From 8a36eaa2ff4a9452a78d799503b920b4e1a0ec31 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sun, 17 Aug 2014 12:03:05 +0200
Subject: ASoC: dmic: Add to SND_SOC_ALL_CODECS

Improve build coverage of the dmic driver.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/codecs/Kconfig | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8838838e25ed..e514e98e48c4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -56,6 +56,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DA7213 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
+	select SND_SOC_DMIC
 	select SND_SOC_BT_SCO
 	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
-- 
cgit v1.2.3


From 371e07ec837464375fe4d7ef3bd13e13cdfbb458 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sun, 17 Aug 2014 16:18:17 +0200
Subject: ASoC: edma-pcm: Include edma-pcm.h

edma_pcm_platform_register() is declared in edma-pcm.h and defined in
edma-pcm.c. To make sure that the function signature matches for both
edma-pcm.c should include edma-pcm.h

Fixes the following sparse warning:
	sound/soc/davinci/edma-pcm.c:48:5: warning: symbol
	 'edma_pcm_platform_register' was not declared. Should it be static?

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/davinci/edma-pcm.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c
index 605e643133db..59e588abe54b 100644
--- a/sound/soc/davinci/edma-pcm.c
+++ b/sound/soc/davinci/edma-pcm.c
@@ -25,6 +25,8 @@
 #include <sound/dmaengine_pcm.h>
 #include <linux/edma.h>
 
+#include "edma-pcm.h"
+
 static const struct snd_pcm_hardware edma_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
-- 
cgit v1.2.3


From 2d15d974618db4ed3adafe9b9fe092db0f5076a0 Mon Sep 17 00:00:00 2001
From: Bard Liao <bardliao@realtek.com>
Date: Wed, 27 Aug 2014 19:50:34 +0800
Subject: ASoC: rt5677: Add DMIC2 clock selection

There are two pins can be used for rt5677's DMIC2 clock. This patch
add the select options for it.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 include/sound/rt5677.h    |  8 +++++++
 sound/soc/codecs/rt5677.c | 57 ++++++++++++++++++++++++++++++++++++++++-------
 sound/soc/codecs/rt5677.h | 10 +++++++++
 3 files changed, 67 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h
index 3da14313bcfc..a676717f74f4 100644
--- a/include/sound/rt5677.h
+++ b/include/sound/rt5677.h
@@ -12,10 +12,18 @@
 #ifndef __LINUX_SND_RT5677_H
 #define __LINUX_SND_RT5677_H
 
+enum rt5677_dmic2_clk {
+	RT5677_DMIC_CLK1 = 0,
+	RT5677_DMIC_CLK2 = 1,
+};
+
+
 struct rt5677_platform_data {
 	/* IN1 IN2 can optionally be differential */
 	bool in1_diff;
 	bool in2_diff;
+	/* DMIC2 clock source selection */
+	enum rt5677_dmic2_clk dmic2_clk_pin;
 };
 
 #endif
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 67f14556462f..f0b751bf1d6c 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -1700,14 +1700,19 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
 
 	SND_SOC_DAPM_INPUT("Haptic Generator"),
 
-	SND_SOC_DAPM_PGA("DMIC1", RT5677_DMIC_CTRL1, RT5677_DMIC_1_EN_SFT, 0,
-		NULL, 0),
-	SND_SOC_DAPM_PGA("DMIC2", RT5677_DMIC_CTRL1, RT5677_DMIC_2_EN_SFT, 0,
-		NULL, 0),
-	SND_SOC_DAPM_PGA("DMIC3", RT5677_DMIC_CTRL1, RT5677_DMIC_3_EN_SFT, 0,
-		NULL, 0),
-	SND_SOC_DAPM_PGA("DMIC4", RT5677_DMIC_CTRL2, RT5677_DMIC_4_EN_SFT, 0,
-		NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC3", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DMIC4", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DMIC1 power", RT5677_DMIC_CTRL1,
+		RT5677_DMIC_1_EN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DMIC2 power", RT5677_DMIC_CTRL1,
+		RT5677_DMIC_2_EN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DMIC3 power", RT5677_DMIC_CTRL1,
+		RT5677_DMIC_3_EN_SFT, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DMIC4 power", RT5677_DMIC_CTRL2,
+		RT5677_DMIC_4_EN_SFT, 0, NULL, 0),
 
 	SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0,
 		set_dmic_clk, SND_SOC_DAPM_PRE_PMU),
@@ -2130,6 +2135,13 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "DMIC L4", NULL, "DMIC CLK" },
 	{ "DMIC R4", NULL, "DMIC CLK" },
 
+	{ "DMIC L1", NULL, "DMIC1 power" },
+	{ "DMIC R1", NULL, "DMIC1 power" },
+	{ "DMIC L3", NULL, "DMIC3 power" },
+	{ "DMIC R3", NULL, "DMIC3 power" },
+	{ "DMIC L4", NULL, "DMIC4 power" },
+	{ "DMIC R4", NULL, "DMIC4 power" },
+
 	{ "BST1", NULL, "IN1P" },
 	{ "BST1", NULL, "IN1N" },
 	{ "BST2", NULL, "IN2P" },
@@ -2793,6 +2805,16 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
 	{ "PDM2R", NULL, "PDM2 R Mux" },
 };
 
+static const struct snd_soc_dapm_route rt5677_dmic2_clk_1[] = {
+	{ "DMIC L2", NULL, "DMIC1 power" },
+	{ "DMIC R2", NULL, "DMIC1 power" },
+};
+
+static const struct snd_soc_dapm_route rt5677_dmic2_clk_2[] = {
+	{ "DMIC L2", NULL, "DMIC2 power" },
+	{ "DMIC R2", NULL, "DMIC2 power" },
+};
+
 static int rt5677_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
@@ -3144,6 +3166,16 @@ static int rt5677_probe(struct snd_soc_codec *codec)
 
 	rt5677->codec = codec;
 
+	if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) {
+		snd_soc_dapm_add_routes(&codec->dapm,
+			rt5677_dmic2_clk_2,
+			ARRAY_SIZE(rt5677_dmic2_clk_2));
+	} else { /*use dmic1 clock by default*/
+		snd_soc_dapm_add_routes(&codec->dapm,
+			rt5677_dmic2_clk_1,
+			ARRAY_SIZE(rt5677_dmic2_clk_1));
+	}
+
 	rt5677_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020);
@@ -3381,6 +3413,15 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 		regmap_update_bits(rt5677->regmap, RT5677_IN1,
 					RT5677_IN_DF2, RT5677_IN_DF2);
 
+	if (rt5677->pdata.dmic2_clk_pin == RT5677_DMIC_CLK2) {
+		regmap_update_bits(rt5677->regmap, RT5677_GEN_CTRL2,
+					RT5677_GPIO5_FUNC_MASK,
+					RT5677_GPIO5_FUNC_DMIC);
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2,
+					RT5677_GPIO5_DIR_MASK,
+					RT5677_GPIO5_DIR_OUT);
+	}
+
 	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
 				      rt5677_dai, ARRAY_SIZE(rt5677_dai));
 }
diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index 863393e62096..8791ab9637f3 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -1363,6 +1363,11 @@
 #define RT5677_SEL_SRC_IB01			(0x1 << 0)
 #define RT5677_SEL_SRC_IB01_SFT			0
 
+/* GPIO Control 2 (0xc1) */
+#define RT5677_GPIO5_DIR_MASK			(0x1 << 14)
+#define RT5677_GPIO5_DIR_IN			(0x0 << 14)
+#define RT5677_GPIO5_DIR_OUT			(0x1 << 14)
+
 /* Virtual DSP Mixer Control (0xf7 0xf8 0xf9) */
 #define RT5677_DSP_IB_01_H			(0x1 << 15)
 #define RT5677_DSP_IB_01_H_SFT			15
@@ -1393,6 +1398,11 @@
 #define RT5677_DSP_IB_9_L			(0x1 << 1)
 #define RT5677_DSP_IB_9_L_SFT			1
 
+/* General Control2 (0xfc)*/
+#define RT5677_GPIO5_FUNC_MASK			(0x1 << 9)
+#define RT5677_GPIO5_FUNC_GPIO			(0x0 << 9)
+#define RT5677_GPIO5_FUNC_DMIC			(0x1 << 9)
+
 /* System Clock Source */
 enum {
 	RT5677_SCLK_S_MCLK,
-- 
cgit v1.2.3


From 75c3daaad5a2f791e0fbad732690130ce1bc55d2 Mon Sep 17 00:00:00 2001
From: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Date: Mon, 1 Sep 2014 08:47:50 +0800
Subject: ASoC: es8328: fix error return code in es8328_codec_probe()

Fix to return a negative error code from the error handling
case instead of 0, as done elsewhere in this function.

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8328.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 7a9f65ad183d..3ff787063304 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -665,6 +665,7 @@ static int es8328_codec_probe(struct snd_soc_codec *codec)
 	es8328->clk = devm_clk_get(codec->dev, NULL);
 	if (IS_ERR(es8328->clk)) {
 		dev_err(codec->dev, "codec clock missing or invalid\n");
+		ret = PTR_ERR(es8328->clk);
 		goto clk_fail;
 	}
 
-- 
cgit v1.2.3


From b43cfb245f7346cbb25c1919577d9607d2adb974 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 2 Sep 2014 22:20:30 +0200
Subject: ASoC: adau1373: Remove unnecessary suspend/resume bias level changes

The ASoC core will only call the suspend/resume callbacks when the device's
DAPM context is idle. Since this driver sets idle_bias_off to true this
means that the device is already in SND_SOC_BIAS_OFF when the suspend
callback is called, so there is no need to manually set this state again.
There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback
since the core will go right back to SND_SOC_BIAS_OFF.

Also drop the regcache_cache_only() calls from the suspend and resume
handlers. There shouldn't be any IO happening after suspend and before
resume.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/adau1373.c | 14 --------------
 1 file changed, 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 1ff7d4d027e9..194756549ef4 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1454,23 +1454,10 @@ static int adau1373_remove(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int adau1373_suspend(struct snd_soc_codec *codec)
-{
-	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	regcache_cache_only(adau1373->regmap, true);
-
-	return ret;
-}
-
 static int adau1373_resume(struct snd_soc_codec *codec)
 {
 	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
 
-	regcache_cache_only(adau1373->regmap, false);
-	adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	regcache_sync(adau1373->regmap);
 
 	return 0;
@@ -1502,7 +1489,6 @@ static const struct regmap_config adau1373_regmap_config = {
 static struct snd_soc_codec_driver adau1373_codec_driver = {
 	.probe =	adau1373_probe,
 	.remove =	adau1373_remove,
-	.suspend =	adau1373_suspend,
 	.resume =	adau1373_resume,
 	.set_bias_level = adau1373_set_bias_level,
 	.idle_bias_off = true,
-- 
cgit v1.2.3


From 8e6fe35eabc64f35eff5844a2e542c403a00db15 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 2 Sep 2014 22:20:31 +0200
Subject: ASoC: lm49453: Remove unnecessary suspend/resume bias level changes

The ASoC core will only call the suspend/resume callbacks when the device's
DAPM context is idle. Since this driver sets idle_bias_off to true this
means that the device is already in SND_SOC_BIAS_OFF when the suspend
callback is called, so there is no need to manually set this state again.
There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback
since the core will go right back to SND_SOC_BIAS_OFF.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/lm49453.c | 14 --------------
 1 file changed, 14 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index 275b3f72f3f4..c1ae5764983f 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -1395,18 +1395,6 @@ static struct snd_soc_dai_driver lm49453_dai[] = {
 	},
 };
 
-static int lm49453_suspend(struct snd_soc_codec *codec)
-{
-	lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int lm49453_resume(struct snd_soc_codec *codec)
-{
-	lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	return 0;
-}
-
 /* power down chip */
 static int lm49453_remove(struct snd_soc_codec *codec)
 {
@@ -1416,8 +1404,6 @@ static int lm49453_remove(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
 	.remove = lm49453_remove,
-	.suspend = lm49453_suspend,
-	.resume = lm49453_resume,
 	.set_bias_level = lm49453_set_bias_level,
 	.controls = lm49453_snd_controls,
 	.num_controls = ARRAY_SIZE(lm49453_snd_controls),
-- 
cgit v1.2.3


From 7d1a99da0861330f02de5c0f59df1d338477cb54 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 2 Sep 2014 22:20:32 +0200
Subject: ASoC: tlv320aic3x: Remove unnecessary suspend/resume bias level
 changes

The ASoC core will only call the suspend/resume callbacks when the device's
DAPM context is idle. Since this driver sets idle_bias_off to true this
means that the device is already in SND_SOC_BIAS_OFF when the suspend
callback is called, so there is no need to manually set this state again.
There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback
since the core will go right back to SND_SOC_BIAS_OFF.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/tlv320aic3x.c | 16 ----------------
 1 file changed, 16 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 64f179ee9834..f2c416d16b6c 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1222,20 +1222,6 @@ static struct snd_soc_dai_driver aic3x_dai = {
 	.symmetric_rates = 1,
 };
 
-static int aic3x_suspend(struct snd_soc_codec *codec)
-{
-	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int aic3x_resume(struct snd_soc_codec *codec)
-{
-	aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-
 static void aic3x_mono_init(struct snd_soc_codec *codec)
 {
 	/* DAC to Mono Line Out default volume and route to Output mixer */
@@ -1429,8 +1415,6 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
 	.idle_bias_off = true,
 	.probe = aic3x_probe,
 	.remove = aic3x_remove,
-	.suspend = aic3x_suspend,
-	.resume = aic3x_resume,
 	.controls = aic3x_snd_controls,
 	.num_controls = ARRAY_SIZE(aic3x_snd_controls),
 	.dapm_widgets = aic3x_dapm_widgets,
-- 
cgit v1.2.3


From a7edeba4cbbd0f3d22d6d54da7c507bda29b2658 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 2 Sep 2014 22:20:33 +0200
Subject: ASoC: wm8804: Remove unnecessary suspend/resume bias level changes

The ASoC core will only call the suspend/resume callbacks when the device's
DAPM context is idle. Since this driver sets idle_bias_off to true this
means that the device is already in SND_SOC_BIAS_OFF when the suspend
callback is called, so there is no need to manually set this state again.
There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback
since the core will go right back to SND_SOC_BIAS_OFF.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8804.c | 19 -------------------
 1 file changed, 19 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 0ea01dfcb6e1..3addc5fe5cb2 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -518,23 +518,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8804_suspend(struct snd_soc_codec *codec)
-{
-	wm8804_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int wm8804_resume(struct snd_soc_codec *codec)
-{
-	wm8804_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	return 0;
-}
-#else
-#define wm8804_suspend NULL
-#define wm8804_resume NULL
-#endif
-
 static int wm8804_remove(struct snd_soc_codec *codec)
 {
 	struct wm8804_priv *wm8804;
@@ -671,8 +654,6 @@ static struct snd_soc_dai_driver wm8804_dai = {
 static struct snd_soc_codec_driver soc_codec_dev_wm8804 = {
 	.probe = wm8804_probe,
 	.remove = wm8804_remove,
-	.suspend = wm8804_suspend,
-	.resume = wm8804_resume,
 	.set_bias_level = wm8804_set_bias_level,
 	.idle_bias_off = true,
 
-- 
cgit v1.2.3


From e02c716d2ec065fd58c2fc8100fd5f359ab61e7e Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Tue, 2 Sep 2014 22:20:34 +0200
Subject: ASoC: wm8995: Remove unnecessary suspend/resume bias level changes

The ASoC core will only call the suspend/resume callbacks when the device's
DAPM context is idle. Since this driver sets idle_bias_off to true this
means that the device is already in SND_SOC_BIAS_OFF when the suspend
callback is called, so there is no need to manually set this state again.
There is also no need to go to SND_SOC_BIAS_STANDBY in the resume callback
since the core will go right back to SND_SOC_BIAS_OFF.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/wm8995.c | 19 -------------------
 1 file changed, 19 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index cae4ac5a5730..1288edeb8c7d 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1998,23 +1998,6 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8995_suspend(struct snd_soc_codec *codec)
-{
-	wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int wm8995_resume(struct snd_soc_codec *codec)
-{
-	wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	return 0;
-}
-#else
-#define wm8995_suspend NULL
-#define wm8995_resume NULL
-#endif
-
 static int wm8995_remove(struct snd_soc_codec *codec)
 {
 	struct wm8995_priv *wm8995;
@@ -2220,8 +2203,6 @@ static struct snd_soc_dai_driver wm8995_dai[] = {
 static struct snd_soc_codec_driver soc_codec_dev_wm8995 = {
 	.probe = wm8995_probe,
 	.remove = wm8995_remove,
-	.suspend = wm8995_suspend,
-	.resume = wm8995_resume,
 	.set_bias_level = wm8995_set_bias_level,
 	.idle_bias_off = true,
 };
-- 
cgit v1.2.3


From 01e0df6647e713469466c7bb6d7157c2e3046192 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:04 +0200
Subject: ASoC: Set card->instantiated to false when removing the card

Set card->instantiated to false when the card is removed to make sure that
operations that expect the card to be fully instantiated do not run anymore
during card removal.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 1b422c5c36c8..ff9d2892f473 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3810,8 +3810,10 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card);
  */
 int snd_soc_unregister_card(struct snd_soc_card *card)
 {
-	if (card->instantiated)
+	if (card->instantiated) {
+		card->instantiated = false;
 		soc_cleanup_card_resources(card);
+	}
 	dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
 
 	return 0;
-- 
cgit v1.2.3


From 1c325f771a88579f227fe017e4ee77d852cf5435 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:05 +0200
Subject: ASoC: Shutdown DAPM contexts when removing a card

Currently when a ASoC sound card is unregistered we leave the individual
components in their current state, just call the remove() callback and leave
it to the drivers to do the proper shutdown/cleanup.

This patch introduces a call to snd_soc_dapm_shutdown() when removing the
card.  This will make sure that all DAPM widgets are properly powered down
and all DAPM contexts are put at the SND_SOC_BIAS_OFF level. This will
ensure that all components are properly powered down when the card is
removed.

Since a lot of drivers manually go to SND_SOC_BIAS_OFF in their remove
callback this will also allow us to remove a bit of duplicated code.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ff9d2892f473..068785fa1a06 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3812,6 +3812,7 @@ int snd_soc_unregister_card(struct snd_soc_card *card)
 {
 	if (card->instantiated) {
 		card->instantiated = false;
+		snd_soc_dapm_shutdown(card);
 		soc_cleanup_card_resources(card);
 	}
 	dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name);
-- 
cgit v1.2.3


From 86dbf2ac6fcb2d2932d4610f2dfe0954aa0633f7 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:06 +0200
Subject: ASoC: Add support for automatically going to BIAS_OFF on suspend

There is a substantial amount of drivers that in go to SND_SOC_BIAS_OFF on
suspend and go back to SND_SOC_BIAS_SUSPEND on resume (Often this is even
the only thing done in the suspend and resume handlers). This patch
introduces a new suspend_bias_off flag, which when set by a driver will let
the ASoC core automatically put the device's DAPM context at the
SND_SOC_BIAS_OFF level during suspend. Once the device is resumed the DAPM
context will go back to SND_SOC_BIAS_STANDBY (if the context is idle,
otherwise to SND_SOC_BIAS_ON).

This will allow us to remove a fair bit of duplicated code from the drivers.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dapm.h |  3 ++-
 include/sound/soc.h      |  1 +
 sound/soc/soc-core.c     |  1 +
 sound/soc/soc-dapm.c     | 20 ++++++++++++++++++--
 4 files changed, 22 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index aac04ff84eea..f955d65c5656 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -587,7 +587,8 @@ struct snd_soc_dapm_context {
 	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
 	unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
-
+	/* Go to BIAS_OFF in suspend if the DAPM context is idle */
+	unsigned int suspend_bias_off:1;
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
 
diff --git a/include/sound/soc.h b/include/sound/soc.h
index ce09302bfd6d..ac99fc083eec 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -848,6 +848,7 @@ struct snd_soc_codec_driver {
 	int (*set_bias_level)(struct snd_soc_codec *,
 			      enum snd_soc_bias_level level);
 	bool idle_bias_off;
+	bool suspend_bias_off;
 
 	void (*seq_notifier)(struct snd_soc_dapm_context *,
 			     enum snd_soc_dapm_type, int);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 068785fa1a06..2bdf9a4ac2b4 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -4402,6 +4402,7 @@ int snd_soc_register_codec(struct device *dev,
 	codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time;
 	codec->dapm.codec = codec;
 	codec->dapm.idle_bias_off = codec_drv->idle_bias_off;
+	codec->dapm.suspend_bias_off = codec_drv->suspend_bias_off;
 	if (codec_drv->seq_notifier)
 		codec->dapm.seq_notifier = codec_drv->seq_notifier;
 	if (codec_drv->set_bias_level)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8348352dc2c6..a2025a6b6a29 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1683,6 +1683,22 @@ static void dapm_power_one_widget(struct snd_soc_dapm_widget *w,
 	}
 }
 
+static bool dapm_idle_bias_off(struct snd_soc_dapm_context *dapm)
+{
+	if (dapm->idle_bias_off)
+		return true;
+
+	switch (snd_power_get_state(dapm->card->snd_card)) {
+	case SNDRV_CTL_POWER_D3hot:
+	case SNDRV_CTL_POWER_D3cold:
+		return dapm->suspend_bias_off;
+	default:
+		break;
+	}
+
+	return false;
+}
+
 /*
  * Scan each dapm widget for complete audio path.
  * A complete path is a route that has valid endpoints i.e.:-
@@ -1706,7 +1722,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 	trace_snd_soc_dapm_start(card);
 
 	list_for_each_entry(d, &card->dapm_list, list) {
-		if (d->idle_bias_off)
+		if (dapm_idle_bias_off(d))
 			d->target_bias_level = SND_SOC_BIAS_OFF;
 		else
 			d->target_bias_level = SND_SOC_BIAS_STANDBY;
@@ -1772,7 +1788,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event)
 		if (d->target_bias_level > bias)
 			bias = d->target_bias_level;
 	list_for_each_entry(d, &card->dapm_list, list)
-		if (!d->idle_bias_off)
+		if (!dapm_idle_bias_off(d))
 			d->target_bias_level = bias;
 
 	trace_snd_soc_dapm_walk_done(card);
-- 
cgit v1.2.3


From a80932979a72ef9d4e66a69520c7588cc6de5699 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:07 +0200
Subject: ASoC: Always run default suspend/resume code

We do a bit more than just running the callbacks during suspend and resume
these days (e.g. call regcache_mark_dirty() during suspend). But this is
only when suspend and resume callbacks are specified for the driver,
otherwise nothing is done. This means that drivers which don't want to do
anything special during suspend and resume, but still want the standard
operations to run, need to provide empty suspend and resume callback
functions (rather than no callbacks). This patch updates the suspend and
resume code to always run standard sequence regardless of whether suspend
and resume handlers are provided.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/soc-core.c | 11 +++++++----
 1 file changed, 7 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 2bdf9a4ac2b4..c612900c80ff 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -637,7 +637,7 @@ int snd_soc_suspend(struct device *dev)
 	list_for_each_entry(codec, &card->codec_dev_list, card_list) {
 		/* If there are paths active then the CODEC will be held with
 		 * bias _ON and should not be suspended. */
-		if (!codec->suspended && codec->driver->suspend) {
+		if (!codec->suspended) {
 			switch (codec->dapm.bias_level) {
 			case SND_SOC_BIAS_STANDBY:
 				/*
@@ -651,8 +651,10 @@ int snd_soc_suspend(struct device *dev)
 						"ASoC: idle_bias_off CODEC on over suspend\n");
 					break;
 				}
+
 			case SND_SOC_BIAS_OFF:
-				codec->driver->suspend(codec);
+				if (codec->driver->suspend)
+					codec->driver->suspend(codec);
 				codec->suspended = 1;
 				codec->cache_sync = 1;
 				if (codec->component.regmap)
@@ -726,11 +728,12 @@ static void soc_resume_deferred(struct work_struct *work)
 		 * left with bias OFF or STANDBY and suspended so we must now
 		 * resume.  Otherwise the suspend was suppressed.
 		 */
-		if (codec->driver->resume && codec->suspended) {
+		if (codec->suspended) {
 			switch (codec->dapm.bias_level) {
 			case SND_SOC_BIAS_STANDBY:
 			case SND_SOC_BIAS_OFF:
-				codec->driver->resume(codec);
+				if (codec->driver->resume)
+					codec->driver->resume(codec);
 				codec->suspended = 0;
 				break;
 			default:
-- 
cgit v1.2.3


From d7858bd647cda68bf832997a280a2f44aec01f1b Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:08 +0200
Subject: ASoC: adau1373: Cleanup manual bias level transitions

The ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF
when removing the CODEC, no need to do it manually anymore.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/adau1373.c | 7 -------
 1 file changed, 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 194756549ef4..7c784ad3e8b2 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1448,12 +1448,6 @@ static int adau1373_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int adau1373_remove(struct snd_soc_codec *codec)
-{
-	adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
 static int adau1373_resume(struct snd_soc_codec *codec)
 {
 	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
@@ -1488,7 +1482,6 @@ static const struct regmap_config adau1373_regmap_config = {
 
 static struct snd_soc_codec_driver adau1373_codec_driver = {
 	.probe =	adau1373_probe,
-	.remove =	adau1373_remove,
 	.resume =	adau1373_resume,
 	.set_bias_level = adau1373_set_bias_level,
 	.idle_bias_off = true,
-- 
cgit v1.2.3


From 0e0f9b960a011a9e3815004f37cc475229170dfd Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:09 +0200
Subject: ASoC: adau17x1: Cleanup manual bias level transitions

Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/adau1761.c | 2 +-
 sound/soc/codecs/adau1781.c | 2 +-
 sound/soc/codecs/adau17x1.c | 8 --------
 sound/soc/codecs/adau17x1.h | 1 -
 4 files changed, 2 insertions(+), 11 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index 848cab839553..5518ebd6947c 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -714,9 +714,9 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec)
 
 static const struct snd_soc_codec_driver adau1761_codec_driver = {
 	.probe = adau1761_codec_probe,
-	.suspend = adau17x1_suspend,
 	.resume	= adau17x1_resume,
 	.set_bias_level	= adau1761_set_bias_level,
+	.suspend_bias_off = true,
 
 	.controls = adau1761_controls,
 	.num_controls = ARRAY_SIZE(adau1761_controls),
diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c
index 045a61413840..e9fc00fb13dd 100644
--- a/sound/soc/codecs/adau1781.c
+++ b/sound/soc/codecs/adau1781.c
@@ -446,9 +446,9 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec)
 
 static const struct snd_soc_codec_driver adau1781_codec_driver = {
 	.probe = adau1781_codec_probe,
-	.suspend = adau17x1_suspend,
 	.resume = adau17x1_resume,
 	.set_bias_level = adau1781_set_bias_level,
+	.suspend_bias_off = true,
 
 	.controls = adau1781_controls,
 	.num_controls = ARRAY_SIZE(adau1781_controls),
diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c
index 0b659704e60c..3e16c1c64115 100644
--- a/sound/soc/codecs/adau17x1.c
+++ b/sound/soc/codecs/adau17x1.c
@@ -815,13 +815,6 @@ int adau17x1_add_routes(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(adau17x1_add_routes);
 
-int adau17x1_suspend(struct snd_soc_codec *codec)
-{
-	codec->driver->set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-EXPORT_SYMBOL_GPL(adau17x1_suspend);
-
 int adau17x1_resume(struct snd_soc_codec *codec)
 {
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
@@ -829,7 +822,6 @@ int adau17x1_resume(struct snd_soc_codec *codec)
 	if (adau->switch_mode)
 		adau->switch_mode(codec->dev);
 
-	codec->driver->set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	regcache_sync(adau->regmap);
 
 	return 0;
diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h
index 3ffabaf4c7a8..e4a557fd7155 100644
--- a/sound/soc/codecs/adau17x1.h
+++ b/sound/soc/codecs/adau17x1.h
@@ -52,7 +52,6 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec,
 	enum adau17x1_micbias_voltage micbias);
 bool adau17x1_readable_register(struct device *dev, unsigned int reg);
 bool adau17x1_volatile_register(struct device *dev, unsigned int reg);
-int adau17x1_suspend(struct snd_soc_codec *codec);
 int adau17x1_resume(struct snd_soc_codec *codec);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
-- 
cgit v1.2.3


From cd5d3a151118cd815be15970db099bcdb3f0ad12 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:10 +0200
Subject: ASoC: adav80x: Cleanup manual bias level transitions

Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner. While we are at it also remove the
regcache_cache_only() calls from suspend/resume as there shouldn't be any IO
between suspend and resume.

Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.

The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/adav80x.c | 23 ++---------------------
 1 file changed, 2 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index c43b93fdf0df..ce3cdca9fc62 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -812,42 +812,23 @@ static int adav80x_probe(struct snd_soc_codec *codec)
 	/* Disable DAC zero flag */
 	regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6);
 
-	return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-}
-
-static int adav80x_suspend(struct snd_soc_codec *codec)
-{
-	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	regcache_cache_only(adav80x->regmap, true);
-
-	return ret;
+	return 0;
 }
 
 static int adav80x_resume(struct snd_soc_codec *codec)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
-	regcache_cache_only(adav80x->regmap, false);
-	adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	regcache_sync(adav80x->regmap);
 
 	return 0;
 }
 
-static int adav80x_remove(struct snd_soc_codec *codec)
-{
-	return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-}
-
 static struct snd_soc_codec_driver adav80x_codec_driver = {
 	.probe = adav80x_probe,
-	.remove = adav80x_remove,
-	.suspend = adav80x_suspend,
 	.resume = adav80x_resume,
 	.set_bias_level = adav80x_set_bias_level,
+	.suspend_bias_off = true,
 
 	.set_pll = adav80x_set_pll,
 	.set_sysclk = adav80x_set_sysclk,
-- 
cgit v1.2.3


From 0f0cc5a775ebe88d9be12489874bd2799b42e242 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:11 +0200
Subject: ASoC: ssm2518: Cleanup manual bias level transitions

Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.

The manual transition to SND_SOC_BIAS_OFF at the end of CODEC probe()
can also be removed as the CODEC is already in OFF state at this point.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ssm2518.c | 13 -------------
 1 file changed, 13 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index e8680bea5f86..67ea55adb307 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -646,17 +646,6 @@ static struct snd_soc_dai_driver ssm2518_dai = {
 	.ops = &ssm2518_dai_ops,
 };
 
-static int ssm2518_probe(struct snd_soc_codec *codec)
-{
-	return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF);
-}
-
-static int ssm2518_remove(struct snd_soc_codec *codec)
-{
-	ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
 static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 	int source, unsigned int freq, int dir)
 {
@@ -727,8 +716,6 @@ static int ssm2518_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 }
 
 static struct snd_soc_codec_driver ssm2518_codec_driver = {
-	.probe = ssm2518_probe,
-	.remove = ssm2518_remove,
 	.set_bias_level = ssm2518_set_bias_level,
 	.set_sysclk = ssm2518_set_sysclk,
 	.idle_bias_off = true,
-- 
cgit v1.2.3


From 85362efb80070bed890602483f71cd103be303c2 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Thu, 4 Sep 2014 19:44:12 +0200
Subject: ASoC: ssm2602: Cleanup manual bias level transitions

Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner. While we are at it also remove the
regcache_cache_only() calls from suspend/resume as there shouldn't be any IO
between suspend and resume.

Since the ASoC core now takes care of setting the bias level to
SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually
anymore either.

The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe()
can also be removed as the core will automatically do this after the CODEC
has been probed.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/ssm2602.c | 24 ++----------------------
 1 file changed, 2 insertions(+), 22 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 484b3bbe8624..0dec13648563 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -502,18 +502,11 @@ static struct snd_soc_dai_driver ssm2602_dai = {
 	.symmetric_samplebits = 1,
 };
 
-static int ssm2602_suspend(struct snd_soc_codec *codec)
-{
-	ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
 static int ssm2602_resume(struct snd_soc_codec *codec)
 {
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 
 	regcache_sync(ssm2602->regmap);
-	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
 }
@@ -586,27 +579,14 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec)
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-
-/* remove everything here */
-static int ssm2602_remove(struct snd_soc_codec *codec)
-{
-	ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
+	return ret;
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
 	.probe =	ssm260x_codec_probe,
-	.remove =	ssm2602_remove,
-	.suspend =	ssm2602_suspend,
 	.resume =	ssm2602_resume,
 	.set_bias_level = ssm2602_set_bias_level,
+	.suspend_bias_off = true,
 
 	.controls = ssm260x_snd_controls,
 	.num_controls = ARRAY_SIZE(ssm260x_snd_controls),
-- 
cgit v1.2.3


From 8d01370f59856a0ac5b222878667d52477b589f0 Mon Sep 17 00:00:00 2001
From: Lars-Peter Clausen <lars@metafoo.de>
Date: Sat, 6 Sep 2014 14:29:32 +0200
Subject: ASoC: es8328: Cleanup manual bias level transitions

Set the CODEC driver's suspend_bias_off flag rather than manually going to
SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes
the code a bit shorter and cleaner.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/es8328.c | 5 ++---
 1 file changed, 2 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
index 3ff787063304..f27325155ace 100644
--- a/sound/soc/codecs/es8328.c
+++ b/sound/soc/codecs/es8328.c
@@ -602,8 +602,6 @@ static int es8328_suspend(struct snd_soc_codec *codec)
 
 	es8328 = snd_soc_codec_get_drvdata(codec);
 
-	es8328_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
 	clk_disable_unprepare(es8328->clk);
 
 	ret = regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
@@ -643,7 +641,6 @@ static int es8328_resume(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	es8328_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	return 0;
 }
 
@@ -712,6 +709,8 @@ static struct snd_soc_codec_driver es8328_codec_driver = {
 	.resume		  = es8328_resume,
 	.remove		  = es8328_remove,
 	.set_bias_level	  = es8328_set_bias_level,
+	.suspend_bias_off = true,
+
 	.controls	  = es8328_snd_controls,
 	.num_controls	  = ARRAY_SIZE(es8328_snd_controls),
 	.dapm_widgets	  = es8328_dapm_widgets,
-- 
cgit v1.2.3


From e5b2791d2a57e9da369bd75ae2a209bcce2ad4d3 Mon Sep 17 00:00:00 2001
From: Oder Chiou <oder_chiou@realtek.com>
Date: Mon, 15 Sep 2014 19:58:44 +0800
Subject: ASoC: rt5677: Revise the wrong name in the header file

The patch revises the wrong name in the header file.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.h | 44 ++++++++++++++++++++++----------------------
 1 file changed, 22 insertions(+), 22 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index 8791ab9637f3..a334eb66cfc1 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -1287,16 +1287,16 @@
 #define RT5677_PLL1_PD_SFT			8
 #define RT5677_PLL1_PD_1			(0x0 << 8)
 #define RT5677_PLL1_PD_2			(0x1 << 8)
-#define RT5671_DAC_OSR_MASK			(0x3 << 6)
-#define RT5671_DAC_OSR_SFT			6
-#define RT5671_DAC_OSR_128			(0x0 << 6)
-#define RT5671_DAC_OSR_64			(0x1 << 6)
-#define RT5671_DAC_OSR_32			(0x2 << 6)
-#define RT5671_ADC_OSR_MASK			(0x3 << 4)
-#define RT5671_ADC_OSR_SFT			4
-#define RT5671_ADC_OSR_128			(0x0 << 4)
-#define RT5671_ADC_OSR_64			(0x1 << 4)
-#define RT5671_ADC_OSR_32			(0x2 << 4)
+#define RT5677_DAC_OSR_MASK			(0x3 << 6)
+#define RT5677_DAC_OSR_SFT			6
+#define RT5677_DAC_OSR_128			(0x0 << 6)
+#define RT5677_DAC_OSR_64			(0x1 << 6)
+#define RT5677_DAC_OSR_32			(0x2 << 6)
+#define RT5677_ADC_OSR_MASK			(0x3 << 4)
+#define RT5677_ADC_OSR_SFT			4
+#define RT5677_ADC_OSR_128			(0x0 << 4)
+#define RT5677_ADC_OSR_64			(0x1 << 4)
+#define RT5677_ADC_OSR_32			(0x2 << 4)
 
 /* Global Clock Control 2 (0x81) */
 #define RT5677_PLL2_PR_SRC_MASK			(0x1 << 15)
@@ -1312,18 +1312,18 @@
 #define RT5677_PLL2_SRC_BCLK4			(0x4 << 12)
 #define RT5677_PLL2_SRC_RCCLK			(0x5 << 12)
 #define RT5677_PLL2_SRC_SLIM			(0x6 << 12)
-#define RT5671_DSP_ASRC_O_SRC			(0x3 << 10)
-#define RT5671_DSP_ASRC_O_SRC_SFT		10
-#define RT5671_DSP_ASRC_O_MCLK			(0x0 << 10)
-#define RT5671_DSP_ASRC_O_PLL1			(0x1 << 10)
-#define RT5671_DSP_ASRC_O_SLIM			(0x2 << 10)
-#define RT5671_DSP_ASRC_O_RCCLK			(0x3 << 10)
-#define RT5671_DSP_ASRC_I_SRC			(0x3 << 8)
-#define RT5671_DSP_ASRC_I_SRC_SFT		8
-#define RT5671_DSP_ASRC_I_MCLK			(0x0 << 8)
-#define RT5671_DSP_ASRC_I_PLL1			(0x1 << 8)
-#define RT5671_DSP_ASRC_I_SLIM			(0x2 << 8)
-#define RT5671_DSP_ASRC_I_RCCLK			(0x3 << 8)
+#define RT5677_DSP_ASRC_O_SRC			(0x3 << 10)
+#define RT5677_DSP_ASRC_O_SRC_SFT		10
+#define RT5677_DSP_ASRC_O_MCLK			(0x0 << 10)
+#define RT5677_DSP_ASRC_O_PLL1			(0x1 << 10)
+#define RT5677_DSP_ASRC_O_SLIM			(0x2 << 10)
+#define RT5677_DSP_ASRC_O_RCCLK			(0x3 << 10)
+#define RT5677_DSP_ASRC_I_SRC			(0x3 << 8)
+#define RT5677_DSP_ASRC_I_SRC_SFT		8
+#define RT5677_DSP_ASRC_I_MCLK			(0x0 << 8)
+#define RT5677_DSP_ASRC_I_PLL1			(0x1 << 8)
+#define RT5677_DSP_ASRC_I_SLIM			(0x2 << 8)
+#define RT5677_DSP_ASRC_I_RCCLK			(0x3 << 8)
 #define RT5677_DSP_CLK_SRC_MASK			(0x1 << 7)
 #define RT5677_DSP_CLK_SRC_SFT			7
 #define RT5677_DSP_CLK_SRC_PLL2			(0x0 << 7)
-- 
cgit v1.2.3


From 44caf7648064502fd1d37d18443ae92c064ebadd Mon Sep 17 00:00:00 2001
From: Oder Chiou <oder_chiou@realtek.com>
Date: Tue, 16 Sep 2014 11:37:39 +0800
Subject: ASoC: rt5677: Add the GPIO function

The patch adds the GPIO function.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 133 ++++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/rt5677.h | 112 ++++++++++++++++++++++++++++++++++++++
 2 files changed, 245 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index f0b751bf1d6c..02bc8bd7caeb 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -19,6 +19,7 @@
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
+#include <linux/gpio.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -3160,6 +3161,135 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
+#ifdef CONFIG_GPIOLIB
+static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip)
+{
+	return container_of(chip, struct rt5677_priv, gpio_chip);
+}
+
+static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+
+	switch (offset) {
+	case RT5677_GPIO1 ... RT5677_GPIO5:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2,
+			0x1 << (offset * 3 + 1), !!value << (offset * 3 + 1));
+		break;
+
+	case RT5677_GPIO6:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3,
+			RT5677_GPIO6_OUT_MASK, !!value << RT5677_GPIO6_OUT_SFT);
+		break;
+
+	default:
+		break;
+	}
+}
+
+static int rt5677_gpio_direction_out(struct gpio_chip *chip,
+				     unsigned offset, int value)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+
+	switch (offset) {
+	case RT5677_GPIO1 ... RT5677_GPIO5:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2,
+			0x3 << (offset * 3 + 1),
+			(0x2 | !!value) << (offset * 3 + 1));
+		break;
+
+	case RT5677_GPIO6:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3,
+			RT5677_GPIO6_DIR_MASK | RT5677_GPIO6_OUT_MASK,
+			RT5677_GPIO6_DIR_OUT | !!value << RT5677_GPIO6_OUT_SFT);
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+	int value, ret;
+
+	ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value);
+	if (ret < 0)
+		return ret;
+
+	return (value & (0x1 << offset)) >> offset;
+}
+
+static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+
+	switch (offset) {
+	case RT5677_GPIO1 ... RT5677_GPIO5:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL2,
+			0x1 << (offset * 3 + 2), 0x0);
+		break;
+
+	case RT5677_GPIO6:
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL3,
+			RT5677_GPIO6_DIR_MASK, RT5677_GPIO6_DIR_IN);
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static struct gpio_chip rt5677_template_chip = {
+	.label			= "rt5677",
+	.owner			= THIS_MODULE,
+	.direction_output	= rt5677_gpio_direction_out,
+	.set			= rt5677_gpio_set,
+	.direction_input	= rt5677_gpio_direction_in,
+	.get			= rt5677_gpio_get,
+	.can_sleep		= 1,
+};
+
+static void rt5677_init_gpio(struct i2c_client *i2c)
+{
+	struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
+	int ret;
+
+	rt5677->gpio_chip = rt5677_template_chip;
+	rt5677->gpio_chip.ngpio = RT5677_GPIO_NUM;
+	rt5677->gpio_chip.dev = &i2c->dev;
+	rt5677->gpio_chip.base = -1;
+
+	ret = gpiochip_add(&rt5677->gpio_chip);
+	if (ret != 0)
+		dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret);
+}
+
+static void rt5677_free_gpio(struct i2c_client *i2c)
+{
+	struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
+	int ret;
+
+	ret = gpiochip_remove(&rt5677->gpio_chip);
+	if (ret != 0)
+		dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret);
+}
+#else
+static void rt5677_init_gpio(struct i2c_client *i2c)
+{
+}
+
+static void rt5677_free_gpio(struct i2c_client *i2c)
+{
+}
+#endif
+
 static int rt5677_probe(struct snd_soc_codec *codec)
 {
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
@@ -3422,6 +3552,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 					RT5677_GPIO5_DIR_OUT);
 	}
 
+	rt5677_init_gpio(i2c);
+
 	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
 				      rt5677_dai, ARRAY_SIZE(rt5677_dai));
 }
@@ -3429,6 +3561,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 static int rt5677_i2c_remove(struct i2c_client *i2c)
 {
 	snd_soc_unregister_codec(&i2c->dev);
+	rt5677_free_gpio(i2c);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index a334eb66cfc1..b61b72cfcbd7 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -1363,10 +1363,109 @@
 #define RT5677_SEL_SRC_IB01			(0x1 << 0)
 #define RT5677_SEL_SRC_IB01_SFT			0
 
+/* GPIO status (0xbf) */
+#define RT5677_GPIO6_STATUS_MASK		(0x1 << 5)
+#define RT5677_GPIO6_STATUS_SFT			5
+#define RT5677_GPIO5_STATUS_MASK		(0x1 << 4)
+#define RT5677_GPIO5_STATUS_SFT			4
+#define RT5677_GPIO4_STATUS_MASK		(0x1 << 3)
+#define RT5677_GPIO4_STATUS_SFT			3
+#define RT5677_GPIO3_STATUS_MASK		(0x1 << 2)
+#define RT5677_GPIO3_STATUS_SFT			2
+#define RT5677_GPIO2_STATUS_MASK		(0x1 << 1)
+#define RT5677_GPIO2_STATUS_SFT			1
+#define RT5677_GPIO1_STATUS_MASK		(0x1 << 0)
+#define RT5677_GPIO1_STATUS_SFT			0
+
+/* GPIO Control 1 (0xc0) */
+#define RT5677_GPIO1_PIN_MASK			(0x1 << 15)
+#define RT5677_GPIO1_PIN_SFT			15
+#define RT5677_GPIO1_PIN_GPIO1			(0x0 << 15)
+#define RT5677_GPIO1_PIN_IRQ			(0x1 << 15)
+#define RT5677_IPTV_MODE_MASK			(0x1 << 14)
+#define RT5677_IPTV_MODE_SFT			14
+#define RT5677_IPTV_MODE_GPIO			(0x0 << 14)
+#define RT5677_IPTV_MODE_IPTV			(0x1 << 14)
+#define RT5677_FUNC_MODE_MASK			(0x1 << 13)
+#define RT5677_FUNC_MODE_SFT			13
+#define RT5677_FUNC_MODE_DMIC_GPIO		(0x0 << 13)
+#define RT5677_FUNC_MODE_JTAG			(0x1 << 13)
+
 /* GPIO Control 2 (0xc1) */
 #define RT5677_GPIO5_DIR_MASK			(0x1 << 14)
+#define RT5677_GPIO5_DIR_SFT			14
 #define RT5677_GPIO5_DIR_IN			(0x0 << 14)
 #define RT5677_GPIO5_DIR_OUT			(0x1 << 14)
+#define RT5677_GPIO5_OUT_MASK			(0x1 << 13)
+#define RT5677_GPIO5_OUT_SFT			13
+#define RT5677_GPIO5_OUT_LO			(0x0 << 13)
+#define RT5677_GPIO5_OUT_HI			(0x1 << 13)
+#define RT5677_GPIO5_P_MASK			(0x1 << 12)
+#define RT5677_GPIO5_P_SFT			12
+#define RT5677_GPIO5_P_NOR			(0x0 << 12)
+#define RT5677_GPIO5_P_INV			(0x1 << 12)
+#define RT5677_GPIO4_DIR_MASK			(0x1 << 11)
+#define RT5677_GPIO4_DIR_SFT			11
+#define RT5677_GPIO4_DIR_IN			(0x0 << 11)
+#define RT5677_GPIO4_DIR_OUT			(0x1 << 11)
+#define RT5677_GPIO4_OUT_MASK			(0x1 << 10)
+#define RT5677_GPIO4_OUT_SFT			10
+#define RT5677_GPIO4_OUT_LO			(0x0 << 10)
+#define RT5677_GPIO4_OUT_HI			(0x1 << 10)
+#define RT5677_GPIO4_P_MASK			(0x1 << 9)
+#define RT5677_GPIO4_P_SFT			9
+#define RT5677_GPIO4_P_NOR			(0x0 << 9)
+#define RT5677_GPIO4_P_INV			(0x1 << 9)
+#define RT5677_GPIO3_DIR_MASK			(0x1 << 8)
+#define RT5677_GPIO3_DIR_SFT			8
+#define RT5677_GPIO3_DIR_IN			(0x0 << 8)
+#define RT5677_GPIO3_DIR_OUT			(0x1 << 8)
+#define RT5677_GPIO3_OUT_MASK			(0x1 << 7)
+#define RT5677_GPIO3_OUT_SFT			7
+#define RT5677_GPIO3_OUT_LO			(0x0 << 7)
+#define RT5677_GPIO3_OUT_HI			(0x1 << 7)
+#define RT5677_GPIO3_P_MASK			(0x1 << 6)
+#define RT5677_GPIO3_P_SFT			6
+#define RT5677_GPIO3_P_NOR			(0x0 << 6)
+#define RT5677_GPIO3_P_INV			(0x1 << 6)
+#define RT5677_GPIO2_DIR_MASK			(0x1 << 5)
+#define RT5677_GPIO2_DIR_SFT			5
+#define RT5677_GPIO2_DIR_IN			(0x0 << 5)
+#define RT5677_GPIO2_DIR_OUT			(0x1 << 5)
+#define RT5677_GPIO2_OUT_MASK			(0x1 << 4)
+#define RT5677_GPIO2_OUT_SFT			4
+#define RT5677_GPIO2_OUT_LO			(0x0 << 4)
+#define RT5677_GPIO2_OUT_HI			(0x1 << 4)
+#define RT5677_GPIO2_P_MASK			(0x1 << 3)
+#define RT5677_GPIO2_P_SFT			3
+#define RT5677_GPIO2_P_NOR			(0x0 << 3)
+#define RT5677_GPIO2_P_INV			(0x1 << 3)
+#define RT5677_GPIO1_DIR_MASK			(0x1 << 2)
+#define RT5677_GPIO1_DIR_SFT			2
+#define RT5677_GPIO1_DIR_IN			(0x0 << 2)
+#define RT5677_GPIO1_DIR_OUT			(0x1 << 2)
+#define RT5677_GPIO1_OUT_MASK			(0x1 << 1)
+#define RT5677_GPIO1_OUT_SFT			1
+#define RT5677_GPIO1_OUT_LO			(0x0 << 1)
+#define RT5677_GPIO1_OUT_HI			(0x1 << 1)
+#define RT5677_GPIO1_P_MASK			(0x1 << 0)
+#define RT5677_GPIO1_P_SFT			0
+#define RT5677_GPIO1_P_NOR			(0x0 << 0)
+#define RT5677_GPIO1_P_INV			(0x1 << 0)
+
+/* GPIO Control 3 (0xc2) */
+#define RT5677_GPIO6_DIR_MASK			(0x1 << 2)
+#define RT5677_GPIO6_DIR_SFT			2
+#define RT5677_GPIO6_DIR_IN			(0x0 << 2)
+#define RT5677_GPIO6_DIR_OUT			(0x1 << 2)
+#define RT5677_GPIO6_OUT_MASK			(0x1 << 1)
+#define RT5677_GPIO6_OUT_SFT			1
+#define RT5677_GPIO6_OUT_LO			(0x0 << 1)
+#define RT5677_GPIO6_OUT_HI			(0x1 << 1)
+#define RT5677_GPIO6_P_MASK			(0x1 << 0)
+#define RT5677_GPIO6_P_SFT			0
+#define RT5677_GPIO6_P_NOR			(0x0 << 0)
+#define RT5677_GPIO6_P_INV			(0x1 << 0)
 
 /* Virtual DSP Mixer Control (0xf7 0xf8 0xf9) */
 #define RT5677_DSP_IB_01_H			(0x1 << 15)
@@ -1428,6 +1527,16 @@ enum {
 	RT5677_AIFS,
 };
 
+enum {
+	RT5677_GPIO1,
+	RT5677_GPIO2,
+	RT5677_GPIO3,
+	RT5677_GPIO4,
+	RT5677_GPIO5,
+	RT5677_GPIO6,
+	RT5677_GPIO_NUM,
+};
+
 struct rt5677_priv {
 	struct snd_soc_codec *codec;
 	struct rt5677_platform_data pdata;
@@ -1441,6 +1550,9 @@ struct rt5677_priv {
 	int pll_src;
 	int pll_in;
 	int pll_out;
+#ifdef CONFIG_GPIOLIB
+	struct gpio_chip gpio_chip;
+#endif
 };
 
 #endif /* __RT5677_H__ */
-- 
cgit v1.2.3


From 5d5e63af998026f0340d1081fb15ad3c26d80c81 Mon Sep 17 00:00:00 2001
From: Axel Lin <axel.lin@ingics.com>
Date: Wed, 17 Sep 2014 20:58:02 +0800
Subject: ASoC: Remove return value checking for gpiochip_remove()

gpiochip_remove() will return void eventually.
Thus this patch removes return value checking for gpiochip_remove().

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/codecs/rt5677.c | 5 +----
 sound/soc/codecs/wm5100.c | 5 +----
 sound/soc/codecs/wm8903.c | 6 +-----
 sound/soc/codecs/wm8962.c | 5 +----
 sound/soc/codecs/wm8996.c | 6 +-----
 5 files changed, 5 insertions(+), 22 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 02bc8bd7caeb..991409f90fd3 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3274,11 +3274,8 @@ static void rt5677_init_gpio(struct i2c_client *i2c)
 static void rt5677_free_gpio(struct i2c_client *i2c)
 {
 	struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
-	int ret;
 
-	ret = gpiochip_remove(&rt5677->gpio_chip);
-	if (ret != 0)
-		dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&rt5677->gpio_chip);
 }
 #else
 static void rt5677_init_gpio(struct i2c_client *i2c)
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 7bb0d36d4c54..a01ad629ed61 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2319,11 +2319,8 @@ static void wm5100_init_gpio(struct i2c_client *i2c)
 static void wm5100_free_gpio(struct i2c_client *i2c)
 {
 	struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c);
-	int ret;
 
-	ret = gpiochip_remove(&wm5100->gpio_chip);
-	if (ret != 0)
-		dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&wm5100->gpio_chip);
 }
 #else
 static void wm5100_init_gpio(struct i2c_client *i2c)
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index aa0984864e76..c038b3e04398 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1877,11 +1877,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903)
 
 static void wm8903_free_gpio(struct wm8903_priv *wm8903)
 {
-	int ret;
-
-	ret = gpiochip_remove(&wm8903->gpio_chip);
-	if (ret != 0)
-		dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&wm8903->gpio_chip);
 }
 #else
 static void wm8903_init_gpio(struct wm8903_priv *wm8903)
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 1098ae32f1f9..9077411e62ce 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3398,11 +3398,8 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec)
 static void wm8962_free_gpio(struct snd_soc_codec *codec)
 {
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
-	int ret;
 
-	ret = gpiochip_remove(&wm8962->gpio_chip);
-	if (ret != 0)
-		dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&wm8962->gpio_chip);
 }
 #else
 static void wm8962_init_gpio(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index f16ff4f56923..b1dcc11c1b23 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2216,11 +2216,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996)
 
 static void wm8996_free_gpio(struct wm8996_priv *wm8996)
 {
-	int ret;
-
-	ret = gpiochip_remove(&wm8996->gpio_chip);
-	if (ret != 0)
-		dev_err(wm8996->dev, "Failed to remove GPIOs: %d\n", ret);
+	gpiochip_remove(&wm8996->gpio_chip);
 }
 #else
 static void wm8996_init_gpio(struct wm8996_priv *wm8996)
-- 
cgit v1.2.3


From 1cc0c054f380c1c477642b5d9d9d9f697f641dbc Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Wed, 1 Oct 2014 16:02:11 +0300
Subject: ASoC: davinci-mcasp: Convert the context save/restore to use array

Instead of individual values use an array to store the registers need to be
saved on suspend and restored on resume.
It is going to be easier to add more registers to save and restore.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/davinci/davinci-mcasp.c | 38 +++++++++++++++++---------------------
 1 file changed, 17 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index c28508da34cf..63e24449eb89 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -42,14 +42,18 @@
 
 #define MCASP_MAX_AFIFO_DEPTH	64
 
+static u32 context_regs[] = {
+	DAVINCI_MCASP_TXFMCTL_REG,
+	DAVINCI_MCASP_RXFMCTL_REG,
+	DAVINCI_MCASP_TXFMT_REG,
+	DAVINCI_MCASP_RXFMT_REG,
+	DAVINCI_MCASP_ACLKXCTL_REG,
+	DAVINCI_MCASP_ACLKRCTL_REG,
+	DAVINCI_MCASP_PDIR_REG,
+};
+
 struct davinci_mcasp_context {
-	u32	txfmtctl;
-	u32	rxfmtctl;
-	u32	txfmt;
-	u32	rxfmt;
-	u32	aclkxctl;
-	u32	aclkrctl;
-	u32	pdir;
+	u32	config_regs[ARRAY_SIZE(context_regs)];
 };
 
 struct davinci_mcasp {
@@ -857,14 +861,10 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 	struct davinci_mcasp_context *context = &mcasp->context;
+	int i;
 
-	context->txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG);
-	context->rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
-	context->txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG);
-	context->rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG);
-	context->aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
-	context->aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG);
-	context->pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG);
+	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+		context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]);
 
 	return 0;
 }
@@ -873,14 +873,10 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 	struct davinci_mcasp_context *context = &mcasp->context;
+	int i;
 
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, context->txfmtctl);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, context->rxfmtctl);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, context->txfmt);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, context->rxfmt);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, context->aclkxctl);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, context->aclkrctl);
-	mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, context->pdir);
+	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+		mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]);
 
 	return 0;
 }
-- 
cgit v1.2.3


From f114ce605daa1fb9d4efa253ea6d5bd4802902af Mon Sep 17 00:00:00 2001
From: Peter Ujfalusi <peter.ujfalusi@ti.com>
Date: Wed, 1 Oct 2014 16:02:12 +0300
Subject: ASoC: davinvi-mcasp: Proper suspend/resume support while audio is
 active

If the board is sent to suspend (deep sleep) the McASP context will be lost.
In case when suspend happens during active audio we need to save and restore
more registers, which was configured during hw_param times as well.
We need to add more config registers, AFIFO control registers and we also
need to save and restore the serializer configuration as well.
Since the number of serializers depends on the SoC we need to allocate the
memory for it based on the num_serializer for the given McASP instance.

With this patch the ongoing stream will resume after resuming from deep
sleep.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/davinci/davinci-mcasp.c | 41 +++++++++++++++++++++++++++++++++++++++
 1 file changed, 41 insertions(+)

(limited to 'sound')

diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 63e24449eb89..5dcacc495438 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -49,11 +49,19 @@ static u32 context_regs[] = {
 	DAVINCI_MCASP_RXFMT_REG,
 	DAVINCI_MCASP_ACLKXCTL_REG,
 	DAVINCI_MCASP_ACLKRCTL_REG,
+	DAVINCI_MCASP_AHCLKXCTL_REG,
+	DAVINCI_MCASP_AHCLKRCTL_REG,
 	DAVINCI_MCASP_PDIR_REG,
+	DAVINCI_MCASP_RXMASK_REG,
+	DAVINCI_MCASP_TXMASK_REG,
+	DAVINCI_MCASP_RXTDM_REG,
+	DAVINCI_MCASP_TXTDM_REG,
 };
 
 struct davinci_mcasp_context {
 	u32	config_regs[ARRAY_SIZE(context_regs)];
+	u32	afifo_regs[2]; /* for read/write fifo control registers */
+	u32	*xrsr_regs; /* for serializer configuration */
 };
 
 struct davinci_mcasp {
@@ -861,11 +869,25 @@ static int davinci_mcasp_suspend(struct snd_soc_dai *dai)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 	struct davinci_mcasp_context *context = &mcasp->context;
+	u32 reg;
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
 		context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]);
 
+	if (mcasp->txnumevt) {
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		context->afifo_regs[0] = mcasp_get_reg(mcasp, reg);
+	}
+	if (mcasp->rxnumevt) {
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		context->afifo_regs[1] = mcasp_get_reg(mcasp, reg);
+	}
+
+	for (i = 0; i < mcasp->num_serializer; i++)
+		context->xrsr_regs[i] = mcasp_get_reg(mcasp,
+						DAVINCI_MCASP_XRSRCTL_REG(i));
+
 	return 0;
 }
 
@@ -873,11 +895,25 @@ static int davinci_mcasp_resume(struct snd_soc_dai *dai)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
 	struct davinci_mcasp_context *context = &mcasp->context;
+	u32 reg;
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
 		mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]);
 
+	if (mcasp->txnumevt) {
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		mcasp_set_reg(mcasp, reg, context->afifo_regs[0]);
+	}
+	if (mcasp->rxnumevt) {
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		mcasp_set_reg(mcasp, reg, context->afifo_regs[1]);
+	}
+
+	for (i = 0; i < mcasp->num_serializer; i++)
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+			      context->xrsr_regs[i]);
+
 	return 0;
 }
 #else
@@ -1195,6 +1231,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	mcasp->op_mode = pdata->op_mode;
 	mcasp->tdm_slots = pdata->tdm_slots;
 	mcasp->num_serializer = pdata->num_serializer;
+#ifdef CONFIG_PM_SLEEP
+	mcasp->context.xrsr_regs = devm_kzalloc(&pdev->dev,
+					sizeof(u32) * mcasp->num_serializer,
+					GFP_KERNEL);
+#endif
 	mcasp->serial_dir = pdata->serial_dir;
 	mcasp->version = pdata->version;
 	mcasp->txnumevt = pdata->txnumevt;
-- 
cgit v1.2.3


From c05a11f7b8b5bc67f2c9f726c52b59f67b1bfe7d Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@freescale.com>
Date: Tue, 30 Sep 2014 16:52:15 -0300
Subject: ASoC: fsl: Do not force codecs selection by SND_SOC_FSL_ASOC_CARD

The wm8962 driver uses the input subsystem, but it is selected by
SND_SOC_FSL_ASOC_CARD, which can be built with CONFIG_INPUT disabled,
resulting in this link error:

ERROR: "input_event" [sound/soc/codecs/snd-soc-wm8962.ko] undefined!
ERROR: "input_register_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined!
ERROR: "devm_input_allocate_device" [sound/soc/codecs/snd-soc-wm8962.ko] undefined!

Do not force the selection of the codecs by SND_SOC_FSL_ASOC_CARD to avoid
such problem.

Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 sound/soc/fsl/Kconfig | 3 ---
 1 file changed, 3 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 7c1da8ede975..0f23d1ae5be7 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -289,9 +289,6 @@ config SND_SOC_FSL_ASOC_CARD
 	select SND_SOC_FSL_ESAI
 	select SND_SOC_FSL_SAI
 	select SND_SOC_FSL_SSI
-	select SND_SOC_CS42XX8_I2C
-	select SND_SOC_SGTL5000
-	select SND_SOC_WM8962
 	help
 	 ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
 	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
-- 
cgit v1.2.3