June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: SOF: Intel: hda: import SOUNDWIRE_INIT namespace
by Bard Liao 23 Jun '20

23 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Make sure the SoundWire driver can be loaded Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 63ca920c8e6e..c0b75d682750 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1207,3 +1207,4 @@ MODULE_LICENSE("Dual BSD/GPL"); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC_I915); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); +MODULE_IMPORT_NS(SOUNDWIRE_INTEL_INIT); -- 2.17.1

1 0

Re: [PATCH 16/19] ASoC: codecs: cs*: merge .digital_mute() into .mute_stream()
by Schulman, James 23 Jun '20

23 Jun '20

On Mon, 23 Jun 2020, Kuninori Morimoto wrote: > From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > snd_soc_dai_digital_mute() is internally using both > mute_stream() (1) or digital_mute() (2), but the difference between > these 2 are only handling direction. > We can merge digital_mute() into mute_stream > > int snd_soc_dai_digital_mute(xxx, int direction) > { > ... > else if (dai->driver->ops->mute_stream) > (1) return dai->driver->ops->mute_stream(xxx, direction); > else if (direction == SNDRV_PCM_STREAM_PLAYBACK && > dai->driver->ops->digital_mute) > (2) return dai->driver->ops->digital_mute(xxx); > ... > } > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > --- > sound/soc/codecs/cs4265.c | 7 +++++-- > sound/soc/codecs/cs4270.c | 7 +++++-- > sound/soc/codecs/cs42l42.c | 7 +++++-- > sound/soc/codecs/cs42l51.c | 7 +++++-- > sound/soc/codecs/cs42l52.c | 7 +++++-- > sound/soc/codecs/cs42l56.c | 7 +++++-- > sound/soc/codecs/cs42xx8.c | 7 +++++-- > sound/soc/codecs/cs4341.c | 7 +++++-- > sound/soc/codecs/cs4349.c | 7 +++++-- > 9 files changed, 45 insertions(+), 18 deletions(-) > > diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c > index 2fb65f246b0c..06182df8948a 100644 > --- a/sound/soc/codecs/cs4265.c > +++ b/sound/soc/codecs/cs4265.c > @@ -378,10 +378,13 @@ static int cs4265_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) > return 0; > } > > -static int cs4265_digital_mute(struct snd_soc_dai *dai, int mute) > +static int cs4265_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > if (mute) { > snd_soc_component_update_bits(component, CS4265_DAC_CTL, > CS4265_DAC_CTL_MUTE, > @@ -498,7 +501,7 @@ static int cs4265_set_bias_level(struct snd_soc_component *component, > > static const struct snd_soc_dai_ops cs4265_ops = { > .hw_params = cs4265_pcm_hw_params, > - .digital_mute = cs4265_digital_mute, > + .mute_stream = cs4265_mute, > .set_fmt = cs4265_set_fmt, > .set_sysclk = cs4265_set_sysclk, > }; > diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c > index 3e8dabc14f05..3c45406339f4 100644 > --- a/sound/soc/codecs/cs4270.c > +++ b/sound/soc/codecs/cs4270.c > @@ -406,12 +406,15 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, > * board does not have the MUTEA or MUTEB pins connected to such circuitry, > * then this function will do nothing. > */ > -static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute) > +static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > struct cs4270_private *cs4270 = snd_soc_component_get_drvdata(component); > int reg6; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > reg6 = snd_soc_component_read(component, CS4270_MUTE); > > if (mute) > @@ -471,7 +474,7 @@ static const struct snd_soc_dai_ops cs4270_dai_ops = { > .hw_params = cs4270_hw_params, > .set_sysclk = cs4270_set_dai_sysclk, > .set_fmt = cs4270_set_dai_fmt, > - .digital_mute = cs4270_dai_mute, > + .mute_stream = cs4270_dai_mute, > }; > > static struct snd_soc_dai_driver cs4270_dai = { > diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c > index 3bc2fa229ef3..4ef52eae1999 100644 > --- a/sound/soc/codecs/cs42l42.c > +++ b/sound/soc/codecs/cs42l42.c > @@ -849,12 +849,15 @@ static int cs42l42_set_sysclk(struct snd_soc_dai *dai, > return 0; > } > > -static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute) > +static int cs42l42_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > unsigned int regval; > u8 fullScaleVol; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > if (mute) { > /* Mark SCLK as not present to turn on the internal > * oscillator. > @@ -909,7 +912,7 @@ static const struct snd_soc_dai_ops cs42l42_ops = { > .hw_params = cs42l42_pcm_hw_params, > .set_fmt = cs42l42_set_dai_fmt, > .set_sysclk = cs42l42_set_sysclk, > - .digital_mute = cs42l42_digital_mute > + .mute_stream = cs42l42_mute > }; > > static struct snd_soc_dai_driver cs42l42_dai = { > diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c > index dde9812490de..b419a578968e 100644 > --- a/sound/soc/codecs/cs42l51.c > +++ b/sound/soc/codecs/cs42l51.c > @@ -484,12 +484,15 @@ static int cs42l51_hw_params(struct snd_pcm_substream *substream, > return 0; > } > > -static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute) > +static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > int reg; > int mask = CS42L51_DAC_OUT_CTL_DACA_MUTE|CS42L51_DAC_OUT_CTL_DACB_MUTE; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > reg = snd_soc_component_read(component, CS42L51_DAC_OUT_CTL); > > if (mute) > @@ -511,7 +514,7 @@ static const struct snd_soc_dai_ops cs42l51_dai_ops = { > .hw_params = cs42l51_hw_params, > .set_sysclk = cs42l51_set_dai_sysclk, > .set_fmt = cs42l51_set_dai_fmt, > - .digital_mute = cs42l51_dai_mute, > + .mute_stream = cs42l51_dai_mute, > }; > > static struct snd_soc_dai_driver cs42l51_dai = { > diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c > index 2ea4cba3be2a..7d2a77b3114b 100644 > --- a/sound/soc/codecs/cs42l52.c > +++ b/sound/soc/codecs/cs42l52.c > @@ -784,10 +784,13 @@ static int cs42l52_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) > return 0; > } > > -static int cs42l52_digital_mute(struct snd_soc_dai *dai, int mute) > +static int cs42l52_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > if (mute) > snd_soc_component_update_bits(component, CS42L52_PB_CTL1, > CS42L52_PB_CTL1_MUTE_MASK, > @@ -865,7 +868,7 @@ static int cs42l52_set_bias_level(struct snd_soc_component *component, > > static const struct snd_soc_dai_ops cs42l52_ops = { > .hw_params = cs42l52_pcm_hw_params, > - .digital_mute = cs42l52_digital_mute, > + .mute_stream = cs42l52_mute, > .set_fmt = cs42l52_set_fmt, > .set_sysclk = cs42l52_set_sysclk, > }; > diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c > index ac569ab3d30f..77749849cf8f 100644 > --- a/sound/soc/codecs/cs42l56.c > +++ b/sound/soc/codecs/cs42l56.c > @@ -800,10 +800,13 @@ static int cs42l56_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) > return 0; > } > > -static int cs42l56_digital_mute(struct snd_soc_dai *dai, int mute) > +static int cs42l56_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > if (mute) { > /* Hit the DSP Mixer first */ > snd_soc_component_update_bits(component, CS42L56_DSP_MUTE_CTL, > @@ -929,7 +932,7 @@ static int cs42l56_set_bias_level(struct snd_soc_component *component, > > static const struct snd_soc_dai_ops cs42l56_ops = { > .hw_params = cs42l56_pcm_hw_params, > - .digital_mute = cs42l56_digital_mute, > + .mute_stream = cs42l56_mute, > .set_fmt = cs42l56_set_dai_fmt, > .set_sysclk = cs42l56_set_sysclk, > }; > diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c > index 94b1adb088fd..b08d44794cdb 100644 > --- a/sound/soc/codecs/cs42xx8.c > +++ b/sound/soc/codecs/cs42xx8.c > @@ -362,13 +362,16 @@ static int cs42xx8_hw_free(struct snd_pcm_substream *substream, > return 0; > } > > -static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) > +static int cs42xx8_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); > u8 dac_unmute = cs42xx8->tx_channels ? > ~((0x1 << cs42xx8->tx_channels) - 1) : 0; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, > mute ? CS42XX8_DACMUTE_ALL : dac_unmute); > > @@ -380,7 +383,7 @@ static const struct snd_soc_dai_ops cs42xx8_dai_ops = { > .set_sysclk = cs42xx8_set_dai_sysclk, > .hw_params = cs42xx8_hw_params, > .hw_free = cs42xx8_hw_free, > - .digital_mute = cs42xx8_digital_mute, > + .mute_stream = cs42xx8_mute, > }; > > static struct snd_soc_dai_driver cs42xx8_dai = { > diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c > index ade7477d04f1..a47ad98eafd6 100644 > --- a/sound/soc/codecs/cs4341.c > +++ b/sound/soc/codecs/cs4341.c > @@ -116,11 +116,14 @@ static int cs4341_hw_params(struct snd_pcm_substream *substream, > CS4341_MODE2_DIF, mode); > } > > -static int cs4341_digital_mute(struct snd_soc_dai *dai, int mute) > +static int cs4341_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > int ret; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > ret = snd_soc_component_update_bits(component, CS4341_REG_VOLA, > CS4341_VOLX_MUTE, > mute ? CS4341_VOLX_MUTE : 0); > @@ -174,7 +177,7 @@ static const struct snd_kcontrol_new cs4341_controls[] = { > static const struct snd_soc_dai_ops cs4341_dai_ops = { > .set_fmt = cs4341_set_fmt, > .hw_params = cs4341_hw_params, > - .digital_mute = cs4341_digital_mute, > + .mute_stream = cs4341_mute, > }; > > static struct snd_soc_dai_driver cs4341_dai = { > diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c > index 3381209a882d..208a94dd4eed 100644 > --- a/sound/soc/codecs/cs4349.c > +++ b/sound/soc/codecs/cs4349.c > @@ -131,11 +131,14 @@ static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, > return 0; > } > > -static int cs4349_digital_mute(struct snd_soc_dai *dai, int mute) > +static int cs4349_mute(struct snd_soc_dai *dai, int mute, int direction) > { > struct snd_soc_component *component = dai->component; > int reg; > > + if (direction != SNDRV_PCM_STREAM_PLAYBACK) > + return 0; > + > reg = 0; > if (mute) > reg = MUTE_AB_MASK; > @@ -236,7 +239,7 @@ static const struct snd_soc_dapm_route cs4349_routes[] = { > static const struct snd_soc_dai_ops cs4349_dai_ops = { > .hw_params = cs4349_pcm_hw_params, > .set_fmt = cs4349_set_dai_fmt, > - .digital_mute = cs4349_digital_mute, > + .mute_stream = cs4349_mute, > }; > > static struct snd_soc_dai_driver cs4349_dai = { > -- > 2.25.1 > > Acked-by: James Schulman <james.schulman(a)cirrus.com> Thanks, James

2 1

[PATCH 00/19] ASoC: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 23 Jun '20

23 Jun '20

Hi Mark ALSA SoC has 2 mute callbacks (= .digital_mute(), .mute_stream()). But the difference between these 2 is very small. .digital_mute() is for Playback .mute_stream() is for Playback/Capture This patch-set merges .digital_mute() into .mute_stream(), and removes .digital_mute(). Kuninori Morimoto (19): ASoC: hdmi-codec: merge .digital_mute() into .mute_stream() ASoC: ti: merge .digital_mute() into .mute_stream() ASoC: spear: merge .digital_mute() into .mute_stream() ASoC: meson: merge .digital_mute() into .mute_stream() ASoC: atmel: merge .digital_mute() into .mute_stream() ASoC: codecs: merge .digital_mute() into .mute_stream() ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream() ASoC: codecs: tas*: merge .digital_mute() into .mute_stream() ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream() ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream() ASoC: codecs: max*: merge .digital_mute() into .mute_stream() ASoC: codecs: alc*: merge .digital_mute() into .mute_stream() ASoC: codecs: wm*: merge .digital_mute() into .mute_stream() ASoC: codecs: es*: merge .digital_mute() into .mute_stream() ASoC: codecs: da*: merge .digital_mute() into .mute_stream() ASoC: codecs: cs*: merge .digital_mute() into .mute_stream() ASoC: codecs: ak*: merge .digital_mute() into .mute_stream() ASoC: soc-dai: remove .digital_mute ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec drivers/gpu/drm/bridge/sii902x.c | 9 ++++--- drivers/gpu/drm/exynos/exynos_hdmi.c | 8 ++++-- drivers/gpu/drm/i2c/tda998x_drv.c | 9 ++++--- drivers/gpu/drm/mediatek/mtk_hdmi.c | 8 ++++-- drivers/gpu/drm/rockchip/cdn-dp-core.c | 9 ++++--- drivers/gpu/drm/sti/sti_hdmi.c | 8 ++++-- drivers/gpu/drm/zte/zx_hdmi.c | 9 ++++--- include/sound/hdmi-codec.h | 3 ++- include/sound/soc-dai.h | 1 - sound/soc/atmel/atmel-classd.c | 10 +++++--- sound/soc/codecs/88pm860x-codec.c | 9 ++++--- sound/soc/codecs/ad193x.c | 7 ++++-- sound/soc/codecs/adau1701.c | 7 ++++-- sound/soc/codecs/ak4458.c | 7 ++++-- sound/soc/codecs/ak4535.c | 8 ++++-- sound/soc/codecs/ak4641.c | 9 ++++--- sound/soc/codecs/alc5623.c | 7 ++++-- sound/soc/codecs/alc5632.c | 7 ++++-- sound/soc/codecs/cpcap.c | 15 ++++++++--- sound/soc/codecs/cq93vc.c | 7 ++++-- sound/soc/codecs/cs4265.c | 7 ++++-- sound/soc/codecs/cs4270.c | 7 ++++-- sound/soc/codecs/cs42l42.c | 7 ++++-- sound/soc/codecs/cs42l51.c | 7 ++++-- sound/soc/codecs/cs42l52.c | 7 ++++-- sound/soc/codecs/cs42l56.c | 7 ++++-- sound/soc/codecs/cs42xx8.c | 7 ++++-- sound/soc/codecs/cs4341.c | 7 ++++-- sound/soc/codecs/cs4349.c | 7 ++++-- sound/soc/codecs/da7210.c | 7 ++++-- sound/soc/codecs/da7213.c | 7 ++++-- sound/soc/codecs/da9055.c | 7 ++++-- sound/soc/codecs/es8316.c | 7 ++++-- sound/soc/codecs/es8328.c | 7 ++++-- sound/soc/codecs/hdmi-codec.c | 13 +++++----- sound/soc/codecs/isabelle.c | 21 +++++++++++----- sound/soc/codecs/jz4770.c | 7 ++++-- sound/soc/codecs/lm49453.c | 35 ++++++++++++++++++-------- sound/soc/codecs/max98088.c | 16 +++++++++--- sound/soc/codecs/max98090.c | 8 ++++-- sound/soc/codecs/max9867.c | 7 ++++-- sound/soc/codecs/ml26124.c | 7 ++++-- sound/soc/codecs/nau8822.c | 7 ++++-- sound/soc/codecs/pcm1681.c | 7 ++++-- sound/soc/codecs/pcm1789.c | 7 ++++-- sound/soc/codecs/pcm179x.c | 7 ++++-- sound/soc/codecs/pcm3168a.c | 7 ++++-- sound/soc/codecs/pcm512x.c | 7 ++++-- sound/soc/codecs/rk3328_codec.c | 7 ++++-- sound/soc/codecs/sgtl5000.c | 7 ++++-- sound/soc/codecs/ssm2518.c | 7 ++++-- sound/soc/codecs/ssm2602.c | 7 ++++-- sound/soc/codecs/ssm4567.c | 7 ++++-- sound/soc/codecs/sta529.c | 7 ++++-- sound/soc/codecs/tas2552.c | 7 ++++-- sound/soc/codecs/tas2562.c | 7 ++++-- sound/soc/codecs/tas2770.c | 7 ++++-- sound/soc/codecs/tas571x.c | 7 ++++-- sound/soc/codecs/tas5720.c | 7 ++++-- sound/soc/codecs/tas6424.c | 7 ++++-- sound/soc/codecs/tfa9879.c | 7 ++++-- sound/soc/codecs/tlv320aic23.c | 7 ++++-- sound/soc/codecs/tlv320aic26.c | 7 ++++-- sound/soc/codecs/tlv320aic31xx.c | 8 ++++-- sound/soc/codecs/tlv320aic32x4.c | 7 ++++-- sound/soc/codecs/tlv320aic3x.c | 7 ++++-- sound/soc/codecs/twl6040.c | 7 ++++-- sound/soc/codecs/uda134x.c | 7 ++++-- sound/soc/codecs/wm8350.c | 7 ++++-- sound/soc/codecs/wm8400.c | 7 ++++-- sound/soc/codecs/wm8510.c | 7 ++++-- sound/soc/codecs/wm8580.c | 7 ++++-- sound/soc/codecs/wm8711.c | 7 ++++-- sound/soc/codecs/wm8728.c | 7 ++++-- sound/soc/codecs/wm8731.c | 7 ++++-- sound/soc/codecs/wm8741.c | 7 ++++-- sound/soc/codecs/wm8750.c | 7 ++++-- sound/soc/codecs/wm8753.c | 9 ++++--- sound/soc/codecs/wm8770.c | 7 ++++-- sound/soc/codecs/wm8776.c | 7 ++++-- sound/soc/codecs/wm8900.c | 7 ++++-- sound/soc/codecs/wm8903.c | 7 ++++-- sound/soc/codecs/wm8904.c | 7 ++++-- sound/soc/codecs/wm8940.c | 7 ++++-- sound/soc/codecs/wm8955.c | 7 ++++-- sound/soc/codecs/wm8960.c | 7 ++++-- sound/soc/codecs/wm8961.c | 7 ++++-- sound/soc/codecs/wm8962.c | 7 ++++-- sound/soc/codecs/wm8971.c | 7 ++++-- sound/soc/codecs/wm8974.c | 7 ++++-- sound/soc/codecs/wm8978.c | 7 ++++-- sound/soc/codecs/wm8983.c | 7 ++++-- sound/soc/codecs/wm8985.c | 7 ++++-- sound/soc/codecs/wm8988.c | 7 ++++-- sound/soc/codecs/wm8990.c | 7 ++++-- sound/soc/codecs/wm8991.c | 7 ++++-- sound/soc/codecs/wm8993.c | 7 ++++-- sound/soc/codecs/wm8994.c | 10 +++++--- sound/soc/codecs/wm8995.c | 9 ++++--- sound/soc/codecs/wm9081.c | 7 ++++-- sound/soc/meson/axg-spdifout.c | 7 ++++-- sound/soc/soc-core.c | 4 +-- sound/soc/soc-dai.c | 4 --- sound/soc/spear/spdif_out.c | 10 +++++--- sound/soc/ti/ams-delta.c | 11 +++++--- 105 files changed, 576 insertions(+), 243 deletions(-) -- 2.25.1

7 25

[RESEND PATCH v2 0/2] Introduce multi PM domains helpers
by Daniel Baluta 23 Jun '20

23 Jun '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Changes since v1: (addressed Ranjani's comments) - enhanced description for dev_multi_pm_attach return value - renamed exit_unroll_pm label to exit_detach_pm Ideally would be to have patch 1/2 merged via power tree and then I will submit again patch 2/2 on sound tree. Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 60 ++++-------------------- 3 files changed, 121 insertions(+), 51 deletions(-) -- 2.17.1

2 6

[PATCH] sound: usb: quirks: add quirk for Samsung USBC Headset (AKG)
by Macpaul Lin 23 Jun '20

23 Jun '20

We've found Samsung USBC Headset (AKG) (VID: 0x04e8, PID: 0xa051) need a tiny delay after each class compliant request. Otherwise the device might not be able to be recognized each times. Signed-off-by: Chihhao Chen <chihhao.chen(a)mediatek.com> Signed-off-by: Macpaul Lin <macpaul.lin(a)mediatek.com> Cc: stable(a)vger.kernel.org --- sound/usb/quirks.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index bca0179..ebba29a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1673,6 +1673,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) usleep_range(1000, 2000); + + /* + * Samsung USBC Headset (AKG) need a tiny delay after each + * class compliant request. (Model number: AAM625R or AAM627R) + */ + if (chip->usb_id == USB_ID(0x04e8, 0xa051) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + usleep_range(5000, 6000); } /* -- 1.7.9.5

2 1

Plantronics Savi 8200 USB Headset Volume Controls
by Roth Richard 23 Jun '20

23 Jun '20

Hi all, I'm currently experiencing the following issue with a (usb) Plantronics Savi 8200 Office Series headset When changing the volume via the buttons on the headset, while the volume changes on the headset itself, the change is not reflected in 'alsamixer', with the (PCM) volume level remaining unchanged from its previous value. However, when changing the volume via 'amixer', the change is realized on the headset. Has anyone used this headset with alsa or able to provide some help (maybe as simple as a missing module) to get this working? Yocto Release: Krogoth Alsa-lib Version: 1.1.0 Headset: Plantronics Savi 8200 Office Series Best Regards and thank you in advance, Richard

1 0

Re: [PATCH v2 1/2] ASoC: fsl_mqs: Don't check clock is NULL before calling clk API
by Markus Elfring 23 Jun '20

23 Jun '20

> In-Reply-To: <cover.1592888591.git.shengjiu.wang(a)nxp.com> I guess that it should be sufficient to specify such a field once for the header information. > Because clk_prepare_enable and clk_disable_unprepare should > check input clock parameter is NULL or not internally, I find this change description unclear. > then we don't need to check them before calling the function. Please use an imperative wording for the commit message. https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… Regards, Markus

2 4

[PATCH] ASoC: fsl_easrc: Fix uninitialized scalar variable in fsl_easrc_set_ctx_format
by Shengjiu Wang 23 Jun '20

23 Jun '20

The "ret" in fsl_easrc_set_ctx_format is not initialized, then the unknown value maybe returned by this function. Fixes: 955ac624058f ("ASoC: fsl_easrc: Add EASRC ASoC CPU DAI drivers") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_easrc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 2f6b3d8bfcfc..03b3aef41d34 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1132,7 +1132,7 @@ static int fsl_easrc_set_ctx_format(struct fsl_asrc_pair *ctx, struct fsl_easrc_ctx_priv *ctx_priv = ctx->private; struct fsl_easrc_data_fmt *in_fmt = &ctx_priv->in_params.fmt; struct fsl_easrc_data_fmt *out_fmt = &ctx_priv->out_params.fmt; - int ret; + int ret = 0; /* Get the bitfield values for input data format */ if (in_raw_format && out_raw_format) { -- 2.21.0

3 2

[PATCH v2 0/2] Fix unchecked return value for clk_prepare_enable
by Shengjiu Wang 23 Jun '20

23 Jun '20

First patch is to remove the check of clock pointer before calling clk API. Second patch is to fix the issue that the return value of clk_prepare_enable is not checked. changes in v2: - split the patch to separate patches Shengjiu Wang (2): ASoC: fsl_mqs: Don't check clock is NULL before calling clk API ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable sound/soc/fsl/fsl_mqs.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) -- 2.21.0

3 5

[PATCH] ASoC: rt1011: fix KASAN out-of-bounds bug in find_next_bit()
by Pierre-Louis Bossart 23 Jun '20

23 Jun '20

From: Fred Oh <fred.oh(a)linux.intel.com> KASAN throws the following warning in rt1011.c: [ 170.777603] BUG: KASAN: stack-out-of-bounds in _find_next_bit.constprop.0+0x3e/0xf0 find_next_bit() relies on unsigned long pointer arguments, but this driver uses a type cast that generates the KASAN warning. Replace find_next_bit() and find_last_bit() with __ffs() and __fls() to pass the value and avoid casting pointers to make the warning go away. Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Fred Oh <fred.oh(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt1011.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index dec5638060c3..098ecf13814d 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1849,13 +1849,13 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, /* Rx slot configuration */ rx_slotnum = hweight_long(rx_mask); - first_bit = find_next_bit((unsigned long *)&rx_mask, 32, 0); - if (rx_slotnum > 1 || rx_slotnum == 0) { + if (rx_slotnum > 1 || !rx_slotnum) { ret = -EINVAL; - dev_dbg(component->dev, "too many rx slots or zero slot\n"); + dev_err(component->dev, "too many rx slots or zero slot\n"); goto _set_tdm_err_; } + first_bit = __ffs(rx_mask); switch (first_bit) { case 0: case 2: @@ -1892,11 +1892,17 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, /* Tx slot configuration */ tx_slotnum = hweight_long(tx_mask); - first_bit = find_next_bit((unsigned long *)&tx_mask, 32, 0); - last_bit = find_last_bit((unsigned long *)&tx_mask, 32); - if (tx_slotnum > 2 || (last_bit-first_bit) > 1) { + if (tx_slotnum > 2 || !tx_slotnum) { ret = -EINVAL; - dev_dbg(component->dev, "too many tx slots or tx slot location error\n"); + dev_err(component->dev, "too many tx slots or zero slot\n"); + goto _set_tdm_err_; + } + + first_bit = __ffs(tx_mask); + last_bit = __fls(tx_mask); + if (last_bit - first_bit > 1) { + ret = -EINVAL; + dev_err(component->dev, "tx slot location error\n"); goto _set_tdm_err_; } base-commit: 39853b1438bf9b07349c8c44b48f6c2eda6f8840 -- 2.20.1

2 1