[PATCH 1/6] ASoC: Indentation fix for null loop operation

[PATCH 1/6] ASoC: Indentation fix for null loop operation

[Mark Brown]

More with the legibility.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 6be6546..988cdff 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -209,7 +209,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 		int val, item, bitmask;
 
 		for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
-		;
+			;
 		val = snd_soc_read(w->codec, e->reg);
 		item = (val >> e->shift_l) & (bitmask - 1);
 
-- 
1.7.5.3

[PATCH 2/6] ASoC: Remove trace for DAPM bias level logging

[Mark Brown]

It's redundant now thanks to the use of the generic trace infrastructure.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c |   18 ------------------
 1 files changed, 0 insertions(+), 18 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 988cdff..4452b57 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -139,24 +139,6 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 	struct snd_soc_card *card = dapm->card;
 	int ret = 0;
 
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		dev_dbg(dapm->dev, "Setting full bias\n");
-		break;
-	case SND_SOC_BIAS_PREPARE:
-		dev_dbg(dapm->dev, "Setting bias prepare\n");
-		break;
-	case SND_SOC_BIAS_STANDBY:
-		dev_dbg(dapm->dev, "Setting standby bias\n");
-		break;
-	case SND_SOC_BIAS_OFF:
-		dev_dbg(dapm->dev, "Setting bias off\n");
-		break;
-	default:
-		dev_err(dapm->dev, "Setting invalid bias %d\n", level);
-		return -EINVAL;
-	}
-
 	trace_snd_soc_bias_level_start(card, level);
 
 	if (card && card->set_bias_level)
-- 
1.7.5.3

[PATCH 3/6] ASoC: Simplify logic in snd_soc_dapm_set_bias_level()

[Mark Brown]

No functional changes but much less indentation.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c |   24 +++++++++++++-----------
 1 files changed, 13 insertions(+), 11 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 4452b57..d989dd7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -143,17 +143,19 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 
 	if (card && card->set_bias_level)
 		ret = card->set_bias_level(card, level);
-	if (ret == 0) {
-		if (dapm->codec && dapm->codec->driver->set_bias_level)
-			ret = dapm->codec->driver->set_bias_level(dapm->codec, level);
-		else
-			dapm->bias_level = level;
-	}
-	if (ret == 0) {
-		if (card && card->set_bias_level_post)
-			ret = card->set_bias_level_post(card, level);
-	}
+	if (ret != 0)
+		goto out;
+
+	if (dapm->codec && dapm->codec->driver->set_bias_level)
+		ret = dapm->codec->driver->set_bias_level(dapm->codec, level);
+	else
+		dapm->bias_level = level;
+	if (ret != 0)
+		goto out;
 
+	if (card && card->set_bias_level_post)
+		ret = card->set_bias_level_post(card, level);
+out:
 	trace_snd_soc_bias_level_done(card, level);
 
 	return ret;
-- 
1.7.5.3

[PATCH 4/6] ASoC: Add context parameter to card DAPM callbacks

[Mark Brown]

The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h              |    2 ++
 sound/soc/atmel/sam9g20_wm8731.c |    1 +
 sound/soc/omap/ams-delta.c       |    3 ++-
 sound/soc/samsung/speyside.c     |    4 ++++
 sound/soc/soc-dapm.c             |    4 ++--
 5 files changed, 11 insertions(+), 3 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 0f29700..f55efc1 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -726,8 +726,10 @@ struct snd_soc_card {
 
 	/* callbacks */
 	int (*set_bias_level)(struct snd_soc_card *,
+			      struct snd_soc_dapm_context *dapm,
 			      enum snd_soc_bias_level level);
 	int (*set_bias_level_post)(struct snd_soc_card *,
+				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level);
 
 	long pmdown_time;
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 95572d2..bad3aa1 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -92,6 +92,7 @@ static struct snd_soc_ops at91sam9g20ek_ops = {
 };
 
 static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
 					enum snd_soc_bias_level level)
 {
 	static int mclk_on;
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 462cbcb..b40095a 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -427,7 +427,8 @@ static struct snd_soc_ops ams_delta_ops = {
 
 /* Board specific codec bias level control */
 static int ams_delta_set_bias_level(struct snd_soc_card *card,
-					enum snd_soc_bias_level level)
+				    struct snd_soc_dapm_context *dapm,
+				    enum snd_soc_bias_level level)
 {
 	struct snd_soc_codec *codec = card->rtd->codec;
 
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 93078b1..99fb033 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -20,11 +20,15 @@
 #define WM8915_HPSEL_GPIO 214
 
 static int speyside_set_bias_level(struct snd_soc_card *card,
+				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level)
 {
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
 	int ret;
 
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
 		ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_MCLK2,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d989dd7..1682834 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -142,7 +142,7 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 	trace_snd_soc_bias_level_start(card, level);
 
 	if (card && card->set_bias_level)
-		ret = card->set_bias_level(card, level);
+		ret = card->set_bias_level(card, dapm, level);
 	if (ret != 0)
 		goto out;
 
@@ -154,7 +154,7 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 		goto out;
 
 	if (card && card->set_bias_level_post)
-		ret = card->set_bias_level_post(card, level);
+		ret = card->set_bias_level_post(card, dapm, level);
 out:
 	trace_snd_soc_bias_level_done(card, level);
 
-- 
1.7.5.3

[PATCH 5/6] ASoC: Only provide a default bias level update for CODEC contexts

[Mark Brown]

This allows the card driver to use the bias level variable more easily in
multi component systems.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c |   11 +++++++----
 1 files changed, 7 insertions(+), 4 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1682834..34106bc 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -146,10 +146,13 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 	if (ret != 0)
 		goto out;
 
-	if (dapm->codec && dapm->codec->driver->set_bias_level)
-		ret = dapm->codec->driver->set_bias_level(dapm->codec, level);
-	else
-		dapm->bias_level = level;
+	if (dapm->codec) {
+		if (dapm->codec->driver->set_bias_level)
+			ret = dapm->codec->driver->set_bias_level(dapm->codec,
+								  level);
+		else
+			dapm->bias_level = level;
+	}
 	if (ret != 0)
 		goto out;
 
-- 
1.7.5.3

[PATCH 6/6] ASoC: Manage Speyside system clocking only in bias management

[Mark Brown]

Now that the CODEC driver supports it defer configuration of the system
clock until bias management which is a much more idiomatic place to do
system power control and makes things a lot more happy when we're using
both interfaces.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/samsung/speyside.c |   50 +++++++++++++++++++++++++++++++++--------
 1 files changed, 40 insertions(+), 10 deletions(-)

diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 99fb033..2515dba 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -42,11 +42,51 @@ static int speyside_set_bias_level(struct snd_soc_card *card,
 			pr_err("Failed to stop FLL\n");
 			return ret;
 		}
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int speyside_set_bias_level_post(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
+					enum snd_soc_bias_level level)
+{
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+			ret = snd_soc_dai_set_pll(codec_dai, 0,
+						  WM8915_FLL_MCLK2,
+						  32768, 48000 * 256);
+			if (ret < 0) {
+				pr_err("Failed to start FLL\n");
+				return ret;
+			}
+
+			ret = snd_soc_dai_set_sysclk(codec_dai,
+						     WM8915_SYSCLK_FLL,
+						     48000 * 256,
+						     SND_SOC_CLOCK_IN);
+			if (ret < 0)
+				return ret;
+		}
+		break;
 
 	default:
 		break;
 	}
 
+	card->dapm.bias_level = level;
+
 	return 0;
 }
 
@@ -70,16 +110,6 @@ static int speyside_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_pll(codec_dai, 0, WM8915_FLL_MCLK2,
-				  32768, 256 * 48000);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8915_SYSCLK_FLL,
-				     256 * 48000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }
 
-- 
1.7.5.3

Re: [PATCH 1/6] ASoC: Indentation fix for null loop operation

[Liam Girdwood]

On 06/06/11 19:18, Mark Brown wrote:
> More with the legibility.
> 
> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> ---
>  sound/soc/soc-dapm.c |    2 +-
>  1 files changed, 1 insertions(+), 1 deletions(-)
> 
> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
> index 6be6546..988cdff 100644
> --- a/sound/soc/soc-dapm.c
> +++ b/sound/soc/soc-dapm.c
> @@ -209,7 +209,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
>  		int val, item, bitmask;
>  
>  		for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
> -		;
> +			;
>  		val = snd_soc_read(w->codec, e->reg);
>  		item = (val >> e->shift_l) & (bitmask - 1);
>  

All


Acked-by: Liam Girdwood <lrg@ti.com>