[alsa-devel] [PATCH v2 00/10] ASoC: soc-pcm cleanup step3

[alsa-devel] [PATCH v2 00/10] ASoC: soc-pcm cleanup step3

[Kuninori Morimoto]

Hi Mark

These are v2 of soc-pcm cleanup step3.
I removed unneeded Multi-Codec patch.

And re-ordered / merged my local tree and add [09/10] patch
to make [10/10] more effective.

Kuninori Morimoto (10):
   1) ASoC: soc-pcm: move dai_get_widget()
   2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()
   3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()
   4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()
   5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute()
   6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state()
   7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code
   8) ASoC: soc-pcm: use goto and remove multi return
   9) ASoC: soc-pcm: merge playback/cature_active into stream_active
  10) ASoC: soc.h: add for_each_pcm_streams()

 include/sound/pcm.h                   |   5 +
 include/sound/soc-dai.h               |   3 +-
 include/sound/soc-dpcm.h              |   9 -
 sound/soc/codecs/cs4271.c             |   4 +-
 sound/soc/dwc/dwc-i2s.c               |   8 +-
 sound/soc/fsl/fsl_asrc_dma.c          |   4 +-
 sound/soc/qcom/lpass-platform.c       |   2 +-
 sound/soc/soc-core.c                  |  48 ++--
 sound/soc/soc-generic-dmaengine-pcm.c |   8 +-
 sound/soc/soc-pcm.c                   | 327 +++++++++-----------------
 sound/soc/sof/sof-audio.c             |  19 +-
 sound/usb/usx2y/usbusx2yaudio.c       |   9 +-
 12 files changed, 162 insertions(+), 284 deletions(-)

-- 
2.17.1

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[alsa-devel] [PATCH v2 01/10] ASoC: soc-pcm: move dai_get_widget()

[Kuninori Morimoto]

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

This patch moves dai_get_widget() to top side.
This is prepare for cleanup soc-pcm.c

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- no change

 sound/soc/soc-pcm.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 6630fadd6e09..a8adc8d06c05 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -82,6 +82,15 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
+static inline
+struct snd_soc_dapm_widget *dai_get_widget(struct snd_soc_dai *dai, int stream)
+{
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return dai->playback_widget;
+	else
+		return dai->capture_widget;
+}
+
 static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd,
 				   int stream, int action)
 {
@@ -1287,15 +1296,6 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 	return NULL;
 }
 
-static inline struct snd_soc_dapm_widget *
-	dai_get_widget(struct snd_soc_dai *dai, int stream)
-{
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
-		return dai->playback_widget;
-	else
-		return dai->capture_widget;
-}
-
 static int widget_in_list(struct snd_soc_dapm_widget_list *list,
 		struct snd_soc_dapm_widget *widget)
 {
-- 
2.17.1

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[alsa-devel] [PATCH v2 02/10] ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()

[Kuninori Morimoto]

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

dpcm_get_be() has very duplicate code.

	dpcm_get_be()
	{
		...
		if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
(1)			/* code for Playback */
		} else  {
(2)			/* code for Capture */
		}
	}

The difference between Playback (1) and Capture (2) code is pointer only
(= "playback_widget" or "caputre_widget").
OTOH, now we already has dai_get_widget() for it.
This means we can merge (1) and (2).
This patch do it and remove duplicated code.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- no change

 sound/soc/soc-pcm.c | 41 ++++++++++++-----------------------------
 1 file changed, 12 insertions(+), 29 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index a8adc8d06c05..e60dfbcf52b4 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1246,47 +1246,30 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card,
 		struct snd_soc_dapm_widget *widget, int stream)
 {
 	struct snd_soc_pcm_runtime *be;
+	struct snd_soc_dapm_widget *w;
 	struct snd_soc_dai *dai;
 	int i;
 
 	dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name);
 
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		for_each_card_rtds(card, be) {
+	for_each_card_rtds(card, be) {
 
-			if (!be->dai_link->no_pcm)
-				continue;
+		if (!be->dai_link->no_pcm)
+			continue;
 
-			dev_dbg(card->dev, "ASoC: try BE : %s\n",
-				be->cpu_dai->playback_widget ?
-				be->cpu_dai->playback_widget->name : "(not set)");
+		w = dai_get_widget(be->cpu_dai, stream);
 
-			if (be->cpu_dai->playback_widget == widget)
-				return be;
+		dev_dbg(card->dev, "ASoC: try BE : %s\n",
+			w ? w->name : "(not set)");
 
-			for_each_rtd_codec_dai(be, i, dai) {
-				if (dai->playback_widget == widget)
-					return be;
-			}
-		}
-	} else {
-
-		for_each_card_rtds(card, be) {
-
-			if (!be->dai_link->no_pcm)
-				continue;
+		if (w == widget)
+			return be;
 
-			dev_dbg(card->dev, "ASoC: try BE %s\n",
-				be->cpu_dai->capture_widget ?
-				be->cpu_dai->capture_widget->name : "(not set)");
+		for_each_rtd_codec_dai(be, i, dai) {
+			w = dai_get_widget(dai, stream);
 
-			if (be->cpu_dai->capture_widget == widget)
+			if (w == widget)
 				return be;
-
-			for_each_rtd_codec_dai(be, i, dai) {
-				if (dai->capture_widget == widget)
-					return be;
-			}
 		}
 	}
 
-- 
2.17.1

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[alsa-devel] [PATCH v2 03/10] ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()

[Kuninori Morimoto]

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

dpcm_end_walk_at_be() has very duplicate code.

	dpcm_end_walk_at_be()
	{
		...
		if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
(1)			/* code for Playback */
		} else  {
(2)			/* code for Capture */
		}
	}

The difference between Playback (1) and Capture (2) code is pointer only
(= "playback_widget" or "caputre_widget").
OTOH, now we already has dai_get_widget() for it.
This means we can merge (1) and (2).
This patch do it and remove duplicated code.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- no change

 sound/soc/soc-pcm.c | 37 ++++++++++++++++---------------------
 1 file changed, 16 insertions(+), 21 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e60dfbcf52b4..62ca275f02b0 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1297,34 +1297,29 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
 {
 	struct snd_soc_card *card = widget->dapm->card;
 	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dapm_widget *w;
 	struct snd_soc_dai *dai;
+	int stream;
 	int i;
 
-	if (dir == SND_SOC_DAPM_DIR_OUT) {
-		for_each_card_rtds(card, rtd) {
-			if (!rtd->dai_link->no_pcm)
-				continue;
+	/* adjust dir to stream */
+	if (dir == SND_SOC_DAPM_DIR_OUT)
+		stream = SNDRV_PCM_STREAM_PLAYBACK;
+	else
+		stream = SNDRV_PCM_STREAM_CAPTURE;
 
-			if (rtd->cpu_dai->playback_widget == widget)
-				return true;
+	for_each_card_rtds(card, rtd) {
+		if (!rtd->dai_link->no_pcm)
+			continue;
 
-			for_each_rtd_codec_dai(rtd, i, dai) {
-				if (dai->playback_widget == widget)
-					return true;
-			}
-		}
-	} else { /* SND_SOC_DAPM_DIR_IN */
-		for_each_card_rtds(card, rtd) {
-			if (!rtd->dai_link->no_pcm)
-				continue;
+		w = dai_get_widget(rtd->cpu_dai, stream);
+		if (w == widget)
+			return true;
 
-			if (rtd->cpu_dai->capture_widget == widget)
+		for_each_rtd_codec_dai(rtd, i, dai) {
+			w = dai_get_widget(dai, stream);
+			if (w == widget)
 				return true;
-
-			for_each_rtd_codec_dai(rtd, i, dai) {
-				if (dai->capture_widget == widget)
-					return true;
-			}
 		}
 	}
 
-- 
2.17.1

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[alsa-devel] [PATCH v2 04/10] ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()

[Kuninori Morimoto]

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

dpcm_end_walk_at_be() and dpcm_get_be() are almost same code.
This patch uses dpcm_get_be() from dpcm_end_walk_at_be().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- no change

 sound/soc/soc-pcm.c | 20 +++-----------------
 1 file changed, 3 insertions(+), 17 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 62ca275f02b0..18fbf27aad01 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1297,10 +1297,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
 {
 	struct snd_soc_card *card = widget->dapm->card;
 	struct snd_soc_pcm_runtime *rtd;
-	struct snd_soc_dapm_widget *w;
-	struct snd_soc_dai *dai;
 	int stream;
-	int i;
 
 	/* adjust dir to stream */
 	if (dir == SND_SOC_DAPM_DIR_OUT)
@@ -1308,20 +1305,9 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget,
 	else
 		stream = SNDRV_PCM_STREAM_CAPTURE;
 
-	for_each_card_rtds(card, rtd) {
-		if (!rtd->dai_link->no_pcm)
-			continue;
-
-		w = dai_get_widget(rtd->cpu_dai, stream);
-		if (w == widget)
-			return true;
-
-		for_each_rtd_codec_dai(rtd, i, dai) {
-			w = dai_get_widget(dai, stream);
-			if (w == widget)
-				return true;
-		}
-	}
+	rtd = dpcm_get_be(card, widget, stream);
+	if (rtd)
+		return true;
 
 	return false;
 }
-- 
2.17.1

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[alsa-devel] [PATCH v2 05/10] ASoC: soc-pcm: remove soc_dpcm_be_digital_mute()

[Kuninori Morimoto]

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

No one is using soc_dpcm_be_digital_mute().
If it exists only by assumption that "it may be necessary someday",
let's remove it now. Otherwise code maintenance will be difficult.
We can revive it when we really needed it.
Let's remove it, so far.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- no change

 include/sound/soc-dpcm.h |  1 -
 sound/soc/soc-pcm.c      | 27 ---------------------------
 2 files changed, 28 deletions(-)

diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index b654ebfc8766..665516387671 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -141,7 +141,6 @@ void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
 	enum snd_soc_dpcm_state state);
 
 /* internal use only */
-int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
 int soc_dpcm_runtime_update(struct snd_soc_card *);
 
 #ifdef CONFIG_DEBUG_FS
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 18fbf27aad01..aafa97bbcd1d 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2719,33 +2719,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
 	mutex_unlock(&card->mutex);
 	return ret;
 }
-int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
-{
-	struct snd_soc_dpcm *dpcm;
-	struct snd_soc_dai *dai;
-
-	for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
-
-		struct snd_soc_pcm_runtime *be = dpcm->be;
-		int i;
-
-		if (be->dai_link->ignore_suspend)
-			continue;
-
-		for_each_rtd_codec_dai(be, i, dai) {
-			struct snd_soc_dai_driver *drv = dai->driver;
-
-			dev_dbg(be->dev, "ASoC: BE digital mute %s\n",
-					 be->dai_link->name);
-
-			if (drv->ops && drv->ops->digital_mute &&
-							dai->playback_active)
-				drv->ops->digital_mute(dai, mute);
-		}
-	}
-
-	return 0;
-}
 
 static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
 {
-- 
2.17.1

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[alsa-devel] [PATCH v2 06/10] ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state()

[Kuninori Morimoto]

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

No one is using snd_soc_dpcm_be_get/set_state().
If it exists only by assumption that "it may be necessary someday",
let's remove it now. Otherwise code maintenance will be difficult.
We can revive it when we really needed it.
Let's remove it, so far.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- no change

 include/sound/soc-dpcm.h |  8 --------
 sound/soc/soc-pcm.c      | 16 ----------------
 2 files changed, 24 deletions(-)

diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
index 665516387671..3e7819d2a6aa 100644
--- a/include/sound/soc-dpcm.h
+++ b/include/sound/soc-dpcm.h
@@ -132,14 +132,6 @@ int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,
 struct snd_pcm_substream *
 	snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream);
 
-/* get the BE runtime state */
-enum snd_soc_dpcm_state
-	snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream);
-
-/* set the BE runtime state */
-void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
-	enum snd_soc_dpcm_state state);
-
 /* internal use only */
 int soc_dpcm_runtime_update(struct snd_soc_card *);
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index aafa97bbcd1d..acc8a7eabfbe 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2957,22 +2957,6 @@ struct snd_pcm_substream *
 }
 EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream);
 
-/* get the BE runtime state */
-enum snd_soc_dpcm_state
-	snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream)
-{
-	return be->dpcm[stream].state;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state);
-
-/* set the BE runtime state */
-void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be,
-		int stream, enum snd_soc_dpcm_state state)
-{
-	be->dpcm[stream].state = state;
-}
-EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state);
-
 /*
  * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
  * are not running, paused or suspended for the specified stream direction.
-- 
2.17.1

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[alsa-devel] [PATCH v2 07/10] ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code

[Kuninori Morimoto]

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

Below functions are doing very similar things, the difference is
used state only.

	snd_soc_dpcm_can_be_free_stop()
	snd_soc_dpcm_can_be_params()

This patch adds common snd_soc_dpcm_check_state(), and use it from
snd_soc_dpcm_can_be_free_stop() / snd_soc_dpcm_can_be_params().
It can reduce duplicate code.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- rename function name to snd_soc_dpcm_check_state()
	- use const for state

 sound/soc/soc-pcm.c | 70 ++++++++++++++++++++++-----------------------
 1 file changed, 35 insertions(+), 35 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index acc8a7eabfbe..fcebc327c0ef 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2957,17 +2957,17 @@ struct snd_pcm_substream *
 }
 EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream);
 
-/*
- * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
- * are not running, paused or suspended for the specified stream direction.
- */
-int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
-		struct snd_soc_pcm_runtime *be, int stream)
+static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe,
+				    struct snd_soc_pcm_runtime *be,
+				    int stream,
+				    const enum snd_soc_dpcm_state *states,
+				    int num_states)
 {
 	struct snd_soc_dpcm *dpcm;
 	int state;
 	int ret = 1;
 	unsigned long flags;
+	int i;
 
 	spin_lock_irqsave(&fe->card->dpcm_lock, flags);
 	for_each_dpcm_fe(be, stream, dpcm) {
@@ -2976,18 +2976,34 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
 			continue;
 
 		state = dpcm->fe->dpcm[stream].state;
-		if (state == SND_SOC_DPCM_STATE_START ||
-			state == SND_SOC_DPCM_STATE_PAUSED ||
-			state == SND_SOC_DPCM_STATE_SUSPEND) {
-			ret = 0;
-			break;
+		for (i = 0; i < num_states; i++) {
+			if (state == states[i]) {
+				ret = 0;
+				break;
+			}
 		}
 	}
 	spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
 
-	/* it's safe to free/stop this BE DAI */
+	/* it's safe to do this BE DAI */
 	return ret;
 }
+
+/*
+ * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE
+ * are not running, paused or suspended for the specified stream direction.
+ */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream)
+{
+	const enum snd_soc_dpcm_state state[] = {
+		SND_SOC_DPCM_STATE_START,
+		SND_SOC_DPCM_STATE_PAUSED,
+		SND_SOC_DPCM_STATE_SUSPEND,
+	};
+
+	return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
+}
 EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
 
 /*
@@ -2997,30 +3013,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop);
 int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
 		struct snd_soc_pcm_runtime *be, int stream)
 {
-	struct snd_soc_dpcm *dpcm;
-	int state;
-	int ret = 1;
-	unsigned long flags;
-
-	spin_lock_irqsave(&fe->card->dpcm_lock, flags);
-	for_each_dpcm_fe(be, stream, dpcm) {
-
-		if (dpcm->fe == fe)
-			continue;
+	const enum snd_soc_dpcm_state state[] = {
+		SND_SOC_DPCM_STATE_START,
+		SND_SOC_DPCM_STATE_PAUSED,
+		SND_SOC_DPCM_STATE_SUSPEND,
+		SND_SOC_DPCM_STATE_PREPARE,
+	};
 
-		state = dpcm->fe->dpcm[stream].state;
-		if (state == SND_SOC_DPCM_STATE_START ||
-			state == SND_SOC_DPCM_STATE_PAUSED ||
-			state == SND_SOC_DPCM_STATE_SUSPEND ||
-			state == SND_SOC_DPCM_STATE_PREPARE) {
-			ret = 0;
-			break;
-		}
-	}
-	spin_unlock_irqrestore(&fe->card->dpcm_lock, flags);
-
-	/* it's safe to change hw_params */
-	return ret;
+	return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state));
 }
 EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params);
 
-- 
2.17.1

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[alsa-devel] [PATCH v2 08/10] ASoC: soc-pcm: use goto and remove multi return

[Kuninori Morimoto]

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

When we use some kind of lock, we need to do unlock.
In that time, multi unlock/return is not good implementation.
This patch add label and use goto at dpcm_fe_dai_open()
to reduce such code.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- no change

 sound/soc/soc-pcm.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fcebc327c0ef..bf1f74c42e1a 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2733,8 +2733,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
 
 	ret = dpcm_path_get(fe, stream, &list);
 	if (ret < 0) {
-		mutex_unlock(&fe->card->mutex);
-		return ret;
+		goto open_end;
 	} else if (ret == 0) {
 		dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
 			fe->dai_link->name, stream ? "capture" : "playback");
@@ -2755,6 +2754,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
 
 	dpcm_clear_pending_state(fe, stream);
 	dpcm_path_put(&list);
+open_end:
 	mutex_unlock(&fe->card->mutex);
 	return ret;
 }
-- 
2.17.1

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[alsa-devel] [PATCH v2 09/10] ASoC: soc-pcm: merge playback/cature_active into stream_active

[Kuninori Morimoto]

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

DAI has playback_active and capture_active to care usage count.
OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE.
But because of this kind of implementation mismatch,
ALSA SoC has many verbose code.

To solve this issue, this patch merge playback_active/capture_active
into stream_active[2];

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- new patch

 include/sound/soc-dai.h   |  3 +--
 sound/soc/codecs/cs4271.c |  4 ++--
 sound/soc/dwc/dwc-i2s.c   |  4 ++--
 sound/soc/soc-core.c      | 17 +++++++++--------
 sound/soc/soc-pcm.c       | 25 ++++++++++++-------------
 5 files changed, 26 insertions(+), 27 deletions(-)

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 04c23ac0dfff..7481e468be39 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -322,8 +322,7 @@ struct snd_soc_dai {
 	struct snd_soc_dai_driver *driver;
 
 	/* DAI runtime info */
-	unsigned int capture_active;		/* stream usage count */
-	unsigned int playback_active;		/* stream usage count */
+	unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */
 
 	unsigned int active;
 
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 04b86a51e055..62f412d6f9f2 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -356,9 +356,9 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
 		 */
 
 		if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
-		     !dai->capture_active) ||
+		     !dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) ||
 		    (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
-		     !dai->playback_active)) {
+		     !dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])) {
 			ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
 						 CS4271_MODE2_PDN,
 						 CS4271_MODE2_PDN);
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index 7eeca2150b2d..a8bff6f08a69 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -427,9 +427,9 @@ static int dw_i2s_resume(struct snd_soc_component *component)
 		clk_enable(dev->clk);
 
 	for_each_component_dais(component, dai) {
-		if (dai->playback_active)
+		if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
 			dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK);
-		if (dai->capture_active)
+		if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
 			dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE);
 	}
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 068d809c349a..13df3175e184 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -365,19 +365,20 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime);
 void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int playback = SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
 
 	dev_dbg(rtd->dev,
 		"ASoC: pop wq checking: %s status: %s waiting: %s\n",
 		codec_dai->driver->playback.stream_name,
-		codec_dai->playback_active ? "active" : "inactive",
+		codec_dai->stream_active[playback] ? "active" : "inactive",
 		rtd->pop_wait ? "yes" : "no");
 
 	/* are we waiting on this codec DAI stream */
 	if (rtd->pop_wait == 1) {
 		rtd->pop_wait = 0;
-		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+		snd_soc_dapm_stream_event(rtd, playback,
 					  SND_SOC_DAPM_STREAM_STOP);
 	}
 
@@ -514,6 +515,7 @@ int snd_soc_suspend(struct device *dev)
 	struct snd_soc_card *card = dev_get_drvdata(dev);
 	struct snd_soc_component *component;
 	struct snd_soc_pcm_runtime *rtd;
+	int playback = SNDRV_PCM_STREAM_PLAYBACK;
 	int i;
 
 	/* If the card is not initialized yet there is nothing to do */
@@ -537,9 +539,8 @@ int snd_soc_suspend(struct device *dev)
 			continue;
 
 		for_each_rtd_codec_dai(rtd, i, dai) {
-			if (dai->playback_active)
-				snd_soc_dai_digital_mute(dai, 1,
-						SNDRV_PCM_STREAM_PLAYBACK);
+			if (dai->stream_active[playback])
+				snd_soc_dai_digital_mute(dai, 1, playback);
 		}
 	}
 
@@ -680,14 +681,14 @@ static void soc_resume_deferred(struct work_struct *work)
 	/* unmute any active DACs */
 	for_each_card_rtds(card, rtd) {
 		struct snd_soc_dai *dai;
+		int playback = SNDRV_PCM_STREAM_PLAYBACK;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
 		for_each_rtd_codec_dai(rtd, i, dai) {
-			if (dai->playback_active)
-				snd_soc_dai_digital_mute(dai, 0,
-						SNDRV_PCM_STREAM_PLAYBACK);
+			if (dai->stream_active[playback])
+				snd_soc_dai_digital_mute(dai, 0, playback);
 		}
 	}
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index bf1f74c42e1a..bbecf8f5f4b9 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -100,15 +100,9 @@ static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd,
 
 	lockdep_assert_held(&rtd->card->pcm_mutex);
 
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		cpu_dai->playback_active += action;
-		for_each_rtd_codec_dai(rtd, i, codec_dai)
-			codec_dai->playback_active += action;
-	} else {
-		cpu_dai->capture_active += action;
-		for_each_rtd_codec_dai(rtd, i, codec_dai)
-			codec_dai->capture_active += action;
-	}
+	cpu_dai->stream_active[stream] += action;
+	for_each_rtd_codec_dai(rtd, i, codec_dai)
+		codec_dai->stream_active[stream] += action;
 
 	cpu_dai->active += action;
 	cpu_dai->component->active += action;
@@ -967,8 +961,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	/* apply codec digital mute */
 	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if ((playback && codec_dai->playback_active == 1) ||
-		    (!playback && codec_dai->capture_active == 1))
+		int playback_active = codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK];
+		int capture_active  = codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE];
+
+		if ((playback && playback_active == 1) ||
+		    (!playback && capture_active == 1))
 			snd_soc_dai_digital_mute(codec_dai, 1,
 						 substream->stream);
 	}
@@ -2634,7 +2631,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 		goto capture;
 
 	/* skip if FE isn't currently playing */
-	if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active)
+	if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] ||
+	    !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
 		goto capture;
 
 	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
@@ -2665,7 +2663,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 		return 0;
 
 	/* skip if FE isn't currently capturing */
-	if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active)
+	if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] ||
+	    !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
 		return 0;
 
 	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
-- 
2.17.1

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[alsa-devel] [PATCH v2 10/10] ASoC: soc.h: add for_each_pcm_streams()

[Kuninori Morimoto]

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

ALSA SoC has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere.
Having for_each_xxxx macro is useful.
This patch adds for_each_pcm_streams() for it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
v1 -> v2

	- macro is implemented at sound/pcm.h
	- care more files

 include/sound/pcm.h                   |  5 ++
 sound/soc/dwc/dwc-i2s.c               |  8 +--
 sound/soc/fsl/fsl_asrc_dma.c          |  4 +-
 sound/soc/qcom/lpass-platform.c       |  2 +-
 sound/soc/soc-core.c                  | 31 ++++-----
 sound/soc/soc-generic-dmaengine-pcm.c |  8 +--
 sound/soc/soc-pcm.c                   | 97 ++++++++++-----------------
 sound/soc/sof/sof-audio.c             | 19 ++----
 sound/usb/usx2y/usbusx2yaudio.c       |  9 +--
 9 files changed, 75 insertions(+), 108 deletions(-)

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index f657ff08f317..2628246b76fa 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -644,6 +644,11 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
 #define snd_pcm_group_for_each_entry(s, substream) \
 	list_for_each_entry(s, &substream->group->substreams, link_list)
 
+#define for_each_pcm_streams(stream)			\
+	for (stream  = SNDRV_PCM_STREAM_PLAYBACK;	\
+	     stream <= SNDRV_PCM_STREAM_LAST;		\
+	     stream++)
+
 /**
  * snd_pcm_running - Check whether the substream is in a running state
  * @substream: substream to check
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
index a8bff6f08a69..515f88456dbd 100644
--- a/sound/soc/dwc/dwc-i2s.c
+++ b/sound/soc/dwc/dwc-i2s.c
@@ -422,15 +422,15 @@ static int dw_i2s_resume(struct snd_soc_component *component)
 {
 	struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component);
 	struct snd_soc_dai *dai;
+	int stream;
 
 	if (dev->capability & DW_I2S_MASTER)
 		clk_enable(dev->clk);
 
 	for_each_component_dais(component, dai) {
-		if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
-			dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK);
-		if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
-			dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE);
+		for_each_pcm_streams(stream)
+			if (dai->stream_active[stream])
+				dw_i2s_config(dev, stream);
 	}
 
 	return 0;
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index ece130f59d15..44e5924be870 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component,
 		return ret;
 	}
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+	for_each_pcm_streams(i) {
 		substream = pcm->streams[i].substream;
 		if (!substream)
 			continue;
@@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component,
 	struct snd_pcm_substream *substream;
 	int i;
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
+	for_each_pcm_streams(i) {
 		substream = pcm->streams[i].substream;
 		if (!substream)
 			continue;
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index b05091c283b7..5d1bc5757169 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -529,7 +529,7 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component,
 	struct snd_pcm_substream *substream;
 	int i;
 
-	for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+	for_each_pcm_streams(i) {
 		substream = pcm->streams[i].substream;
 		if (substream) {
 			snd_dma_free_pages(&substream->dma_buffer);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 13df3175e184..6f3fc22e0562 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -432,6 +432,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
 	struct snd_soc_component *component;
 	struct device *dev;
 	int ret;
+	int stream;
 
 	/*
 	 * for rtd->dev
@@ -466,10 +467,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
 
 	rtd->dev = dev;
 	INIT_LIST_HEAD(&rtd->list);
-	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
-	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
-	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
-	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
+	for_each_pcm_streams(stream) {
+		INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients);
+		INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients);
+	}
 	dev_set_drvdata(dev, rtd);
 	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
 
@@ -559,17 +560,14 @@ int snd_soc_suspend(struct device *dev)
 	snd_soc_flush_all_delayed_work(card);
 
 	for_each_card_rtds(card, rtd) {
+		int stream;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		snd_soc_dapm_stream_event(rtd,
-					  SNDRV_PCM_STREAM_PLAYBACK,
-					  SND_SOC_DAPM_STREAM_SUSPEND);
-
-		snd_soc_dapm_stream_event(rtd,
-					  SNDRV_PCM_STREAM_CAPTURE,
-					  SND_SOC_DAPM_STREAM_SUSPEND);
+		for_each_pcm_streams(stream)
+			snd_soc_dapm_stream_event(rtd, stream,
+						  SND_SOC_DAPM_STREAM_SUSPEND);
 	}
 
 	/* Recheck all endpoints too, their state is affected by suspend */
@@ -665,17 +663,14 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	for_each_card_rtds(card, rtd) {
+		int stream;
 
 		if (rtd->dai_link->ignore_suspend)
 			continue;
 
-		snd_soc_dapm_stream_event(rtd,
-					  SNDRV_PCM_STREAM_PLAYBACK,
-					  SND_SOC_DAPM_STREAM_RESUME);
-
-		snd_soc_dapm_stream_event(rtd,
-					  SNDRV_PCM_STREAM_CAPTURE,
-					  SND_SOC_DAPM_STREAM_RESUME);
+		for_each_pcm_streams(stream)
+			snd_soc_dapm_stream_event(rtd, stream,
+						  SND_SOC_DAPM_STREAM_RESUME);
 	}
 
 	/* unmute any active DACs */
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 2cc25651661c..d6b4831e8aec 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -237,7 +237,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component,
 		max_buffer_size = SIZE_MAX;
 	}
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
+	for_each_pcm_streams(i) {
 		substream = rtd->pcm->streams[i].substream;
 		if (!substream)
 			continue;
@@ -371,8 +371,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
 		dev = config->dma_dev;
 	}
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
-	     i++) {
+	for_each_pcm_streams(i) {
 		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
 			name = "rx-tx";
 		else
@@ -401,8 +400,7 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
 {
 	unsigned int i;
 
-	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
-	     i++) {
+	for_each_pcm_streams(i) {
 		if (!pcm->chan[i])
 			continue;
 		dma_release_channel(pcm->chan[i]);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index bbecf8f5f4b9..d69b53e21a18 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2612,6 +2612,7 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
 static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 {
 	struct snd_soc_dapm_widget_list *list;
+	int stream;
 	int count, paths;
 
 	if (!fe->dai_link->dynamic)
@@ -2625,69 +2626,42 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
 	dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n",
 		new ? "new" : "old", fe->dai_link->name);
 
-	/* skip if FE doesn't have playback capability */
-	if (!snd_soc_dai_stream_valid(fe->cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK) ||
-	    !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK))
-		goto capture;
-
-	/* skip if FE isn't currently playing */
-	if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] ||
-	    !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
-		goto capture;
-
-	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
-	if (paths < 0) {
-		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
-			 fe->dai_link->name,  "playback");
-		return paths;
-	}
-
-	/* update any playback paths */
-	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new);
-	if (count) {
-		if (new)
-			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
-		else
-			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
+	for_each_pcm_streams(stream) {
 
-		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
-		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
-	}
-
-	dpcm_path_put(&list);
+		/* skip if FE doesn't have playback/capture capability */
+		if (!snd_soc_dai_stream_valid(fe->cpu_dai,   stream) ||
+		    !snd_soc_dai_stream_valid(fe->codec_dai, stream))
+			continue;
 
-capture:
-	/* skip if FE doesn't have capture capability */
-	if (!snd_soc_dai_stream_valid(fe->cpu_dai,   SNDRV_PCM_STREAM_CAPTURE) ||
-	    !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE))
-		return 0;
+		/* skip if FE isn't currently playing/capturing */
+		if (!fe->cpu_dai->stream_active[stream] ||
+		    !fe->codec_dai->stream_active[stream])
+			continue;
 
-	/* skip if FE isn't currently capturing */
-	if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] ||
-	    !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
-		return 0;
+		paths = dpcm_path_get(fe, stream, &list);
+		if (paths < 0) {
+			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
+				 fe->dai_link->name,
+				 stream == SNDRV_PCM_STREAM_PLAYBACK ?
+				 "playback" : "capture");
+			return paths;
+		}
 
-	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
-	if (paths < 0) {
-		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
-			 fe->dai_link->name,  "capture");
-		return paths;
-	}
+		/* update any playback/capture paths */
+		count = dpcm_process_paths(fe, stream, &list, new);
+		if (count) {
+			if (new)
+				dpcm_run_new_update(fe, stream);
+			else
+				dpcm_run_old_update(fe, stream);
 
-	/* update any old capture paths */
-	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new);
-	if (count) {
-		if (new)
-			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
-		else
-			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
+			dpcm_clear_pending_state(fe, stream);
+			dpcm_be_disconnect(fe, stream);
+		}
 
-		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
-		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
+		dpcm_path_put(&list);
 	}
 
-	dpcm_path_put(&list);
-
 	return 0;
 }
 
@@ -3117,19 +3091,18 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
 {
 	struct snd_soc_pcm_runtime *fe = file->private_data;
 	ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
+	int stream;
 	char *buf;
 
 	buf = kmalloc(out_count, GFP_KERNEL);
 	if (!buf)
 		return -ENOMEM;
 
-	if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK))
-		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK,
-					buf + offset, out_count - offset);
-
-	if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE))
-		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE,
-					buf + offset, out_count - offset);
+	for_each_pcm_streams(stream)
+		if (snd_soc_dai_stream_valid(fe->cpu_dai, stream))
+			offset += dpcm_show_state(fe, stream,
+						  buf + offset,
+						  out_count - offset);
 
 	ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
 
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index 75f2ef2bd94b..fc4ed2a8a914 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -23,7 +23,7 @@ bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev)
 	int dir;
 
 	list_for_each_entry(spcm, &sdev->pcm_list, list) {
-		for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) {
+		for_each_pcm_streams(dir) {
 			substream = spcm->stream[dir].substream;
 			if (!substream || !substream->runtime)
 				continue;
@@ -71,7 +71,7 @@ int sof_set_hw_params_upon_resume(struct device *dev)
 	 * have been suspended.
 	 */
 	list_for_each_entry(spcm, &sdev->pcm_list, list) {
-		for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) {
+		for_each_pcm_streams(dir) {
 			/*
 			 * do not reset hw_params upon resume for streams that
 			 * were kept running during suspend
@@ -319,16 +319,11 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp,
 	int dir;
 
 	list_for_each_entry(spcm, &sdev->pcm_list, list) {
-		dir = SNDRV_PCM_STREAM_PLAYBACK;
-		if (spcm->stream[dir].comp_id == comp_id) {
-			*direction = dir;
-			return spcm;
-		}
-
-		dir = SNDRV_PCM_STREAM_CAPTURE;
-		if (spcm->stream[dir].comp_id == comp_id) {
-			*direction = dir;
-			return spcm;
+		for_each_pcm_streams(dir) {
+			if (spcm->stream[dir].comp_id == comp_id) {
+				*direction = dir;
+				return spcm;
+			}
 		}
 	}
 
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 772f6f3ccbb1..37d290fe9d43 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -906,11 +906,12 @@ static const struct snd_pcm_ops snd_usX2Y_pcm_ops =
  */
 static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream)
 {
-	kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]);
-	usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL;
+	int stream;
 
-	kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]);
-	usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL;
+	for_each_pcm_streams(stream) {
+		kfree(usX2Y_substream[stream]);
+		usX2Y_substream[stream] = NULL;
+	}
 }
 
 static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm)
-- 
2.17.1

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Re: [alsa-devel] [PATCH v2 10/10] ASoC: soc.h: add for_each_pcm_streams()

[Pierre-Louis Bossart]

On 2/13/20 8:36 PM, Kuninori Morimoto wrote:
> 
> From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> 
> ALSA SoC has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere.
> Having for_each_xxxx macro is useful.
> This patch adds for_each_pcm_streams() for it.
> 
> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> ---
> v1 -> v2
> 
> 	- macro is implemented at sound/pcm.h
> 	- care more files
> 
>   include/sound/pcm.h                   |  5 ++
>   sound/soc/dwc/dwc-i2s.c               |  8 +--
>   sound/soc/fsl/fsl_asrc_dma.c          |  4 +-
>   sound/soc/qcom/lpass-platform.c       |  2 +-
>   sound/soc/soc-core.c                  | 31 ++++-----
>   sound/soc/soc-generic-dmaengine-pcm.c |  8 +--
>   sound/soc/soc-pcm.c                   | 97 ++++++++++-----------------
>   sound/soc/sof/sof-audio.c             | 19 ++----
>   sound/usb/usx2y/usbusx2yaudio.c       |  9 +--

Morimoto-san, could you split this last patch in core changes and then 
one patch for each hardware?

It's not fun for people doing backports to deal with completely 
unrelated platforms. If we have one patch per hardware, then it's much 
easier to avoid conflicts and it helps minimize the number of patches to 
be maintained.

Thank you!

>   9 files changed, 75 insertions(+), 108 deletions(-)
> 
> diff --git a/include/sound/pcm.h b/include/sound/pcm.h
> index f657ff08f317..2628246b76fa 100644
> --- a/include/sound/pcm.h
> +++ b/include/sound/pcm.h
> @@ -644,6 +644,11 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
>   #define snd_pcm_group_for_each_entry(s, substream) \
>   	list_for_each_entry(s, &substream->group->substreams, link_list)
>   
> +#define for_each_pcm_streams(stream)			\
> +	for (stream  = SNDRV_PCM_STREAM_PLAYBACK;	\
> +	     stream <= SNDRV_PCM_STREAM_LAST;		\
> +	     stream++)
> +
>   /**
>    * snd_pcm_running - Check whether the substream is in a running state
>    * @substream: substream to check
> diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c
> index a8bff6f08a69..515f88456dbd 100644
> --- a/sound/soc/dwc/dwc-i2s.c
> +++ b/sound/soc/dwc/dwc-i2s.c
> @@ -422,15 +422,15 @@ static int dw_i2s_resume(struct snd_soc_component *component)
>   {
>   	struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component);
>   	struct snd_soc_dai *dai;
> +	int stream;
>   
>   	if (dev->capability & DW_I2S_MASTER)
>   		clk_enable(dev->clk);
>   
>   	for_each_component_dais(component, dai) {
> -		if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
> -			dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK);
> -		if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
> -			dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE);
> +		for_each_pcm_streams(stream)
> +			if (dai->stream_active[stream])
> +				dw_i2s_config(dev, stream);
>   	}
>   
>   	return 0;
> diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
> index ece130f59d15..44e5924be870 100644
> --- a/sound/soc/fsl/fsl_asrc_dma.c
> +++ b/sound/soc/fsl/fsl_asrc_dma.c
> @@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component,
>   		return ret;
>   	}
>   
> -	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
> +	for_each_pcm_streams(i) {
>   		substream = pcm->streams[i].substream;
>   		if (!substream)
>   			continue;
> @@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component,
>   	struct snd_pcm_substream *substream;
>   	int i;
>   
> -	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) {
> +	for_each_pcm_streams(i) {
>   		substream = pcm->streams[i].substream;
>   		if (!substream)
>   			continue;
> diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
> index b05091c283b7..5d1bc5757169 100644
> --- a/sound/soc/qcom/lpass-platform.c
> +++ b/sound/soc/qcom/lpass-platform.c
> @@ -529,7 +529,7 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component,
>   	struct snd_pcm_substream *substream;
>   	int i;
>   
> -	for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
> +	for_each_pcm_streams(i) {
>   		substream = pcm->streams[i].substream;
>   		if (substream) {
>   			snd_dma_free_pages(&substream->dma_buffer);
> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
> index 13df3175e184..6f3fc22e0562 100644
> --- a/sound/soc/soc-core.c
> +++ b/sound/soc/soc-core.c
> @@ -432,6 +432,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
>   	struct snd_soc_component *component;
>   	struct device *dev;
>   	int ret;
> +	int stream;
>   
>   	/*
>   	 * for rtd->dev
> @@ -466,10 +467,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
>   
>   	rtd->dev = dev;
>   	INIT_LIST_HEAD(&rtd->list);
> -	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
> -	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients);
> -	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients);
> -	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
> +	for_each_pcm_streams(stream) {
> +		INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients);
> +		INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients);
> +	}
>   	dev_set_drvdata(dev, rtd);
>   	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
>   
> @@ -559,17 +560,14 @@ int snd_soc_suspend(struct device *dev)
>   	snd_soc_flush_all_delayed_work(card);
>   
>   	for_each_card_rtds(card, rtd) {
> +		int stream;
>   
>   		if (rtd->dai_link->ignore_suspend)
>   			continue;
>   
> -		snd_soc_dapm_stream_event(rtd,
> -					  SNDRV_PCM_STREAM_PLAYBACK,
> -					  SND_SOC_DAPM_STREAM_SUSPEND);
> -
> -		snd_soc_dapm_stream_event(rtd,
> -					  SNDRV_PCM_STREAM_CAPTURE,
> -					  SND_SOC_DAPM_STREAM_SUSPEND);
> +		for_each_pcm_streams(stream)
> +			snd_soc_dapm_stream_event(rtd, stream,
> +						  SND_SOC_DAPM_STREAM_SUSPEND);
>   	}
>   
>   	/* Recheck all endpoints too, their state is affected by suspend */
> @@ -665,17 +663,14 @@ static void soc_resume_deferred(struct work_struct *work)
>   	}
>   
>   	for_each_card_rtds(card, rtd) {
> +		int stream;
>   
>   		if (rtd->dai_link->ignore_suspend)
>   			continue;
>   
> -		snd_soc_dapm_stream_event(rtd,
> -					  SNDRV_PCM_STREAM_PLAYBACK,
> -					  SND_SOC_DAPM_STREAM_RESUME);
> -
> -		snd_soc_dapm_stream_event(rtd,
> -					  SNDRV_PCM_STREAM_CAPTURE,
> -					  SND_SOC_DAPM_STREAM_RESUME);
> +		for_each_pcm_streams(stream)
> +			snd_soc_dapm_stream_event(rtd, stream,
> +						  SND_SOC_DAPM_STREAM_RESUME);
>   	}
>   
>   	/* unmute any active DACs */
> diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
> index 2cc25651661c..d6b4831e8aec 100644
> --- a/sound/soc/soc-generic-dmaengine-pcm.c
> +++ b/sound/soc/soc-generic-dmaengine-pcm.c
> @@ -237,7 +237,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component,
>   		max_buffer_size = SIZE_MAX;
>   	}
>   
> -	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) {
> +	for_each_pcm_streams(i) {
>   		substream = rtd->pcm->streams[i].substream;
>   		if (!substream)
>   			continue;
> @@ -371,8 +371,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm,
>   		dev = config->dma_dev;
>   	}
>   
> -	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
> -	     i++) {
> +	for_each_pcm_streams(i) {
>   		if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX)
>   			name = "rx-tx";
>   		else
> @@ -401,8 +400,7 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm)
>   {
>   	unsigned int i;
>   
> -	for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE;
> -	     i++) {
> +	for_each_pcm_streams(i) {
>   		if (!pcm->chan[i])
>   			continue;
>   		dma_release_channel(pcm->chan[i]);
> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
> index bbecf8f5f4b9..d69b53e21a18 100644
> --- a/sound/soc/soc-pcm.c
> +++ b/sound/soc/soc-pcm.c
> @@ -2612,6 +2612,7 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
>   static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
>   {
>   	struct snd_soc_dapm_widget_list *list;
> +	int stream;
>   	int count, paths;
>   
>   	if (!fe->dai_link->dynamic)
> @@ -2625,69 +2626,42 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new)
>   	dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n",
>   		new ? "new" : "old", fe->dai_link->name);
>   
> -	/* skip if FE doesn't have playback capability */
> -	if (!snd_soc_dai_stream_valid(fe->cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK) ||
> -	    !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK))
> -		goto capture;
> -
> -	/* skip if FE isn't currently playing */
> -	if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] ||
> -	    !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])
> -		goto capture;
> -
> -	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
> -	if (paths < 0) {
> -		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
> -			 fe->dai_link->name,  "playback");
> -		return paths;
> -	}
> -
> -	/* update any playback paths */
> -	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new);
> -	if (count) {
> -		if (new)
> -			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
> -		else
> -			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK);
> +	for_each_pcm_streams(stream) {
>   
> -		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK);
> -		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
> -	}
> -
> -	dpcm_path_put(&list);
> +		/* skip if FE doesn't have playback/capture capability */
> +		if (!snd_soc_dai_stream_valid(fe->cpu_dai,   stream) ||
> +		    !snd_soc_dai_stream_valid(fe->codec_dai, stream))
> +			continue;
>   
> -capture:
> -	/* skip if FE doesn't have capture capability */
> -	if (!snd_soc_dai_stream_valid(fe->cpu_dai,   SNDRV_PCM_STREAM_CAPTURE) ||
> -	    !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE))
> -		return 0;
> +		/* skip if FE isn't currently playing/capturing */
> +		if (!fe->cpu_dai->stream_active[stream] ||
> +		    !fe->codec_dai->stream_active[stream])
> +			continue;
>   
> -	/* skip if FE isn't currently capturing */
> -	if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] ||
> -	    !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE])
> -		return 0;
> +		paths = dpcm_path_get(fe, stream, &list);
> +		if (paths < 0) {
> +			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
> +				 fe->dai_link->name,
> +				 stream == SNDRV_PCM_STREAM_PLAYBACK ?
> +				 "playback" : "capture");
> +			return paths;
> +		}
>   
> -	paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
> -	if (paths < 0) {
> -		dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
> -			 fe->dai_link->name,  "capture");
> -		return paths;
> -	}
> +		/* update any playback/capture paths */
> +		count = dpcm_process_paths(fe, stream, &list, new);
> +		if (count) {
> +			if (new)
> +				dpcm_run_new_update(fe, stream);
> +			else
> +				dpcm_run_old_update(fe, stream);
>   
> -	/* update any old capture paths */
> -	count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new);
> -	if (count) {
> -		if (new)
> -			dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE);
> -		else
> -			dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE);
> +			dpcm_clear_pending_state(fe, stream);
> +			dpcm_be_disconnect(fe, stream);
> +		}
>   
> -		dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE);
> -		dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE);
> +		dpcm_path_put(&list);
>   	}
>   
> -	dpcm_path_put(&list);
> -
>   	return 0;
>   }
>   
> @@ -3117,19 +3091,18 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf,
>   {
>   	struct snd_soc_pcm_runtime *fe = file->private_data;
>   	ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0;
> +	int stream;
>   	char *buf;
>   
>   	buf = kmalloc(out_count, GFP_KERNEL);
>   	if (!buf)
>   		return -ENOMEM;
>   
> -	if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK))
> -		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK,
> -					buf + offset, out_count - offset);
> -
> -	if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE))
> -		offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE,
> -					buf + offset, out_count - offset);
> +	for_each_pcm_streams(stream)
> +		if (snd_soc_dai_stream_valid(fe->cpu_dai, stream))
> +			offset += dpcm_show_state(fe, stream,
> +						  buf + offset,
> +						  out_count - offset);
>   
>   	ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset);
>   
> diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
> index 75f2ef2bd94b..fc4ed2a8a914 100644
> --- a/sound/soc/sof/sof-audio.c
> +++ b/sound/soc/sof/sof-audio.c
> @@ -23,7 +23,7 @@ bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev)
>   	int dir;
>   
>   	list_for_each_entry(spcm, &sdev->pcm_list, list) {
> -		for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) {
> +		for_each_pcm_streams(dir) {
>   			substream = spcm->stream[dir].substream;
>   			if (!substream || !substream->runtime)
>   				continue;
> @@ -71,7 +71,7 @@ int sof_set_hw_params_upon_resume(struct device *dev)
>   	 * have been suspended.
>   	 */
>   	list_for_each_entry(spcm, &sdev->pcm_list, list) {
> -		for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) {
> +		for_each_pcm_streams(dir) {
>   			/*
>   			 * do not reset hw_params upon resume for streams that
>   			 * were kept running during suspend
> @@ -319,16 +319,11 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp,
>   	int dir;
>   
>   	list_for_each_entry(spcm, &sdev->pcm_list, list) {
> -		dir = SNDRV_PCM_STREAM_PLAYBACK;
> -		if (spcm->stream[dir].comp_id == comp_id) {
> -			*direction = dir;
> -			return spcm;
> -		}
> -
> -		dir = SNDRV_PCM_STREAM_CAPTURE;
> -		if (spcm->stream[dir].comp_id == comp_id) {
> -			*direction = dir;
> -			return spcm;
> +		for_each_pcm_streams(dir) {
> +			if (spcm->stream[dir].comp_id == comp_id) {
> +				*direction = dir;
> +				return spcm;
> +			}
>   		}
>   	}
>   
> diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
> index 772f6f3ccbb1..37d290fe9d43 100644
> --- a/sound/usb/usx2y/usbusx2yaudio.c
> +++ b/sound/usb/usx2y/usbusx2yaudio.c
> @@ -906,11 +906,12 @@ static const struct snd_pcm_ops snd_usX2Y_pcm_ops =
>    */
>   static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream)
>   {
> -	kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]);
> -	usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL;
> +	int stream;
>   
> -	kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]);
> -	usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL;
> +	for_each_pcm_streams(stream) {
> +		kfree(usX2Y_substream[stream]);
> +		usX2Y_substream[stream] = NULL;
> +	}
>   }
>   
>   static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm)
> 
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Re: [alsa-devel] [PATCH v2 00/10] ASoC: soc-pcm cleanup step3

[Ranjani Sridharan]

On Fri, 2020-02-14 at 11:35 +0900, Kuninori Morimoto wrote:
> Hi Mark
> 
> These are v2 of soc-pcm cleanup step3.
> I removed unneeded Multi-Codec patch.
> 
> And re-ordered / merged my local tree and add [09/10] patch
> to make [10/10] more effective.
> 
> Kuninori Morimoto (10):
>    1) ASoC: soc-pcm: move dai_get_widget()
>    2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()
>    3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()
>    4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()
>    5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute()
>    6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state()
>    7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate
> code
>    8) ASoC: soc-pcm: use goto and remove multi return
>    9) ASoC: soc-pcm: merge playback/cature_active into stream_active
>   10) ASoC: soc.h: add for_each_pcm_streams()
> 
LGTM, thanks Morimoto-sanm, for the whole V2 series

Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>

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Re: [alsa-devel] [PATCH v2 10/10] ASoC: soc.h: add for_each_pcm_streams()

[Kuninori Morimoto]

Hi Pierre-Louis

> > ALSA SoC has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere.
> > Having for_each_xxxx macro is useful.
> > This patch adds for_each_pcm_streams() for it.
> > 
> > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
> > ---
> > v1 -> v2
> > 
> > 	- macro is implemented at sound/pcm.h
> > 	- care more files
> > 
> >   include/sound/pcm.h                   |  5 ++
> >   sound/soc/dwc/dwc-i2s.c               |  8 +--
> >   sound/soc/fsl/fsl_asrc_dma.c          |  4 +-
> >   sound/soc/qcom/lpass-platform.c       |  2 +-
> >   sound/soc/soc-core.c                  | 31 ++++-----
> >   sound/soc/soc-generic-dmaengine-pcm.c |  8 +--
> >   sound/soc/soc-pcm.c                   | 97 ++++++++++-----------------
> >   sound/soc/sof/sof-audio.c             | 19 ++----
> >   sound/usb/usx2y/usbusx2yaudio.c       |  9 +--
> 
> Morimoto-san, could you split this last patch in core changes and then
> one patch for each hardware?
> 
> It's not fun for people doing backports to deal with completely
> unrelated platforms. If we have one patch per hardware, then it's much
> easier to avoid conflicts and it helps minimize the number of patches
> to be maintained.

Ahh, OK.
Will do in v3

Thank you for your help !!
Best regards
---
Kuninori Morimoto
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