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From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
To: Mark Brown <broonie@kernel.org>
Cc: Linux-ALSA <alsa-devel@alsa-project.org>
Subject: [PATCH v3 12/21] ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream()
Date: 09 Jul 2020 10:56:39 +0900	[thread overview]
Message-ID: <87zh89wix5.wl-kuninori.morimoto.gx@renesas.com> (raw)
In-Reply-To: <87h7uhxxk6.wl-kuninori.morimoto.gx@renesas.com>

From: Kuninori Morimoto <kuninori.morimoto.gx@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@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
 sound/soc/codecs/pcm1681.c  | 5 +++--
 sound/soc/codecs/pcm1789.c  | 5 +++--
 sound/soc/codecs/pcm179x.c  | 5 +++--
 sound/soc/codecs/pcm3168a.c | 5 +++--
 sound/soc/codecs/pcm512x.c  | 5 +++--
 5 files changed, 15 insertions(+), 10 deletions(-)

diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 4767e158cd5e..07ed8fded471 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -147,7 +147,7 @@ static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute)
+static int pcm1681_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct pcm1681_private *priv = snd_soc_component_get_drvdata(component);
@@ -205,7 +205,8 @@ static int pcm1681_hw_params(struct snd_pcm_substream *substream,
 static const struct snd_soc_dai_ops pcm1681_dai_ops = {
 	.set_fmt	= pcm1681_set_dai_fmt,
 	.hw_params	= pcm1681_hw_params,
-	.digital_mute	= pcm1681_digital_mute,
+	.mute_stream	= pcm1681_mute,
+	.no_capture_mute = 1,
 };
 
 static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = {
diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c
index 8df6447c76a6..620dec172ce7 100644
--- a/sound/soc/codecs/pcm1789.c
+++ b/sound/soc/codecs/pcm1789.c
@@ -60,7 +60,7 @@ static int pcm1789_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int pcm1789_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int pcm1789_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	struct pcm1789_private *priv = snd_soc_component_get_drvdata(component);
@@ -167,8 +167,9 @@ static int pcm1789_trigger(struct snd_pcm_substream *substream, int cmd,
 static const struct snd_soc_dai_ops pcm1789_dai_ops = {
 	.set_fmt	= pcm1789_set_dai_fmt,
 	.hw_params	= pcm1789_hw_params,
-	.digital_mute	= pcm1789_digital_mute,
+	.mute_stream	= pcm1789_mute,
 	.trigger	= pcm1789_trigger,
+	.no_capture_mute = 1,
 };
 
 static const DECLARE_TLV_DB_SCALE(pcm1789_dac_tlv, -12000, 50, 1);
diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c
index 9e70b7385c69..ee60373d7d25 100644
--- a/sound/soc/codecs/pcm179x.c
+++ b/sound/soc/codecs/pcm179x.c
@@ -76,7 +76,7 @@ static int pcm179x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int pcm179x_digital_mute(struct snd_soc_dai *dai, int mute)
+static int pcm179x_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct pcm179x_private *priv = snd_soc_component_get_drvdata(component);
@@ -145,7 +145,8 @@ static int pcm179x_hw_params(struct snd_pcm_substream *substream,
 static const struct snd_soc_dai_ops pcm179x_dai_ops = {
 	.set_fmt	= pcm179x_set_dai_fmt,
 	.hw_params	= pcm179x_hw_params,
-	.digital_mute	= pcm179x_digital_mute,
+	.mute_stream	= pcm179x_mute,
+	.no_capture_mute = 1,
 };
 
 static const DECLARE_TLV_DB_SCALE(pcm179x_dac_tlv, -12000, 50, 1);
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 9711fab296eb..5e445fee4ef5 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -290,7 +290,7 @@ static int pcm3168a_reset(struct pcm3168a_priv *pcm3168a)
 			PCM3168A_MRST_MASK | PCM3168A_SRST_MASK);
 }
 
-static int pcm3168a_digital_mute(struct snd_soc_dai *dai, int mute)
+static int pcm3168a_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
@@ -570,8 +570,9 @@ static const struct snd_soc_dai_ops pcm3168a_dai_ops = {
 	.set_fmt	= pcm3168a_set_dai_fmt,
 	.set_sysclk	= pcm3168a_set_dai_sysclk,
 	.hw_params	= pcm3168a_hw_params,
-	.digital_mute	= pcm3168a_digital_mute,
+	.mute_stream	= pcm3168a_mute,
 	.set_tdm_slot	= pcm3168a_set_tdm_slot,
+	.no_capture_mute = 1,
 };
 
 static struct snd_soc_dai_driver pcm3168a_dais[] = {
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 4cbef9affffd..8153d3d01654 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -1394,7 +1394,7 @@ static int pcm512x_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
 	return 0;
 }
 
-static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute)
+static int pcm512x_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component);
@@ -1445,8 +1445,9 @@ static const struct snd_soc_dai_ops pcm512x_dai_ops = {
 	.startup = pcm512x_dai_startup,
 	.hw_params = pcm512x_hw_params,
 	.set_fmt = pcm512x_set_fmt,
-	.digital_mute = pcm512x_digital_mute,
+	.mute_stream = pcm512x_mute,
 	.set_bclk_ratio = pcm512x_set_bclk_ratio,
+	.no_capture_mute = 1,
 };
 
 static struct snd_soc_dai_driver pcm512x_dai = {
-- 
2.25.1


  parent reply	other threads:[~2020-07-09  2:03 UTC|newest]

Thread overview: 26+ messages / expand[flat|nested]  mbox.gz  Atom feed  top
2020-07-09  1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
2020-07-09  1:55 ` [PATCH v3 01/21] ASoC: hdmi-codec: return -ENOTSUPP for digital_mute Kuninori Morimoto
2020-07-09  1:55 ` [PATCH v3 02/21] ASoC: soc-dai.c: add .no_capture_mute support Kuninori Morimoto
2020-07-09  1:55 ` [PATCH v3 03/21] ASoC: hdmi-codec: merge .digital_mute() into .mute_stream() Kuninori Morimoto
2020-07-09  1:55 ` [PATCH v3 04/21] ASoC: ti: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 05/21] ASoC: spear: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 06/21] ASoC: meson: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 07/21] ASoC: atmel: " Kuninori Morimoto
2020-07-16 20:44   ` Mark Brown
2020-07-16 23:27     ` Kuninori Morimoto
2020-07-16 23:59       ` Mark Brown
2020-07-09  1:56 ` [PATCH v3 08/21] ASoC: codecs: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 09/21] ASoC: codecs: tlv*: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 10/21] ASoC: codecs: tas*: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 11/21] ASoC: codecs: ssm*: " Kuninori Morimoto
2020-07-09  1:56 ` Kuninori Morimoto [this message]
2020-07-09  1:56 ` [PATCH v3 13/21] ASoC: codecs: max*: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 14/21] ASoC: codecs: alc*: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 15/21] ASoC: codecs: wm*: " Kuninori Morimoto
2020-07-09  1:56 ` [PATCH v3 16/21] ASoC: codecs: es*: " Kuninori Morimoto
2020-07-09  1:57 ` [PATCH v3 17/21] ASoC: codecs: da*: " Kuninori Morimoto
2020-07-09  1:57 ` [PATCH v3 18/21] ASoC: codecs: cs*: " Kuninori Morimoto
2020-07-09  1:57 ` [PATCH v3 19/21] ASoC: codecs: ak*: " Kuninori Morimoto
2020-07-09  1:57 ` [PATCH v3 20/21] ASoC: soc-dai: remove .digital_mute Kuninori Morimoto
2020-07-09  1:57 ` [PATCH v3 21/21] ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec Kuninori Morimoto
2020-07-16 23:56 ` [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Mark Brown

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