Re: [PATCH 16/19] ASoC: codecs: cs*: merge .digital_mute() into .mute_stream()

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On Mon, 23 Jun 2020, Kuninori Morimoto wrote:

> From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
>
> 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@xxxxxxxxxxx>
> ---
> 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@xxxxxxxxxx>

Thanks,
James



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