* [PATCH v3 01/21] ASoC: hdmi-codec: return -ENOTSUPP for digital_mute
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
@ 2020-07-09 1:55 ` Kuninori Morimoto
2020-07-09 1:55 ` [PATCH v3 02/21] ASoC: soc-dai.c: add .no_capture_mute support Kuninori Morimoto
` (20 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:55 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() will return -ENOTSUPP if driver doesn't
support mute.
In hdmi-codec case, hdmi_codec_digital_mute() will be used for it,
and each driver has .digital_mute() callback.
hdmi_codec_digital_mute() want to return -ENOTSUPP to follow it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
sound/soc/codecs/hdmi-codec.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index f005751da2cc..926ab447a96b 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -566,7 +566,7 @@ static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute)
return hcp->hcd.ops->digital_mute(dai->dev->parent,
hcp->hcd.data, mute);
- return 0;
+ return -ENOTSUPP;
}
static const struct snd_soc_dai_ops hdmi_codec_i2s_dai_ops = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 02/21] ASoC: soc-dai.c: add .no_capture_mute support
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
2020-07-09 1:55 ` [PATCH v3 01/21] ASoC: hdmi-codec: return -ENOTSUPP for digital_mute Kuninori Morimoto
@ 2020-07-09 1:55 ` Kuninori Morimoto
2020-07-09 1:55 ` [PATCH v3 03/21] ASoC: hdmi-codec: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (19 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:55 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
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);
...
}
To prepare merging mute_stream()/digital_mute(),
this patch adds .no_capture_mute support to emulate .digital_mute().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
include/sound/soc-dai.h | 3 +++
sound/soc/soc-dai.c | 8 +++++++-
2 files changed, 10 insertions(+), 1 deletion(-)
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 212257e84fac..e0e061b8e9bd 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -280,6 +280,9 @@ struct snd_soc_dai_ops {
*/
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
struct snd_soc_dai *);
+
+ /* bit field */
+ unsigned int no_capture_mute:1;
};
struct snd_soc_cdai_ops {
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index b05e18b63a1c..458d2ea44329 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -298,8 +298,14 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
{
int ret = -ENOTSUPP;
+ /*
+ * ignore if direction was CAPTURE
+ * and it had .no_capture_mute flag
+ */
if (dai->driver->ops &&
- dai->driver->ops->mute_stream)
+ dai->driver->ops->mute_stream &&
+ (direction == SNDRV_PCM_STREAM_PLAYBACK ||
+ !dai->driver->ops->no_capture_mute))
ret = dai->driver->ops->mute_stream(dai, mute, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops &&
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 03/21] ASoC: hdmi-codec: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
2020-07-09 1:55 ` [PATCH v3 01/21] ASoC: hdmi-codec: return -ENOTSUPP for digital_mute Kuninori Morimoto
2020-07-09 1:55 ` [PATCH v3 02/21] ASoC: soc-dai.c: add .no_capture_mute support Kuninori Morimoto
@ 2020-07-09 1:55 ` Kuninori Morimoto
2020-07-09 1:55 ` [PATCH v3 04/21] ASoC: ti: " Kuninori Morimoto
` (18 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:55 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
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>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
drivers/gpu/drm/bridge/sii902x.c | 7 ++++---
drivers/gpu/drm/exynos/exynos_hdmi.c | 6 ++++--
drivers/gpu/drm/i2c/tda998x_drv.c | 7 ++++---
drivers/gpu/drm/mediatek/mtk_hdmi.c | 6 ++++--
drivers/gpu/drm/rockchip/cdn-dp-core.c | 7 ++++---
drivers/gpu/drm/sti/sti_hdmi.c | 6 ++++--
drivers/gpu/drm/zte/zx_hdmi.c | 7 ++++---
include/sound/hdmi-codec.h | 6 +++++-
sound/soc/codecs/hdmi-codec.c | 21 +++++++++++++++------
9 files changed, 48 insertions(+), 25 deletions(-)
diff --git a/drivers/gpu/drm/bridge/sii902x.c b/drivers/gpu/drm/bridge/sii902x.c
index 6dad025f8da7..c751baf3d064 100644
--- a/drivers/gpu/drm/bridge/sii902x.c
+++ b/drivers/gpu/drm/bridge/sii902x.c
@@ -672,8 +672,8 @@ 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);
@@ -724,9 +724,10 @@ 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,
+ .no_capture_mute = 1,
};
static int sii902x_audio_codec_init(struct sii902x *sii902x,
diff --git a/drivers/gpu/drm/exynos/exynos_hdmi.c b/drivers/gpu/drm/exynos/exynos_hdmi.c
index 95dd399aa9cc..68d7b1ce1b7c 100644
--- a/drivers/gpu/drm/exynos/exynos_hdmi.c
+++ b/drivers/gpu/drm/exynos/exynos_hdmi.c
@@ -1604,7 +1604,8 @@ 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);
@@ -1634,8 +1635,9 @@ 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,
+ .no_capture_mute = 1,
};
static int hdmi_register_audio_device(struct hdmi_context *hdata)
diff --git a/drivers/gpu/drm/i2c/tda998x_drv.c b/drivers/gpu/drm/i2c/tda998x_drv.c
index 9517f522dcb9..3010a4536da3 100644
--- a/drivers/gpu/drm/i2c/tda998x_drv.c
+++ b/drivers/gpu/drm/i2c/tda998x_drv.c
@@ -1133,8 +1133,8 @@ 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);
@@ -1162,8 +1162,9 @@ 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,
+ .no_capture_mute = 1,
};
static int tda998x_audio_codec_init(struct tda998x_priv *priv,
diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c
index 5feb760617cb..37b4420a0b22 100644
--- a/drivers/gpu/drm/mediatek/mtk_hdmi.c
+++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c
@@ -1647,7 +1647,8 @@ 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);
@@ -1692,9 +1693,10 @@ 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,
+ .no_capture_mute = 1,
};
static int mtk_hdmi_register_audio_driver(struct device *dev)
diff --git a/drivers/gpu/drm/rockchip/cdn-dp-core.c b/drivers/gpu/drm/rockchip/cdn-dp-core.c
index c634b95b50f7..a4a45daf93f2 100644
--- a/drivers/gpu/drm/rockchip/cdn-dp-core.c
+++ b/drivers/gpu/drm/rockchip/cdn-dp-core.c
@@ -817,8 +817,8 @@ 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;
@@ -849,8 +849,9 @@ 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,
+ .no_capture_mute = 1,
};
static int cdn_dp_audio_codec_init(struct cdn_dp_device *dp,
diff --git a/drivers/gpu/drm/sti/sti_hdmi.c b/drivers/gpu/drm/sti/sti_hdmi.c
index 5b15c4974e6b..008f07923bbc 100644
--- a/drivers/gpu/drm/sti/sti_hdmi.c
+++ b/drivers/gpu/drm/sti/sti_hdmi.c
@@ -1191,7 +1191,8 @@ 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);
@@ -1219,8 +1220,9 @@ 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,
+ .no_capture_mute = 1,
};
static int sti_hdmi_register_audio_driver(struct device *dev,
diff --git a/drivers/gpu/drm/zte/zx_hdmi.c b/drivers/gpu/drm/zte/zx_hdmi.c
index 76a16d997a23..cd79ca0a92a9 100644
--- a/drivers/gpu/drm/zte/zx_hdmi.c
+++ b/drivers/gpu/drm/zte/zx_hdmi.c
@@ -439,8 +439,8 @@ 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);
@@ -468,8 +468,9 @@ 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,
+ .no_capture_mute = 1,
};
static struct hdmi_codec_pdata zx_hdmi_codec_pdata = {
diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h
index 83b17682e01c..17eebd34835a 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.
@@ -99,6 +100,9 @@ struct hdmi_codec_ops {
int (*hook_plugged_cb)(struct device *dev, void *data,
hdmi_codec_plugged_cb fn,
struct device *codec_dev);
+
+ /* bit field */
+ unsigned int no_capture_mute:1;
};
/* HDMI codec initalization data */
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 926ab447a96b..bc760a81e217 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -558,13 +558,22 @@ 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);
+ /*
+ * ignore if direction was CAPTURE
+ * and it had .no_capture_mute flag
+ * see
+ * snd_soc_dai_digital_mute()
+ */
+ if (hcp->hcd.ops->mute_stream &&
+ (direction == SNDRV_PCM_STREAM_PLAYBACK ||
+ !hcp->hcd.ops->no_capture_mute))
+ return hcp->hcd.ops->mute_stream(dai->dev->parent,
+ hcp->hcd.data,
+ mute, direction);
return -ENOTSUPP;
}
@@ -574,14 +583,14 @@ 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
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 04/21] ASoC: ti: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (2 preceding siblings ...)
2020-07-09 1:55 ` [PATCH v3 03/21] ASoC: hdmi-codec: merge .digital_mute() into .mute_stream() Kuninori Morimoto
@ 2020-07-09 1:55 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 05/21] ASoC: spear: " Kuninori Morimoto
` (17 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:55 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/ti/ams-delta.c | 9 +++++----
1 file changed, 5 insertions(+), 4 deletions(-)
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
index e17cd5e939f0..5c47de96c529 100644
--- a/sound/soc/ti/ams-delta.c
+++ b/sound/soc/ti/ams-delta.c
@@ -420,7 +420,7 @@ 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;
@@ -439,18 +439,19 @@ 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,
+ .no_capture_mute = 1,
};
/* 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
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 05/21] ASoC: spear: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (3 preceding siblings ...)
2020-07-09 1:55 ` [PATCH v3 04/21] ASoC: ti: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 06/21] ASoC: meson: " Kuninori Morimoto
` (16 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/spear/spdif_out.c | 8 +++++---
1 file changed, 5 insertions(+), 3 deletions(-)
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
index 58d5843811f9..38f9fff5be6b 100644
--- a/sound/soc/spear/spdif_out.c
+++ b/sound/soc/spear/spdif_out.c
@@ -188,7 +188,7 @@ 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;
@@ -229,7 +229,8 @@ 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;
}
@@ -250,11 +251,12 @@ 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,
.hw_params = spdif_out_hw_params,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver spdif_out_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 06/21] ASoC: meson: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (4 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 05/21] ASoC: spear: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 07/21] ASoC: atmel: " Kuninori Morimoto
` (15 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/meson/axg-spdifout.c | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c
index 7ce6aa97ddf7..e769a5ee6e27 100644
--- a/sound/soc/meson/axg-spdifout.c
+++ b/sound/soc/meson/axg-spdifout.c
@@ -108,7 +108,7 @@ 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);
@@ -285,10 +285,11 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver axg_spdifout_dai_drv[] = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 07/21] ASoC: atmel: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (5 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 06/21] ASoC: meson: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-16 20:44 ` Mark Brown
2020-07-09 1:56 ` [PATCH v3 08/21] ASoC: codecs: " Kuninori Morimoto
` (14 subsequent siblings)
21 siblings, 1 reply; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
---
sound/soc/atmel/atmel-classd.c | 7 ++++---
1 file changed, 4 insertions(+), 3 deletions(-)
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index e98601eccfa3..5deabb506a79 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/sound/soc/atmel/atmel-classd.c
@@ -327,8 +327,8 @@ 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;
@@ -469,12 +469,13 @@ 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,
.prepare = atmel_classd_codec_dai_prepare,
.trigger = atmel_classd_codec_dai_trigger,
+ .no_capture_mute = 1,
};
#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi"
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* Re: [PATCH v3 07/21] ASoC: atmel: merge .digital_mute() into .mute_stream()
2020-07-09 1:56 ` [PATCH v3 07/21] ASoC: atmel: " Kuninori Morimoto
@ 2020-07-16 20:44 ` Mark Brown
2020-07-16 23:27 ` Kuninori Morimoto
0 siblings, 1 reply; 26+ messages in thread
From: Mark Brown @ 2020-07-16 20:44 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: Linux-ALSA
[-- Attachment #1: Type: text/plain, Size: 377 bytes --]
On Thu, Jul 09, 2020 at 10:56:15AM +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.
At least this patch needs a rebase against current code, possibly others.
[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [PATCH v3 07/21] ASoC: atmel: merge .digital_mute() into .mute_stream()
2020-07-16 20:44 ` Mark Brown
@ 2020-07-16 23:27 ` Kuninori Morimoto
2020-07-16 23:59 ` Mark Brown
0 siblings, 1 reply; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-16 23:27 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
Hi Mark
> > 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.
>
> At least this patch needs a rebase against current code, possibly others.
I noticed it yesterday.
OK, I will rebase and re-post v4
Thank you for your help !!
Best regards
---
Kuninori Morimoto
^ permalink raw reply [flat|nested] 26+ messages in thread
* Re: [PATCH v3 07/21] ASoC: atmel: merge .digital_mute() into .mute_stream()
2020-07-16 23:27 ` Kuninori Morimoto
@ 2020-07-16 23:59 ` Mark Brown
0 siblings, 0 replies; 26+ messages in thread
From: Mark Brown @ 2020-07-16 23:59 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: Linux-ALSA
[-- Attachment #1: Type: text/plain, Size: 583 bytes --]
On Fri, Jul 17, 2020 at 08:27:46AM +0900, Kuninori Morimoto wrote:
> > > 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.
> > At least this patch needs a rebase against current code, possibly others.
> I noticed it yesterday.
> OK, I will rebase and re-post v4
Turned out it was just that one - everything else (except the final
removal patches) was fine, git fixed up the remaining issues itself.
> Thank you for your help !!
Thank you for your hard work!
[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 26+ messages in thread
* [PATCH v3 08/21] ASoC: codecs: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (6 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 07/21] ASoC: atmel: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 09/21] ASoC: codecs: tlv*: " Kuninori Morimoto
` (13 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/88pm860x-codec.c | 8 +++++---
sound/soc/codecs/ad193x.c | 5 +++--
sound/soc/codecs/adau1701.c | 5 +++--
sound/soc/codecs/cpcap.c | 11 +++++++----
sound/soc/codecs/cq93vc.c | 5 +++--
sound/soc/codecs/isabelle.c | 15 +++++++++------
sound/soc/codecs/jz4770.c | 5 +++--
sound/soc/codecs/lm49453.c | 25 +++++++++++++++----------
sound/soc/codecs/ml26124.c | 5 +++--
sound/soc/codecs/nau8822.c | 5 +++--
sound/soc/codecs/rk3328_codec.c | 5 +++--
sound/soc/codecs/sgtl5000.c | 5 +++--
sound/soc/codecs/sta529.c | 5 +++--
sound/soc/codecs/tfa9879.c | 5 +++--
sound/soc/codecs/twl6040.c | 5 +++--
sound/soc/codecs/uda134x.c | 5 +++--
16 files changed, 72 insertions(+), 47 deletions(-)
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 068914d0ef3d..cac7e557edc8 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -902,7 +902,7 @@ 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_stream(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;
@@ -1136,17 +1136,19 @@ 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_stream,
.hw_params = pm860x_pcm_hw_params,
.set_fmt = pm860x_pcm_set_dai_fmt,
.set_sysclk = pm860x_set_dai_sysclk,
+ .no_capture_mute = 1,
};
static const struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
- .digital_mute = pm860x_digital_mute,
+ .mute_stream = pm860x_mute_stream,
.hw_params = pm860x_i2s_hw_params,
.set_fmt = pm860x_i2s_set_dai_fmt,
.set_sysclk = pm860x_set_dai_sysclk,
+ .no_capture_mute = 1,
};
#define PM860X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 980e024a5720..f37ab7eda615 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -143,7 +143,7 @@ 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);
@@ -371,10 +371,11 @@ 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,
+ .no_capture_mute = 1,
};
/* codec DAI instance */
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 115e296b2ad6..68130eaa64a4 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -573,7 +573,7 @@ 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_stream(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
unsigned int mask = ADAU1701_DSPCTRL_DAM;
@@ -631,8 +631,9 @@ 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_stream,
.startup = adau1701_startup,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver adau1701_dai = {
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
index d7f05b384f1f..f046987ee4cd 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);
@@ -1237,7 +1237,8 @@ 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,
+ .no_capture_mute = 1,
};
static int cpcap_voice_hw_params(struct snd_pcm_substream *substream,
@@ -1370,7 +1371,8 @@ 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);
@@ -1391,7 +1393,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cpcap_dai[] = {
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index b0cc61178a41..0aae5790222a 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -30,7 +30,7 @@ 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;
@@ -87,8 +87,9 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cq93vc_dai = {
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 3626f70f7768..79afced75d76 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -860,7 +860,7 @@ 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)
{
snd_soc_component_update_bits(dai->component, ISABELLE_DAC1_SOFTRAMP_REG,
BIT(4), (mute ? BIT(4) : 0));
@@ -868,7 +868,7 @@ static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute)
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)
{
snd_soc_component_update_bits(dai->component, ISABELLE_DAC2_SOFTRAMP_REG,
BIT(4), (mute ? BIT(4) : 0));
@@ -876,7 +876,7 @@ static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute)
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)
{
snd_soc_component_update_bits(dai->component, ISABELLE_DAC3_SOFTRAMP_REG,
BIT(4), (mute ? BIT(4) : 0));
@@ -1014,19 +1014,22 @@ 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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
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..e98f04749c11 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_stream(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);
@@ -753,7 +753,8 @@ 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_stream,
+ .no_capture_mute = 1,
};
#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..06ab61f6f719 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1218,35 +1218,35 @@ 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)
{
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)
{
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)
{
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)
{
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)
{
snd_soc_component_update_bits(dai->component, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6),
(mute ? (BIT(7)|BIT(6)) : 0));
@@ -1288,35 +1288,40 @@ 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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
/* LM49453 dai structure. */
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 55823bc95d06..70c17be455ca 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -372,7 +372,7 @@ 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);
@@ -492,9 +492,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver ml26124_dai = {
diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c
index 79928ddeb7a1..609aeeb27818 100644
--- a/sound/soc/codecs/nau8822.c
+++ b/sound/soc/codecs/nau8822.c
@@ -900,7 +900,7 @@ 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;
@@ -967,10 +967,11 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver nau8822_dai = {
diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c
index 115706a55577..940a2fa933ed 100644
--- a/sound/soc/codecs/rk3328_codec.c
+++ b/sound/soc/codecs/rk3328_codec.c
@@ -107,7 +107,7 @@ 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_stream(struct snd_soc_dai *dai, int mute, int direction)
{
struct rk3328_codec_priv *rk3328 =
snd_soc_component_get_drvdata(dai->component);
@@ -316,9 +316,10 @@ 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_stream,
.startup = rk3328_pcm_startup,
.shutdown = rk3328_pcm_shutdown,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver rk3328_dai[] = {
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index eb08976a7d06..4d6ff8114622 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -775,7 +775,7 @@ 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_stream(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;
@@ -1160,9 +1160,10 @@ 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_stream,
.set_fmt = sgtl5000_set_dai_fmt,
.set_sysclk = sgtl5000_set_dai_sysclk,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver sgtl5000_dai = {
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 2881a0f7bb39..97b5f34027c0 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -251,7 +251,7 @@ 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;
@@ -291,7 +291,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver sta529_dai = {
diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c
index abc114a3ae2b..3d8e8c2276f0 100644
--- a/sound/soc/codecs/tfa9879.c
+++ b/sound/soc/codecs/tfa9879.c
@@ -93,7 +93,7 @@ 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_stream(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
@@ -251,8 +251,9 @@ 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_stream,
.set_fmt = tfa9879_set_fmt,
+ .no_capture_mute = 1,
};
#define TFA9879_RATES SNDRV_PCM_RATE_8000_96000
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index f34637afee51..b37203336c4e 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -997,7 +997,7 @@ 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_stream(struct snd_soc_dai *dai, int mute, int direction)
{
switch (dai->id) {
case TWL6040_DAI_LEGACY:
@@ -1020,7 +1020,8 @@ 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_stream,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver twl6040_dai[] = {
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 1cc7f56912dc..bf9182cedb82 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -117,7 +117,7 @@ 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;
@@ -416,9 +416,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver uda134x_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 09/21] ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (7 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 08/21] ASoC: codecs: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 10/21] ASoC: codecs: tas*: " Kuninori Morimoto
` (12 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/tlv320aic23.c | 5 +++--
sound/soc/codecs/tlv320aic26.c | 5 +++--
sound/soc/codecs/tlv320aic31xx.c | 6 ++++--
sound/soc/codecs/tlv320aic32x4.c | 5 +++--
sound/soc/codecs/tlv320aic3x.c | 5 +++--
5 files changed, 16 insertions(+), 10 deletions(-)
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index d22f75e8fb6a..68a78f0da402 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -404,7 +404,7 @@ 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;
@@ -512,9 +512,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver tlv320aic23_dai = {
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 032b39735643..35a4d6cf936d 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -134,7 +134,7 @@ 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);
@@ -211,9 +211,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver aic26_dai = {
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 31daa60695bd..aee23b1abe6a 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -972,7 +972,8 @@ 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;
@@ -1378,7 +1379,8 @@ 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,
+ .no_capture_mute = 1,
};
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..467802875c13 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -812,7 +812,7 @@ 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;
@@ -866,9 +866,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver aic32x4_dai = {
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 6860743ecdca..6d066bc58ac8 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1216,7 +1216,7 @@ 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;
@@ -1481,10 +1481,11 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver aic3x_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 10/21] ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (8 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 09/21] ASoC: codecs: tlv*: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 11/21] ASoC: codecs: ssm*: " Kuninori Morimoto
` (11 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/tas2552.c | 5 +++--
sound/soc/codecs/tas2562.c | 5 +++--
sound/soc/codecs/tas2770.c | 5 +++--
sound/soc/codecs/tas571x.c | 5 +++--
sound/soc/codecs/tas5720.c | 5 +++--
sound/soc/codecs/tas6424.c | 5 +++--
6 files changed, 18 insertions(+), 12 deletions(-)
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 529c0fb93f9b..3a153526e47f 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -465,7 +465,7 @@ 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;
@@ -519,7 +519,8 @@ 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,
+ .no_capture_mute = 1,
};
/* Formats supported by TAS2552 driver. */
diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c
index 5c28af370bd4..e74628061040 100644
--- a/sound/soc/codecs/tas2562.c
+++ b/sound/soc/codecs/tas2562.c
@@ -394,7 +394,7 @@ 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;
@@ -612,7 +612,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver tas2562_dai[] = {
diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c
index 54c8135fe43c..4538b2d0216f 100644
--- a/sound/soc/codecs/tas2770.c
+++ b/sound/soc/codecs/tas2770.c
@@ -189,7 +189,7 @@ 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;
@@ -530,10 +530,11 @@ 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,
+ .no_capture_mute = 1,
};
#define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 5b7f9fcf6cbf..835a723ce5bc 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -301,7 +301,7 @@ 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;
@@ -354,7 +354,8 @@ 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,
+ .no_capture_mute = 1,
};
diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c
index e159f839d928..139ac5e683bf 100644
--- a/sound/soc/codecs/tas5720.c
+++ b/sound/soc/codecs/tas5720.c
@@ -199,7 +199,7 @@ 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;
@@ -604,7 +604,8 @@ 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,
+ .no_capture_mute = 1,
};
/*
diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c
index aaba39295079..6198138e693a 100644
--- a/sound/soc/codecs/tas6424.c
+++ b/sound/soc/codecs/tas6424.c
@@ -252,7 +252,7 @@ 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);
@@ -382,7 +382,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver tas6424_dai[] = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 11/21] ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (9 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 10/21] ASoC: codecs: tas*: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 12/21] ASoC: codecs: pcm*: " Kuninori Morimoto
` (10 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/ssm2518.c | 5 +++--
sound/soc/codecs/ssm2602.c | 5 +++--
sound/soc/codecs/ssm4567.c | 5 +++--
3 files changed, 9 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index c47e3c4762fe..09449c6c4024 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -388,7 +388,7 @@ 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;
@@ -623,9 +623,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver ssm2518_dai = {
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 464a4d7873bb..905160246614 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -338,7 +338,7 @@ 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);
@@ -505,9 +505,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver ssm2602_dai = {
diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index bb4958bb8fe9..811b1a2c404a 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/sound/soc/codecs/ssm4567.c
@@ -220,7 +220,7 @@ 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;
@@ -390,9 +390,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver ssm4567_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 12/21] ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (10 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 11/21] ASoC: codecs: ssm*: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 13/21] ASoC: codecs: max*: " Kuninori Morimoto
` (9 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/pcm1681.c | 5 +++--
sound/soc/codecs/pcm1789.c | 5 +++--
sound/soc/codecs/pcm179x.c | 5 +++--
sound/soc/codecs/pcm3168a.c | 5 +++--
sound/soc/codecs/pcm512x.c | 5 +++--
5 files changed, 15 insertions(+), 10 deletions(-)
diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c
index 4767e158cd5e..07ed8fded471 100644
--- a/sound/soc/codecs/pcm1681.c
+++ b/sound/soc/codecs/pcm1681.c
@@ -147,7 +147,7 @@ static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
-static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute)
+static int pcm1681_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
struct pcm1681_private *priv = snd_soc_component_get_drvdata(component);
@@ -205,7 +205,8 @@ static int pcm1681_hw_params(struct snd_pcm_substream *substream,
static const struct snd_soc_dai_ops pcm1681_dai_ops = {
.set_fmt = pcm1681_set_dai_fmt,
.hw_params = pcm1681_hw_params,
- .digital_mute = pcm1681_digital_mute,
+ .mute_stream = pcm1681_mute,
+ .no_capture_mute = 1,
};
static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = {
diff --git a/sound/soc/codecs/pcm1789.c b/sound/soc/codecs/pcm1789.c
index 8df6447c76a6..620dec172ce7 100644
--- a/sound/soc/codecs/pcm1789.c
+++ b/sound/soc/codecs/pcm1789.c
@@ -60,7 +60,7 @@ static int pcm1789_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
-static int pcm1789_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+static int pcm1789_mute(struct snd_soc_dai *codec_dai, int mute, int direction)
{
struct snd_soc_component *component = codec_dai->component;
struct pcm1789_private *priv = snd_soc_component_get_drvdata(component);
@@ -167,8 +167,9 @@ static int pcm1789_trigger(struct snd_pcm_substream *substream, int cmd,
static const struct snd_soc_dai_ops pcm1789_dai_ops = {
.set_fmt = pcm1789_set_dai_fmt,
.hw_params = pcm1789_hw_params,
- .digital_mute = pcm1789_digital_mute,
+ .mute_stream = pcm1789_mute,
.trigger = pcm1789_trigger,
+ .no_capture_mute = 1,
};
static const DECLARE_TLV_DB_SCALE(pcm1789_dac_tlv, -12000, 50, 1);
diff --git a/sound/soc/codecs/pcm179x.c b/sound/soc/codecs/pcm179x.c
index 9e70b7385c69..ee60373d7d25 100644
--- a/sound/soc/codecs/pcm179x.c
+++ b/sound/soc/codecs/pcm179x.c
@@ -76,7 +76,7 @@ static int pcm179x_set_dai_fmt(struct snd_soc_dai *codec_dai,
return 0;
}
-static int pcm179x_digital_mute(struct snd_soc_dai *dai, int mute)
+static int pcm179x_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
struct pcm179x_private *priv = snd_soc_component_get_drvdata(component);
@@ -145,7 +145,8 @@ static int pcm179x_hw_params(struct snd_pcm_substream *substream,
static const struct snd_soc_dai_ops pcm179x_dai_ops = {
.set_fmt = pcm179x_set_dai_fmt,
.hw_params = pcm179x_hw_params,
- .digital_mute = pcm179x_digital_mute,
+ .mute_stream = pcm179x_mute,
+ .no_capture_mute = 1,
};
static const DECLARE_TLV_DB_SCALE(pcm179x_dac_tlv, -12000, 50, 1);
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 9711fab296eb..5e445fee4ef5 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -290,7 +290,7 @@ static int pcm3168a_reset(struct pcm3168a_priv *pcm3168a)
PCM3168A_MRST_MASK | PCM3168A_SRST_MASK);
}
-static int pcm3168a_digital_mute(struct snd_soc_dai *dai, int mute)
+static int pcm3168a_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component);
@@ -570,8 +570,9 @@ static const struct snd_soc_dai_ops pcm3168a_dai_ops = {
.set_fmt = pcm3168a_set_dai_fmt,
.set_sysclk = pcm3168a_set_dai_sysclk,
.hw_params = pcm3168a_hw_params,
- .digital_mute = pcm3168a_digital_mute,
+ .mute_stream = pcm3168a_mute,
.set_tdm_slot = pcm3168a_set_tdm_slot,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver pcm3168a_dais[] = {
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 4cbef9affffd..8153d3d01654 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -1394,7 +1394,7 @@ static int pcm512x_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
return 0;
}
-static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute)
+static int pcm512x_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component);
@@ -1445,8 +1445,9 @@ static const struct snd_soc_dai_ops pcm512x_dai_ops = {
.startup = pcm512x_dai_startup,
.hw_params = pcm512x_hw_params,
.set_fmt = pcm512x_set_fmt,
- .digital_mute = pcm512x_digital_mute,
+ .mute_stream = pcm512x_mute,
.set_bclk_ratio = pcm512x_set_bclk_ratio,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver pcm512x_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 13/21] ASoC: codecs: max*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (11 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 12/21] ASoC: codecs: pcm*: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 14/21] ASoC: codecs: alc*: " Kuninori Morimoto
` (8 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/max98088.c | 12 ++++++++----
sound/soc/codecs/max98090.c | 6 ++++--
sound/soc/codecs/max9867.c | 5 +++--
3 files changed, 15 insertions(+), 8 deletions(-)
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 1f1817634a41..4be24e7f51c8 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1274,7 +1274,8 @@ 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;
@@ -1289,7 +1290,8 @@ 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;
@@ -1354,14 +1356,16 @@ 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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver max98088_dai[] = {
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index a61c5638652d..945a79e4f3eb 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2017,7 +2017,8 @@ 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;
@@ -2347,8 +2348,9 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver max98090_dai[] = {
diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c
index c72cb2888c21..fcb31144d69c 100644
--- a/sound/soc/codecs/max9867.c
+++ b/sound/soc/codecs/max9867.c
@@ -283,7 +283,7 @@ 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);
@@ -393,9 +393,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver max9867_dai[] = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 14/21] ASoC: codecs: alc*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (12 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 13/21] ASoC: codecs: max*: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 15/21] ASoC: codecs: wm*: " Kuninori Morimoto
` (7 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/alc5623.c | 5 +++--
sound/soc/codecs/alc5632.c | 5 +++--
2 files changed, 6 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index c70c49bb4a3e..3d1761a531f5 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -737,7 +737,7 @@ 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;
@@ -829,10 +829,11 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver alc5623_dai = {
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index f49543163f69..9d6dcd3ffa57 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -902,7 +902,7 @@ 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
@@ -1005,10 +1005,11 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver alc5632_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 15/21] ASoC: codecs: wm*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (13 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 14/21] ASoC: codecs: alc*: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:56 ` [PATCH v3 16/21] ASoC: codecs: es*: " Kuninori Morimoto
` (6 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
---
sound/soc/codecs/wm8350.c | 5 +++--
sound/soc/codecs/wm8400.c | 5 +++--
sound/soc/codecs/wm8510.c | 5 +++--
sound/soc/codecs/wm8580.c | 5 +++--
sound/soc/codecs/wm8711.c | 5 +++--
sound/soc/codecs/wm8728.c | 5 +++--
sound/soc/codecs/wm8731.c | 5 +++--
sound/soc/codecs/wm8741.c | 5 +++--
sound/soc/codecs/wm8750.c | 5 +++--
sound/soc/codecs/wm8753.c | 8 +++++---
sound/soc/codecs/wm8770.c | 5 +++--
sound/soc/codecs/wm8776.c | 5 +++--
sound/soc/codecs/wm8900.c | 5 +++--
sound/soc/codecs/wm8903.c | 5 +++--
sound/soc/codecs/wm8904.c | 5 +++--
sound/soc/codecs/wm8940.c | 5 +++--
sound/soc/codecs/wm8955.c | 5 +++--
sound/soc/codecs/wm8960.c | 5 +++--
sound/soc/codecs/wm8961.c | 5 +++--
sound/soc/codecs/wm8962.c | 5 +++--
sound/soc/codecs/wm8971.c | 5 +++--
sound/soc/codecs/wm8974.c | 5 +++--
sound/soc/codecs/wm8978.c | 5 +++--
sound/soc/codecs/wm8983.c | 7 ++++---
sound/soc/codecs/wm8985.c | 7 ++++---
sound/soc/codecs/wm8988.c | 5 +++--
sound/soc/codecs/wm8990.c | 5 +++--
sound/soc/codecs/wm8991.c | 7 ++++---
sound/soc/codecs/wm8993.c | 5 +++--
sound/soc/codecs/wm8994.c | 9 ++++++---
sound/soc/codecs/wm8995.c | 8 +++++---
sound/soc/codecs/wm9081.c | 5 +++--
32 files changed, 106 insertions(+), 70 deletions(-)
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 7fe7c1e91882..a6aa212fa0c8 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -942,7 +942,7 @@ 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;
@@ -1426,11 +1426,12 @@ 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,
.set_clkdiv = wm8350_set_clkdiv,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8350_dai = {
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 35ffa7765c85..bf5e77c86aed 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1092,7 +1092,7 @@ 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;
@@ -1222,11 +1222,12 @@ 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,
.set_pll = wm8400_set_dai_pll,
+ .no_capture_mute = 1,
};
/*
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 63a877a8ee2b..73c4a8b9f59e 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -487,7 +487,7 @@ 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;
@@ -547,10 +547,11 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8510_dai = {
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d1fc529d20e7..85ad2f03cfd0 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -800,7 +800,7 @@ 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;
@@ -866,7 +866,8 @@ 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,
+ .no_capture_mute = 1,
};
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..bc4d161c59e5 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -204,7 +204,7 @@ 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;
@@ -329,9 +329,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8711_dai = {
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index bb5521f544ba..2cd58d133899 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -69,7 +69,7 @@ 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);
@@ -192,8 +192,9 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8728_dai = {
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index cae2cc38d93c..304bf725a613 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -366,7 +366,7 @@ 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;
@@ -546,9 +546,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8731_dai = {
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 328df81ee839..0e3994326936 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -364,7 +364,7 @@ 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;
@@ -386,7 +386,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8741_dai = {
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 970941f8ae81..9491817020d8 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -606,7 +606,7 @@ 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;
@@ -660,9 +660,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8750_dai = {
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a1b6765c8f23..ae9d85622006 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1295,7 +1295,7 @@ 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;
@@ -1380,20 +1380,22 @@ 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,
.set_sysclk = wm8753_set_dai_sysclk,
+ .no_capture_mute = 1,
};
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,
.set_sysclk = wm8753_set_dai_sysclk,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8753_dai[] = {
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index d51be2531e2e..1176a6ad269d 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -472,7 +472,7 @@ 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;
@@ -538,10 +538,11 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8770_dai = {
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index f174d7ce2b13..554acf56130c 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -309,7 +309,7 @@ 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;
@@ -361,10 +361,11 @@ 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,
+ .no_capture_mute = 1,
};
static const struct snd_soc_dai_ops wm8776_adc_ops = {
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 3921af63adf2..a9a6d766a176 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -967,7 +967,7 @@ 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;
@@ -997,7 +997,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8900_dai = {
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 5de663d61ba6..ac373fb1658f 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1307,7 +1307,7 @@ 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;
@@ -1737,9 +1737,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8903_dai = {
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 3f0e49c51fd5..6bfdeda858e4 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1846,7 +1846,7 @@ 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;
@@ -1962,7 +1962,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8904_dai = {
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 41d87e172775..016cd8aeef37 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -452,7 +452,7 @@ 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;
@@ -664,10 +664,11 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8940_dai = {
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 73c192f58382..28c1d4515dd5 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -745,7 +745,7 @@ 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;
@@ -848,7 +848,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8955_dai = {
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 9dca6e28032a..e6f04fc01a50 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -878,7 +878,7 @@ 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;
@@ -1315,11 +1315,12 @@ 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,
.set_sysclk = wm8960_set_dai_sysclk,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8960_dai = {
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index d11a38a0b283..9e31ba4173a4 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -698,7 +698,7 @@ 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);
@@ -806,9 +806,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8961_dai = {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 6ef022295f55..174a288fa9bf 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2918,7 +2918,7 @@ 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;
@@ -2951,7 +2951,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8962_dai = {
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 2cefb795da03..21ae55c32a6d 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -536,7 +536,7 @@ 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;
@@ -602,9 +602,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8971_dai = {
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 953d12e4576f..89c6507d5566 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -559,7 +559,7 @@ 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;
@@ -616,11 +616,12 @@ 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,
.set_sysclk = wm8974_set_dai_sysclk,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8974_dai = {
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index e01ba5394527..a7acb8981715 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -836,7 +836,7 @@ 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;
@@ -893,10 +893,11 @@ 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,
+ .no_capture_mute = 1,
};
/* Also supports 12kHz */
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index 78e1a302c54c..d1d2d408ad95 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -557,7 +557,7 @@ 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;
@@ -943,11 +943,12 @@ 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,
- .set_pll = wm8983_set_pll
+ .set_pll = wm8983_set_pll,
+ .no_capture_mute = 1,
};
#define WM8983_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index 62f2c603eb2d..3f27482349b2 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -649,7 +649,7 @@ 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;
@@ -1072,11 +1072,12 @@ 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,
- .set_pll = wm8985_set_pll
+ .set_pll = wm8985_set_pll,
+ .no_capture_mute = 1,
};
#define WM8985_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 646cfd8b2693..d2c2d0d943f0 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -707,7 +707,7 @@ 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;
@@ -766,7 +766,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8988_dai = {
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 13bca8ebf677..938940777e5d 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -998,7 +998,7 @@ 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;
@@ -1152,11 +1152,12 @@ 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,
.set_sysclk = wm8990_set_dai_sysclk,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8990_dai = {
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index ba71c2f59511..16bc8609d0d2 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1064,7 +1064,7 @@ 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;
@@ -1196,10 +1196,11 @@ 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
+ .set_pll = wm8991_set_dai_pll,
+ .no_capture_mute = 1,
};
/*
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 207c0211caa9..447e6b04b6fb 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1330,7 +1330,7 @@ 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;
@@ -1444,9 +1444,10 @@ 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,
+ .no_capture_mute = 1,
};
#define WM8993_RATES SNDRV_PCM_RATE_8000_48000
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 75242ec47406..42f81b4e1e79 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3110,7 +3110,8 @@ 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;
@@ -3187,18 +3188,20 @@ 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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = {
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 276ffa84cc31..67e4f6b3a860 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1417,7 +1417,7 @@ 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;
@@ -2094,18 +2094,20 @@ 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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
static const struct snd_soc_dai_ops wm8995_aif3_dai_ops = {
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index be5c9c2b0162..27ee343925da 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1147,7 +1147,7 @@ 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;
@@ -1232,8 +1232,9 @@ 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,
+ .no_capture_mute = 1,
};
/* We report two channels because the CODEC processes a stereo signal, even
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 16/21] ASoC: codecs: es*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (14 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 15/21] ASoC: codecs: wm*: " Kuninori Morimoto
@ 2020-07-09 1:56 ` Kuninori Morimoto
2020-07-09 1:57 ` [PATCH v3 17/21] ASoC: codecs: da*: " Kuninori Morimoto
` (5 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:56 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/es8316.c | 5 +++--
sound/soc/codecs/es8328.c | 5 +++--
2 files changed, 6 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 36eef1fb3d18..c474812182be 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -507,7 +507,7 @@ 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)
{
snd_soc_component_update_bits(dai->component, ES8316_DAC_SET1, 0x20,
mute ? 0x20 : 0);
@@ -522,7 +522,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver es8316_dai = {
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index fdf64c29f563..1acb3dc412bc 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -449,7 +449,7 @@ 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)
{
return snd_soc_component_update_bits(dai->component, ES8328_DACCONTROL3,
ES8328_DACCONTROL3_DACMUTE,
@@ -692,9 +692,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver es8328_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 17/21] ASoC: codecs: da*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (15 preceding siblings ...)
2020-07-09 1:56 ` [PATCH v3 16/21] ASoC: codecs: es*: " Kuninori Morimoto
@ 2020-07-09 1:57 ` Kuninori Morimoto
2020-07-09 1:57 ` [PATCH v3 18/21] ASoC: codecs: cs*: " Kuninori Morimoto
` (4 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:57 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
---
sound/soc/codecs/da7210.c | 5 +++--
sound/soc/codecs/da7213.c | 5 +++--
sound/soc/codecs/da9055.c | 5 +++--
3 files changed, 9 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 0c99dcf242e4..46d1c4d38cce 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -924,7 +924,7 @@ 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;
@@ -1034,7 +1034,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver da7210_dai = {
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index fe93ec702645..72402467adcc 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1332,7 +1332,7 @@ 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;
@@ -1528,7 +1528,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver da7213_dai = {
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index e93436ccb674..b0d9ca6de685 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1211,7 +1211,7 @@ 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;
@@ -1324,7 +1324,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver da9055_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 18/21] ASoC: codecs: cs*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (16 preceding siblings ...)
2020-07-09 1:57 ` [PATCH v3 17/21] ASoC: codecs: da*: " Kuninori Morimoto
@ 2020-07-09 1:57 ` Kuninori Morimoto
2020-07-09 1:57 ` [PATCH v3 19/21] ASoC: codecs: ak*: " Kuninori Morimoto
` (3 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:57 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/cs4265.c | 5 +++--
sound/soc/codecs/cs4270.c | 5 +++--
sound/soc/codecs/cs42l42.c | 5 +++--
sound/soc/codecs/cs42l51.c | 5 +++--
sound/soc/codecs/cs42l52.c | 5 +++--
sound/soc/codecs/cs42l56.c | 5 +++--
sound/soc/codecs/cs42xx8.c | 5 +++--
sound/soc/codecs/cs4341.c | 5 +++--
sound/soc/codecs/cs4349.c | 5 +++--
9 files changed, 27 insertions(+), 18 deletions(-)
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 2fb65f246b0c..d76be44f46b4 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -378,7 +378,7 @@ 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;
@@ -498,9 +498,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs4265_dai[] = {
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index bd17c806d543..ddd95c8269ed 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -406,7 +406,7 @@ 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);
@@ -471,7 +471,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs4270_dai = {
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index d391b5074904..210fcbedf241 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -849,7 +849,7 @@ 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;
@@ -909,7 +909,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs42l42_dai = {
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index dde9812490de..764f2ef8f59d 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -484,7 +484,7 @@ 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;
@@ -511,7 +511,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs42l51_dai = {
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 2ea4cba3be2a..f772628f233e 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -784,7 +784,7 @@ 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;
@@ -865,9 +865,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs42l52_dai = {
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index ac569ab3d30f..97024a6ac96d 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -800,7 +800,7 @@ 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;
@@ -929,9 +929,10 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs42l56_dai = {
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index 94b1adb088fd..5d6ef660f851 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -362,7 +362,7 @@ 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);
@@ -380,7 +380,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs42xx8_dai = {
diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c
index ade7477d04f1..f566604de78c 100644
--- a/sound/soc/codecs/cs4341.c
+++ b/sound/soc/codecs/cs4341.c
@@ -116,7 +116,7 @@ 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;
@@ -174,7 +174,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs4341_dai = {
diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c
index 3381209a882d..fd5526319779 100644
--- a/sound/soc/codecs/cs4349.c
+++ b/sound/soc/codecs/cs4349.c
@@ -131,7 +131,7 @@ 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;
@@ -236,7 +236,8 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver cs4349_dai = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 19/21] ASoC: codecs: ak*: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (17 preceding siblings ...)
2020-07-09 1:57 ` [PATCH v3 18/21] ASoC: codecs: cs*: " Kuninori Morimoto
@ 2020-07-09 1:57 ` Kuninori Morimoto
2020-07-09 1:57 ` [PATCH v3 20/21] ASoC: soc-dai: remove .digital_mute Kuninori Morimoto
` (2 subsequent siblings)
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:57 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
---
sound/soc/codecs/ak4458.c | 5 +++--
sound/soc/codecs/ak4535.c | 6 ++++--
sound/soc/codecs/ak4641.c | 8 +++++---
3 files changed, 12 insertions(+), 7 deletions(-)
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index f180cb5dfe4f..0c51ba8f6072 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -401,7 +401,7 @@ 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);
@@ -495,8 +495,9 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver ak4458_dai = {
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index f5ad1f59eb46..91e7a57c43da 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -309,10 +309,11 @@ 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 (!mute)
snd_soc_component_write(component, AK4535_DAC, mute_reg & ~0x20);
else
@@ -348,8 +349,9 @@ 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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver ak4535_dai = {
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 2d5b640aab58..77004cd7caa3 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -405,7 +405,7 @@ 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;
@@ -467,15 +467,17 @@ 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,
+ .no_capture_mute = 1,
};
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,
+ .no_capture_mute = 1,
};
static struct snd_soc_dai_driver ak4641_dai[] = {
--
2.25.1
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 20/21] ASoC: soc-dai: remove .digital_mute
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (18 preceding siblings ...)
2020-07-09 1:57 ` [PATCH v3 19/21] ASoC: codecs: ak*: " Kuninori Morimoto
@ 2020-07-09 1:57 ` Kuninori Morimoto
2020-07-09 1:57 ` [PATCH v3 21/21] ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec Kuninori Morimoto
2020-07-16 23:56 ` [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Mark Brown
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:57 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
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>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.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 e0e061b8e9bd..2c5a49411276 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 458d2ea44329..c89a1929d141 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -307,10 +307,6 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
(direction == SNDRV_PCM_STREAM_PLAYBACK ||
!dai->driver->ops->no_capture_mute))
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
^ permalink raw reply related [flat|nested] 26+ messages in thread* [PATCH v3 21/21] ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (19 preceding siblings ...)
2020-07-09 1:57 ` [PATCH v3 20/21] ASoC: soc-dai: remove .digital_mute Kuninori Morimoto
@ 2020-07-09 1:57 ` Kuninori Morimoto
2020-07-16 23:56 ` [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Mark Brown
21 siblings, 0 replies; 26+ messages in thread
From: Kuninori Morimoto @ 2020-07-09 1:57 UTC (permalink / raw)
To: Mark Brown; +Cc: Linux-ALSA
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>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.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 7c58e45c1c3f..defd96b14c28 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
^ permalink raw reply related [flat|nested] 26+ messages in thread* Re: [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream()
2020-07-09 1:55 [PATCH v3 00/21] ASoC: merge .digital_mute() into .mute_stream() Kuninori Morimoto
` (20 preceding siblings ...)
2020-07-09 1:57 ` [PATCH v3 21/21] ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec Kuninori Morimoto
@ 2020-07-16 23:56 ` Mark Brown
21 siblings, 0 replies; 26+ messages in thread
From: Mark Brown @ 2020-07-16 23:56 UTC (permalink / raw)
To: Kuninori Morimoto; +Cc: Linux-ALSA
On 09 Jul 2020 10:55:05 +0900, Kuninori Morimoto wrote:
> These are v3 patch-set.
> ALSA SoC has 2 mute callbacks (= .digital_mute(), .mute_stream()).
> But the difference between these 2 are very small.
> .digital_mute() is for Playback
> .mute_stream() is for Playback/Capture
>
> This patch-set adds new .no_capture_mute flag and emulate
> .digital_mute() by .mute_stream().
>
> [...]
Applied to
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
Thanks!
[01/21] ASoC: hdmi-codec: return -ENOTSUPP for digital_mute
commit: e07e49c0d1e3693facf588142c4cbde45904b3f8
[02/21] ASoC: soc-dai.c: add .no_capture_mute support
commit: 350d993510115e3d9e78f1b3359bff7b68e88418
[03/21] ASoC: hdmi-codec: merge .digital_mute() into .mute_stream()
commit: d789710fb248df0c2279a785c7b9beb313629c0a
[04/21] ASoC: ti: merge .digital_mute() into .mute_stream()
commit: 059374fe9ea5dd1e5a1d6b14741baf994dbc8dd2
[05/21] ASoC: spear: merge .digital_mute() into .mute_stream()
commit: 1ff123965bd00f1aacbd4549b03a6a5b98432eaf
[06/21] ASoC: meson: merge .digital_mute() into .mute_stream()
commit: f795ff78e266492a5278cbef89d3422bdcf591b0
[07/21] ASoC: atmel: merge .digital_mute() into .mute_stream()
(no commit info)
[08/21] ASoC: codecs: merge .digital_mute() into .mute_stream()
commit: 54b5927067dd79b58032a4938ee13b233f1a368c
[09/21] ASoC: codecs: tlv*: merge .digital_mute() into .mute_stream()
commit: 960af79d31495b81bf9406f2f1fda425cb49342b
[10/21] ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()
commit: 38803ce7b53bd7588e8ad899a73fe21e8741723b
[11/21] ASoC: codecs: ssm*: merge .digital_mute() into .mute_stream()
commit: bd63ed76890f38246a42cee715a437774bedf70b
[12/21] ASoC: codecs: pcm*: merge .digital_mute() into .mute_stream()
commit: 1eb2c43d26a585bbbe196360e921c5ffe9d00fd1
[13/21] ASoC: codecs: max*: merge .digital_mute() into .mute_stream()
commit: 18e028e22ebfe362656f534c540d7b3c6c094bd0
[14/21] ASoC: codecs: alc*: merge .digital_mute() into .mute_stream()
commit: 4c66c2fc106f04899e778eec5f7a33a6fef4471e
[15/21] ASoC: codecs: wm*: merge .digital_mute() into .mute_stream()
commit: 26d3c16e62e9f07b99d3886bec05effe5aadcbe5
[16/21] ASoC: codecs: es*: merge .digital_mute() into .mute_stream()
commit: 8667d948a6185fcde88ebc89e090fa574a44c366
[17/21] ASoC: codecs: da*: merge .digital_mute() into .mute_stream()
commit: f39c0540d6941b2390cea20f413b620adcc3be86
[18/21] ASoC: codecs: cs*: merge .digital_mute() into .mute_stream()
commit: 03c0f1b5e1e01afdff8bca8ea6c599ff9a349ed6
[19/21] ASoC: codecs: ak*: merge .digital_mute() into .mute_stream()
commit: 50891431aaadf784980f8649d676327928bce72f
[20/21] ASoC: soc-dai: remove .digital_mute
(no commit info)
[21/21] ASoC: soc-core: snd_soc_dai_digital_mute() for both CPU/Codec
(no commit info)
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
^ permalink raw reply [flat|nested] 26+ messages in thread