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[PATCH v3 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 01 Oct '24

01 Oct '24

Hi Mark This is v3 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v2 -> v3 - tidyup Intel files v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r091lwhg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 583 deletions(-) -- 2.43.0

1 13

[PATCH v2 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 01 Oct '24

01 Oct '24

Hi Mark This is v2 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 583 deletions(-) -- 2.43.0

2 15

[PATCH] sound: soc: codecs: tlv320aic3x: Fix codec gpio-reset
by Benjamin Hahn 01 Oct '24

01 Oct '24

The TLV320AIC3007 requires a hardware reset after power-up for proper operation. After all power supplies are at their specified values, the RESET pin must be driven low for at least 10 ns. Even though the datasheet sais min 10ns, I needed more than 10ns when testing this out. 15ns worked for me. Signed-off-by: Benjamin Hahn <B.Hahn(a)phytec.de> --- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 56e795a00e22..d002fc0b99b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1367,8 +1367,11 @@ static int aic3x_regulator_event(struct notifier_block *nb, * Put codec to reset and require cache sync as at least one * of the supplies was disabled */ - if (aic3x->gpio_reset) + if (aic3x->gpio_reset) { gpiod_set_value(aic3x->gpio_reset, 1); + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + } regcache_mark_dirty(aic3x->regmap); } @@ -1813,6 +1816,10 @@ int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver gpiod_set_consumer_name(aic3x->gpio_reset, "tlv320aic3x reset"); + /* CODEC datasheet says minimum of 10ns */ + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; --- base-commit: ad46e8f95e931e113cb98253daf6d443ac244cde change-id: 20240930-wip-bhahn-tlv320aic3x_fix_reset-cca918a4889d Best regards, -- Benjamin Hahn <B.Hahn(a)phytec.de>

2 1

[PATCH] ASoC: fsl: Use maple tree register cache
by Mark Brown 01 Oct '24

01 Oct '24

Several of the NXP drivers use regmaps with a rbtree register cache. Since the maple tree cache is uisng a generally more modern data structure which makes implementation choices more suitable for modern systems let's convert these drivers to it. This should have no practical impact. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index a6cbaa6364c7..910b4ce7671c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -169,7 +169,7 @@ static const struct regmap_config fsl_aud2htx_regmap_config = { .readable_reg = fsl_aud2htx_readable_reg, .volatile_reg = fsl_aud2htx_volatile_reg, .writeable_reg = fsl_aud2htx_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static const struct of_device_id fsl_aud2htx_dt_ids[] = { diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f30912091..5149e732f763 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1748,7 +1748,7 @@ static const struct regmap_config fsl_easrc_regmap_config = { .rd_table = &fsl_easrc_readable_table, .wr_table = &fsl_easrc_writeable_table, .volatile_table = &fsl_easrc_volatileable_table, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; #ifdef DEBUG diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 22b240a70ad4..c0dc9bebb77b 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -955,7 +955,7 @@ static const struct regmap_config fsl_micfil_regmap_config = { .readable_reg = fsl_micfil_readable_reg, .volatile_reg = fsl_micfil_volatile_reg, .writeable_reg = fsl_micfil_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; /* END OF REGMAP */ --- base-commit: 98f7e32f20d28ec452afb208f9cffc08448a2652 change-id: 20240924-asoc-imx-maple-abb62f013533 Best regards, -- Mark Brown <broonie(a)kernel.org>

3 3

[PATCH] ASoC: meson: axg-iface: set continuous rates
by Jerome Brunet 01 Oct '24

01 Oct '24

The axg TDM HW does not depend on a selected set of rates. The hardware itself, just takes an input clock and work with it, regardless of its rate. In this way, the rates TDM can take are continuous. What might force the use of specific rate are the PLL available as clock and/or the codecs facing the TDM HW. Either way, this constraint does not belong in the TDM interface driver. Allow any rate as far as TDM is concerned by setting SNDRV_PCM_RATE_CONTINUOUS with an interval it has been tested with. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 12 +++++++++--- sound/soc/meson/axg-tdm.h | 2 -- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 62057c71f742..09103eef2a97 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -442,14 +442,18 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Playback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_PAD, @@ -461,7 +465,9 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Loopback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_LOOPBACK, diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 1a17f546ce6e..acfcd48f8a00 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -15,8 +15,6 @@ #define AXG_TDM_NUM_LANES 4 #define AXG_TDM_CHANNEL_MAX 128 -#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_768000) #define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ --- base-commit: cd87a98b53518e44cf3c1a7c1c07c869ce33bf83 change-id: 20240920-asoc-axg-iface-continuous-b94cfa234787 Best regards, -- Jerome

2 1

[PATCH] ASoC: tas2781: Fix redundant parameter assignment
by Tang Bin 01 Oct '24

01 Oct '24

In these functions, the variable 'rc' is redundant, thus remove it. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/tas2781-i2c.c | 35 ++++++++-------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a8d97dd7..68887799e 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -650,7 +650,6 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; @@ -659,9 +658,7 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, @@ -673,7 +670,6 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG; @@ -681,9 +677,7 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, reg = TAS2563_RUNTIME_RE_REG; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, @@ -696,7 +690,6 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); @@ -707,9 +700,7 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, else return -1; dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, @@ -721,13 +712,10 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, @@ -739,13 +727,10 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_nop_get( @@ -1115,14 +1100,12 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); int dev_id = ucontrol->value.integer.value[0]; - int max = tas_priv->ndev - 1, rc; + int max = tas_priv->ndev - 1; dev_id = clamp(dev_id, 0, max); guard(mutex)(&tas_priv->codec_lock); - rc = tasdev_chn_switch(tas_priv, dev_id); - - return rc; + return tasdev_chn_switch(tas_priv, dev_id); } static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) @@ -1339,10 +1322,8 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) i++; } - rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, + return snd_soc_add_component_controls(priv->codec, cali_ctrls, nctrls < i ? nctrls : i); - - return rc; } static void tasdevice_fw_ready(const struct firmware *fmw, -- 2.33.0

2 1

[PATCH v2] ALSA: hda/conexant: fix some typos
by Oldherl Oh 30 Sep '24

30 Sep '24

Fix some typos in patch_conexant.c Signed-off-by: Oldherl Oh <me(a)oldherl.one> --- sound/pci/hda/patch_conexant.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e851785ff058..ade42a8209c2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -166,18 +166,18 @@ static void cxt_init_gpio_led(struct hda_codec *codec) static void cx_fixup_headset_recog(struct hda_codec *codec) { - unsigned int mic_persent; + unsigned int mic_present; /* fix some headset type recognize fail issue, such as EDIFIER headset */ - /* set micbiasd output current comparator threshold from 66% to 55%. */ + /* set micbias output current comparator threshold from 66% to 55%. */ snd_hda_codec_write(codec, 0x1c, 0, 0x320, 0x010); - /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias registor + /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias register * value adjustment trim from 2.2K ohms to 2.0K ohms. */ snd_hda_codec_write(codec, 0x1c, 0, 0x3b0, 0xe10); /* fix reboot headset type recognize fail issue */ - mic_persent = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); - if (mic_persent & AC_PINSENSE_PRESENCE) + mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); + if (mic_present & AC_PINSENSE_PRESENCE) /* enable headset mic VREF */ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24); else @@ -249,9 +249,9 @@ static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_ { unsigned int mic_present; - /* In cx8070 and sn6140, the node 16 can only be config to headphone or disabled, - * the node 19 can only be config to microphone or disabled. - * Check hp&mic tag to process headset pulgin&plugout. + /* In cx8070 and sn6140, the node 16 can only be configured to headphone or disabled, + * the node 19 can only be configured to microphone or disabled. + * Check hp&mic tag to process headset plugin & plugout. */ mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); if (!(mic_present & AC_PINSENSE_PRESENCE)) /* mic plugout */ -- 2.46.2

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ucm2: USB-Audio: add Steinberg UR22C (USB0499:172f)
by GitHub pull_request - opened 30 Sep '24

30 Sep '24

alsa-project/alsa-ucm-conf pull request #446 was opened from CoelacanthusHex: Similar to UR24C and UR44C, but with 2 inputs 2 outputs. Signed-off-by: Celeste Liu <CoelacanthusHex(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/446 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/446.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: Fix the push button function for the ALC257
by Oder Chiou 30 Sep '24

30 Sep '24

The patch "Fix noise from speakers on Lenovo IdeaPad 3 15IAU7" caused that the headset push button cannot work properly in case of ALC257. This patch reverted it to correct the side effect. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- 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 f787ff4182d4..86db1a4ec24c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -587,6 +587,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: case 0x10ec0283: case 0x10ec0285: -- 2.34.1

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Fix some typos in hda/conexant
by Oldherl Oh 30 Sep '24

30 Sep '24

I'm trying to understand Conexant driver code in order to solve my microphone detection problem. As I am new to kernel development, I have made this simple typo-fixing-only patch as an exercise. Hope it is not disturbing.

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