[PATCH 1/7] ASoC: ad1836: Replace direct snd_soc_codec dapm field access

[PATCH 1/7] ASoC: ad1836: 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/ad1836.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 685998d..95f0bec 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -251,7 +251,7 @@ static int ad1836_resume(struct snd_soc_codec *codec)
 static int ad1836_probe(struct snd_soc_codec *codec)
 {
 	struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int num_dacs, num_adcs;
 	int ret = 0;
 	int i;
-- 
1.8.0

[PATCH 2/7] ASoC: adau17x1: 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/adau1761.c | 26 ++++++++++++--------------
 sound/soc/codecs/adau1781.c |  9 ++++-----
 sound/soc/codecs/adau17x1.c | 20 ++++++++++----------
 3 files changed, 26 insertions(+), 29 deletions(-)

diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index 5ba2461..2f12477 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -482,6 +482,7 @@ static enum adau1761_output_mode adau1761_get_lineout_mode(
 
 static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau1761_platform_data *pdata = codec->dev->platform_data;
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	enum adau1761_digmic_jackdet_pin_mode mode;
@@ -514,21 +515,18 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
 		if (ret)
 			return ret;
 	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1761_no_dmic_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes,
 			ARRAY_SIZE(adau1761_no_dmic_routes));
 		if (ret)
 			return ret;
 		break;
 	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_DIGMIC:
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-			adau1761_dmic_widgets,
+		ret = snd_soc_dapm_new_controls(dapm, adau1761_dmic_widgets,
 			ARRAY_SIZE(adau1761_dmic_widgets));
 		if (ret)
 			return ret;
 
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1761_dmic_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1761_dmic_routes,
 			ARRAY_SIZE(adau1761_dmic_routes));
 		if (ret)
 			return ret;
@@ -546,6 +544,7 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
 
 static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	struct adau1761_platform_data *pdata = codec->dev->platform_data;
 	enum adau1761_output_mode mode;
@@ -576,12 +575,12 @@ static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
 	}
 
 	if (mode == ADAU1761_OUTPUT_MODE_HEADPHONE_CAPLESS) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			adau1761_capless_dapm_widgets,
 			ARRAY_SIZE(adau1761_capless_dapm_widgets));
 		if (ret)
 			return ret;
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
+		ret = snd_soc_dapm_add_routes(dapm,
 			adau1761_capless_dapm_routes,
 			ARRAY_SIZE(adau1761_capless_dapm_routes));
 	} else {
@@ -589,12 +588,12 @@ static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
 			ARRAY_SIZE(adau1761_mono_controls));
 		if (ret)
 			return ret;
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			adau1761_mono_dapm_widgets,
 			ARRAY_SIZE(adau1761_mono_dapm_widgets));
 		if (ret)
 			return ret;
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
+		ret = snd_soc_dapm_add_routes(dapm,
 			adau1761_mono_dapm_routes,
 			ARRAY_SIZE(adau1761_mono_dapm_routes));
 	}
@@ -639,6 +638,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg)
 
 static int adau1761_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau1761_platform_data *pdata = codec->dev->platform_data;
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	int ret;
@@ -691,14 +691,12 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec)
 		return ret;
 
 	if (adau->type == ADAU1761) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-			adau1761_dapm_widgets,
+		ret = snd_soc_dapm_new_controls(dapm, adau1761_dapm_widgets,
 			ARRAY_SIZE(adau1761_dapm_widgets));
 		if (ret)
 			return ret;
 
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1761_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1761_dapm_routes,
 			ARRAY_SIZE(adau1761_dapm_routes));
 		if (ret)
 			return ret;
diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c
index 9c01ef0..fde9068 100644
--- a/sound/soc/codecs/adau1781.c
+++ b/sound/soc/codecs/adau1781.c
@@ -382,6 +382,7 @@ static int adau1781_set_input_mode(struct adau *adau, unsigned int reg,
 
 static int adau1781_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau1781_platform_data *pdata = dev_get_platdata(codec->dev);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	int ret;
@@ -402,19 +403,17 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec)
 	}
 
 	if (pdata && pdata->use_dmic) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			adau1781_dmic_dapm_widgets,
 			ARRAY_SIZE(adau1781_dmic_dapm_widgets));
 		if (ret)
 			return ret;
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1781_dmic_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1781_dmic_dapm_routes,
 			ARRAY_SIZE(adau1781_dmic_dapm_routes));
 		if (ret)
 			return ret;
 	} else {
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1781_adc_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1781_adc_dapm_routes,
 			ARRAY_SIZE(adau1781_adc_dapm_routes));
 		if (ret)
 			return ret;
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index fa2e690..fcf05b2 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -155,6 +155,7 @@ static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	struct snd_soc_dapm_update update;
@@ -188,7 +189,7 @@ static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol,
 		update.reg = reg;
 		update.val = val;
 
-		snd_soc_dapm_mux_update_power(&codec->dapm, kcontrol,
+		snd_soc_dapm_mux_update_power(dapm, kcontrol,
 				ucontrol->value.enumerated.item[0], e, &update);
 	}
 
@@ -444,8 +445,8 @@ static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
 static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai,
 		int clk_id, unsigned int freq, int dir)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(dai->codec);
-	struct snd_soc_dapm_context *dapm = &dai->codec->dapm;
 
 	switch (clk_id) {
 	case ADAU17X1_CLK_SRC_MCLK:
@@ -804,6 +805,7 @@ EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 
 int adau17x1_add_widgets(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
@@ -811,14 +813,13 @@ int adau17x1_add_widgets(struct snd_soc_codec *codec)
 		ARRAY_SIZE(adau17x1_controls));
 	if (ret)
 		return ret;
-	ret = snd_soc_dapm_new_controls(&codec->dapm, adau17x1_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, adau17x1_dapm_widgets,
 		ARRAY_SIZE(adau17x1_dapm_widgets));
 	if (ret)
 		return ret;
 
 	if (adau17x1_has_dsp(adau)) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-			adau17x1_dsp_dapm_widgets,
+		ret = snd_soc_dapm_new_controls(dapm, adau17x1_dsp_dapm_widgets,
 			ARRAY_SIZE(adau17x1_dsp_dapm_widgets));
 		if (ret)
 			return ret;
@@ -840,21 +841,20 @@ EXPORT_SYMBOL_GPL(adau17x1_add_widgets);
 
 int adau17x1_add_routes(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	ret = snd_soc_dapm_add_routes(&codec->dapm, adau17x1_dapm_routes,
+	ret = snd_soc_dapm_add_routes(dapm, adau17x1_dapm_routes,
 		ARRAY_SIZE(adau17x1_dapm_routes));
 	if (ret)
 		return ret;
 
 	if (adau17x1_has_dsp(adau)) {
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau17x1_dsp_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau17x1_dsp_dapm_routes,
 			ARRAY_SIZE(adau17x1_dsp_dapm_routes));
 	} else {
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau17x1_no_dsp_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes,
 			ARRAY_SIZE(adau17x1_no_dsp_dapm_routes));
 	}
 	return ret;
-- 
1.8.0

[PATCH 3/7] ASoC: adau1977: 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() and all
remaining access to codec->dapm with snd_soc_codec_get_dapm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
---
 sound/soc/codecs/adau1977.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c
index e54d8d5..9bdd15f 100644
--- a/sound/soc/codecs/adau1977.c
+++ b/sound/soc/codecs/adau1977.c
@@ -485,7 +485,7 @@ static int adau1977_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 = adau1977_power_enable(adau1977);
 		break;
 	case SND_SOC_BIAS_OFF:
@@ -848,12 +848,13 @@ static int adau1977_set_sysclk(struct snd_soc_codec *codec,
 
 static int adau1977_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	switch (adau1977->type) {
 	case ADAU1977:
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			adau1977_micbias_dapm_widgets,
 			ARRAY_SIZE(adau1977_micbias_dapm_widgets));
 		if (ret < 0)
-- 
1.8.0

[PATCH 4/7] ASoC: adav80x: 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/adav80x.c | 10 ++++++----
 1 file changed, 6 insertions(+), 4 deletions(-)

diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 260a652..36d8425 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -539,7 +539,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec,
 			      unsigned int freq, int dir)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	if (dir == SND_SOC_CLOCK_IN) {
 		switch (clk_id) {
@@ -622,6 +622,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec,
 static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
 		int source, unsigned int freq_in, unsigned int freq_out)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int pll_ctrl1 = 0;
 	unsigned int pll_ctrl2 = 0;
@@ -687,7 +688,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
 
 		adav80x->pll_src = source;
 
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_sync(dapm);
 	}
 
 	return 0;
@@ -800,11 +801,12 @@ static struct snd_soc_dai_driver adav80x_dais[] = {
 
 static int adav80x_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
 	/* Force PLLs on for SYSCLK output */
-	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
-	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+	snd_soc_dapm_force_enable_pin(dapm, "PLL1");
+	snd_soc_dapm_force_enable_pin(dapm, "PLL2");
 
 	/* Power down S/PDIF receiver, since it is currently not supported */
 	regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20);
-- 
1.8.0

[PATCH 5/7] ASoC: ssm2518: 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/ssm2518.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index 13c6ab0..f30de76 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -510,7 +510,7 @@ static int ssm2518_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 = ssm2518_set_power(ssm2518, true);
 		break;
 	case SND_SOC_BIAS_OFF:
-- 
1.8.0

[PATCH 6/7] ASoC: ssm2602: 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/ssm2602.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 296a140..69a773a 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -523,8 +523,8 @@ static int ssm2602_resume(struct snd_soc_codec *codec)
 
 static int ssm2602_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V,
@@ -548,7 +548,7 @@ static int ssm2602_codec_probe(struct snd_soc_codec *codec)
 
 static int ssm2604_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);
 	int ret;
 
 	ret = snd_soc_dapm_new_controls(dapm, ssm2604_dapm_widgets,
-- 
1.8.0

[PATCH 7/7] ASoC: ssm4567: 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/ssm4567.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c
index 643bcff..938d2cb 100644
--- a/sound/soc/codecs/ssm4567.c
+++ b/sound/soc/codecs/ssm4567.c
@@ -353,7 +353,7 @@ static int ssm4567_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 = ssm4567_set_power(ssm4567, true);
 		break;
 	case SND_SOC_BIAS_OFF:
-- 
1.8.0

Re: [PATCH 1/7] ASoC: ad1836: Replace direct snd_soc_codec dapm field access

[Mark Brown]

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On Mon, May 04, 2015 at 06:46:08PM +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 with snd_soc_codec_get_dapm().

Applied all, thanks.

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