[PATCH 00/19] ASoC: merge .digital_mute() into .mute_stream()

[PATCH 00/19] ASoC: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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

[PATCH 01/19] ASoC: hdmi-codec: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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);
		...
	}

For hdmi-codec, we need to update struct hdmi_codec_ops,
and all its users in the same time.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
 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 ++-
 sound/soc/codecs/hdmi-codec.c          | 13 +++++++------
 9 files changed, 51 insertions(+), 25 deletions(-)

diff --git a/drivers/gpu/drm/bridge/sii902x.c b/drivers/gpu/drm/bridge/sii902x.c
index 6dad025f8da7..68b850f3a027 100644
--- a/drivers/gpu/drm/bridge/sii902x.c
+++ b/drivers/gpu/drm/bridge/sii902x.c
@@ -672,11 +672,14 @@ static void sii902x_audio_shutdown(struct device *dev, void *data)
 	clk_disable_unprepare(sii902x->audio.mclk);
 }
 
-static int sii902x_audio_digital_mute(struct device *dev,
-				      void *data, bool enable)
+static int sii902x_audio_mute(struct device *dev, void *data,
+			      bool enable, int direction)
 {
 	struct sii902x *sii902x = dev_get_drvdata(dev);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	mutex_lock(&sii902x->mutex);
 
 	sii902x_mute(sii902x, enable);
@@ -724,7 +727,7 @@ static int sii902x_audio_get_dai_id(struct snd_soc_component *component,
 static const struct hdmi_codec_ops sii902x_audio_codec_ops = {
 	.hw_params = sii902x_audio_hw_params,
 	.audio_shutdown = sii902x_audio_shutdown,
-	.digital_mute = sii902x_audio_digital_mute,
+	.mute_stream = sii902x_audio_mute,
 	.get_eld = sii902x_audio_get_eld,
 	.get_dai_id = sii902x_audio_get_dai_id,
 };
diff --git a/drivers/gpu/drm/exynos/exynos_hdmi.c b/drivers/gpu/drm/exynos/exynos_hdmi.c
index 95dd399aa9cc..f0f8fb9c62ec 100644
--- a/drivers/gpu/drm/exynos/exynos_hdmi.c
+++ b/drivers/gpu/drm/exynos/exynos_hdmi.c
@@ -1604,10 +1604,14 @@ static int hdmi_audio_hw_params(struct device *dev, void *data,
 	return 0;
 }
 
-static int hdmi_audio_digital_mute(struct device *dev, void *data, bool mute)
+static int hdmi_audio_mute(struct device *dev, void *data,
+			   bool mute, int direction)
 {
 	struct hdmi_context *hdata = dev_get_drvdata(dev);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	mutex_lock(&hdata->mutex);
 
 	hdata->audio.mute = mute;
@@ -1634,7 +1638,7 @@ static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf,
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = hdmi_audio_hw_params,
 	.audio_shutdown = hdmi_audio_shutdown,
-	.digital_mute = hdmi_audio_digital_mute,
+	.mute_stream = hdmi_audio_mute,
 	.get_eld = hdmi_audio_get_eld,
 };
 
diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c
index 9517f522dcb9..e24cbfbb154a 100644
--- a/drivers/gpu/drm/i2c/tda998x_drv.c
+++ b/drivers/gpu/drm/i2c/tda998x_drv.c
@@ -1133,11 +1133,14 @@ static void tda998x_audio_shutdown(struct device *dev, void *data)
 	mutex_unlock(&priv->audio_mutex);
 }
 
-static int tda998x_audio_digital_mute(struct device *dev, void *data,
-				      bool enable)
+static int tda998x_audio_mute_stream(struct device *dev, void *data,
+				     bool enable, int direction)
 {
 	struct tda998x_priv *priv = dev_get_drvdata(dev);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	mutex_lock(&priv->audio_mutex);
 
 	tda998x_audio_mute(priv, enable);
@@ -1162,7 +1165,7 @@ static int tda998x_audio_get_eld(struct device *dev, void *data,
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = tda998x_audio_hw_params,
 	.audio_shutdown = tda998x_audio_shutdown,
-	.digital_mute = tda998x_audio_digital_mute,
+	.mute_stream = tda998x_audio_mute_stream,
 	.get_eld = tda998x_audio_get_eld,
 };
 
diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c
index 5feb760617cb..7721b4762758 100644
--- a/drivers/gpu/drm/mediatek/mtk_hdmi.c
+++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c
@@ -1647,10 +1647,14 @@ static void mtk_hdmi_audio_shutdown(struct device *dev, void *data)
 }
 
 static int
-mtk_hdmi_audio_digital_mute(struct device *dev, void *data, bool enable)
+mtk_hdmi_audio_mute(struct device *dev, void *data,
+		    bool enable, int direction)
 {
 	struct mtk_hdmi *hdmi = dev_get_drvdata(dev);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	dev_dbg(dev, "%s(%d)\n", __func__, enable);
 
 	if (enable)
@@ -1692,7 +1696,7 @@ static const struct hdmi_codec_ops mtk_hdmi_audio_codec_ops = {
 	.hw_params = mtk_hdmi_audio_hw_params,
 	.audio_startup = mtk_hdmi_audio_startup,
 	.audio_shutdown = mtk_hdmi_audio_shutdown,
-	.digital_mute = mtk_hdmi_audio_digital_mute,
+	.mute_stream = mtk_hdmi_audio_mute,
 	.get_eld = mtk_hdmi_audio_get_eld,
 	.hook_plugged_cb = mtk_hdmi_audio_hook_plugged_cb,
 };
diff --git a/drivers/gpu/drm/rockchip/cdn-dp-core.c b/drivers/gpu/drm/rockchip/cdn-dp-core.c
index c634b95b50f7..aacde58b8fe8 100644
--- a/drivers/gpu/drm/rockchip/cdn-dp-core.c
+++ b/drivers/gpu/drm/rockchip/cdn-dp-core.c
@@ -817,12 +817,15 @@ static void cdn_dp_audio_shutdown(struct device *dev, void *data)
 	mutex_unlock(&dp->lock);
 }
 
-static int cdn_dp_audio_digital_mute(struct device *dev, void *data,
-				     bool enable)
+static int cdn_dp_audio_mute_stream(struct device *dev, void *data,
+				    bool enable, int direction)
 {
 	struct cdn_dp_device *dp = dev_get_drvdata(dev);
 	int ret;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	mutex_lock(&dp->lock);
 	if (!dp->active) {
 		ret = -ENODEV;
@@ -849,7 +852,7 @@ static int cdn_dp_audio_get_eld(struct device *dev, void *data,
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = cdn_dp_audio_hw_params,
 	.audio_shutdown = cdn_dp_audio_shutdown,
-	.digital_mute = cdn_dp_audio_digital_mute,
+	.mute_stream = cdn_dp_audio_mute_stream,
 	.get_eld = cdn_dp_audio_get_eld,
 };
 
diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c
index 5b15c4974e6b..471ac2139a64 100644
--- a/drivers/gpu/drm/sti/sti_hdmi.c
+++ b/drivers/gpu/drm/sti/sti_hdmi.c
@@ -1191,10 +1191,14 @@ static int hdmi_audio_hw_params(struct device *dev,
 	return 0;
 }
 
-static int hdmi_audio_digital_mute(struct device *dev, void *data, bool enable)
+static int hdmi_audio_mute(struct device *dev, void *data,
+			   bool enable, int direction)
 {
 	struct sti_hdmi *hdmi = dev_get_drvdata(dev);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	DRM_DEBUG_DRIVER("%s\n", enable ? "enable" : "disable");
 
 	if (enable)
@@ -1219,7 +1223,7 @@ static int hdmi_audio_get_eld(struct device *dev, void *data, uint8_t *buf, size
 static const struct hdmi_codec_ops audio_codec_ops = {
 	.hw_params = hdmi_audio_hw_params,
 	.audio_shutdown = hdmi_audio_shutdown,
-	.digital_mute = hdmi_audio_digital_mute,
+	.mute_stream = hdmi_audio_mute,
 	.get_eld = hdmi_audio_get_eld,
 };
 
diff --git a/drivers/gpu/drm/zte/zx_hdmi.c b/drivers/gpu/drm/zte/zx_hdmi.c
index 76a16d997a23..bb4ab395f71d 100644
--- a/drivers/gpu/drm/zte/zx_hdmi.c
+++ b/drivers/gpu/drm/zte/zx_hdmi.c
@@ -439,11 +439,14 @@ static int zx_hdmi_audio_hw_params(struct device *dev,
 	return zx_hdmi_infoframe_trans(hdmi, &frame, FSEL_AUDIO);
 }
 
-static int zx_hdmi_audio_digital_mute(struct device *dev, void *data,
-				      bool enable)
+static int zx_hdmi_audio_mute(struct device *dev, void *data,
+			      bool enable, int direction)
 {
 	struct zx_hdmi *hdmi = dev_get_drvdata(dev);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (enable)
 		hdmi_writeb_mask(hdmi, TPI_AUD_CONFIG, TPI_AUD_MUTE,
 				 TPI_AUD_MUTE);
@@ -468,7 +471,7 @@ static const struct hdmi_codec_ops zx_hdmi_codec_ops = {
 	.audio_startup = zx_hdmi_audio_startup,
 	.hw_params = zx_hdmi_audio_hw_params,
 	.audio_shutdown = zx_hdmi_audio_shutdown,
-	.digital_mute = zx_hdmi_audio_digital_mute,
+	.mute_stream = zx_hdmi_audio_mute,
 	.get_eld = zx_hdmi_audio_get_eld,
 };
 
diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h
index 83b17682e01c..4b7cc86bae95 100644
--- a/include/sound/hdmi-codec.h
+++ b/include/sound/hdmi-codec.h
@@ -76,7 +76,8 @@ struct hdmi_codec_ops {
 	 * Mute/unmute HDMI audio stream.
 	 * Optional
 	 */
-	int (*digital_mute)(struct device *dev, void *data, bool enable);
+	int (*mute_stream)(struct device *dev, void *data,
+			   bool enable, int direction);
 
 	/*
 	 * Provides EDID-Like-Data from connected HDMI device.
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index f005751da2cc..422ed5cd427f 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -558,13 +558,14 @@ static int hdmi_codec_i2s_set_fmt(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute)
+static int hdmi_codec_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
 
-	if (hcp->hcd.ops->digital_mute)
-		return hcp->hcd.ops->digital_mute(dai->dev->parent,
-						  hcp->hcd.data, mute);
+	if (hcp->hcd.ops->mute_stream)
+		return hcp->hcd.ops->mute_stream(dai->dev->parent,
+						 hcp->hcd.data,
+						 mute, direction);
 
 	return 0;
 }
@@ -574,14 +575,14 @@ static const struct snd_soc_dai_ops hdmi_codec_i2s_dai_ops = {
 	.shutdown	= hdmi_codec_shutdown,
 	.hw_params	= hdmi_codec_hw_params,
 	.set_fmt	= hdmi_codec_i2s_set_fmt,
-	.digital_mute	= hdmi_codec_digital_mute,
+	.mute_stream	= hdmi_codec_mute,
 };
 
 static const struct snd_soc_dai_ops hdmi_codec_spdif_dai_ops = {
 	.startup	= hdmi_codec_startup,
 	.shutdown	= hdmi_codec_shutdown,
 	.hw_params	= hdmi_codec_hw_params,
-	.digital_mute	= hdmi_codec_digital_mute,
+	.mute_stream	= hdmi_codec_mute,
 };
 
 #define HDMI_RATES	(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
-- 
2.25.1

[PATCH 02/19] ASoC: ti: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/ti/ams-delta.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index e17cd5e939f0..4fd3ec39bde4 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -420,10 +420,13 @@ static struct snd_soc_ops ams_delta_ops;
  * Shares hardware with codec config pulse generation */
 static bool ams_delta_muted = 1;
 
-static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
+static int ams_delta_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	int apply;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (ams_delta_muted == mute)
 		return 0;
 
@@ -439,18 +442,18 @@ static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
 
 /* Our codec DAI probably doesn't have its own .ops structure */
 static const struct snd_soc_dai_ops ams_delta_dai_ops = {
-	.digital_mute = ams_delta_digital_mute,
+	.mute_stream = ams_delta_mute,
 };
 
 /* Will be used if the codec ever has its own digital_mute function */
 static int ams_delta_startup(struct snd_pcm_substream *substream)
 {
-	return ams_delta_digital_mute(NULL, 0);
+	return ams_delta_digital_mute(NULL, 0, substream->stream);
 }
 
 static void ams_delta_shutdown(struct snd_pcm_substream *substream)
 {
-	ams_delta_digital_mute(NULL, 1);
+	ams_delta_digital_mute(NULL, 1, substream->stream);
 }
 
 
-- 
2.25.1

[PATCH 03/19] ASoC: spear: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/spear/spdif_out.c | 10 +++++++---
 1 file changed, 7 insertions(+), 3 deletions(-)

diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index 58d5843811f9..c53c59f3b359 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -188,11 +188,14 @@ static int spdif_out_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
-static int spdif_digital_mute(struct snd_soc_dai *dai, int mute)
+static int spdif_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
 	u32 val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	host->saved_params.mute = mute;
 	val = readl(host->io_base + SPDIF_OUT_CTRL);
 	val &= ~SPDIF_OPMODE_MASK;
@@ -229,7 +232,8 @@ static int spdif_mute_put(struct snd_kcontrol *kcontrol,
 	if (host->saved_params.mute == ucontrol->value.integer.value[0])
 		return 0;
 
-	spdif_digital_mute(cpu_dai, ucontrol->value.integer.value[0]);
+	spdif_mute(cpu_dai, ucontrol->value.integer.value[0],
+		   SNDRV_PCM_STREAM_PLAYBACK);
 
 	return 1;
 }
@@ -250,7 +254,7 @@ static int spdif_soc_dai_probe(struct snd_soc_dai *dai)
 }
 
 static const struct snd_soc_dai_ops spdif_out_dai_ops = {
-	.digital_mute	= spdif_digital_mute,
+	.mute_stream	= spdif_mute,
 	.startup	= spdif_out_startup,
 	.shutdown	= spdif_out_shutdown,
 	.trigger	= spdif_out_trigger,
-- 
2.25.1

[PATCH 04/19] ASoC: meson: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/meson/axg-spdifout.c | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c
index 7ce6aa97ddf7..b5dcbfc35a1b 100644
--- a/sound/soc/meson/axg-spdifout.c
+++ b/sound/soc/meson/axg-spdifout.c
@@ -108,10 +108,13 @@ static int axg_spdifout_trigger(struct snd_pcm_substream *substream, int cmd,
 	}
 }
 
-static int axg_spdifout_digital_mute(struct snd_soc_dai *dai, int mute)
+static int axg_spdifout_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct axg_spdifout *priv = snd_soc_dai_get_drvdata(dai);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	/* Use spdif valid bit to perform digital mute */
 	regmap_update_bits(priv->map, SPDIFOUT_CTRL0, SPDIFOUT_CTRL0_VSET,
 			   mute ? SPDIFOUT_CTRL0_VSET : 0);
@@ -285,7 +288,7 @@ static void axg_spdifout_shutdown(struct snd_pcm_substream *substream,
 
 static const struct snd_soc_dai_ops axg_spdifout_ops = {
 	.trigger	= axg_spdifout_trigger,
-	.digital_mute	= axg_spdifout_digital_mute,
+	.mute_stream	= axg_spdifout_mute,
 	.hw_params	= axg_spdifout_hw_params,
 	.startup	= axg_spdifout_startup,
 	.shutdown	= axg_spdifout_shutdown,
-- 
2.25.1

[PATCH 05/19] ASoC: atmel: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/atmel/atmel-classd.c | 10 +++++++---
 1 file changed, 7 insertions(+), 3 deletions(-)

diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index e98601eccfa3..59231c0371b8 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/sound/soc/atmel/atmel-classd.c
@@ -327,12 +327,16 @@ static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream,
 	return clk_prepare_enable(dd->gclk);
 }
 
-static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai,
-	int mute)
+static int atmel_classd_codec_dai_mute(struct snd_soc_dai *codec_dai,
+				       int mute,
+				       int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	u32 mask, val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK;
 
 	if (mute)
@@ -469,7 +473,7 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream,
 }
 
 static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = {
-	.digital_mute	= atmel_classd_codec_dai_digital_mute,
+	.mute_stream	= atmel_classd_codec_dai_mute,
 	.startup	= atmel_classd_codec_dai_startup,
 	.shutdown	= atmel_classd_codec_dai_shutdown,
 	.hw_params	= atmel_classd_codec_dai_hw_params,
-- 
2.25.1

[PATCH 06/19] ASoC: codecs: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/codecs/88pm860x-codec.c |  9 +++++---
 sound/soc/codecs/ad193x.c         |  7 +++++--
 sound/soc/codecs/adau1701.c       |  7 +++++--
 sound/soc/codecs/cpcap.c          | 15 +++++++++----
 sound/soc/codecs/cq93vc.c         |  7 +++++--
 sound/soc/codecs/isabelle.c       | 21 +++++++++++++------
 sound/soc/codecs/jz4770.c         |  7 +++++--
 sound/soc/codecs/lm49453.c        | 35 ++++++++++++++++++++++---------
 sound/soc/codecs/ml26124.c        |  7 +++++--
 sound/soc/codecs/nau8822.c        |  7 +++++--
 sound/soc/codecs/rk3328_codec.c   |  7 +++++--
 sound/soc/codecs/sgtl5000.c       |  7 +++++--
 sound/soc/codecs/sta529.c         |  7 +++++--
 sound/soc/codecs/tfa9879.c        |  7 +++++--
 sound/soc/codecs/twl6040.c        |  7 +++++--
 sound/soc/codecs/uda134x.c        |  7 +++++--
 16 files changed, 117 insertions(+), 47 deletions(-)

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 068914d0ef3d..c668029258b0 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -902,11 +902,14 @@ static const struct snd_soc_dapm_route pm860x_dapm_routes[] = {
  * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
  * These bits can also be used to mute.
  */
-static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int pm860x_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		data = mask;
 	snd_soc_component_update_bits(component, PM860X_DAC_OFFSET, mask, data);
@@ -1136,14 +1139,14 @@ static int pm860x_set_bias_level(struct snd_soc_component *component,
 }
 
 static const struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
-	.digital_mute	= pm860x_digital_mute,
+	.mute_stream	= pm860x_mute,
 	.hw_params	= pm860x_pcm_hw_params,
 	.set_fmt	= pm860x_pcm_set_dai_fmt,
 	.set_sysclk	= pm860x_set_dai_sysclk,
 };
 
 static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
-	.digital_mute	= pm860x_digital_mute,
+	.mute_stream	= pm860x_mute,
 	.hw_params	= pm860x_i2s_hw_params,
 	.set_fmt	= pm860x_i2s_set_dai_fmt,
 	.set_sysclk	= pm860x_set_dai_sysclk,
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 980e024a5720..5af815f1028d 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -143,10 +143,13 @@ static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x)
  * DAI ops entries
  */
 
-static int ad193x_mute(struct snd_soc_dai *dai, int mute)
+static int ad193x_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(dai->component);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL2,
 				    AD193X_DAC_MASTER_MUTE,
@@ -371,7 +374,7 @@ static int ad193x_startup(struct snd_pcm_substream *substream,
 static const struct snd_soc_dai_ops ad193x_dai_ops = {
 	.startup = ad193x_startup,
 	.hw_params = ad193x_hw_params,
-	.digital_mute = ad193x_mute,
+	.mute_stream = ad193x_mute,
 	.set_tdm_slot = ad193x_set_tdm_slot,
 	.set_sysclk	= ad193x_set_dai_sysclk,
 	.set_fmt = ad193x_set_dai_fmt,
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 115e296b2ad6..2aa41f91386f 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -573,13 +573,16 @@ static int adau1701_set_bias_level(struct snd_soc_component *component,
 	return 0;
 }
 
-static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute)
+static int adau1701_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	unsigned int mask = ADAU1701_DSPCTRL_DAM;
 	struct adau1701 *adau1701 = snd_soc_component_get_drvdata(component);
 	unsigned int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = 0;
 	else
@@ -631,7 +634,7 @@ static int adau1701_startup(struct snd_pcm_substream *substream,
 static const struct snd_soc_dai_ops adau1701_dai_ops = {
 	.set_fmt	= adau1701_set_dai_fmt,
 	.hw_params	= adau1701_hw_params,
-	.digital_mute	= adau1701_digital_mute,
+	.mute_stream	= adau1701_mute,
 	.startup	= adau1701_startup,
 };
 
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
index d7f05b384f1f..24a1d1196d3a 100644
--- a/sound/soc/codecs/cpcap.c
+++ b/sound/soc/codecs/cpcap.c
@@ -1216,7 +1216,7 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return regmap_update_bits(cpcap->regmap, reg, mask, val);
 }
 
-static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute)
+static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
@@ -1224,6 +1224,9 @@ static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute)
 	static const u16 mask = BIT(CPCAP_BIT_ST_DAC_SW);
 	u16 val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = 0;
 	else
@@ -1237,7 +1240,7 @@ static const struct snd_soc_dai_ops cpcap_dai_hifi_ops = {
 	.hw_params	= cpcap_hifi_hw_params,
 	.set_sysclk	= cpcap_hifi_set_dai_sysclk,
 	.set_fmt	= cpcap_hifi_set_dai_fmt,
-	.digital_mute	= cpcap_hifi_set_mute,
+	.mute_stream	= cpcap_hifi_set_mute,
 };
 
 static int cpcap_voice_hw_params(struct snd_pcm_substream *substream,
@@ -1370,7 +1373,8 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute)
+static int cpcap_voice_set_mute(struct snd_soc_dai *dai,
+				int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component);
@@ -1378,6 +1382,9 @@ static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute)
 	static const u16 mask = BIT(CPCAP_BIT_CDC_SW);
 	u16 val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = 0;
 	else
@@ -1391,7 +1398,7 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = {
 	.hw_params	= cpcap_voice_hw_params,
 	.set_sysclk	= cpcap_voice_set_dai_sysclk,
 	.set_fmt	= cpcap_voice_set_dai_fmt,
-	.digital_mute	= cpcap_voice_set_mute,
+	.mute_stream	= cpcap_voice_set_mute,
 };
 
 static struct snd_soc_dai_driver cpcap_dai[] = {
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index b0cc61178a41..4a41cc551bdd 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -30,11 +30,14 @@ static const struct snd_kcontrol_new cq93vc_snd_controls[] = {
 	SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0),
 };
 
-static int cq93vc_mute(struct snd_soc_dai *dai, int mute)
+static int cq93vc_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u8 reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		reg = DAVINCI_VC_REG09_MUTE;
 	else
@@ -87,7 +90,7 @@ static int cq93vc_set_bias_level(struct snd_soc_component *component,
 #define CQ93VC_FORMATS	(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE)
 
 static const struct snd_soc_dai_ops cq93vc_dai_ops = {
-	.digital_mute	= cq93vc_mute,
+	.mute_stream	= cq93vc_mute,
 	.set_sysclk	= cq93vc_set_dai_sysclk,
 };
 
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 3626f70f7768..204b89d6aa4a 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -860,24 +860,33 @@ static const struct snd_soc_dapm_route isabelle_intercon[] = {
 	{ "LINEOUT2", NULL, "LINEOUT2 Driver" },
 };
 
-static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute)
+static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, ISABELLE_DAC1_SOFTRAMP_REG,
 			BIT(4), (mute ? BIT(4) : 0));
 
 	return 0;
 }
 
-static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute)
+static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, ISABELLE_DAC2_SOFTRAMP_REG,
 			BIT(4), (mute ? BIT(4) : 0));
 
 	return 0;
 }
 
-static int isabelle_line_mute(struct snd_soc_dai *dai, int mute)
+static int isabelle_line_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, ISABELLE_DAC3_SOFTRAMP_REG,
 			BIT(4), (mute ? BIT(4) : 0));
 
@@ -1014,19 +1023,19 @@ static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 static const struct snd_soc_dai_ops isabelle_hs_dai_ops = {
 	.hw_params	= isabelle_hw_params,
 	.set_fmt	= isabelle_set_dai_fmt,
-	.digital_mute	= isabelle_hs_mute,
+	.mute_stream	= isabelle_hs_mute,
 };
 
 static const struct snd_soc_dai_ops isabelle_hf_dai_ops = {
 	.hw_params	= isabelle_hw_params,
 	.set_fmt	= isabelle_set_dai_fmt,
-	.digital_mute	= isabelle_hf_mute,
+	.mute_stream	= isabelle_hf_mute,
 };
 
 static const struct snd_soc_dai_ops isabelle_line_dai_ops = {
 	.hw_params	= isabelle_hw_params,
 	.set_fmt	= isabelle_set_dai_fmt,
-	.digital_mute	= isabelle_line_mute,
+	.mute_stream	= isabelle_line_mute,
 };
 
 static const struct snd_soc_dai_ops isabelle_ul_dai_ops = {
diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c
index 34775aa62402..c9e2ac155421 100644
--- a/sound/soc/codecs/jz4770.c
+++ b/sound/soc/codecs/jz4770.c
@@ -264,7 +264,7 @@ static int jz4770_codec_pcm_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static int jz4770_codec_digital_mute(struct snd_soc_dai *dai, int mute)
+static int jz4770_codec_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *codec = dai->component;
 	struct jz_codec *jz_codec = snd_soc_component_get_drvdata(codec);
@@ -272,6 +272,9 @@ static int jz4770_codec_digital_mute(struct snd_soc_dai *dai, int mute)
 	unsigned int val;
 	int change, err;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	change = snd_soc_component_update_bits(codec, JZ4770_CODEC_REG_CR_DAC,
 					       REG_CR_DAC_MUTE,
 					       mute ? REG_CR_DAC_MUTE : 0);
@@ -753,7 +756,7 @@ static const struct snd_soc_dai_ops jz4770_codec_dai_ops = {
 	.shutdown	= jz4770_codec_shutdown,
 	.hw_params	= jz4770_codec_hw_params,
 	.trigger	= jz4770_codec_pcm_trigger,
-	.digital_mute	= jz4770_codec_digital_mute,
+	.mute_stream	= jz4770_codec_mute,
 };
 
 #define JZ_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE  | \
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index f864b07cb0b8..32a384d0ec18 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1218,36 +1218,51 @@ static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 	return 0;
 }
 
-static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0),
 			    (mute ? (BIT(1)|BIT(0)) : 0));
 	return 0;
 }
 
-static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2),
 			    (mute ? (BIT(3)|BIT(2)) : 0));
 	return 0;
 }
 
-static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4),
 			    (mute ? (BIT(5)|BIT(4)) : 0));
 	return 0;
 }
 
-static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(4),
 			    (mute ? BIT(4) : 0));
 	return 0;
 }
 
-static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute)
+static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6),
 			    (mute ? (BIT(7)|BIT(6)) : 0));
 	return 0;
@@ -1288,35 +1303,35 @@ static const struct snd_soc_dai_ops lm49453_headset_dai_ops = {
 	.hw_params	= lm49453_hw_params,
 	.set_sysclk	= lm49453_set_dai_sysclk,
 	.set_fmt	= lm49453_set_dai_fmt,
-	.digital_mute	= lm49453_hp_mute,
+	.mute_stream	= lm49453_hp_mute,
 };
 
 static const struct snd_soc_dai_ops lm49453_speaker_dai_ops = {
 	.hw_params	= lm49453_hw_params,
 	.set_sysclk	= lm49453_set_dai_sysclk,
 	.set_fmt	= lm49453_set_dai_fmt,
-	.digital_mute	= lm49453_ls_mute,
+	.mute_stream	= lm49453_ls_mute,
 };
 
 static const struct snd_soc_dai_ops lm49453_haptic_dai_ops = {
 	.hw_params	= lm49453_hw_params,
 	.set_sysclk	= lm49453_set_dai_sysclk,
 	.set_fmt	= lm49453_set_dai_fmt,
-	.digital_mute	= lm49453_ha_mute,
+	.mute_stream	= lm49453_ha_mute,
 };
 
 static const struct snd_soc_dai_ops lm49453_ep_dai_ops = {
 	.hw_params	= lm49453_hw_params,
 	.set_sysclk	= lm49453_set_dai_sysclk,
 	.set_fmt	= lm49453_set_dai_fmt,
-	.digital_mute	= lm49453_ep_mute,
+	.mute_stream	= lm49453_ep_mute,
 };
 
 static const struct snd_soc_dai_ops lm49453_lineout_dai_ops = {
 	.hw_params	= lm49453_hw_params,
 	.set_sysclk	= lm49453_set_dai_sysclk,
 	.set_fmt	= lm49453_set_dai_fmt,
-	.digital_mute	= lm49453_lo_mute,
+	.mute_stream	= lm49453_lo_mute,
 };
 
 /* LM49453 dai structure. */
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 55823bc95d06..cbfd3919e31c 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -372,11 +372,14 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int ml26124_mute(struct snd_soc_dai *dai, int mute)
+static int ml26124_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct ml26124_priv *priv = snd_soc_component_get_drvdata(component);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	switch (priv->substream->stream) {
 	case SNDRV_PCM_STREAM_CAPTURE:
 		snd_soc_component_update_bits(component, ML26124_REC_PLYBAK_RUN, BIT(0), 1);
@@ -492,7 +495,7 @@ static int ml26124_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops ml26124_dai_ops = {
 	.hw_params	= ml26124_hw_params,
-	.digital_mute	= ml26124_mute,
+	.mute_stream	= ml26124_mute,
 	.set_fmt	= ml26124_set_dai_fmt,
 	.set_sysclk	= ml26124_set_dai_sysclk,
 };
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
index 79928ddeb7a1..9e7937f889cd 100644
--- a/sound/soc/codecs/nau8822.c
+++ b/sound/soc/codecs/nau8822.c
@@ -900,10 +900,13 @@ static int nau8822_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int nau8822_mute(struct snd_soc_dai *dai, int mute)
+static int nau8822_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;
+
 	dev_dbg(component->dev, "%s: %d\n", __func__, mute);
 
 	if (mute)
@@ -967,7 +970,7 @@ static int nau8822_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops nau8822_dai_ops = {
 	.hw_params	= nau8822_hw_params,
-	.digital_mute	= nau8822_mute,
+	.mute_stream	= nau8822_mute,
 	.set_fmt	= nau8822_set_dai_fmt,
 	.set_sysclk	= nau8822_set_dai_sysclk,
 	.set_pll	= nau8822_set_pll,
diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c
index 115706a55577..7f235d0da3f5 100644
--- a/sound/soc/codecs/rk3328_codec.c
+++ b/sound/soc/codecs/rk3328_codec.c
@@ -107,12 +107,15 @@ static int rk3328_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static int rk3328_digital_mute(struct snd_soc_dai *dai, int mute)
+static int rk3328_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct rk3328_codec_priv *rk3328 =
 		snd_soc_component_get_drvdata(dai->component);
 	unsigned int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = HPOUTL_MUTE | HPOUTR_MUTE;
 	else
@@ -316,7 +319,7 @@ static void rk3328_pcm_shutdown(struct snd_pcm_substream *substream,
 static const struct snd_soc_dai_ops rk3328_dai_ops = {
 	.hw_params = rk3328_hw_params,
 	.set_fmt = rk3328_set_dai_fmt,
-	.digital_mute = rk3328_digital_mute,
+	.mute_stream = rk3328_mute,
 	.startup = rk3328_pcm_startup,
 	.shutdown = rk3328_pcm_shutdown,
 };
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index eb08976a7d06..9a87c08caca3 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -775,11 +775,14 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 };
 
 /* mute the codec used by alsa core */
-static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int sgtl5000_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	u16 i2s_pwr = SGTL5000_I2S_IN_POWERUP;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	/*
 	 * During 'digital mute' do not mute DAC
 	 * because LINE_IN would be muted aswell. We want to mute
@@ -1160,7 +1163,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops sgtl5000_ops = {
 	.hw_params = sgtl5000_pcm_hw_params,
-	.digital_mute = sgtl5000_digital_mute,
+	.mute_stream = sgtl5000_mute,
 	.set_fmt = sgtl5000_set_dai_fmt,
 	.set_sysclk = sgtl5000_set_dai_sysclk,
 };
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 2881a0f7bb39..3bcb55b240df 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -251,10 +251,13 @@ static int sta529_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int sta529_mute(struct snd_soc_dai *dai, int mute)
+static int sta529_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	u8 val = 0;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val |= CODEC_MUTE_VAL;
 
@@ -291,7 +294,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 static const struct snd_soc_dai_ops sta529_dai_ops = {
 	.hw_params	=	sta529_hw_params,
 	.set_fmt	=	sta529_set_dai_fmt,
-	.digital_mute	=	sta529_mute,
+	.mute_stream	=	sta529_mute,
 };
 
 static struct snd_soc_dai_driver sta529_dai = {
diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c
index abc114a3ae2b..d59a2b1e380c 100644
--- a/sound/soc/codecs/tfa9879.c
+++ b/sound/soc/codecs/tfa9879.c
@@ -93,10 +93,13 @@ static int tfa9879_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int tfa9879_digital_mute(struct snd_soc_dai *dai, int mute)
+static int tfa9879_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;
+
 	snd_soc_component_update_bits(component, TFA9879_MISC_CONTROL,
 				      TFA9879_S_MUTE_MASK,
 				      !!mute << TFA9879_S_MUTE_SHIFT);
@@ -251,7 +254,7 @@ static const struct regmap_config tfa9879_regmap = {
 
 static const struct snd_soc_dai_ops tfa9879_dai_ops = {
 	.hw_params = tfa9879_hw_params,
-	.digital_mute = tfa9879_digital_mute,
+	.mute_stream = tfa9879_mute,
 	.set_fmt = tfa9879_set_fmt,
 };
 
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index f34637afee51..9fead4faa739 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -997,8 +997,11 @@ static void twl6040_mute_path(struct snd_soc_component *component, enum twl6040_
 	}
 }
 
-static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute)
+static int twl6040_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	switch (dai->id) {
 	case TWL6040_DAI_LEGACY:
 		twl6040_mute_path(dai->component, TWL6040_DAI_DL1, mute);
@@ -1020,7 +1023,7 @@ static const struct snd_soc_dai_ops twl6040_dai_ops = {
 	.hw_params	= twl6040_hw_params,
 	.prepare	= twl6040_prepare,
 	.set_sysclk	= twl6040_set_dai_sysclk,
-	.digital_mute	= twl6040_digital_mute,
+	.mute_stream	= twl6040_mute,
 };
 
 static struct snd_soc_dai_driver twl6040_dai[] = {
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 1cc7f56912dc..a58291581813 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -117,12 +117,15 @@ static inline void uda134x_reset(struct snd_soc_component *component)
 	regmap_update_bits(uda134x->regmap, UDA134X_STATUS0, mask, 0);
 }
 
-static int uda134x_mute(struct snd_soc_dai *dai, int mute)
+static int uda134x_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(dai->component);
 	unsigned int mask = 1<<2;
 	unsigned int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	pr_debug("%s mute: %d\n", __func__, mute);
 
 	if (mute)
@@ -416,7 +419,7 @@ static const struct snd_soc_dai_ops uda134x_dai_ops = {
 	.startup	= uda134x_startup,
 	.shutdown	= uda134x_shutdown,
 	.hw_params	= uda134x_hw_params,
-	.digital_mute	= uda134x_mute,
+	.mute_stream	= uda134x_mute,
 	.set_sysclk	= uda134x_set_dai_sysclk,
 	.set_fmt	= uda134x_set_dai_fmt,
 };
-- 
2.25.1

[PATCH 07/19] ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 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 +++++--
 5 files changed, 26 insertions(+), 10 deletions(-)

diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index d22f75e8fb6a..74a11ccd9ca2 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -404,11 +404,14 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
 		aic23->requested_adc = 0;
 }
 
-static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	reg = snd_soc_component_read(component, TLV320AIC23_DIGT);
 	if (mute)
 		reg |= TLV320AIC23_DACM_MUTE;
@@ -512,7 +515,7 @@ static const struct snd_soc_dai_ops tlv320aic23_dai_ops = {
 	.prepare	= tlv320aic23_pcm_prepare,
 	.hw_params	= tlv320aic23_hw_params,
 	.shutdown	= tlv320aic23_shutdown,
-	.digital_mute	= tlv320aic23_mute,
+	.mute_stream	= tlv320aic23_mute,
 	.set_fmt	= tlv320aic23_set_dai_fmt,
 	.set_sysclk	= tlv320aic23_set_dai_sysclk,
 };
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 032b39735643..afc2a6bf0da4 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -134,12 +134,15 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
 /**
  * aic26_mute - Mute control to reduce noise when changing audio format
  */
-static int aic26_mute(struct snd_soc_dai *dai, int mute)
+static int aic26_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct aic26 *aic26 = snd_soc_component_get_drvdata(component);
 	u16 reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n",
 		dai, mute);
 
@@ -211,7 +214,7 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 
 static const struct snd_soc_dai_ops aic26_dai_ops = {
 	.hw_params	= aic26_hw_params,
-	.digital_mute	= aic26_mute,
+	.mute_stream	= aic26_mute,
 	.set_sysclk	= aic26_set_sysclk,
 	.set_fmt	= aic26_set_fmt,
 };
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 31daa60695bd..e7f68bc46826 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -972,10 +972,14 @@ static int aic31xx_hw_params(struct snd_pcm_substream *substream,
 	return aic31xx_setup_pll(component, params);
 }
 
-static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute)
+static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute,
+			    int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute) {
 		snd_soc_component_update_bits(component, AIC31XX_DACMUTE,
 				    AIC31XX_DACMUTE_MASK,
@@ -1378,7 +1382,7 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = {
 	.hw_params	= aic31xx_hw_params,
 	.set_sysclk	= aic31xx_set_dai_sysclk,
 	.set_fmt	= aic31xx_set_dai_fmt,
-	.digital_mute	= aic31xx_dac_mute,
+	.mute_stream	= aic31xx_dac_mute,
 };
 
 static struct snd_soc_dai_driver dac31xx_dai_driver[] = {
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 8682daec016e..19a0a02ee909 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -812,10 +812,13 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
+static int aic32x4_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;
+
 	snd_soc_component_update_bits(component, AIC32X4_DACMUTE,
 				AIC32X4_MUTEON, mute ? AIC32X4_MUTEON : 0);
 
@@ -866,7 +869,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops aic32x4_ops = {
 	.hw_params = aic32x4_hw_params,
-	.digital_mute = aic32x4_mute,
+	.mute_stream = aic32x4_mute,
 	.set_fmt = aic32x4_set_dai_fmt,
 	.set_sysclk = aic32x4_set_dai_sysclk,
 };
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 6860743ecdca..8b3d5af987cc 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1216,12 +1216,15 @@ static int aic3x_prepare(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int aic3x_mute(struct snd_soc_dai *dai, int mute)
+static int aic3x_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u8 ldac_reg = snd_soc_component_read(component, LDAC_VOL) & ~MUTE_ON;
 	u8 rdac_reg = snd_soc_component_read(component, RDAC_VOL) & ~MUTE_ON;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute) {
 		snd_soc_component_write(component, LDAC_VOL, ldac_reg | MUTE_ON);
 		snd_soc_component_write(component, RDAC_VOL, rdac_reg | MUTE_ON);
@@ -1481,7 +1484,7 @@ static int aic3x_set_bias_level(struct snd_soc_component *component,
 static const struct snd_soc_dai_ops aic3x_dai_ops = {
 	.hw_params	= aic3x_hw_params,
 	.prepare	= aic3x_prepare,
-	.digital_mute	= aic3x_mute,
+	.mute_stream	= aic3x_mute,
 	.set_sysclk	= aic3x_set_dai_sysclk,
 	.set_fmt	= aic3x_set_dai_fmt,
 	.set_tdm_slot	= aic3x_set_dai_tdm_slot,
-- 
2.25.1

[PATCH 08/19] ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 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 +++++--
 6 files changed, 30 insertions(+), 12 deletions(-)

diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 529c0fb93f9b..32610af4d5e7 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -465,11 +465,14 @@ static int tas2552_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static int tas2552_mute(struct snd_soc_dai *dai, int mute)
+static int tas2552_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	u8 cfg1_reg = 0;
 	struct snd_soc_component *component = dai->component;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		cfg1_reg |= TAS2552_MUTE;
 
@@ -519,7 +522,7 @@ static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = {
 	.set_sysclk	= tas2552_set_dai_sysclk,
 	.set_fmt	= tas2552_set_dai_fmt,
 	.set_tdm_slot	= tas2552_set_dai_tdm_slot,
-	.digital_mute = tas2552_mute,
+	.mute_stream	= tas2552_mute,
 };
 
 /* Formats supported by TAS2552 driver. */
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index 7fae88655a0f..c818be9536be 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -334,10 +334,13 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
-static int tas2562_mute(struct snd_soc_dai *dai, int mute)
+static int tas2562_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;
+
 	return snd_soc_component_update_bits(component, TAS2562_PWR_CTRL,
 					     TAS2562_MODE_MASK,
 					     mute ? TAS2562_MUTE : 0);
@@ -552,7 +555,7 @@ static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = {
 	.hw_params	= tas2562_hw_params,
 	.set_fmt	= tas2562_set_dai_fmt,
 	.set_tdm_slot	= tas2562_set_dai_tdm_slot,
-	.digital_mute	= tas2562_mute,
+	.mute_stream	= tas2562_mute,
 };
 
 static struct snd_soc_dai_driver tas2562_dai[] = {
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index 54c8135fe43c..60ef721fb456 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -189,11 +189,14 @@ static const struct snd_soc_dapm_route tas2770_audio_map[] = {
 	{"VSENSE", "Switch", "VMON"},
 };
 
-static int tas2770_mute(struct snd_soc_dai *dai, int mute)
+static int tas2770_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;
+
 	if (mute)
 		ret = snd_soc_component_update_bits(component,
 			TAS2770_PWR_CTRL,
@@ -530,7 +533,7 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai,
 }
 
 static struct snd_soc_dai_ops tas2770_dai_ops = {
-	.digital_mute = tas2770_mute,
+	.mute_stream = tas2770_mute,
 	.hw_params  = tas2770_hw_params,
 	.set_fmt    = tas2770_set_fmt,
 	.set_tdm_slot = tas2770_set_dai_tdm_slot,
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 5b7f9fcf6cbf..a65a874fa974 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -301,12 +301,15 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream,
 				  TAS571X_SDI_FMT_MASK, val);
 }
 
-static int tas571x_mute(struct snd_soc_dai *dai, int mute)
+static int tas571x_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u8 sysctl2;
 	int ret;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	sysctl2 = mute ? TAS571X_SYS_CTRL_2_SDN_MASK : 0;
 
 	ret = snd_soc_component_update_bits(component,
@@ -354,7 +357,7 @@ static int tas571x_set_bias_level(struct snd_soc_component *component,
 static const struct snd_soc_dai_ops tas571x_dai_ops = {
 	.set_fmt	= tas571x_set_dai_fmt,
 	.hw_params	= tas571x_hw_params,
-	.digital_mute	= tas571x_mute,
+	.mute_stream	= tas571x_mute,
 };
 
 
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index e159f839d928..b445f4cf035e 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -199,11 +199,14 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	return ret;
 }
 
-static int tas5720_mute(struct snd_soc_dai *dai, int mute)
+static int tas5720_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, TAS5720_DIGITAL_CTRL2_REG,
 				  TAS5720_MUTE, mute ? TAS5720_MUTE : 0);
 	if (ret < 0) {
@@ -604,7 +607,7 @@ static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = {
 	.hw_params	= tas5720_hw_params,
 	.set_fmt	= tas5720_set_dai_fmt,
 	.set_tdm_slot	= tas5720_set_dai_tdm_slot,
-	.digital_mute	= tas5720_mute,
+	.mute_stream	= tas5720_mute,
 };
 
 /*
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index aaba39295079..33b97a603a1d 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -252,12 +252,15 @@ static int tas6424_set_dai_tdm_slot(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static int tas6424_mute(struct snd_soc_dai *dai, int mute)
+static int tas6424_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tas6424_data *tas6424 = snd_soc_component_get_drvdata(component);
 	unsigned int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	dev_dbg(component->dev, "%s() mute=%d\n", __func__, mute);
 
 	if (tas6424->mute_gpio) {
@@ -382,7 +385,7 @@ static const struct snd_soc_dai_ops tas6424_speaker_dai_ops = {
 	.hw_params	= tas6424_hw_params,
 	.set_fmt	= tas6424_set_dai_fmt,
 	.set_tdm_slot	= tas6424_set_dai_tdm_slot,
-	.digital_mute	= tas6424_mute,
+	.mute_stream	= tas6424_mute,
 };
 
 static struct snd_soc_dai_driver tas6424_dai[] = {
-- 
2.25.1

[PATCH 09/19] ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/codecs/ssm2518.c | 7 +++++--
 sound/soc/codecs/ssm2602.c | 7 +++++--
 sound/soc/codecs/ssm4567.c | 7 +++++--
 3 files changed, 15 insertions(+), 6 deletions(-)

diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index c47e3c4762fe..da4ed07b0912 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -388,11 +388,14 @@ static int ssm2518_hw_params(struct snd_pcm_substream *substream,
 				SSM2518_POWER1_MCS_MASK, mcs << 1);
 }
 
-static int ssm2518_mute(struct snd_soc_dai *dai, int mute)
+static int ssm2518_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct ssm2518 *ssm2518 = snd_soc_component_get_drvdata(dai->component);
 	unsigned int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = SSM2518_MUTE_CTRL_MUTE_MASTER;
 	else
@@ -623,7 +626,7 @@ static int ssm2518_startup(struct snd_pcm_substream *substream,
 static const struct snd_soc_dai_ops ssm2518_dai_ops = {
 	.startup = ssm2518_startup,
 	.hw_params	= ssm2518_hw_params,
-	.digital_mute	= ssm2518_mute,
+	.mute_stream	= ssm2518_mute,
 	.set_fmt	= ssm2518_set_dai_fmt,
 	.set_tdm_slot	= ssm2518_set_tdm_slot,
 };
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 464a4d7873bb..7a3c068b16dc 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -338,10 +338,13 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
+static int ssm2602_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct ssm2602_priv *ssm2602 = snd_soc_component_get_drvdata(dai->component);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
 				    APDIGI_ENABLE_DAC_MUTE,
@@ -505,7 +508,7 @@ static int ssm2602_set_bias_level(struct snd_soc_component *component,
 static const struct snd_soc_dai_ops ssm2602_dai_ops = {
 	.startup	= ssm2602_startup,
 	.hw_params	= ssm2602_hw_params,
-	.digital_mute	= ssm2602_mute,
+	.mute_stream	= ssm2602_mute,
 	.set_sysclk	= ssm2602_set_dai_sysclk,
 	.set_fmt	= ssm2602_set_dai_fmt,
 };
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index bb4958bb8fe9..a59f485c2d7f 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/sound/soc/codecs/ssm4567.c
@@ -220,11 +220,14 @@ static int ssm4567_hw_params(struct snd_pcm_substream *substream,
 				SSM4567_DAC_FS_MASK, dacfs);
 }
 
-static int ssm4567_mute(struct snd_soc_dai *dai, int mute)
+static int ssm4567_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct ssm4567 *ssm4567 = snd_soc_component_get_drvdata(dai->component);
 	unsigned int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	val = mute ? SSM4567_DAC_MUTE : 0;
 	return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL,
 			SSM4567_DAC_MUTE, val);
@@ -390,7 +393,7 @@ static int ssm4567_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops ssm4567_dai_ops = {
 	.hw_params	= ssm4567_hw_params,
-	.digital_mute	= ssm4567_mute,
+	.mute_stream	= ssm4567_mute,
 	.set_fmt	= ssm4567_set_dai_fmt,
 	.set_tdm_slot	= ssm4567_set_tdm_slot,
 };
-- 
2.25.1

[PATCH 10/19] ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 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 +++++--
 5 files changed, 25 insertions(+), 10 deletions(-)

diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 4767e158cd5e..0f641d29b8b3 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -147,12 +147,15 @@ 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);
 	int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = PCM1681_SOFT_MUTE_ALL;
 	else
@@ -205,7 +208,7 @@ 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,
 };
 
 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..eb887802c6d0 100644
--- a/sound/soc/codecs/pcm1789.c
+++ b/sound/soc/codecs/pcm1789.c
@@ -60,11 +60,14 @@ 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);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	return regmap_update_bits(priv->regmap, PCM1789_SOFT_MUTE,
 				  PCM1789_MUTE_MASK,
 				  mute ? 0 : PCM1789_MUTE_MASK);
@@ -167,7 +170,7 @@ 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,
 };
 
diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c
index 9e70b7385c69..cc944250ad71 100644
--- a/sound/soc/codecs/pcm179x.c
+++ b/sound/soc/codecs/pcm179x.c
@@ -76,12 +76,15 @@ 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);
 	int ret;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	ret = regmap_update_bits(priv->regmap, PCM179X_SOFT_MUTE,
 				 PCM179X_MUTE_MASK, !!mute);
 	if (ret < 0)
@@ -145,7 +148,7 @@ 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,
 };
 
 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..e9756c45e15d 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -290,11 +290,14 @@ 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);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	regmap_write(pcm3168a->regmap, PCM3168A_DAC_MUTE, mute ? 0xff : 0);
 
 	return 0;
@@ -570,7 +573,7 @@ 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,
 };
 
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 4cbef9affffd..5a82a59d6336 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -1394,13 +1394,16 @@ 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);
 	int ret;
 	unsigned int mute_det;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	mutex_lock(&pcm512x->mutex);
 
 	if (mute) {
@@ -1445,7 +1448,7 @@ 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,
 };
 
-- 
2.25.1

[PATCH 11/19] ASoC: codecs: max*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/codecs/max98088.c | 16 ++++++++++++----
 sound/soc/codecs/max98090.c |  8 ++++++--
 sound/soc/codecs/max9867.c  |  7 +++++--
 3 files changed, 23 insertions(+), 8 deletions(-)

diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 1f1817634a41..60096e732517 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1274,11 +1274,15 @@ static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai,
        return 0;
 }
 
-static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int max98088_dai1_mute(struct snd_soc_dai *codec_dai, int mute,
+			      int direction)
 {
        struct snd_soc_component *component = codec_dai->component;
        int reg;
 
+       if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+                return 0;
+
        if (mute)
                reg = M98088_DAI_MUTE;
        else
@@ -1289,11 +1293,15 @@ static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int mute)
        return 0;
 }
 
-static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int max98088_dai2_mute(struct snd_soc_dai *codec_dai, int mute,
+			      int direction)
 {
        struct snd_soc_component *component = codec_dai->component;
        int reg;
 
+       if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+               return 0;
+
        if (mute)
                reg = M98088_DAI_MUTE;
        else
@@ -1354,14 +1362,14 @@ static const struct snd_soc_dai_ops max98088_dai1_ops = {
        .set_sysclk = max98088_dai_set_sysclk,
        .set_fmt = max98088_dai1_set_fmt,
        .hw_params = max98088_dai1_hw_params,
-       .digital_mute = max98088_dai1_digital_mute,
+       .mute_stream = max98088_dai1_mute,
 };
 
 static const struct snd_soc_dai_ops max98088_dai2_ops = {
        .set_sysclk = max98088_dai_set_sysclk,
        .set_fmt = max98088_dai2_set_fmt,
        .hw_params = max98088_dai2_hw_params,
-       .digital_mute = max98088_dai2_digital_mute,
+       .mute_stream = max98088_dai2_mute,
 };
 
 static struct snd_soc_dai_driver max98088_dai[] = {
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index a61c5638652d..05a1a7f13b36 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2017,11 +2017,15 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai,
 	return 0;
 }
 
-static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int max98090_dai_mute(struct snd_soc_dai *codec_dai, int mute,
+			     int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	int regval;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	regval = mute ? M98090_DVM_MASK : 0;
 	snd_soc_component_update_bits(component, M98090_REG_DAI_PLAYBACK_LEVEL,
 		M98090_DVM_MASK, regval);
@@ -2347,7 +2351,7 @@ static const struct snd_soc_dai_ops max98090_dai_ops = {
 	.set_fmt = max98090_dai_set_fmt,
 	.set_tdm_slot = max98090_set_tdm_slot,
 	.hw_params = max98090_dai_hw_params,
-	.digital_mute = max98090_dai_digital_mute,
+	.mute_stream = max98090_dai_mute,
 	.trigger = max98090_dai_trigger,
 };
 
diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c
index c72cb2888c21..b4ac9f696a5d 100644
--- a/sound/soc/codecs/max9867.c
+++ b/sound/soc/codecs/max9867.c
@@ -283,11 +283,14 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int max9867_mute(struct snd_soc_dai *dai, int mute)
+static int max9867_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	return regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL,
 				  1 << 6, !!mute << 6);
 }
@@ -393,7 +396,7 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai,
 static const struct snd_soc_dai_ops max9867_dai_ops = {
 	.set_sysclk	= max9867_set_dai_sysclk,
 	.set_fmt	= max9867_dai_set_fmt,
-	.digital_mute	= max9867_mute,
+	.mute_stream	= max9867_mute,
 	.startup	= max9867_startup,
 	.hw_params	= max9867_dai_hw_params,
 };
-- 
2.25.1

[PATCH 12/19] ASoC: codecs: alc*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/codecs/alc5623.c | 7 +++++--
 sound/soc/codecs/alc5632.c | 7 +++++--
 2 files changed, 10 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index c70c49bb4a3e..de7cabaa211c 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -737,12 +737,15 @@ static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int alc5623_mute(struct snd_soc_dai *dai, int mute)
+static int alc5623_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
 	u16 mute_reg = snd_soc_component_read(component, ALC5623_MISC_CTRL) & ~hp_mute;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		mute_reg |= hp_mute;
 
@@ -829,7 +832,7 @@ static int alc5623_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops alc5623_dai_ops = {
 		.hw_params = alc5623_pcm_hw_params,
-		.digital_mute = alc5623_mute,
+		.mute_stream = alc5623_mute,
 		.set_fmt = alc5623_set_dai_fmt,
 		.set_sysclk = alc5623_set_dai_sysclk,
 		.set_pll = alc5623_set_dai_pll,
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index f49543163f69..f90bd77438b8 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -902,13 +902,16 @@ static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int alc5632_mute(struct snd_soc_dai *dai, int mute)
+static int alc5632_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L
 						|ALC5632_MISC_HP_DEPOP_MUTE_R;
 	u16 mute_reg = snd_soc_component_read(component, ALC5632_MISC_CTRL) & ~hp_mute;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		mute_reg |= hp_mute;
 
@@ -1005,7 +1008,7 @@ static int alc5632_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops alc5632_dai_ops = {
 		.hw_params = alc5632_pcm_hw_params,
-		.digital_mute = alc5632_mute,
+		.mute_stream = alc5632_mute,
 		.set_fmt = alc5632_set_dai_fmt,
 		.set_sysclk = alc5632_set_dai_sysclk,
 		.set_pll = alc5632_set_dai_pll,
-- 
2.25.1

[PATCH 13/19] ASoC: codecs: wm*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 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 +++++--
 32 files changed, 164 insertions(+), 67 deletions(-)

diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 7fe7c1e91882..470f2655418f 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -942,11 +942,14 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8350_mute(struct snd_soc_dai *dai, int mute)
+static int wm8350_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	unsigned int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = WM8350_DAC_MUTE_ENA;
 	else
@@ -1426,7 +1429,7 @@ EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect);
 
 static const struct snd_soc_dai_ops wm8350_dai_ops = {
 	 .hw_params	= wm8350_pcm_hw_params,
-	 .digital_mute	= wm8350_mute,
+	 .mute_stream	= wm8350_mute,
 	 .set_fmt	= wm8350_set_dai_fmt,
 	 .set_sysclk	= wm8350_set_dai_sysclk,
 	 .set_pll	= wm8350_set_fll,
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 2551eb0f1868..5709a2c45c3e 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1104,11 +1104,14 @@ static int wm8400_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8400_mute(struct snd_soc_dai *dai, int mute)
+static int wm8400_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 val = snd_soc_component_read(component, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
 	else
@@ -1234,7 +1237,7 @@ static int wm8400_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8400_dai_ops = {
 	.hw_params = wm8400_hw_params,
-	.digital_mute = wm8400_mute,
+	.mute_stream = wm8400_mute,
 	.set_fmt = wm8400_set_dai_fmt,
 	.set_clkdiv = wm8400_set_dai_clkdiv,
 	.set_sysclk = wm8400_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 63a877a8ee2b..b480dfe47267 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -487,11 +487,14 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8510_mute(struct snd_soc_dai *dai, int mute)
+static int wm8510_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8510_DAC) & 0xffbf;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8510_DAC, mute_reg | 0x40);
 	else
@@ -547,7 +550,7 @@ static int wm8510_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8510_dai_ops = {
 	.hw_params	= wm8510_pcm_hw_params,
-	.digital_mute	= wm8510_mute,
+	.mute_stream	= wm8510_mute,
 	.set_fmt	= wm8510_set_dai_fmt,
 	.set_clkdiv	= wm8510_set_dai_clkdiv,
 	.set_pll	= wm8510_set_dai_pll,
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d1fc529d20e7..63284d3a9518 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -800,11 +800,14 @@ static int wm8580_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	return 0;
 }
 
-static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm8580_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	unsigned int reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	reg = snd_soc_component_read(component, WM8580_DAC_CONTROL5);
 
 	if (mute)
@@ -866,7 +869,7 @@ static const struct snd_soc_dai_ops wm8580_dai_ops_playback = {
 	.set_fmt	= wm8580_set_paif_dai_fmt,
 	.set_clkdiv	= wm8580_set_dai_clkdiv,
 	.set_pll	= wm8580_set_dai_pll,
-	.digital_mute	= wm8580_digital_mute,
+	.mute_stream	= wm8580_mute,
 };
 
 static const struct snd_soc_dai_ops wm8580_dai_ops_capture = {
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 8a0f93f54b60..cbbb46a6ccac 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -204,11 +204,14 @@ static void wm8711_shutdown(struct snd_pcm_substream *substream,
 	}
 }
 
-static int wm8711_mute(struct snd_soc_dai *dai, int mute)
+static int wm8711_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8711_APDIGI) & 0xfff7;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8711_APDIGI, mute_reg | 0x8);
 	else
@@ -329,7 +332,7 @@ static const struct snd_soc_dai_ops wm8711_ops = {
 	.prepare = wm8711_pcm_prepare,
 	.hw_params = wm8711_hw_params,
 	.shutdown = wm8711_shutdown,
-	.digital_mute = wm8711_mute,
+	.mute_stream = wm8711_mute,
 	.set_sysclk = wm8711_set_dai_sysclk,
 	.set_fmt = wm8711_set_dai_fmt,
 };
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index bb5521f544ba..b33c50e82725 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -69,11 +69,14 @@ static const struct snd_soc_dapm_route wm8728_intercon[] = {
 	{"VOUTR", NULL, "DAC"},
 };
 
-static int wm8728_mute(struct snd_soc_dai *dai, int mute)
+static int wm8728_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8728_DACCTL);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8728_DACCTL, mute_reg | 1);
 	else
@@ -192,7 +195,7 @@ static int wm8728_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8728_dai_ops = {
 	.hw_params	= wm8728_hw_params,
-	.digital_mute	= wm8728_mute,
+	.mute_stream	= wm8728_mute,
 	.set_fmt	= wm8728_set_dai_fmt,
 };
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index cae2cc38d93c..9b6b2e95d3cc 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -366,11 +366,14 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8731_mute(struct snd_soc_dai *dai, int mute)
+static int wm8731_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8731_APDIGI) & 0xfff7;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8731_APDIGI, mute_reg | 0x8);
 	else
@@ -546,7 +549,7 @@ static int wm8731_startup(struct snd_pcm_substream *substream,
 static const struct snd_soc_dai_ops wm8731_dai_ops = {
 	.startup	= wm8731_startup,
 	.hw_params	= wm8731_hw_params,
-	.digital_mute	= wm8731_mute,
+	.mute_stream	= wm8731_mute,
 	.set_sysclk	= wm8731_set_dai_sysclk,
 	.set_fmt	= wm8731_set_dai_fmt,
 };
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 328df81ee839..283b875a344d 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -364,10 +364,13 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int wm8741_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm8741_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(component, WM8741_VOLUME_CONTROL,
 			WM8741_SOFT_MASK, !!mute << WM8741_SOFT_SHIFT);
 	return 0;
@@ -386,7 +389,7 @@ static const struct snd_soc_dai_ops wm8741_dai_ops = {
 	.hw_params	= wm8741_hw_params,
 	.set_sysclk	= wm8741_set_dai_sysclk,
 	.set_fmt	= wm8741_set_dai_fmt,
-	.digital_mute   = wm8741_mute,
+	.mute_stream	= wm8741_mute,
 };
 
 static struct snd_soc_dai_driver wm8741_dai = {
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 970941f8ae81..a8d6195e27f9 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -606,11 +606,14 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8750_mute(struct snd_soc_dai *dai, int mute)
+static int wm8750_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8750_ADCDAC) & 0xfff7;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8750_ADCDAC, mute_reg | 0x8);
 	else
@@ -660,7 +663,7 @@ static int wm8750_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8750_dai_ops = {
 	.hw_params	= wm8750_pcm_hw_params,
-	.digital_mute	= wm8750_mute,
+	.mute_stream	= wm8750_mute,
 	.set_fmt	= wm8750_set_dai_fmt,
 	.set_sysclk	= wm8750_set_dai_sysclk,
 };
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a1b6765c8f23..09d8d8177f2f 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1295,12 +1295,15 @@ static int wm8753_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return wm8753_voice_write_dai_fmt(component, fmt);
 };
 
-static int wm8753_mute(struct snd_soc_dai *dai, int mute)
+static int wm8753_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8753_DAC) & 0xfff7;
 	struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	/* the digital mute covers the HiFi and Voice DAC's on the WM8753.
 	 * make sure we check if they are not both active when we mute */
 	if (mute && wm8753->dai_func == 1) {
@@ -1380,7 +1383,7 @@ static int wm8753_set_bias_level(struct snd_soc_component *component,
  */
 static const struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
 	.hw_params	= wm8753_i2s_hw_params,
-	.digital_mute	= wm8753_mute,
+	.mute_stream	= wm8753_mute,
 	.set_fmt	= wm8753_hifi_set_dai_fmt,
 	.set_clkdiv	= wm8753_set_dai_clkdiv,
 	.set_pll	= wm8753_set_dai_pll,
@@ -1389,7 +1392,7 @@ static const struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
 
 static const struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = {
 	.hw_params	= wm8753_pcm_hw_params,
-	.digital_mute	= wm8753_mute,
+	.mute_stream	= wm8753_mute,
 	.set_fmt	= wm8753_voice_set_dai_fmt,
 	.set_clkdiv	= wm8753_set_dai_clkdiv,
 	.set_pll	= wm8753_set_dai_pll,
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index d51be2531e2e..9242231a9886 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -472,10 +472,13 @@ static int wm8770_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8770_mute(struct snd_soc_dai *dai, int mute)
+static int wm8770_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	component = dai->component;
 	return snd_soc_component_update_bits(component, WM8770_DACMUTE, 0x10,
 				   !!mute << 4);
@@ -538,7 +541,7 @@ static int wm8770_set_bias_level(struct snd_soc_component *component,
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 static const struct snd_soc_dai_ops wm8770_dai_ops = {
-	.digital_mute = wm8770_mute,
+	.mute_stream = wm8770_mute,
 	.hw_params = wm8770_hw_params,
 	.set_fmt = wm8770_set_fmt,
 	.set_sysclk = wm8770_set_sysclk,
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index f174d7ce2b13..367612e2fc25 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -309,10 +309,13 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8776_mute(struct snd_soc_dai *dai, int mute)
+static int wm8776_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;
+
 	return snd_soc_component_write(component, WM8776_DACMUTE, !!mute);
 }
 
@@ -361,7 +364,7 @@ static int wm8776_set_bias_level(struct snd_soc_component *component,
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 static const struct snd_soc_dai_ops wm8776_dac_ops = {
-	.digital_mute	= wm8776_mute,
+	.mute_stream	= wm8776_mute,
 	.hw_params      = wm8776_hw_params,
 	.set_fmt        = wm8776_set_fmt,
 	.set_sysclk     = wm8776_set_sysclk,
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 3921af63adf2..428c39e560ff 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -967,11 +967,14 @@ static int wm8900_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm8900_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	u16 reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	reg = snd_soc_component_read(component, WM8900_REG_DACCTRL);
 
 	if (mute)
@@ -997,7 +1000,7 @@ static const struct snd_soc_dai_ops wm8900_dai_ops = {
 	.set_clkdiv	= wm8900_set_dai_clkdiv,
 	.set_pll	= wm8900_set_dai_pll,
 	.set_fmt	= wm8900_set_dai_fmt,
-	.digital_mute	= wm8900_digital_mute,
+	.mute_stream	= wm8900_mute,
 };
 
 static struct snd_soc_dai_driver wm8900_dai = {
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 5de663d61ba6..2cb1c5b29f8f 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1307,11 +1307,14 @@ static int wm8903_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int wm8903_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm8903_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	u16 reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	reg = snd_soc_component_read(component, WM8903_DAC_DIGITAL_1);
 
 	if (mute)
@@ -1737,7 +1740,7 @@ static irqreturn_t wm8903_irq(int irq, void *data)
 
 static const struct snd_soc_dai_ops wm8903_dai_ops = {
 	.hw_params	= wm8903_hw_params,
-	.digital_mute	= wm8903_digital_mute,
+	.mute_stream	= wm8903_mute,
 	.set_fmt	= wm8903_set_dai_fmt,
 	.set_sysclk	= wm8903_set_dai_sysclk,
 };
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3f0e49c51fd5..eba04e40d782 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1846,11 +1846,14 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	return 0;
 }
 
-static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm8904_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = WM8904_DAC_MUTE;
 	else
@@ -1962,7 +1965,7 @@ static const struct snd_soc_dai_ops wm8904_dai_ops = {
 	.set_tdm_slot = wm8904_set_tdm_slot,
 	.set_pll = wm8904_set_fll,
 	.hw_params = wm8904_hw_params,
-	.digital_mute = wm8904_digital_mute,
+	.mute_stream = wm8904_mute,
 };
 
 static struct snd_soc_dai_driver wm8904_dai = {
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 41d87e172775..165fe1abc323 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -452,11 +452,14 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static int wm8940_mute(struct snd_soc_dai *dai, int mute)
+static int wm8940_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8940_DAC) & 0xffbf;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		mute_reg |= 0x40;
 
@@ -664,7 +667,7 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 static const struct snd_soc_dai_ops wm8940_dai_ops = {
 	.hw_params = wm8940_i2s_hw_params,
 	.set_sysclk = wm8940_set_dai_sysclk,
-	.digital_mute = wm8940_mute,
+	.mute_stream = wm8940_mute,
 	.set_fmt = wm8940_set_dai_fmt,
 	.set_clkdiv = wm8940_set_dai_clkdiv,
 	.set_pll = wm8940_set_dai_pll,
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 73c192f58382..12ffd1660189 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -745,11 +745,14 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 }
 
 
-static int wm8955_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm8955_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	int val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = WM8955_DACMU;
 	else
@@ -848,7 +851,7 @@ static const struct snd_soc_dai_ops wm8955_dai_ops = {
 	.set_sysclk = wm8955_set_sysclk,
 	.set_fmt = wm8955_set_fmt,
 	.hw_params = wm8955_hw_params,
-	.digital_mute = wm8955_digital_mute,
+	.mute_stream = wm8955_mute,
 };
 
 static struct snd_soc_dai_driver wm8955_dai = {
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 96c4400e92f8..13819cbca22a 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -878,10 +878,13 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+static int wm8960_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, WM8960_DACCTL1, 0x8, 0x8);
 	else
@@ -1315,7 +1318,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
 static const struct snd_soc_dai_ops wm8960_dai_ops = {
 	.hw_params = wm8960_hw_params,
 	.hw_free = wm8960_hw_free,
-	.digital_mute = wm8960_mute,
+	.mute_stream = wm8960_mute,
 	.set_fmt = wm8960_set_dai_fmt,
 	.set_clkdiv = wm8960_set_dai_clkdiv,
 	.set_pll = wm8960_set_dai_pll,
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index d11a38a0b283..84ecfc90a1f5 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -698,11 +698,14 @@ static int wm8961_set_tristate(struct snd_soc_dai *dai, int tristate)
 	return snd_soc_component_write(component, WM8961_ADDITIONAL_CONTROL_2, reg);
 }
 
-static int wm8961_digital_mute(struct snd_soc_dai *dai, int mute)
+static int wm8961_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 reg = snd_soc_component_read(component, WM8961_ADC_DAC_CONTROL_1);
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		reg |= WM8961_DACMU;
 	else
@@ -806,7 +809,7 @@ static const struct snd_soc_dai_ops wm8961_dai_ops = {
 	.hw_params = wm8961_hw_params,
 	.set_sysclk = wm8961_set_sysclk,
 	.set_fmt = wm8961_set_fmt,
-	.digital_mute = wm8961_digital_mute,
+	.mute_stream = wm8961_mute,
 	.set_tristate = wm8961_set_tristate,
 	.set_clkdiv = wm8961_set_clkdiv,
 };
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 6ef022295f55..b3dfac8f77d3 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2918,11 +2918,14 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s
 	return 0;
 }
 
-static int wm8962_mute(struct snd_soc_dai *dai, int mute)
+static int wm8962_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	int val, ret;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		val = WM8962_DAC_MUTE | WM8962_DAC_MUTE_ALT;
 	else
@@ -2951,7 +2954,7 @@ static const struct snd_soc_dai_ops wm8962_dai_ops = {
 	.hw_params = wm8962_hw_params,
 	.set_sysclk = wm8962_set_dai_sysclk,
 	.set_fmt = wm8962_set_dai_fmt,
-	.digital_mute = wm8962_mute,
+	.mute_stream = wm8962_mute,
 };
 
 static struct snd_soc_dai_driver wm8962_dai = {
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 2cefb795da03..345e6c2ce22f 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -536,11 +536,14 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8971_mute(struct snd_soc_dai *dai, int mute)
+static int wm8971_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8971_ADCDAC) & 0xfff7;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8971_ADCDAC, mute_reg | 0x8);
 	else
@@ -602,7 +605,7 @@ static int wm8971_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8971_dai_ops = {
 	.hw_params	= wm8971_pcm_hw_params,
-	.digital_mute	= wm8971_mute,
+	.mute_stream	= wm8971_mute,
 	.set_fmt	= wm8971_set_dai_fmt,
 	.set_sysclk	= wm8971_set_dai_sysclk,
 };
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 953d12e4576f..d3231989d4a2 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -559,11 +559,14 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8974_mute(struct snd_soc_dai *dai, int mute)
+static int wm8974_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8974_DAC) & 0xffbf;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
 	else
@@ -616,7 +619,7 @@ static int wm8974_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8974_ops = {
 	.hw_params = wm8974_pcm_hw_params,
-	.digital_mute = wm8974_mute,
+	.mute_stream = wm8974_mute,
 	.set_fmt = wm8974_set_dai_fmt,
 	.set_clkdiv = wm8974_set_dai_clkdiv,
 	.set_pll = wm8974_set_dai_pll,
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index e01ba5394527..fe491ab0d2b8 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -836,10 +836,13 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8978_mute(struct snd_soc_dai *dai, int mute)
+static int wm8978_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;
+
 	dev_dbg(component->dev, "%s: %d\n", __func__, mute);
 
 	if (mute)
@@ -893,7 +896,7 @@ static int wm8978_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8978_dai_ops = {
 	.hw_params	= wm8978_hw_params,
-	.digital_mute	= wm8978_mute,
+	.mute_stream	= wm8978_mute,
 	.set_fmt	= wm8978_set_dai_fmt,
 	.set_clkdiv	= wm8978_set_dai_clkdiv,
 	.set_sysclk	= wm8978_set_dai_sysclk,
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index 78e1a302c54c..1b91a1636823 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -557,10 +557,13 @@ static bool wm8983_writeable(struct device *dev, unsigned int reg)
 	}
 }
 
-static int wm8983_dac_mute(struct snd_soc_dai *dai, int mute)
+static int wm8983_dac_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;
+
 	return snd_soc_component_update_bits(component, WM8983_DAC_CONTROL,
 				   WM8983_SOFTMUTE_MASK,
 				   !!mute << WM8983_SOFTMUTE_SHIFT);
@@ -943,7 +946,7 @@ static int wm8983_probe(struct snd_soc_component *component)
 }
 
 static const struct snd_soc_dai_ops wm8983_dai_ops = {
-	.digital_mute = wm8983_dac_mute,
+	.mute_stream = wm8983_dac_mute,
 	.hw_params = wm8983_hw_params,
 	.set_fmt = wm8983_set_fmt,
 	.set_sysclk = wm8983_set_sysclk,
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 62f2c603eb2d..61b25f82bddd 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -649,10 +649,13 @@ static int wm8985_reset(struct snd_soc_component *component)
 	return snd_soc_component_write(component, WM8985_SOFTWARE_RESET, 0x0);
 }
 
-static int wm8985_dac_mute(struct snd_soc_dai *dai, int mute)
+static int wm8985_dac_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;
+
 	return snd_soc_component_update_bits(component, WM8985_DAC_CONTROL,
 				   WM8985_SOFTMUTE_MASK,
 				   !!mute << WM8985_SOFTMUTE_SHIFT);
@@ -1072,7 +1075,7 @@ static int wm8985_probe(struct snd_soc_component *component)
 }
 
 static const struct snd_soc_dai_ops wm8985_dai_ops = {
-	.digital_mute = wm8985_dac_mute,
+	.mute_stream = wm8985_dac_mute,
 	.hw_params = wm8985_hw_params,
 	.set_fmt = wm8985_set_fmt,
 	.set_sysclk = wm8985_set_sysclk,
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 646cfd8b2693..7e7c561f88ab 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -707,11 +707,14 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8988_mute(struct snd_soc_dai *dai, int mute)
+static int wm8988_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, WM8988_ADCDAC) & 0xfff7;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, WM8988_ADCDAC, mute_reg | 0x8);
 	else
@@ -766,7 +769,7 @@ static const struct snd_soc_dai_ops wm8988_ops = {
 	.hw_params = wm8988_pcm_hw_params,
 	.set_fmt = wm8988_set_dai_fmt,
 	.set_sysclk = wm8988_set_dai_sysclk,
-	.digital_mute = wm8988_mute,
+	.mute_stream = wm8988_mute,
 };
 
 static struct snd_soc_dai_driver wm8988_dai = {
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 13bca8ebf677..22613df4969d 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -998,11 +998,14 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8990_mute(struct snd_soc_dai *dai, int mute)
+static int wm8990_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	val  = snd_soc_component_read(component, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE;
 
 	if (mute)
@@ -1152,7 +1155,7 @@ static int wm8990_set_bias_level(struct snd_soc_component *component,
  */
 static const struct snd_soc_dai_ops wm8990_dai_ops = {
 	.hw_params	= wm8990_hw_params,
-	.digital_mute	= wm8990_mute,
+	.mute_stream	= wm8990_mute,
 	.set_fmt	= wm8990_set_dai_fmt,
 	.set_clkdiv	= wm8990_set_dai_clkdiv,
 	.set_pll	= wm8990_set_dai_pll,
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index ba71c2f59511..1830eb9ada28 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1064,11 +1064,14 @@ static int wm8991_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8991_mute(struct snd_soc_dai *dai, int mute)
+static int wm8991_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 val;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	val  = snd_soc_component_read(component, WM8991_DAC_CTRL) & ~WM8991_DAC_MUTE;
 	if (mute)
 		snd_soc_component_write(component, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE);
@@ -1196,7 +1199,7 @@ static int wm8991_set_bias_level(struct snd_soc_component *component,
 
 static const struct snd_soc_dai_ops wm8991_ops = {
 	.hw_params = wm8991_hw_params,
-	.digital_mute = wm8991_mute,
+	.mute_stream = wm8991_mute,
 	.set_fmt = wm8991_set_dai_fmt,
 	.set_clkdiv = wm8991_set_dai_clkdiv,
 	.set_pll = wm8991_set_dai_pll
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 207c0211caa9..fbce5e4692fa 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1330,11 +1330,14 @@ static int wm8993_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8993_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm8993_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	unsigned int reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	reg = snd_soc_component_read(component, WM8993_DAC_CTRL);
 
 	if (mute)
@@ -1444,7 +1447,7 @@ static const struct snd_soc_dai_ops wm8993_ops = {
 	.set_sysclk = wm8993_set_sysclk,
 	.set_fmt = wm8993_set_dai_fmt,
 	.hw_params = wm8993_hw_params,
-	.digital_mute = wm8993_digital_mute,
+	.mute_stream = wm8993_mute,
 	.set_pll = wm8993_set_fll,
 	.set_tdm_slot = wm8993_set_tdm_slot,
 };
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 5e1ce243feb1..fa188d422f0b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3110,12 +3110,16 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream,
 	return snd_soc_component_update_bits(component, aif1_reg, WM8994_AIF1_WL_MASK, aif1);
 }
 
-static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute,
+			   int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	int mute_reg;
 	int reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	switch (codec_dai->id) {
 	case 1:
 		mute_reg = WM8994_AIF1_DAC1_FILTERS_1;
@@ -3187,7 +3191,7 @@ static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = {
 	.set_sysclk	= wm8994_set_dai_sysclk,
 	.set_fmt	= wm8994_set_dai_fmt,
 	.hw_params	= wm8994_hw_params,
-	.digital_mute	= wm8994_aif_mute,
+	.mute_stream	= wm8994_aif_mute,
 	.set_pll	= wm8994_set_fll,
 	.set_tristate	= wm8994_set_tristate,
 };
@@ -3196,7 +3200,7 @@ static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
 	.set_sysclk	= wm8994_set_dai_sysclk,
 	.set_fmt	= wm8994_set_dai_fmt,
 	.hw_params	= wm8994_hw_params,
-	.digital_mute   = wm8994_aif_mute,
+	.mute_stream	= wm8994_aif_mute,
 	.set_pll	= wm8994_set_fll,
 	.set_tristate	= wm8994_set_tristate,
 };
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 276ffa84cc31..a79906fa0cea 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1417,11 +1417,14 @@ static bool wm8995_volatile(struct device *dev, unsigned int reg)
 	}
 }
 
-static int wm8995_aif_mute(struct snd_soc_dai *dai, int mute)
+static int wm8995_aif_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	int mute_reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	switch (dai->id) {
 	case 0:
 		mute_reg = WM8995_AIF1_DAC1_FILTERS_1;
@@ -2094,7 +2097,7 @@ static const struct snd_soc_dai_ops wm8995_aif1_dai_ops = {
 	.set_sysclk = wm8995_set_dai_sysclk,
 	.set_fmt = wm8995_set_dai_fmt,
 	.hw_params = wm8995_hw_params,
-	.digital_mute = wm8995_aif_mute,
+	.mute_stream = wm8995_aif_mute,
 	.set_pll = wm8995_set_fll,
 	.set_tristate = wm8995_set_tristate,
 };
@@ -2103,7 +2106,7 @@ static const struct snd_soc_dai_ops wm8995_aif2_dai_ops = {
 	.set_sysclk = wm8995_set_dai_sysclk,
 	.set_fmt = wm8995_set_dai_fmt,
 	.hw_params = wm8995_hw_params,
-	.digital_mute = wm8995_aif_mute,
+	.mute_stream = wm8995_aif_mute,
 	.set_pll = wm8995_set_fll,
 	.set_tristate = wm8995_set_tristate,
 };
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index be5c9c2b0162..60e20aeb81ae 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1147,11 +1147,14 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int wm9081_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
 {
 	struct snd_soc_component *component = codec_dai->component;
 	unsigned int reg;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	reg = snd_soc_component_read(component, WM9081_DAC_DIGITAL_2);
 
 	if (mute)
@@ -1232,7 +1235,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
 static const struct snd_soc_dai_ops wm9081_dai_ops = {
 	.hw_params = wm9081_hw_params,
 	.set_fmt = wm9081_set_dai_fmt,
-	.digital_mute = wm9081_digital_mute,
+	.mute_stream = wm9081_mute,
 	.set_tdm_slot = wm9081_set_tdm_slot,
 };
 
-- 
2.25.1

[PATCH 14/19] ASoC: codecs: es*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/codecs/es8316.c | 7 +++++--
 sound/soc/codecs/es8328.c | 7 +++++--
 2 files changed, 10 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 36eef1fb3d18..6e0066fc9e26 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -507,8 +507,11 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int es8316_mute(struct snd_soc_dai *dai, int mute)
+static int es8316_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	snd_soc_component_update_bits(dai->component, ES8316_DAC_SET1, 0x20,
 			    mute ? 0x20 : 0);
 	return 0;
@@ -522,7 +525,7 @@ static const struct snd_soc_dai_ops es8316_ops = {
 	.hw_params = es8316_pcm_hw_params,
 	.set_fmt = es8316_set_dai_fmt,
 	.set_sysclk = es8316_set_dai_sysclk,
-	.digital_mute = es8316_mute,
+	.mute_stream = es8316_mute,
 };
 
 static struct snd_soc_dai_driver es8316_dai = {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index fdf64c29f563..b8ac7b31e2b9 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -449,8 +449,11 @@ static const struct snd_soc_dapm_route es8328_dapm_routes[] = {
 	{ "ROUT2", NULL, "Right Out 2" },
 };
 
-static int es8328_mute(struct snd_soc_dai *dai, int mute)
+static int es8328_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	return snd_soc_component_update_bits(dai->component, ES8328_DACCONTROL3,
 			ES8328_DACCONTROL3_DACMUTE,
 			mute ? ES8328_DACCONTROL3_DACMUTE : 0);
@@ -692,7 +695,7 @@ static int es8328_set_bias_level(struct snd_soc_component *component,
 static const struct snd_soc_dai_ops es8328_dai_ops = {
 	.startup	= es8328_startup,
 	.hw_params	= es8328_hw_params,
-	.digital_mute	= es8328_mute,
+	.mute_stream	= es8328_mute,
 	.set_sysclk	= es8328_set_sysclk,
 	.set_fmt	= es8328_set_dai_fmt,
 };
-- 
2.25.1

[PATCH 15/19] ASoC: codecs: da*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/codecs/da7210.c | 7 +++++--
 sound/soc/codecs/da7213.c | 7 +++++--
 sound/soc/codecs/da9055.c | 7 +++++--
 3 files changed, 15 insertions(+), 6 deletions(-)

diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 0c99dcf242e4..8e5e5cd4bcbe 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -924,11 +924,14 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 	return 0;
 }
 
-static int da7210_mute(struct snd_soc_dai *dai, int mute)
+static int da7210_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u8 mute_reg = snd_soc_component_read(component, DA7210_DAC_HPF) & 0xFB;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (mute)
 		snd_soc_component_write(component, DA7210_DAC_HPF, mute_reg | 0x4);
 	else
@@ -1034,7 +1037,7 @@ static const struct snd_soc_dai_ops da7210_dai_ops = {
 	.set_fmt	= da7210_set_dai_fmt,
 	.set_sysclk	= da7210_set_dai_sysclk,
 	.set_pll	= da7210_set_dai_pll,
-	.digital_mute	= da7210_mute,
+	.mute_stream	= da7210_mute,
 };
 
 static struct snd_soc_dai_driver da7210_dai = {
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index cc4ae7b311b4..55f8097112e1 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1321,10 +1321,13 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	return 0;
 }
 
-static int da7213_mute(struct snd_soc_dai *dai, int mute)
+static int da7213_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, DA7213_DAC_L_CTRL,
 				    DA7213_MUTE_EN, DA7213_MUTE_EN);
@@ -1507,7 +1510,7 @@ static int da7213_set_component_pll(struct snd_soc_component *component,
 static const struct snd_soc_dai_ops da7213_dai_ops = {
 	.hw_params	= da7213_hw_params,
 	.set_fmt	= da7213_set_dai_fmt,
-	.digital_mute	= da7213_mute,
+	.mute_stream	= da7213_mute,
 };
 
 static struct snd_soc_dai_driver da7213_dai = {
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index e93436ccb674..e388b1c0ba19 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1211,10 +1211,13 @@ static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	return 0;
 }
 
-static int da9055_mute(struct snd_soc_dai *dai, int mute)
+static int da9055_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, DA9055_DAC_L_CTRL,
 				    DA9055_DAC_L_MUTE_EN, DA9055_DAC_L_MUTE_EN);
@@ -1324,7 +1327,7 @@ static const struct snd_soc_dai_ops da9055_dai_ops = {
 	.set_fmt	= da9055_set_dai_fmt,
 	.set_sysclk	= da9055_set_dai_sysclk,
 	.set_pll	= da9055_set_dai_pll,
-	.digital_mute	= da9055_mute,
+	.mute_stream	= da9055_mute,
 };
 
 static struct snd_soc_dai_driver da9055_dai = {
-- 
2.25.1

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

[Kuninori Morimoto]

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>
---
 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

[PATCH 17/19] ASoC: codecs: ak*: merge .digital_mute() into .mute_stream()

[Kuninori Morimoto]

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>
---
 sound/soc/codecs/ak4458.c | 7 +++++--
 sound/soc/codecs/ak4535.c | 8 ++++++--
 sound/soc/codecs/ak4641.c | 9 ++++++---
 3 files changed, 17 insertions(+), 7 deletions(-)

diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index f180cb5dfe4f..7ffae5950142 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -401,13 +401,16 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 static const int att_speed[] = { 4080, 2040, 510, 255 };
 
-static int ak4458_set_dai_mute(struct snd_soc_dai *dai, int mute)
+static int ak4458_set_dai_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component);
 	int nfs, ndt, ret, reg;
 	int ats;
 
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	nfs = ak4458->fs;
 
 	reg = snd_soc_component_read(component, AK4458_0B_CONTROL7);
@@ -495,7 +498,7 @@ static const struct snd_soc_dai_ops ak4458_dai_ops = {
 	.startup        = ak4458_startup,
 	.hw_params	= ak4458_hw_params,
 	.set_fmt	= ak4458_set_dai_fmt,
-	.digital_mute	= ak4458_set_dai_mute,
+	.mute_stream	= ak4458_set_dai_mute,
 	.set_tdm_slot	= ak4458_set_tdm_slot,
 };
 
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index f5ad1f59eb46..37bbe239a515 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -309,10 +309,14 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-static int ak4535_mute(struct snd_soc_dai *dai, int mute)
+static int ak4535_mute(struct snd_soc_dai *dai, int mute, int direction)
 {
 	struct snd_soc_component *component = dai->component;
 	u16 mute_reg = snd_soc_component_read(component, AK4535_DAC);
+
+	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
 	if (!mute)
 		snd_soc_component_write(component, AK4535_DAC, mute_reg & ~0x20);
 	else
@@ -348,7 +352,7 @@ static int ak4535_set_bias_level(struct snd_soc_component *component,
 static const struct snd_soc_dai_ops ak4535_dai_ops = {
 	.hw_params	= ak4535_hw_params,
 	.set_fmt	= ak4535_set_dai_fmt,
-	.digital_mute	= ak4535_mute,
+	.mute_stream	= ak4535_mute,
 	.set_sysclk	= ak4535_set_dai_sysclk,
 };
 
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 2d5b640aab58..2602970a824e 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -405,10 +405,13 @@ static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return snd_soc_component_write(component, AK4641_MODE1, mode1);
 }
 
-static int ak4641_mute(struct snd_soc_dai *dai, int mute)
+static int ak4641_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;
+
 	return snd_soc_component_update_bits(component, AK4641_DAC, 0x20, mute ? 0x20 : 0);
 }
 
@@ -467,14 +470,14 @@ static int ak4641_set_bias_level(struct snd_soc_component *component,
 static const struct snd_soc_dai_ops ak4641_i2s_dai_ops = {
 	.hw_params    = ak4641_i2s_hw_params,
 	.set_fmt      = ak4641_i2s_set_dai_fmt,
-	.digital_mute = ak4641_mute,
+	.mute_stream  = ak4641_mute,
 	.set_sysclk   = ak4641_set_dai_sysclk,
 };
 
 static const struct snd_soc_dai_ops ak4641_pcm_dai_ops = {
 	.hw_params    = NULL, /* rates are controlled by BT chip */
 	.set_fmt      = ak4641_pcm_set_dai_fmt,
-	.digital_mute = ak4641_mute,
+	.mute_stream  = ak4641_mute,
 	.set_sysclk   = ak4641_set_dai_sysclk,
 };
 
-- 
2.25.1

[PATCH 18/19] ASoC: soc-dai: remove .digital_mute

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

All drivers are now using .mute_stream.
Let's remove .digital_mute.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
 include/sound/soc-dai.h | 1 -
 sound/soc/soc-dai.c     | 4 ----
 2 files changed, 5 deletions(-)

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 212257e84fac..a3db208cd062 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -246,7 +246,6 @@ struct snd_soc_dai_ops {
 	 * DAI digital mute - optional.
 	 * Called by soc-core to minimise any pops.
 	 */
-	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
 	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
 
 	/*
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index b05e18b63a1c..b0fc3d8b1798 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -301,10 +301,6 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 	if (dai->driver->ops &&
 	    dai->driver->ops->mute_stream)
 		ret = dai->driver->ops->mute_stream(dai, mute, direction);
-	else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
-		 dai->driver->ops &&
-		 dai->driver->ops->digital_mute)
-		ret = dai->driver->ops->digital_mute(dai, mute);
 
 	return soc_dai_ret(dai, ret);
 }
-- 
2.25.1

[PATCH 19/19] ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec

[Kuninori Morimoto]

From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>

snd_soc_dai_digital_mute() is used for both CPU and Codec.
For example, soc_pcm_prepare() / soc_pcm_hw_free() are caring
both CPU and Codec.
But, soc_resume_deferred() / snd_soc_suspend() are not.
This patch cares it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
 sound/soc/soc-core.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 62c0c9482018..d84d91d8e3e0 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -548,7 +548,7 @@ int snd_soc_suspend(struct device *dev)
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		for_each_rtd_codec_dais(rtd, i, dai) {
+		for_each_rtd_dais(rtd, i, dai) {
 			if (snd_soc_dai_stream_active(dai, playback))
 				snd_soc_dai_digital_mute(dai, 1, playback);
 		}
@@ -687,7 +687,7 @@ static void soc_resume_deferred(struct work_struct *work)
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		for_each_rtd_codec_dais(rtd, i, dai) {
+		for_each_rtd_dais(rtd, i, dai) {
 			if (snd_soc_dai_stream_active(dai, playback))
 				snd_soc_dai_digital_mute(dai, 0, playback);
 		}
-- 
2.25.1

Re: [PATCH 00/19] ASoC: merge .digital_mute() into .mute_stream()

[Peter Ujfalusi]

Hi Morimoto-san,

On 23/06/2020 4.16, Kuninori Morimoto wrote:
> 
> 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().

Instead of finding the patch to ack, I just checked the series and:
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>

- Péter

> 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(-)
> 


Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki.
Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

Re: [PATCH 13/19] ASoC: codecs: wm*: merge .digital_mute() into .mute_stream()

[Charles Keepax]

On Tue, Jun 23, 2020 at 10:20:43AM +0900, Kuninori Morimoto wrote:
> 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>
> ---

Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>

Thanks,
Charles

Re: [PATCH 05/19] ASoC: atmel: merge .digital_mute() into .mute_stream()

[Alexandre Belloni]

On 23/06/2020 10:19:56+0900, Kuninori Morimoto wrote:
> 
> 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: Alexandre Belloni <alexandre.belloni@bootlin.com>

> ---
>  sound/soc/atmel/atmel-classd.c | 10 +++++++---
>  1 file changed, 7 insertions(+), 3 deletions(-)
> 
> diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
> index e98601eccfa3..59231c0371b8 100644
> --- a/sound/soc/atmel/atmel-classd.c
> +++ b/sound/soc/atmel/atmel-classd.c
> @@ -327,12 +327,16 @@ static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream,
>  	return clk_prepare_enable(dd->gclk);
>  }
>  
> -static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai,
> -	int mute)
> +static int atmel_classd_codec_dai_mute(struct snd_soc_dai *codec_dai,
> +				       int mute,
> +				       int direction)
>  {
>  	struct snd_soc_component *component = codec_dai->component;
>  	u32 mask, val;
>  
> +	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
> +		return 0;
> +
>  	mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK;
>  
>  	if (mute)
> @@ -469,7 +473,7 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream,
>  }
>  
>  static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = {
> -	.digital_mute	= atmel_classd_codec_dai_digital_mute,
> +	.mute_stream	= atmel_classd_codec_dai_mute,
>  	.startup	= atmel_classd_codec_dai_startup,
>  	.shutdown	= atmel_classd_codec_dai_shutdown,
>  	.hw_params	= atmel_classd_codec_dai_hw_params,
> -- 
> 2.25.1
> 

-- 
Alexandre Belloni, Bootlin
Embedded Linux and Kernel engineering
https://bootlin.com

RE: [PATCH 15/19] ASoC: codecs: da*: merge .digital_mute() into .mute_stream()

[Adam Thomson]

On 23 June 2020 02:21, Kuninori Morimoto wrote:

> 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: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>

> ---
>  sound/soc/codecs/da7210.c | 7 +++++--
>  sound/soc/codecs/da7213.c | 7 +++++--
>  sound/soc/codecs/da9055.c | 7 +++++--
>  3 files changed, 15 insertions(+), 6 deletions(-)
> 
> diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
> index 0c99dcf242e4..8e5e5cd4bcbe 100644
> --- a/sound/soc/codecs/da7210.c
> +++ b/sound/soc/codecs/da7210.c
> @@ -924,11 +924,14 @@ static int da7210_set_dai_fmt(struct snd_soc_dai
> *codec_dai, u32 fmt)
>  	return 0;
>  }
> 
> -static int da7210_mute(struct snd_soc_dai *dai, int mute)
> +static int da7210_mute(struct snd_soc_dai *dai, int mute, int direction)
>  {
>  	struct snd_soc_component *component = dai->component;
>  	u8 mute_reg = snd_soc_component_read(component,
> DA7210_DAC_HPF) & 0xFB;
> 
> +	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
> +		return 0;
> +
>  	if (mute)
>  		snd_soc_component_write(component, DA7210_DAC_HPF,
> mute_reg | 0x4);
>  	else
> @@ -1034,7 +1037,7 @@ static const struct snd_soc_dai_ops da7210_dai_ops = {
>  	.set_fmt	= da7210_set_dai_fmt,
>  	.set_sysclk	= da7210_set_dai_sysclk,
>  	.set_pll	= da7210_set_dai_pll,
> -	.digital_mute	= da7210_mute,
> +	.mute_stream	= da7210_mute,
>  };
> 
>  static struct snd_soc_dai_driver da7210_dai = {
> diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
> index cc4ae7b311b4..55f8097112e1 100644
> --- a/sound/soc/codecs/da7213.c
> +++ b/sound/soc/codecs/da7213.c
> @@ -1321,10 +1321,13 @@ static int da7213_set_dai_fmt(struct snd_soc_dai
> *codec_dai, unsigned int fmt)
>  	return 0;
>  }
> 
> -static int da7213_mute(struct snd_soc_dai *dai, int mute)
> +static int da7213_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,
> DA7213_DAC_L_CTRL,
>  				    DA7213_MUTE_EN, DA7213_MUTE_EN);
> @@ -1507,7 +1510,7 @@ static int da7213_set_component_pll(struct
> snd_soc_component *component,
>  static const struct snd_soc_dai_ops da7213_dai_ops = {
>  	.hw_params	= da7213_hw_params,
>  	.set_fmt	= da7213_set_dai_fmt,
> -	.digital_mute	= da7213_mute,
> +	.mute_stream	= da7213_mute,
>  };
> 
>  static struct snd_soc_dai_driver da7213_dai = {
> diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
> index e93436ccb674..e388b1c0ba19 100644
> --- a/sound/soc/codecs/da9055.c
> +++ b/sound/soc/codecs/da9055.c
> @@ -1211,10 +1211,13 @@ static int da9055_set_dai_fmt(struct snd_soc_dai
> *codec_dai, unsigned int fmt)
>  	return 0;
>  }
> 
> -static int da9055_mute(struct snd_soc_dai *dai, int mute)
> +static int da9055_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,
> DA9055_DAC_L_CTRL,
>  				    DA9055_DAC_L_MUTE_EN,
> DA9055_DAC_L_MUTE_EN);
> @@ -1324,7 +1327,7 @@ static const struct snd_soc_dai_ops da9055_dai_ops = {
>  	.set_fmt	= da9055_set_dai_fmt,
>  	.set_sysclk	= da9055_set_dai_sysclk,
>  	.set_pll	= da9055_set_dai_pll,
> -	.digital_mute	= da9055_mute,
> +	.mute_stream	= da9055_mute,
>  };
> 
>  static struct snd_soc_dai_driver da9055_dai = {
> --
> 2.25.1

Re: [PATCH 01/19] ASoC: hdmi-codec: merge .digital_mute() into .mute_stream()

[Pierre-Louis Bossart]

> -static int sii902x_audio_digital_mute(struct device *dev,
> -				      void *data, bool enable)
> +static int sii902x_audio_mute(struct device *dev, void *data,
> +			      bool enable, int direction)
>   {
>   	struct sii902x *sii902x = dev_get_drvdata(dev);
>   
> +	if (direction != SNDRV_PCM_STREAM_PLAYBACK)
> +		return 0;
> +

Thanks for this clean-up Morimoto-san.

One question: in all the codec changes the return is now 0.

But in the previous implementation based on digital_mute, when the wrong 
direction was used it would have been -ENOTSUPP.

int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
			     int direction)
{
	int ret = -ENOTSUPP;

	if (dai->driver->ops &&
	    dai->driver->ops->mute_stream)
		ret = dai->driver->ops->mute_stream(dai, mute, direction);
	else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
		 dai->driver->ops &&
		 dai->driver->ops->digital_mute)
		ret = dai->driver->ops->digital_mute(dai, mute);

<<< none of the branches taken for capture + digital_mute supported, so 
return -ENOTSUPP.

	return soc_dai_ret(dai, ret);
}

Was this change intentional?

We also want to check why this return value is only tested in 
soc-dapm.c, if this digital_mute can fail then we are missing tests left 
and right - maybe that's a follow-up change?

Re: [PATCH 18/19] ASoC: soc-dai: remove .digital_mute

[Dan Murphy]

Kuninori

On 6/22/20 8:21 PM, Kuninori Morimoto wrote:
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
>
> All drivers are now using .mute_stream.
> Let's remove .digital_mute.
>
> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> ---
>   include/sound/soc-dai.h | 1 -
>   sound/soc/soc-dai.c     | 4 ----
>   2 files changed, 5 deletions(-)
>
> diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
> index 212257e84fac..a3db208cd062 100644
> --- a/include/sound/soc-dai.h
> +++ b/include/sound/soc-dai.h
> @@ -246,7 +246,6 @@ struct snd_soc_dai_ops {
>   	 * DAI digital mute - optional.
>   	 * Called by soc-core to minimise any pops.
>   	 */
> -	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
>   	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
>   
>   	/*
> diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
> index b05e18b63a1c..b0fc3d8b1798 100644
> --- a/sound/soc/soc-dai.c
> +++ b/sound/soc/soc-dai.c
> @@ -301,10 +301,6 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
>   	if (dai->driver->ops &&
>   	    dai->driver->ops->mute_stream)
>   		ret = dai->driver->ops->mute_stream(dai, mute, direction);
> -	else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
> -		 dai->driver->ops &&
> -		 dai->driver->ops->digital_mute)
> -		ret = dai->driver->ops->digital_mute(dai, mute);

Instead of putting the direction check in every codec driver can't the 
check just part of the function? And then maybe a flag in dai_ops to 
override that call to mute?

if (direction != SNDRV_PCM_STREAM_PLAYBACK)
         return 0;

I did not look at every codec driver as there are too many.

Dan