[PATCH 01/11] ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access

[PATCH 01/11] ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
All patches in this series have a dependency on topic/dapm
---
 sound/soc/codecs/88pm860x-codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index c0b2686..ee31fa7 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1140,7 +1140,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Enable Audio PLL & Audio section */
 			data = AUDIO_PLL | AUDIO_SECTION_ON;
 			pm860x_reg_write(pm860x->i2c, REG_MISC2, data);
-- 
1.8.0

[PATCH 02/11] ASoC: ab8500: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/ab8500-codec.c | 20 +++++++++++---------
 1 file changed, 11 insertions(+), 9 deletions(-)

diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index 88ca9cb..c7d243d 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -1209,6 +1209,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
 	struct device *dev = codec->dev;
 	bool apply_fir, apply_iir;
@@ -1234,15 +1235,14 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 	apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR;
 	apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;
 
-	status = snd_soc_dapm_force_enable_pin(&codec->dapm,
-					"ANC Configure Input");
+	status = snd_soc_dapm_force_enable_pin(dapm, "ANC Configure Input");
 	if (status < 0) {
 		dev_err(dev,
 			"%s: ERROR: Failed to enable power (status = %d)!\n",
 			__func__, status);
 		goto cleanup;
 	}
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_sync(dapm);
 
 	anc_configure(codec, apply_fir, apply_iir);
 
@@ -1259,8 +1259,8 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 			drvdata->anc_status =  ANC_IIR_CONFIGURED;
 	}
 
-	status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
-	snd_soc_dapm_sync(&codec->dapm);
+	status = snd_soc_dapm_disable_pin(dapm, "ANC Configure Input");
+	snd_soc_dapm_sync(dapm);
 
 cleanup:
 	mutex_unlock(&drvdata->ctrl_lock);
@@ -1947,6 +1947,7 @@ static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec)
 static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
 			struct amic_settings *amics)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	u8 value8;
 	unsigned int value;
 	int status;
@@ -1973,15 +1974,15 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
 	dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic1a_micbias));
 	route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias];
-	status = snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status = snd_soc_dapm_add_routes(dapm, route, 1);
 	dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic1b_micbias));
 	route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias];
-	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status |= snd_soc_dapm_add_routes(dapm, route, 1);
 	dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic2_micbias));
 	route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias];
-	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status |= snd_soc_dapm_add_routes(dapm, route, 1);
 	if (status < 0) {
 		dev_err(codec->dev,
 			"%s: Failed to add AMic-regulator DAPM-routes (%d).\n",
@@ -2461,6 +2462,7 @@ static void ab8500_codec_of_probe(struct device *dev, struct device_node *np,
 
 static int ab8500_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct device *dev = codec->dev;
 	struct device_node *np = dev->of_node;
 	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev);
@@ -2541,7 +2543,7 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
 		&ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value;
 	drvdata->sid_fir_values = (long *)fc->value;
 
-	(void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+	snd_soc_dapm_disable_pin(dapm, "ANC Configure Input");
 
 	mutex_init(&drvdata->ctrl_lock);
 
-- 
1.8.0

[PATCH 03/11] ASoC: ak4641: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/ak4641.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 3b22b58..2d0ff45 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -412,7 +412,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			if (pdata && gpio_is_valid(pdata->gpio_power))
 				gpio_set_value(pdata->gpio_power, 1);
 			mdelay(1);
-- 
1.8.0

[PATCH 04/11] ASoC: cx20442: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/cx20442.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index 13041cc..d6f4abb 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -333,7 +333,7 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level != SND_SOC_BIAS_STANDBY)
+		if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_STANDBY)
 			break;
 		if (IS_ERR(cx20442->por))
 			err = PTR_ERR(cx20442->por);
@@ -341,7 +341,7 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
 			err = regulator_enable(cx20442->por);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level != SND_SOC_BIAS_PREPARE)
+		if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_PREPARE)
 			break;
 		if (IS_ERR(cx20442->por))
 			err = PTR_ERR(cx20442->por);
-- 
1.8.0

[PATCH 05/11] ASoC: es8328: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/es8328.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 996e3f4..6a09101 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -536,7 +536,7 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, ES8328_CONTROL1,
 					ES8328_CONTROL1_VMIDSEL_MASK |
 					ES8328_CONTROL1_ENREF,
-- 
1.8.0

[PATCH 06/11] ASoC: jz4740: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/jz4740.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index 8425d26..9363fdb 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -258,7 +258,7 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* The only way to clear the suspend flag is to reset the codec */
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			jz4740_codec_wakeup(regmap);
 
 		mask = JZ4740_CODEC_1_VREF_DISABLE |
-- 
1.8.0

[PATCH 07/11] ASoC: ml26124: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/ml26124.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index f1d5778..62dda24 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -523,7 +523,7 @@ static int ml26124_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* VMID ON */
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
 					    ML26124_VMID, ML26124_VMID);
 			msleep(500);
-- 
1.8.0

[PATCH 08/11] ASoC: uda134x: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/uda134x.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index dbecbc0..913edf2 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -477,6 +477,7 @@ static struct snd_soc_dai_driver uda134x_dai = {
 
 static int uda134x_soc_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct uda134x_priv *uda134x;
 	struct uda134x_platform_data *pd = codec->component.card->dev->platform_data;
 	const struct snd_soc_dapm_widget *widgets;
@@ -525,7 +526,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec)
 		num_widgets = ARRAY_SIZE(uda1340_dapm_widgets);
 	}
 
-	ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets);
+	ret = snd_soc_dapm_new_controls(dapm, widgets, num_widgets);
 	if (ret) {
 		printk(KERN_ERR "%s failed to register dapm controls: %d",
 			__func__, ret);
-- 
1.8.0

[PATCH 09/11] ASoC: uda1380: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Also drop the unnecessary check at the beginning of the
uda1380_set_bias_level() which compares the current level to the target
level and aborts if they are the same. Since the core will not call the
set_bias_level() callback if we already are in the target state the result
of the check is always false.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/uda1380.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 4f761a2..6e159f5 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -590,9 +590,6 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 	int reg;
 	struct uda1380_platform_data *pdata = codec->dev->platform_data;
 
-	if (codec->dapm.bias_level == level)
-		return 0;
-
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
@@ -600,7 +597,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 		uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			if (gpio_is_valid(pdata->gpio_power)) {
 				gpio_set_value(pdata->gpio_power, 1);
 				mdelay(1);
-- 
1.8.0

[PATCH 10/11] ASoC: sgtl5000: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/sgtl5000.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 69bc87d..e673f6c 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -948,7 +948,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regulator_bulk_enable(
 						ARRAY_SIZE(sgtl5000->supplies),
 						sgtl5000->supplies);
-- 
1.8.0

[PATCH 11/11] ASoC: sirf-audio-codec: Replace direct snd_soc_codec dapm field access

[Lars-Peter Clausen]

The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/sirf-audio-codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c
index 0a8e43c..29cb442 100644
--- a/sound/soc/codecs/sirf-audio-codec.c
+++ b/sound/soc/codecs/sirf-audio-codec.c
@@ -395,7 +395,7 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = {
 
 static int sirf_audio_codec_probe(struct snd_soc_codec *codec)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	pm_runtime_enable(codec->dev);
 
-- 
1.8.0

Re: [PATCH 01/11] ASoC: 88pm860x: Replace direct snd_soc_codec dapm field access

[Mark Brown]

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On Mon, May 11, 2015 at 09:42:26AM +0200, Lars-Peter Clausen wrote:
> The dapm field of the snd_soc_codec struct is eventually going to be
> removed, in preparation for this replace all manual access to
> codec->dapm.bias_level with snd_soc_codec_get_bias_level().

Applied all, thanks.

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