_suspend_all(cpcm->pcm); From 564778d7b1ea465f9487eedeece7527a033549c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Feb 2022 15:56:26 +0000 Subject: [PATCH 32/52] ASoC: ops: Fix stereo change notifications in snd_soc_put_volsw() When writing out a stereo control we discard the change notification from the first channel, meaning that events are only generated based on changes to the second channel. Ensure that we report a change if either channel has changed. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220201155629.120510-2-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 9833611b83d1f..2ce73e3391f4b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -308,7 +308,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, unsigned int sign_bit = mc->sign_bit; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - int err; + int err, ret; bool type_2r = false; unsigned int val2 = 0; unsigned int val, val_mask; @@ -350,12 +350,18 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, err = snd_soc_component_update_bits(component, reg, val_mask, val); if (err < 0) return err; + ret = err; - if (type_2r) + if (type_2r) { err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); + val2); + /* Don't discard any error code or drop change flag */ + if (ret == 0 || err < 0) { + ret = err; + } + } - return err; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw); From 7f3d90a3519680dfa23e750f80bfdefc0f5eda4a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Feb 2022 15:56:27 +0000 Subject: [PATCH 33/52] ASoC: ops: Fix stereo change notifications in snd_soc_put_volsw_sx() When writing out a stereo control we discard the change notification from the first channel, meaning that events are only generated based on changes to the second channel. Ensure that we report a change if either channel has changed. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220201155629.120510-3-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2ce73e3391f4b..4987c58cd59a6 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -427,6 +427,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int min = mc->min; unsigned int mask = (1U << (fls(min + max) - 1)) - 1; int err = 0; + int ret; unsigned int val, val_mask; if (ucontrol->value.integer.value[0] < 0) @@ -443,6 +444,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, err = snd_soc_component_update_bits(component, reg, val_mask, val); if (err < 0) return err; + ret = err; if (snd_soc_volsw_is_stereo(mc)) { unsigned int val2; @@ -453,6 +455,11 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, err = snd_soc_component_update_bits(component, reg2, val_mask, val2); + + /* Don't discard any error code or drop change flag */ + if (ret == 0 || err < 0) { + ret = err; + } } return err; } From 650204ded3703b5817bd4b6a77fa47d333c4f902 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Feb 2022 15:56:28 +0000 Subject: [PATCH 34/52] ASoC: ops: Fix stereo change notifications in snd_soc_put_volsw_range() When writing out a stereo control we discard the change notification from the first channel, meaning that events are only generated based on changes to the second channel. Ensure that we report a change if either channel has changed. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220201155629.120510-4-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 4987c58cd59a6..2e57f814bcc28 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -519,7 +519,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int ret; + int err, ret; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -528,9 +528,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - ret = snd_soc_component_update_bits(component, reg, val_mask, val); - if (ret < 0) - return ret; + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + ret = err; if (snd_soc_volsw_is_stereo(mc)) { if (invert) @@ -540,8 +541,12 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - ret = snd_soc_component_update_bits(component, rreg, val_mask, + err = snd_soc_component_update_bits(component, rreg, val_mask, val); + /* Don't discard any error code or drop change flag */ + if (ret == 0 || err < 0) { + ret = err; + } } return ret; From 2b7c46369f09c358164d31d17e5695185403185e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Feb 2022 15:56:29 +0000 Subject: [PATCH 35/52] ASoC: ops: Fix stereo change notifications in snd_soc_put_xr_sx() When writing out a stereo control we discard the change notification from the first channel, meaning that events are only generated based on changes to the second channel. Ensure that we report a change if either channel has changed. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220201155629.120510-5-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2e57f814bcc28..03ea9591fb164 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -895,6 +895,7 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned long mask = (1UL<nbits)-1; long max = mc->max; long val = ucontrol->value.integer.value[0]; + int ret = 0; unsigned int i; if (val < mc->min || val > mc->max) @@ -909,9 +910,11 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, regmask, regval); if (err < 0) return err; + if (err > 0) + ret = err; } - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_xr_sx); From 7fa5c33d043160eba3be9fb8e21588dff2a467c7 Mon Sep 17 00:00:00 2001 From: V sujith kumar Reddy Date: Tue, 1 Feb 2022 02:02:15 +0530 Subject: [PATCH 36/52] ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver Maxim codec driver already enabling/disabling spk_en_gpio in form of sd_mode gpio hence remove such gpio access control from machine driver to avoid conflict Signed-off-by: V sujith kumar Reddy Link: https://lore.kernel.org/r/20220131203225.1418648-1-vsujithkumar.reddy@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach.h | 1 - sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index fd6299844ebe4..c855f50d6b34c 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,7 +21,6 @@ #include #define EN_SPKR_GPIO_GB 0x11F -#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 07de461426559..4cc431e54fe1a 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NK, + .gpio_spkr_en = EN_SPKR_GPIO_NONE, }; static struct acp_card_drvdata sof_rt5682s_max_data = { @@ -47,7 +47,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NK, + .gpio_spkr_en = EN_SPKR_GPIO_NONE, }; static const struct snd_kcontrol_new acp_controls[] = { From 946eb87114af37c9c13c618a7c1cdaca936905fa Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Mon, 7 Feb 2022 08:09:23 -0800 Subject: [PATCH 37/52] ASoC: Revert "ASoC: mediatek: Check for error clk pointer" This reverts commit 9de2b9286a6d ("ASoC: mediatek: Check for error clk pointer"). With this patch in the tree, Chromebooks running the affected hardware no longer boot. Bisect points to this patch, and reverting it fixes the problem. An analysis of the code with this patch applied shows: ret = init_clks(pdev, clk); if (ret) return ERR_PTR(ret); ... for (j = 0; j < MAX_CLKS && data->clk_id[j]; j++) { struct clk *c = clk[data->clk_id[j]]; if (IS_ERR(c)) { dev_err(&pdev->dev, "%s: clk unavailable\n", data->name); return ERR_CAST(c); } scpd->clk[j] = c; } Not all clocks in the clk_names array have to be present. Only the clocks in the data->clk_id array are actually needed. The code already checks if the required clocks are available and bails out if not. The assumption that all clocks have to be present is wrong, and commit 9de2b9286a6d ("ASoC: mediatek: Check for error clk pointer") needs to be reverted. Cc: Jiasheng Jiang Cc: Mark Brown Cc: James Liao Cc: Kevin Hilman Cc: Matthias Brugger Reported-by: Frank Wunderlich Reported-by: Daniel Golle Fixes: 9de2b9286a6d ("ASoC: mediatek: Check for error clk pointer") Signed-off-by: Guenter Roeck Link: https://lore.kernel.org/r/20220207160923.3911501-1-linux@roeck-us.net Signed-off-by: Mark Brown --- drivers/soc/mediatek/mtk-scpsys.c | 15 ++++----------- 1 file changed, 4 insertions(+), 11 deletions(-) diff --git a/drivers/soc/mediatek/mtk-scpsys.c b/drivers/soc/mediatek/mtk-scpsys.c index 670cc82d17dc2..ca75b14931ec9 100644 --- a/drivers/soc/mediatek/mtk-scpsys.c +++ b/drivers/soc/mediatek/mtk-scpsys.c @@ -411,17 +411,12 @@ static int scpsys_power_off(struct generic_pm_domain *genpd) return ret; } -static int init_clks(struct platform_device *pdev, struct clk **clk) +static void init_clks(struct platform_device *pdev, struct clk **clk) { int i; - for (i = CLK_NONE + 1; i < CLK_MAX; i++) { + for (i = CLK_NONE + 1; i < CLK_MAX; i++) clk[i] = devm_clk_get(&pdev->dev, clk_names[i]); - if (IS_ERR(clk[i])) - return PTR_ERR(clk[i]); - } - - return 0; } static struct scp *init_scp(struct platform_device *pdev, @@ -431,7 +426,7 @@ static struct scp *init_scp(struct platform_device *pdev, { struct genpd_onecell_data *pd_data; struct resource *res; - int i, j, ret; + int i, j; struct scp *scp; struct clk *clk[CLK_MAX]; @@ -486,9 +481,7 @@ static struct scp *init_scp(struct platform_device *pdev, pd_data->num_domains = num; - ret = init_clks(pdev, clk); - if (ret) - return ERR_PTR(ret); + init_clks(pdev, clk); for (i = 0; i < num; i++) { struct scp_domain *scpd = &scp->domains[i]; From 307f31452078792aab94a729fce33200c6e42dc4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Martin=20Povi=C5=A1er?= Date: Fri, 4 Feb 2022 10:53:01 +0100 Subject: [PATCH 38/52] ASoC: tas2770: Insert post reset delay MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Per TAS2770 datasheet there must be a 1 ms delay from reset to first command. So insert delays into the driver where appropriate. Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Martin Povišer Link: https://lore.kernel.org/r/20220204095301.5554-1-povik+lin@cutebit.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2770.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 6549e7fef3e32..c5ea3b115966b 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -38,10 +38,12 @@ static void tas2770_reset(struct tas2770_priv *tas2770) gpiod_set_value_cansleep(tas2770->reset_gpio, 0); msleep(20); gpiod_set_value_cansleep(tas2770->reset_gpio, 1); + usleep_range(1000, 2000); } snd_soc_component_write(tas2770->component, TAS2770_SW_RST, TAS2770_RST); + usleep_range(1000, 2000); } static int tas2770_set_bias_level(struct snd_soc_component *component, @@ -110,6 +112,7 @@ static int tas2770_codec_resume(struct snd_soc_component *component) if (tas2770->sdz_gpio) { gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); + usleep_range(1000, 2000); } else { ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, @@ -510,8 +513,10 @@ static int tas2770_codec_probe(struct snd_soc_component *component) tas2770->component = component; - if (tas2770->sdz_gpio) + if (tas2770->sdz_gpio) { gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); + usleep_range(1000, 2000); + } tas2770_reset(tas2770); From d7b530fdc45e75a54914a194c4becd9672a4e24f Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Mon, 7 Feb 2022 17:29:58 +0200 Subject: [PATCH 39/52] ASoC: rt5682s: do not block workqueue if card is unbound MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The current rt5682s_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Shuming Fan Link: https://lore.kernel.org/r/20220207153000.3452802-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682s.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index efa1016831dd2..1e662d1be2b3e 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -824,11 +824,13 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682s_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682s->component) - usleep_range(10000, 15000); - - while (!rt5682s->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682s->component || !rt5682s->component->card || + !rt5682s->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682s->jdet_mutex); mutex_lock(&rt5682s->calibrate_mutex); From a6d78661dc903d90a327892bbc34268f3a5f4b9c Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Mon, 7 Feb 2022 17:29:59 +0200 Subject: [PATCH 40/52] ASoC: rt5668: do not block workqueue if card is unbound MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The current rt5668_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Shuming Fan Link: https://lore.kernel.org/r/20220207153000.3452802-2-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5668.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index fb09715bf9328..5b12cbf2ba215 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1022,11 +1022,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type; - while (!rt5668->component) - usleep_range(10000, 15000); - - while (!rt5668->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5668->component || !rt5668->component->card || + !rt5668->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5668->calibrate_mutex); From 4c33de0673ced9c7c37b3bbd9bfe0fda72340b2a Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Mon, 7 Feb 2022 17:30:00 +0200 Subject: [PATCH 41/52] ASoC: rt5682: do not block workqueue if card is unbound MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The current rt5682_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Shuming Fan Link: https://lore.kernel.org/r/20220207153000.3452802-3-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 0a0ec4a021e11..be68d573a4906 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1092,11 +1092,13 @@ void rt5682_jack_detect_handler(struct work_struct *work) struct snd_soc_dapm_context *dapm; int val, btn_type; - while (!rt5682->component) - usleep_range(10000, 15000); - - while (!rt5682->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682->component || !rt5682->component->card || + !rt5682->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(15)); + return; + } dapm = snd_soc_component_get_dapm(rt5682->component); From e4e3a93c6e267572ca2345d8d86053e166843a8c Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih Date: Tue, 8 Feb 2022 11:12:42 +0800 Subject: [PATCH 42/52] MAINTAINERS: update cros_ec_codec maintainers Updates cros_ec_codec maintainers. Signed-off-by: Tzung-Bi Shih Acked-By: Cheng-Yi Chiang Acked-By: Benson Leung Link: https://lore.kernel.org/r/20220208031242.227563-1-tzungbi@google.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 1 + MAINTAINERS | 1 + 2 files changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index 77adbebed8242..c3e9f3485449e 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,6 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang + - Tzung-Bi Shih description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/MAINTAINERS b/MAINTAINERS index 09734251e1de9..9256c3540a0ff 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4476,6 +4476,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang +M: Tzung-Bi Shih R: Guenter Roeck S: Maintained F: Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml From 8e1741c658996a16bd096e077dae0da2460a997f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Feb 2022 13:33:43 +0100 Subject: [PATCH 43/52] ALSA: memalloc: Fix dma_need_sync() checks dma_need_sync() checks each DMA address. Fix the incorrect usages for non-contiguous and non-coherent page allocations. Fortunately, there are no actual call sites that need manual syncs yet. Fixes: a25684a95646 ("ALSA: memalloc: Support for non-contiguous page allocation") Fixes: 73325f60e2ed ("ALSA: memalloc: Support for non-coherent page allocation") Cc: Reported-by: Ezequiel Garcia Link: https://lore.kernel.org/r/20220210123344.8756-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index d1fcd1d5adae3..f98f694e3ce48 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -511,7 +511,8 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) DEFAULT_GFP, 0); if (!sgt) return NULL; - dmab->dev.need_sync = dma_need_sync(dmab->dev.dev, dmab->dev.dir); + dmab->dev.need_sync = dma_need_sync(dmab->dev.dev, + sg_dma_address(sgt->sgl)); p = dma_vmap_noncontiguous(dmab->dev.dev, size, sgt); if (p) dmab->private_data = sgt; @@ -671,9 +672,13 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = { */ static void *snd_dma_noncoherent_alloc(struct snd_dma_buffer *dmab, size_t size) { - dmab->dev.need_sync = dma_need_sync(dmab->dev.dev, dmab->dev.dir); - return dma_alloc_noncoherent(dmab->dev.dev, size, &dmab->addr, - dmab->dev.dir, DEFAULT_GFP); + void *p; + + p = dma_alloc_noncoherent(dmab->dev.dev, size, &dmab->addr, + dmab->dev.dir, DEFAULT_GFP); + if (p) + dmab->dev.need_sync = dma_need_sync(dmab->dev.dev, dmab->addr); + return p; } static void snd_dma_noncoherent_free(struct snd_dma_buffer *dmab) From 3e16dc50d77dc3494275a241fac250c94bf45206 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Feb 2022 13:33:44 +0100 Subject: [PATCH 44/52] ALSA: memalloc: invalidate SG pages before sync It seems that calling invalidate_kernel_vmap_range() is more correct to be called before dma_sync_*(), judging from the other thread: https://lore.kernel.org/all/20220111085958.GA22795@lst.de/ Although this won't matter much in practice, let's fix the call order for consistency. Fixes: a25684a95646 ("ALSA: memalloc: Support for non-contiguous page allocation") Reported-by: Ezequiel Garcia Cc: Link: https://lore.kernel.org/r/20220210123344.8756-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index f98f694e3ce48..6fd763d4d15b1 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -541,9 +541,9 @@ static void snd_dma_noncontig_sync(struct snd_dma_buffer *dmab, if (mode == SNDRV_DMA_SYNC_CPU) { if (dmab->dev.dir == DMA_TO_DEVICE) return; + invalidate_kernel_vmap_range(dmab->area, dmab->bytes); dma_sync_sgtable_for_cpu(dmab->dev.dev, dmab->private_data, dmab->dev.dir); - invalidate_kernel_vmap_range(dmab->area, dmab->bytes); } else { if (dmab->dev.dir == DMA_FROM_DEVICE) return; From c8d251f51ee61df06ee0e419348d8c9160bbfb86 Mon Sep 17 00:00:00 2001 From: Stephen Boyd Date: Wed, 9 Feb 2022 15:25:20 -0800 Subject: [PATCH 45/52] ASoC: qcom: Actually clear DMA interrupt register for HDMI In commit da0363f7bfd3 ("ASoC: qcom: Fix for DMA interrupt clear reg overwriting") we changed regmap_write() to regmap_update_bits() so that we can avoid overwriting bits that we didn't intend to modify. Unfortunately this change breaks the case where a register is writable but not readable, which is exactly how the HDMI irq clear register is designed (grep around LPASS_HDMITX_APP_IRQCLEAR_REG to see how it's write only). That's because regmap_update_bits() tries to read the register from the hardware and if it isn't readable it looks in the regmap cache to see what was written there last time to compare against what we want to write there. Eventually, we're unable to modify this register at all because the bits that we're trying to set are already set in the cache. This is doubly bad for the irq clear register because you have to write the bit to clear an interrupt. Given the irq is level triggered, we see an interrupt storm upon plugging in an HDMI cable and starting audio playback. The irq storm is so great that performance degrades significantly, leading to CPU soft lockups. Fix it by using regmap_write_bits() so that we really do write the bits in the clear register that we want to. This brings the number of irqs handled by lpass_dma_interrupt_handler() down from ~150k/sec to ~10/sec. Fixes: da0363f7bfd3 ("ASoC: qcom: Fix for DMA interrupt clear reg overwriting") Cc: Srinivasa Rao Mandadapu Cc: Srinivas Kandagatla Signed-off-by: Stephen Boyd Link: https://lore.kernel.org/r/20220209232520.4017634-1-swboyd@chromium.org Signed-off-by: Mark Brown --- sound/soc/qcom/lpass-platform.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index a59e9d20cb46b..4b1773c1fb95f 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -524,7 +524,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); + ret = regmap_write_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -665,7 +665,7 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_PER(chan) | val)); + rv = regmap_write_bits(map, reg, mask, (LPAIF_IRQ_PER(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -676,7 +676,7 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_XRUN(chan) | val)); + rv = regmap_write_bits(map, reg, mask, (LPAIF_IRQ_XRUN(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +688,7 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_ERR(chan) | val)); + rv = regmap_write_bits(map, reg, mask, (LPAIF_IRQ_ERR(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); From a887f9c7a4d37a8e874ba8415a42a92a1b5139fc Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 10 Feb 2022 17:20:51 +0000 Subject: [PATCH 46/52] ASoC: wm_adsp: Correct control read size when parsing compressed buffer When parsing the compressed stream the whole buffer descriptor is now read in a single cs_dsp_coeff_read_ctrl; on older firmwares this descriptor is just 4 bytes but on more modern firmwares it is 24 bytes. The current code reads the full 24 bytes regardless, this was working but reading junk for the last 20 bytes. However commit f444da38ac92 ("firmware: cs_dsp: Add offset to cs_dsp read/write") added a size check into cs_dsp_coeff_read_ctrl, causing the older firmwares to now return an error. Update the code to only read the amount of data appropriate for the firmware loaded. Fixes: 04ae08596737 ("ASoC: wm_adsp: Switch to using wm_coeff_read_ctrl for compressed buffers") Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220210172053.22782-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3672e3d1703e..0582585236a28 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1441,7 +1441,8 @@ static int wm_adsp_buffer_parse_coeff(struct cs_dsp_coeff_ctl *cs_ctl) int ret, i; for (i = 0; i < 5; ++i) { - ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, &coeff_v1, sizeof(coeff_v1)); + ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, &coeff_v1, + min(cs_ctl->len, sizeof(coeff_v1))); if (ret < 0) return ret; From 19d20c7a29bf2e46ff1ab8e8c4fcd2da8a4f38e2 Mon Sep 17 00:00:00 2001 From: Matteo Martelli Date: Fri, 11 Feb 2022 23:49:13 +0100 Subject: [PATCH 47/52] ALSA: usb-audio: revert to IMPLICIT_FB_FIXED_DEV for M-Audio FastTrack Ultra Commit 83b7dcbc51c930fc2079ab6c6fc9d719768321f1 introduced a generic implicit feedback parser, which fails to execute for M-Audio FastTrack Ultra sound cards. The issue is with the ENDPOINT_SYNCTYPE check in add_generic_implicit_fb() where the SYNCTYPE is ADAPTIVE instead of ASYNC. The reason is that the sync type of the FastTrack output endpoints are set to adaptive in the quirks table since commit 65f04443c96dbda11b8fff21d6390e082846aa3c. Fixes: 83b7dcbc51c9 ("ALSA: usb-audio: Add generic implicit fb parsing") Signed-off-by: Matteo Martelli Cc: Link: https://lore.kernel.org/r/20220211224913.20683-2-matteomartelli3@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/implicit.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 70319c822c10b..2d444ec742029 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -47,13 +47,13 @@ struct snd_usb_implicit_fb_match { static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Generic matching */ IMPLICIT_FB_GENERIC_DEV(0x0499, 0x1509), /* Steinberg UR22 */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2080), /* M-Audio FastTrack Ultra */ - IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2081), /* M-Audio FastTrack Ultra */ IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2030), /* M-Audio Fast Track C400 */ IMPLICIT_FB_GENERIC_DEV(0x0763, 0x2031), /* M-Audio Fast Track C600 */ /* Fixed EP */ /* FIXME: check the availability of generic matching */ + IMPLICIT_FB_FIXED_DEV(0x0763, 0x2080, 0x81, 2), /* M-Audio FastTrack Ultra */ + IMPLICIT_FB_FIXED_DEV(0x0763, 0x2081, 0x81, 2), /* M-Audio FastTrack Ultra */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0001, 0x81, 2), /* Solid State Logic SSL2 */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */ From c07f2c7b45413a9e50ba78630fda04ecfa17b4f2 Mon Sep 17 00:00:00 2001 From: Yu Huang Date: Sun, 13 Feb 2022 00:08:33 +0800 Subject: [PATCH 48/52] ALSA: hda/realtek: Add quirk for Legion Y9000X 2019 Legion Y9000X 2019 has the same speaker with Y9000X 2020, but with a different quirk address. Add one quirk entry to make the speaker work on Y9000X 2019 too. Signed-off-by: Yu Huang Cc: Link: https://lore.kernel.org/r/20220212160835.165065-1-diwang90@gmail.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 8315bf7d4c381..9473fb76ff197 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9170,6 +9170,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x383d, "Legion Y9000X 2019", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), SND_PCI_QUIRK(0x17aa, 0x3847, "Legion 7 16ACHG6", ALC287_FIXUP_LEGION_16ACHG6), SND_PCI_QUIRK(0x17aa, 0x384a, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), From 6317f7449348a897483a2b4841f7a9190745c81b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Feb 2022 11:00:19 +0100 Subject: [PATCH 49/52] ALSA: hda: Fix regression on forced probe mask option The forced probe mask via probe_mask 0x100 bit doesn't work any longer as expected since the bus init code was moved and it's clearing the codec_mask value that was set beforehand. This patch fixes the long-time regression by moving the check_probe_mask() call. Fixes: a41d122449be ("ALSA: hda - Embed bus into controller object") Reported-by: dmummenschanz@web.de Cc: Link: https://lore.kernel.org/r/trinity-f018660b-95c9-442b-a2a8-c92a56eb07ed-1644345967148@3c-app-webde-bap22 Link: https://lore.kernel.org/r/20220214100020.8870-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4b0338c4c5437..18b795220b52c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1798,8 +1798,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, assign_position_fix(chip, check_position_fix(chip, position_fix[dev])); - check_probe_mask(chip, dev); - if (single_cmd < 0) /* allow fallback to single_cmd at errors */ chip->fallback_to_single_cmd = 1; else /* explicitly set to single_cmd or not */ @@ -1825,6 +1823,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, chip->bus.core.needs_damn_long_delay = 1; } + check_probe_mask(chip, dev); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { dev_err(card->dev, "Error creating device [card]!\n"); From dd8e5b161d7fb9cefa1f1d6e35a39b9e1563c8d3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Feb 2022 11:00:20 +0100 Subject: [PATCH 50/52] ALSA: hda: Fix missing codec probe on Shenker Dock 15 By some unknown reason, BIOS on Shenker Dock 15 doesn't set up the codec mask properly for the onboard audio. Let's set the forced codec mask to enable the codec discovery. Reported-by: dmummenschanz@web.de Cc: Link: https://lore.kernel.org/r/trinity-f018660b-95c9-442b-a2a8-c92a56eb07ed-1644345967148@3c-app-webde-bap22 Link: https://lore.kernel.org/r/20220214100020.8870-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 18b795220b52c..917ad9d375b17 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1615,6 +1615,7 @@ static const struct snd_pci_quirk probe_mask_list[] = { /* forced codec slots */ SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103), SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), + SND_PCI_QUIRK(0x1558, 0x0351, "Schenker Dock 15", 0x105), /* WinFast VP200 H (Teradici) user reported broken communication */ SND_PCI_QUIRK(0x3a21, 0x040d, "WinFast VP200 H", 0x101), {} From 9a5adeb28b77416446658e75bdef3bbe5fb92a83 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Feb 2022 13:57:11 +0100 Subject: [PATCH 51/52] ALSA: usb-audio: Don't abort resume upon errors The default mixer resume code treats the errors at restoring the modified mixer items as a fatal error, and it returns back to the caller. This ends up in the resume failure, and the device will be come unavailable, although basically those errors are intermittent and can be safely ignored. The problem itself has been present from the beginning, but it didn't hit usually because the code tries to resume only the modified items. But now with the recent commit to forcibly initialize each item at the probe time, the problem surfaced more often, hence it appears as a regression. This patch fixes the regression simply by ignoring the errors at resume. Fixes: b96681bd5827 ("ALSA: usb-audio: Initialize every feature unit once at probe time") Cc: BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215561 Link: https://lore.kernel.org/r/20220214125711.20531-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 630766ba259fd..a5641956ef102 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3678,17 +3678,14 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) err = snd_usb_set_cur_mix_value(cval, c + 1, idx, cval->cache_val[idx]); if (err < 0) - return err; + break; } idx++; } } else { /* master */ - if (cval->cached) { - err = snd_usb_set_cur_mix_value(cval, 0, 0, *cval->cache_val); - if (err < 0) - return err; - } + if (cval->cached) + snd_usb_set_cur_mix_value(cval, 0, 0, *cval->cache_val); } return 0; From 2a845837e3d0ddaed493b4c5c4643d7f0542804d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Feb 2022 14:04:10 +0100 Subject: [PATCH 52/52] ALSA: hda/realtek: Fix deadlock by COEF mutex The recently introduced coef_mutex for Realtek codec seems causing a deadlock when the relevant code is invoked from the power-off state; then the HD-audio core tries to power-up internally, and this kicks off the codec runtime PM code that tries to take the same coef_mutex. In order to avoid the deadlock, do the temporary power up/down around the coef_mutex acquisition and release. This assures that the power-up sequence runs before the mutex, hence no re-entrance will happen. Fixes: b837a9f5ab3b ("ALSA: hda: realtek: Fix race at concurrent COEF updates") Reported-and-tested-by: Julian Wollrath Cc: Link: https://lore.kernel.org/r/20220214132838.4db10fca@schienar Link: https://lore.kernel.org/r/20220214130410.21230-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 39 +++++++++++++++++++++-------------- 1 file changed, 24 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9473fb76ff197..3a42457984e98 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -138,6 +138,22 @@ struct alc_spec { * COEF access helper functions */ +static void coef_mutex_lock(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + snd_hda_power_up_pm(codec); + mutex_lock(&spec->coef_mutex); +} + +static void coef_mutex_unlock(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + mutex_unlock(&spec->coef_mutex); + snd_hda_power_down_pm(codec); +} + static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx) { @@ -151,12 +167,11 @@ static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx) { - struct alc_spec *spec = codec->spec; unsigned int val; - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); val = __alc_read_coefex_idx(codec, nid, coef_idx); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); return val; } @@ -173,11 +188,9 @@ static void __alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx, unsigned int coef_val) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); __alc_write_coefex_idx(codec, nid, coef_idx, coef_val); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } #define alc_write_coef_idx(codec, coef_idx, coef_val) \ @@ -198,11 +211,9 @@ static void alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx, unsigned int mask, unsigned int bits_set) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); __alc_update_coefex_idx(codec, nid, coef_idx, mask, bits_set); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } #define alc_update_coef_idx(codec, coef_idx, mask, bits_set) \ @@ -235,9 +246,7 @@ struct coef_fw { static void alc_process_coef_fw(struct hda_codec *codec, const struct coef_fw *fw) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); for (; fw->nid; fw++) { if (fw->mask == (unsigned short)-1) __alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val); @@ -245,7 +254,7 @@ static void alc_process_coef_fw(struct hda_codec *codec, __alc_update_coefex_idx(codec, fw->nid, fw->idx, fw->mask, fw->val); } - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } /*