Merge 3.14-rc5 into driver-core-next

We want the fixes in here.

24 files changed, 412 insertions, 261 deletions

diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 54f74f8cbb75..4544d8eb1452 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -11,7 +11,7 @@ config SND_BF5XX_I2S
 
 config SND_BF5XX_SOC_SSM2602
 	tristate "SoC SSM2602 Audio Codec Add-On Card support"
-	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+	depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
 	select SND_BF5XX_SOC_I2S if !BF60x
 	select SND_BF6XX_SOC_I2S if BF60x
 	select SND_SOC_SSM2602
@@ -21,10 +21,9 @@ config SND_BF5XX_SOC_SSM2602
 
 config SND_SOC_BFIN_EVAL_ADAU1701
 	tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
-	depends on SND_BF5XX_I2S
+	depends on SND_BF5XX_I2S && I2C
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_ADAU1701
-	select I2C
 	help
 	  Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ
 	  board connected to one of the Blackfin evaluation boards like the
@@ -45,7 +44,7 @@ config SND_SOC_BFIN_EVAL_ADAU1373
 
 config SND_SOC_BFIN_EVAL_ADAV80X
 	tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
-	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+	depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_ADAV80X
 	help
@@ -58,7 +57,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X
 
 config SND_BF5XX_SOC_AD1836
 	tristate "SoC AD1836 Audio support for BF5xx"
-	depends on SND_BF5XX_I2S
+	depends on SND_BF5XX_I2S && SPI_MASTER
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_AD1836
 	help
@@ -66,7 +65,7 @@ config SND_BF5XX_SOC_AD1836
 
 config SND_BF5XX_SOC_AD193X
 	tristate "SoC AD193X Audio support for Blackfin"
-	depends on SND_BF5XX_I2S
+	depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_AD193X
 	help
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 7257a8885f42..34d965a4a040 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -57,8 +57,8 @@ static const u16 ad1980_reg[] = {
 static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line",
 		"Stereo Mix", "Mono Mix", "Phone"};
 
-static const struct soc_enum ad1980_cap_src =
-	SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel);
+static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src,
+			    AC97_REC_SEL, 8, 0, ad1980_rec_sel);
 
 static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = {
 SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index f295b6569910..f4d965ebc29e 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = {
 	},
 };
 
+static bool da732x_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case DA732X_REG_HPL_DAC_OFF_CNTL:
+	case DA732X_REG_HPR_DAC_OFF_CNTL:
+		return true;
+	default:
+		return false;
+	}
+}
+
 static const struct regmap_config da732x_regmap = {
 	.reg_bits		= 8,
 	.val_bits		= 8,
 
 	.max_register		= DA732X_MAX_REG,
+	.volatile_reg		= da732x_volatile,
 	.reg_defaults		= da732x_reg_cache,
 	.num_reg_defaults	= ARRAY_SIZE(da732x_reg_cache),
 	.cache_type		= REGCACHE_RBTREE,
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index 52b79a487ac7..422812613a28 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1523,8 +1523,15 @@ static int da9055_remove(struct i2c_client *client)
 	return 0;
 }
 
+/*
+ * DO NOT change the device Ids. The naming is intentionally specific as both
+ * the CODEC and PMIC parts of this chip are instantiated separately as I2C
+ * devices (both have configurable I2C addresses, and are to all intents and
+ * purposes separate). As a result there are specific DA9055 Ids for CODEC
+ * and PMIC, which must be different to operate together.
+ */
 static const struct i2c_device_id da9055_i2c_id[] = {
-	{ "da9055", 0 },
+	{ "da9055-codec", 0 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
@@ -1532,7 +1539,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id);
 /* I2C codec control layer */
 static struct i2c_driver da9055_i2c_driver = {
 	.driver = {
-		.name = "da9055",
+		.name = "da9055-codec",
 		.owner = THIS_MODULE,
 	},
 	.probe		= da9055_i2c_probe,
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index 5839048ec467..cb736ddc446d 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -140,13 +140,17 @@ static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"};
 static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"};
 
 static const struct soc_enum isabelle_rx1_enum[] = {
-	SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts),
-	SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts),
+	SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3,
+			ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts),
+	SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5,
+			ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts),
 };
 
 static const struct soc_enum isabelle_rx2_enum[] = {
-	SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts),
-	SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts),
+	SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2,
+			ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts),
+	SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4,
+			ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts),
 };
 
 /* Headset DAC playback switches */
@@ -161,13 +165,17 @@ static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"};
 static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"};
 
 static const struct soc_enum isabelle_atx_enum[] = {
-	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts),
-	SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts),
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7,
+			ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts),
+	SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0,
+			ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts),
 };
 
 static const struct soc_enum isabelle_vtx_enum[] = {
-	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts),
-	SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts),
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6,
+			ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts),
+	SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0,
+			ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts),
 };
 
 static const struct snd_kcontrol_new atx_mux_controls =
@@ -183,17 +191,13 @@ static const char *isabelle_amic1_texts[] = {
 /* Left analog microphone selection */
 static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"};
 
-static const struct soc_enum isabelle_amic1_enum[] = {
-	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5,
-			ARRAY_SIZE(isabelle_amic1_texts),
-			isabelle_amic1_texts),
-};
+static SOC_ENUM_SINGLE_DECL(isabelle_amic1_enum,
+			    ISABELLE_AMIC_CFG_REG, 5,
+			    isabelle_amic1_texts);
 
-static const struct soc_enum isabelle_amic2_enum[] = {
-	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4,
-			ARRAY_SIZE(isabelle_amic2_texts),
-			isabelle_amic2_texts),
-};
+static SOC_ENUM_SINGLE_DECL(isabelle_amic2_enum,
+			    ISABELLE_AMIC_CFG_REG, 4,
+			    isabelle_amic2_texts);
 
 static const struct snd_kcontrol_new amic1_control =
 	SOC_DAPM_ENUM("Route", isabelle_amic1_enum);
@@ -206,16 +210,20 @@ static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"};
 static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"};
 
 static const struct soc_enum isabelle_st_audio_enum[] = {
-	SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1,
+	SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7,
+			ARRAY_SIZE(isabelle_st_audio_texts),
 			isabelle_st_audio_texts),
-	SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1,
+	SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7,
+			ARRAY_SIZE(isabelle_st_audio_texts),
 			isabelle_st_audio_texts),
 };
 
 static const struct soc_enum isabelle_st_voice_enum[] = {
-	SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1,
+	SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7,
+			ARRAY_SIZE(isabelle_st_voice_texts),
 			isabelle_st_voice_texts),
-	SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1,
+	SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7,
+			ARRAY_SIZE(isabelle_st_voice_texts),
 			isabelle_st_voice_texts),
 };
 
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 51f9b3d16b41..9f714ea86613 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -336,6 +336,7 @@ static bool max98090_readable_register(struct device *dev, unsigned int reg)
 	case M98090_REG_RECORD_TDM_SLOT:
 	case M98090_REG_SAMPLE_RATE:
 	case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E:
+	case M98090_REG_REVISION_ID:
 		return true;
 	default:
 		return false;
@@ -1769,16 +1770,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			ret = regcache_sync(max98090->regmap);
-
-			if (ret != 0) {
-				dev_err(codec->dev,
-					"Failed to sync cache: %d\n", ret);
-				return ret;
-			}
-		}
-
 		if (max98090->jack_state == M98090_JACK_STATE_HEADSET) {
 			/*
 			 * Set to normal bias level.
@@ -1792,6 +1783,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = regcache_sync(max98090->regmap);
+			if (ret != 0) {
+				dev_err(codec->dev,
+					"Failed to sync cache: %d\n", ret);
+				return ret;
+			}
+		}
+		break;
+
 	case SND_SOC_BIAS_OFF:
 		/* Set internal pull-up to lowest power mode */
 		snd_soc_update_bits(codec, M98090_REG_JACK_DETECT,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index a3fb41179636..886924934aa5 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2093,6 +2093,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id);
 #ifdef CONFIG_ACPI
 static struct acpi_device_id rt5640_acpi_match[] = {
 	{ "INT33CA", 0 },
+	{ "10EC5640", 0 },
 	{ },
 };
 MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match);
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 06edb396e733..2735361a4c3c 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -187,42 +187,42 @@ static const unsigned int sta32x_limiter_drc_release_tlv[] = {
 	13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0),
 };
 
-static const struct soc_enum sta32x_drc_ac_enum =
-	SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
-			2, sta32x_drc_ac);
-static const struct soc_enum sta32x_auto_eq_enum =
-	SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
-			3, sta32x_auto_eq_mode);
-static const struct soc_enum sta32x_auto_gc_enum =
-	SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
-			4, sta32x_auto_gc_mode);
-static const struct soc_enum sta32x_auto_xo_enum =
-	SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
-			16, sta32x_auto_xo_mode);
-static const struct soc_enum sta32x_preset_eq_enum =
-	SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
-			32, sta32x_preset_eq_mode);
-static const struct soc_enum sta32x_limiter_ch1_enum =
-	SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
-			3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter_ch2_enum =
-	SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
-			3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter_ch3_enum =
-	SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
-			3, sta32x_limiter_select);
-static const struct soc_enum sta32x_limiter1_attack_rate_enum =
-	SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT,
-			16, sta32x_limiter_attack_rate);
-static const struct soc_enum sta32x_limiter2_attack_rate_enum =
-	SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT,
-			16, sta32x_limiter_attack_rate);
-static const struct soc_enum sta32x_limiter1_release_rate_enum =
-	SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT,
-			16, sta32x_limiter_release_rate);
-static const struct soc_enum sta32x_limiter2_release_rate_enum =
-	SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT,
-			16, sta32x_limiter_release_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_drc_ac_enum,
+			    STA32X_CONFD, STA32X_CONFD_DRC_SHIFT,
+			    sta32x_drc_ac);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_eq_enum,
+			    STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT,
+			    sta32x_auto_eq_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_gc_enum,
+			    STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT,
+			    sta32x_auto_gc_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_auto_xo_enum,
+			    STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT,
+			    sta32x_auto_xo_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_preset_eq_enum,
+			    STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT,
+			    sta32x_preset_eq_mode);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch1_enum,
+			    STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT,
+			    sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch2_enum,
+			    STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT,
+			    sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch3_enum,
+			    STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT,
+			    sta32x_limiter_select);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_attack_rate_enum,
+			    STA32X_L1AR, STA32X_LxA_SHIFT,
+			    sta32x_limiter_attack_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_attack_rate_enum,
+			    STA32X_L2AR, STA32X_LxA_SHIFT,
+			    sta32x_limiter_attack_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_release_rate_enum,
+			    STA32X_L1AR, STA32X_LxR_SHIFT,
+			    sta32x_limiter_release_rate);
+static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum,
+			    STA32X_L2AR, STA32X_LxR_SHIFT,
+			    sta32x_limiter_release_rate);
 
 /* byte array controls for setting biquad, mixer, scaling coefficients;
  * for biquads all five coefficients need to be set in one go,
@@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec)
 
 static int sta32x_cache_sync(struct snd_soc_codec *codec)
 {
-	struct sta32x_priv *sta32x = codec->control_data;
+	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
 	unsigned int mute;
 	int rc;
 
@@ -434,7 +434,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0,
 SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum),
 SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum),
 SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
-SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum),
+SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum),
 
 /* depending on mode, the attack/release thresholds have
  * two different enum definitions; provide both
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 48dc7d2fee36..6d684d934f4d 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -117,19 +117,23 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
 static const char *wm8400_digital_sidetone[] =
 	{"None", "Left ADC", "Right ADC", "Reserved"};
 
-static const struct soc_enum wm8400_left_digital_sidetone_enum =
-SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
-		WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8400_left_digital_sidetone_enum,
+			    WM8400_DIGITAL_SIDE_TONE,
+			    WM8400_ADC_TO_DACL_SHIFT,
+			    wm8400_digital_sidetone);
 
-static const struct soc_enum wm8400_right_digital_sidetone_enum =
-SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
-		WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone);
+static SOC_ENUM_SINGLE_DECL(wm8400_right_digital_sidetone_enum,
+			    WM8400_DIGITAL_SIDE_TONE,
+			    WM8400_ADC_TO_DACR_SHIFT,
+			    wm8400_digital_sidetone);
 
 static const char *wm8400_adcmode[] =
 	{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
 
-static const struct soc_enum wm8400_right_adcmode_enum =
-SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode);
+static SOC_ENUM_SINGLE_DECL(wm8400_right_adcmode_enum,
+			    WM8400_ADC_CTRL,
+			    WM8400_ADC_HPF_CUT_SHIFT,
+			    wm8400_adcmode);
 
 static const struct snd_kcontrol_new wm8400_snd_controls[] = {
 /* INMIXL */
@@ -422,9 +426,10 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
 static const char *wm8400_ainlmux[] =
 	{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
 
-static const struct soc_enum wm8400_ainlmux_enum =
-SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT,
-	ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux);
+static SOC_ENUM_SINGLE_DECL(wm8400_ainlmux_enum,
+			    WM8400_INPUT_MIXER1,
+			    WM8400_AINLMODE_SHIFT,
+			    wm8400_ainlmux);
 
 static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls =
 SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum);
@@ -435,9 +440,10 @@ SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum);
 static const char *wm8400_ainrmux[] =
 	{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
 
-static const struct soc_enum wm8400_ainrmux_enum =
-SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT,
-	ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux);
+static SOC_ENUM_SINGLE_DECL(wm8400_ainrmux_enum,
+			    WM8400_INPUT_MIXER1,
+			    WM8400_AINRMODE_SHIFT,
+			    wm8400_ainrmux);
 
 static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls =
 SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum);
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 89a18d82f303..5bce21013485 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -196,8 +196,8 @@ static const char *ain_text[] = {
 	"AIN5", "AIN6", "AIN7", "AIN8"
 };
 
-static const struct soc_enum ain_enum =
-	SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text);
+static SOC_ENUM_DOUBLE_DECL(ain_enum,
+			    WM8770_ADCMUX, 0, 4, ain_text);
 
 static const struct snd_kcontrol_new ain_mux =
 	SOC_DAPM_ENUM("Capture Mux", ain_enum);
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index e98bc7038a08..43c2201cb901 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -304,53 +304,53 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1);
 
 static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" };
 
-static const struct soc_enum mic_bias_level =
-SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt);
+static SOC_ENUM_SINGLE_DECL(mic_bias_level,
+			    WM8900_REG_INCTL, 8, mic_bias_level_txt);
 
 static const char *dac_mute_rate_txt[] = { "Fast", "Slow" };
 
-static const struct soc_enum dac_mute_rate =
-SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt);
+static SOC_ENUM_SINGLE_DECL(dac_mute_rate,
+			    WM8900_REG_DACCTRL, 7, dac_mute_rate_txt);
 
 static const char *dac_deemphasis_txt[] = {
 	"Disabled", "32kHz", "44.1kHz", "48kHz"
 };
 
-static const struct soc_enum dac_deemphasis =
-SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt);
+static SOC_ENUM_SINGLE_DECL(dac_deemphasis,
+			    WM8900_REG_DACCTRL, 4, dac_deemphasis_txt);
 
 static const char *adc_hpf_cut_txt[] = {
 	"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"
 };
 
-static const struct soc_enum adc_hpf_cut =
-SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt);
+static SOC_ENUM_SINGLE_DECL(adc_hpf_cut,
+			    WM8900_REG_ADCCTRL, 5, adc_hpf_cut_txt);
 
 static const char *lr_txt[] = {
 	"Left", "Right"
 };
 
-static const struct soc_enum aifl_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(aifl_src,
+			    WM8900_REG_AUDIO1, 15, lr_txt);
 
-static const struct soc_enum aifr_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(aifr_src,
+			    WM8900_REG_AUDIO1, 14, lr_txt);
 
-static const struct soc_enum dacl_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(dacl_src,
+			    WM8900_REG_AUDIO2, 15, lr_txt);
 
-static const struct soc_enum dacr_src =
-SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt);
+static SOC_ENUM_SINGLE_DECL(dacr_src,
+			    WM8900_REG_AUDIO2, 14, lr_txt);
 
 static const char *sidetone_txt[] = {
 	"Disabled", "Left ADC", "Right ADC"
 };
 
-static const struct soc_enum dacl_sidetone =
-SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt);
+static SOC_ENUM_SINGLE_DECL(dacl_sidetone,
+			    WM8900_REG_SIDETONE, 2, sidetone_txt);
 
-static const struct soc_enum dacr_sidetone =
-SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt);
+static SOC_ENUM_SINGLE_DECL(dacr_sidetone,
+			    WM8900_REG_SIDETONE, 0, sidetone_txt);
 
 static const struct snd_kcontrol_new wm8900_snd_controls[] = {
 SOC_ENUM("Mic Bias Level", mic_bias_level),
@@ -496,8 +496,8 @@ SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0),
 
 static const char *wm8900_lp_mux[] = { "Disabled", "Enabled" };
 
-static const struct soc_enum wm8900_lineout2_lp_mux =
-SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm8900_lp_mux);
+static SOC_ENUM_SINGLE_DECL(wm8900_lineout2_lp_mux,
+			    WM8900_REG_LOUTMIXCTL1, 1, wm8900_lp_mux);
 
 static const struct snd_kcontrol_new wm8900_lineout2_lp =
 SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux);
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index b7488f190d2b..d4248e00160e 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
 
 			data32 &= 0xffffff;
 
-			wm8994_bulk_write(codec->control_data,
+			wm8994_bulk_write(wm8994->wm8994,
 					  data32 & 0xffffff,
 					  block_len / 2,
 					  (void *)(data + 8));
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 433d59a0f3ef..2ee23a39622c 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec)
 	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 
 	wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index b9be9cbc4603..adb72063d44e 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -265,21 +265,21 @@ static const char *sidetone_hpf_text[] = {
 	"2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz"
 };
 
-static const struct soc_enum sidetone_hpf =
-	SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text);
+static SOC_ENUM_SINGLE_DECL(sidetone_hpf,
+			    WM8994_SIDETONE, 7, sidetone_hpf_text);
 
 static const char *adc_hpf_text[] = {
 	"HiFi", "Voice 1", "Voice 2", "Voice 3"
 };
 
-static const struct soc_enum aif1adc1_hpf =
-	SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif1adc1_hpf,
+			    WM8994_AIF1_ADC1_FILTERS, 13, adc_hpf_text);
 
-static const struct soc_enum aif1adc2_hpf =
-	SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif1adc2_hpf,
+			    WM8994_AIF1_ADC2_FILTERS, 13, adc_hpf_text);
 
-static const struct soc_enum aif2adc_hpf =
-	SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text);
+static SOC_ENUM_SINGLE_DECL(aif2adc_hpf,
+			    WM8994_AIF2_ADC_FILTERS, 13, adc_hpf_text);
 
 static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0);
 static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
@@ -501,39 +501,39 @@ static const char *aif_chan_src_text[] = {
 	"Left", "Right"
 };
 
-static const struct soc_enum aif1adcl_src =
-	SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1adcl_src,
+			    WM8994_AIF1_CONTROL_1, 15, aif_chan_src_text);
 
-static const struct soc_enum aif1adcr_src =
-	SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1adcr_src,
+			    WM8994_AIF1_CONTROL_1, 14, aif_chan_src_text);
 
-static const struct soc_enum aif2adcl_src =
-	SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2adcl_src,
+			    WM8994_AIF2_CONTROL_1, 15, aif_chan_src_text);
 
-static const struct soc_enum aif2adcr_src =
-	SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2adcr_src,
+			    WM8994_AIF2_CONTROL_1, 14, aif_chan_src_text);
 
-static const struct soc_enum aif1dacl_src =
-	SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1dacl_src,
+			    WM8994_AIF1_CONTROL_2, 15, aif_chan_src_text);
 
-static const struct soc_enum aif1dacr_src =
-	SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif1dacr_src,
+			    WM8994_AIF1_CONTROL_2, 14, aif_chan_src_text);
 
-static const struct soc_enum aif2dacl_src =
-	SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacl_src,
+			    WM8994_AIF2_CONTROL_2, 15, aif_chan_src_text);
 
-static const struct soc_enum aif2dacr_src =
-	SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacr_src,
+			    WM8994_AIF2_CONTROL_2, 14, aif_chan_src_text);
 
 static const char *osr_text[] = {
 	"Low Power", "High Performance",
 };
 
-static const struct soc_enum dac_osr =
-	SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(dac_osr,
+			    WM8994_OVERSAMPLING, 0, osr_text);
 
-static const struct soc_enum adc_osr =
-	SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text);
+static SOC_ENUM_SINGLE_DECL(adc_osr,
+			    WM8994_OVERSAMPLING, 1, osr_text);
 
 static const struct snd_kcontrol_new wm8994_snd_controls[] = {
 SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME,
@@ -690,17 +690,20 @@ static const char *wm8958_ng_text[] = {
 	"30ms", "125ms", "250ms", "500ms",
 };
 
-static const struct soc_enum wm8958_aif1dac1_ng_hold =
-	SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE,
-			WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac1_ng_hold,
+			    WM8958_AIF1_DAC1_NOISE_GATE,
+			    WM8958_AIF1DAC1_NG_THR_SHIFT,
+			    wm8958_ng_text);
 
-static const struct soc_enum wm8958_aif1dac2_ng_hold =
-	SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE,
-			WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac2_ng_hold,
+			    WM8958_AIF1_DAC2_NOISE_GATE,
+			    WM8958_AIF1DAC2_NG_THR_SHIFT,
+			    wm8958_ng_text);
 
-static const struct soc_enum wm8958_aif2dac_ng_hold =
-	SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE,
-			WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif2dac_ng_hold,
+			    WM8958_AIF2_DAC_NOISE_GATE,
+			    WM8958_AIF2DAC_NG_THR_SHIFT,
+			    wm8958_ng_text);
 
 static const struct snd_kcontrol_new wm8958_snd_controls[] = {
 SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv),
@@ -1341,8 +1344,8 @@ static const char *adc_mux_text[] = {
 	"DMIC",
 };
 
-static const struct soc_enum adc_enum =
-	SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text);
+static SOC_ENUM_SINGLE_DECL(adc_enum,
+			    0, 0, adc_mux_text);
 
 static const struct snd_kcontrol_new adcl_mux =
 	SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum);
@@ -1478,14 +1481,14 @@ static const char *sidetone_text[] = {
 	"ADC/DMIC1", "DMIC2",
 };
 
-static const struct soc_enum sidetone1_enum =
-	SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone1_enum,
+			    WM8994_SIDETONE, 0, sidetone_text);
 
 static const struct snd_kcontrol_new sidetone1_mux =
 	SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum);
 
-static const struct soc_enum sidetone2_enum =
-	SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text);
+static SOC_ENUM_SINGLE_DECL(sidetone2_enum,
+			    WM8994_SIDETONE, 1, sidetone_text);
 
 static const struct snd_kcontrol_new sidetone2_mux =
 	SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum);
@@ -1498,22 +1501,24 @@ static const char *loopback_text[] = {
 	"None", "ADCDAT",
 };
 
-static const struct soc_enum aif1_loopback_enum =
-	SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2,
-			loopback_text);
+static SOC_ENUM_SINGLE_DECL(aif1_loopback_enum,
+			    WM8994_AIF1_CONTROL_2,
+			    WM8994_AIF1_LOOPBACK_SHIFT,
+			    loopback_text);
 
 static const struct snd_kcontrol_new aif1_loopback =
 	SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum);
 
-static const struct soc_enum aif2_loopback_enum =
-	SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2,
-			loopback_text);
+static SOC_ENUM_SINGLE_DECL(aif2_loopback_enum,
+			    WM8994_AIF2_CONTROL_2,
+			    WM8994_AIF2_LOOPBACK_SHIFT,
+			    loopback_text);
 
 static const struct snd_kcontrol_new aif2_loopback =
 	SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum);
 
-static const struct soc_enum aif1dac_enum =
-	SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text);
+static SOC_ENUM_SINGLE_DECL(aif1dac_enum,
+			    WM8994_POWER_MANAGEMENT_6, 0, aif1dac_text);
 
 static const struct snd_kcontrol_new aif1dac_mux =
 	SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum);
@@ -1522,8 +1527,8 @@ static const char *aif2dac_text[] = {
 	"AIF2DACDAT", "AIF3DACDAT",
 };
 
-static const struct soc_enum aif2dac_enum =
-	SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text);
+static SOC_ENUM_SINGLE_DECL(aif2dac_enum,
+			    WM8994_POWER_MANAGEMENT_6, 1, aif2dac_text);
 
 static const struct snd_kcontrol_new aif2dac_mux =
 	SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum);
@@ -1532,8 +1537,8 @@ static const char *aif2adc_text[] = {
 	"AIF2ADCDAT", "AIF3DACDAT",
 };
 
-static const struct soc_enum aif2adc_enum =
-	SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text);
+static SOC_ENUM_SINGLE_DECL(aif2adc_enum,
+			    WM8994_POWER_MANAGEMENT_6, 2, aif2adc_text);
 
 static const struct snd_kcontrol_new aif2adc_mux =
 	SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum);
@@ -1542,14 +1547,14 @@ static const char *aif3adc_text[] = {
 	"AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", "Mono PCM",
 };
 
-static const struct soc_enum wm8994_aif3adc_enum =
-	SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text);
+static SOC_ENUM_SINGLE_DECL(wm8994_aif3adc_enum,
+			    WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text);
 
 static const struct snd_kcontrol_new wm8994_aif3adc_mux =
 	SOC_DAPM_ENUM("AIF3ADC Mux", wm8994_aif3adc_enum);
 
-static const struct soc_enum wm8958_aif3adc_enum =
-	SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 4, aif3adc_text);
+static SOC_ENUM_SINGLE_DECL(wm8958_aif3adc_enum,
+			    WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text);
 
 static const struct snd_kcontrol_new wm8958_aif3adc_mux =
 	SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum);
@@ -1558,8 +1563,8 @@ static const char *mono_pcm_out_text[] = {
 	"None", "AIF2ADCL", "AIF2ADCR",
 };
 
-static const struct soc_enum mono_pcm_out_enum =
-	SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 9, 3, mono_pcm_out_text);
+static SOC_ENUM_SINGLE_DECL(mono_pcm_out_enum,
+			    WM8994_POWER_MANAGEMENT_6, 9, mono_pcm_out_text);
 
 static const struct snd_kcontrol_new mono_pcm_out_mux =
 	SOC_DAPM_ENUM("Mono PCM Out Mux", mono_pcm_out_enum);
@@ -1569,14 +1574,14 @@ static const char *aif2dac_src_text[] = {
 };
 
 /* Note that these two control shouldn't be simultaneously switched to AIF3 */
-static const struct soc_enum aif2dacl_src_enum =
-	SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 7, 2, aif2dac_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacl_src_enum,
+			    WM8994_POWER_MANAGEMENT_6, 7, aif2dac_src_text);
 
 static const struct snd_kcontrol_new aif2dacl_src_mux =
 	SOC_DAPM_ENUM("AIF2DACL Mux", aif2dacl_src_enum);
 
-static const struct soc_enum aif2dacr_src_enum =
-	SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 8, 2, aif2dac_src_text);
+static SOC_ENUM_SINGLE_DECL(aif2dacr_src_enum,
+			    WM8994_POWER_MANAGEMENT_6, 8, aif2dac_src_text);
 
 static const struct snd_kcontrol_new aif2dacr_src_mux =
 	SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 70ff3772079f..5e3bc3c6801a 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -399,6 +399,7 @@ static struct platform_driver davinci_evm_driver = {
 	.driver		= {
 		.name	= "davinci_evm",
 		.owner	= THIS_MODULE,
+		.pm	= &snd_soc_pm_ops,
 		.of_match_table = of_match_ptr(davinci_evm_dt_ids),
 	},
 };
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index b7858bfa0295..670afa29e30d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -263,7 +263,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 					 unsigned int fmt)
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
 
+	pm_runtime_get_sync(mcasp->dev);
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_B:
 	case SND_SOC_DAIFMT_AC97:
@@ -317,7 +319,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		break;
 
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
+		goto out;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -354,10 +357,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		break;
 
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
+		break;
 	}
-
-	return 0;
+out:
+	pm_runtime_put_sync(mcasp->dev);
+	return ret;
 }
 
 static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
@@ -448,7 +453,7 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
 	return 0;
 }
 
-static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
+static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 				    int channels)
 {
 	int i;
@@ -524,12 +529,18 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream,
 	return 0;
 }
 
-static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
+static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream)
 {
 	int i, active_slots;
 	u32 mask = 0;
 	u32 busel = 0;
 
+	if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) {
+		dev_err(mcasp->dev, "tdm slot %d not supported\n",
+			mcasp->tdm_slots);
+		return -EINVAL;
+	}
+
 	active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots;
 	for (i = 0; i < active_slots; i++)
 		mask |= (1 << i);
@@ -539,35 +550,21 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream)
 	if (!mcasp->dat_port)
 		busel = TXSEL;
 
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		/* bit stream is MSB first  with no delay */
-		/* DSP_B mode */
-		mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
-		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
-
-		if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
-			mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
-				       FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
-		else
-			printk(KERN_ERR "playback tdm slot %d not supported\n",
-				mcasp->tdm_slots);
-	} else {
-		/* bit stream is MSB first with no delay */
-		/* DSP_B mode */
-		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
-		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
-
-		if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32))
-			mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
-				       FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
-		else
-			printk(KERN_ERR "capture tdm slot %d not supported\n",
-				mcasp->tdm_slots);
-	}
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+	mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+		       FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF));
+
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+	mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+		       FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF));
+
+	return 0;
 }
 
 /* S/PDIF */
-static void davinci_hw_dit_param(struct davinci_mcasp *mcasp)
+static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp)
 {
 	/* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0
 	   and LSB first */
@@ -589,6 +586,8 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp)
 
 	/* Enable the DIT */
 	mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN);
+
+	return 0;
 }
 
 static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
@@ -605,13 +604,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	u8 slots = mcasp->tdm_slots;
 	u8 active_serializers;
 	int channels;
+	int ret;
 	struct snd_interval *pcm_channels = hw_param_interval(params,
 					SNDRV_PCM_HW_PARAM_CHANNELS);
 	channels = pcm_channels->min;
 
 	active_serializers = (channels + slots - 1) / slots;
 
-	if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL)
+	if (mcasp_common_hw_param(mcasp, substream->stream, channels) == -EINVAL)
 		return -EINVAL;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		fifo_level = mcasp->txnumevt * active_serializers;
@@ -619,9 +619,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		fifo_level = mcasp->rxnumevt * active_serializers;
 
 	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
-		davinci_hw_dit_param(mcasp);
+		ret = mcasp_dit_hw_param(mcasp);
 	else
-		davinci_hw_param(mcasp, substream->stream);
+		ret = mcasp_i2s_hw_param(mcasp, substream->stream);
+
+	if (ret)
+		return ret;
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_U8:
@@ -678,19 +681,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		ret = pm_runtime_get_sync(mcasp->dev);
-		if (IS_ERR_VALUE(ret))
-			dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n");
 		davinci_mcasp_start(mcasp, substream->stream);
 		break;
-
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		davinci_mcasp_stop(mcasp, substream->stream);
-		ret = pm_runtime_put_sync(mcasp->dev);
-		if (IS_ERR_VALUE(ret))
-			dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n");
-		break;
-
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		davinci_mcasp_stop(mcasp, substream->stream);
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index d0c72ed261e7..c84026c99134 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA,
 			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask));
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB,
-			   ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask));
+			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask));
 
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
 			   ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots));
@@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA,
 			   ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask));
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB,
-			   ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask));
+			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
 
 	esai_priv->slot_width = slot_width;
 
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 9c9f957fcae1..75e14033e8d8 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -322,7 +322,7 @@
 #define ESAI_xSMB_xS_SHIFT	0
 #define ESAI_xSMB_xS_WIDTH	16
 #define ESAI_xSMB_xS_MASK	(((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT)
-#define ESAI_xSMB_xS(v)		(((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK)
+#define ESAI_xSMB_xS(v)		(((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK)
 
 /* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */
 #define ESAI_PRRC_PDC_SHIFT	0
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index 79cee782dbbf..a2fd7321b5a9 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = {
 	.driver = {
 		.name = "imx_mc13783",
 		.owner = THIS_MODULE,
-		.pm = &snd_soc_pm_ops,
 	},
 	.probe = imx_mc13783_probe,
 	.remove = imx_mc13783_remove
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index f2beae78969f..1cb22dd034eb 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -33,8 +33,7 @@ struct imx_sgtl5000_data {
 
 static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct imx_sgtl5000_data *data = container_of(rtd->card,
-					struct imx_sgtl5000_data, card);
+	struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card);
 	struct device *dev = rtd->card->dev;
 	int ret;
 
@@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev)
 	data->card.dapm_widgets = imx_sgtl5000_dapm_widgets;
 	data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets);
 
+	platform_set_drvdata(pdev, &data->card);
+	snd_soc_card_set_drvdata(&data->card, data);
+
 	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
 		goto fail;
 	}
 
-	platform_set_drvdata(pdev, data);
 	of_node_put(ssi_np);
 	of_node_put(codec_np);
 
@@ -184,7 +185,8 @@ fail:
 
 static int imx_sgtl5000_remove(struct platform_device *pdev)
 {
-	struct imx_sgtl5000_data *data = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card);
 
 	clk_put(data->codec_clk);
 
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 3fd76bc391de..3a3d17ce6ba4 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
 {
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
 	struct imx_priv *priv = &card_priv;
-	struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
 	struct device *dev = &priv->pdev->dev;
 	unsigned int pll_out;
 	int ret;
@@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card)
 {
 	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
 	struct imx_priv *priv = &card_priv;
-	struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev);
+	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
 	struct device *dev = &priv->pdev->dev;
 	int ret;
 
@@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev)
 	data->card.late_probe = imx_wm8962_late_probe;
 	data->card.set_bias_level = imx_wm8962_set_bias_level;
 
+	platform_set_drvdata(pdev, &data->card);
+	snd_soc_card_set_drvdata(&data->card, data);
+
 	ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
 		goto clk_fail;
 	}
 
-	platform_set_drvdata(pdev, data);
 	of_node_put(ssi_np);
 	of_node_put(codec_np);
 
@@ -289,7 +291,8 @@ fail:
 
 static int imx_wm8962_remove(struct platform_device *pdev)
 {
-	struct imx_wm8962_data *data = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
 
 	if (!IS_ERR(data->codec_clk))
 		clk_disable_unprepare(data->codec_clk);
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 454f41cfc828..350757400391 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -59,7 +59,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750
 	select SND_SOC_WM8750
 	select SND_S3C2412_SOC_I2S
 	help
-	  Sat Y if you want to add support for SoC audio on the Jive.
+	  Say Y if you want to add support for SoC audio on the Jive.
 
 config SND_SOC_SAMSUNG_SMDK_WM8580
 	tristate "SoC I2S Audio support for WM8580 on SMDK"
@@ -145,11 +145,11 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380
 
 config SND_SOC_SAMSUNG_SMDK_WM9713
 	tristate "SoC AC97 Audio support for SMDK with WM9713"
-	depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210)
+	depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110)
 	select SND_SOC_WM9713
 	select SND_SAMSUNG_AC97
 	help
-	  Sat Y if you want to add support for SoC audio on the SMDK.
+	  Say Y if you want to add support for SoC audio on the SMDK.
 
 config SND_SOC_SMARTQ
 	tristate "SoC I2S Audio support for SmartQ board"
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index dc8ff13187f7..b9dc6acbba8c 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1218,7 +1218,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 			ret = regulator_allow_bypass(w->regulator, false);
 			if (ret != 0)
 				dev_warn(w->dapm->dev,
-					 "ASoC: Failed to bypass %s: %d\n",
+					 "ASoC: Failed to unbypass %s: %d\n",
 					 w->name, ret);
 		}
 
@@ -1228,7 +1228,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 			ret = regulator_allow_bypass(w->regulator, true);
 			if (ret != 0)
 				dev_warn(w->dapm->dev,
-					 "ASoC: Failed to unbypass %s: %d\n",
+					 "ASoC: Failed to bypass %s: %d\n",
 					 w->name, ret);
 		}
 
@@ -3210,15 +3210,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
 	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 	const char *pin = (const char *)kcontrol->private_value;
 
-	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
-
 	if (ucontrol->value.integer.value[0])
 		snd_soc_dapm_enable_pin(&card->dapm, pin);
 	else
 		snd_soc_dapm_disable_pin(&card->dapm, pin);
 
-	mutex_unlock(&card->dapm_mutex);
-
 	snd_soc_dapm_sync(&card->dapm);
 	return 0;
 }
@@ -3248,7 +3244,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 			ret = regulator_allow_bypass(w->regulator, true);
 			if (ret != 0)
 				dev_warn(w->dapm->dev,
-					 "ASoC: Failed to unbypass %s: %d\n",
+					 "ASoC: Failed to bypass %s: %d\n",
 					 w->name, ret);
 		}
 		break;
@@ -3767,23 +3763,52 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
 }
 
 /**
+ * snd_soc_dapm_enable_pin_unlocked - enable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Enables input/output pin and its parents or children widgets iff there is
+ * a valid audio route and active audio stream.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+				   const char *pin)
+{
+	return snd_soc_dapm_set_pin(dapm, pin, 1);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin_unlocked);
+
+/**
  * snd_soc_dapm_enable_pin - enable pin.
  * @dapm: DAPM context
  * @pin: pin name
  *
  * Enables input/output pin and its parents or children widgets iff there is
  * a valid audio route and active audio stream.
+ *
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
 int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin)
 {
-	return snd_soc_dapm_set_pin(dapm, pin, 1);
+	int ret;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+	ret = snd_soc_dapm_set_pin(dapm, pin, 1);
+
+	mutex_unlock(&dapm->card->dapm_mutex);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
 
 /**
- * snd_soc_dapm_force_enable_pin - force a pin to be enabled
+ * snd_soc_dapm_force_enable_pin_unlocked - force a pin to be enabled
  * @dapm: DAPM context
  * @pin: pin name
  *
@@ -3791,11 +3816,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
  * intended for use with microphone bias supplies used in microphone
  * jack detection.
  *
+ * Requires external locking.
+ *
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
-				  const char *pin)
+int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+					 const char *pin)
 {
 	struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true);
 
@@ -3811,25 +3838,103 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin_unlocked);
+
+/**
+ * snd_soc_dapm_force_enable_pin - force a pin to be enabled
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Enables input/output pin regardless of any other state.  This is
+ * intended for use with microphone bias supplies used in microphone
+ * jack detection.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
+				  const char *pin)
+{
+	int ret;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+	ret = snd_soc_dapm_force_enable_pin_unlocked(dapm, pin);
+
+	mutex_unlock(&dapm->card->dapm_mutex);
+
+	return ret;
+}
 EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin);
 
 /**
+ * snd_soc_dapm_disable_pin_unlocked - disable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Disables input/output pin and its parents or children widgets.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_disable_pin_unlocked(struct snd_soc_dapm_context *dapm,
+				    const char *pin)
+{
+	return snd_soc_dapm_set_pin(dapm, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin_unlocked);
+
+/**
  * snd_soc_dapm_disable_pin - disable pin.
  * @dapm: DAPM context
  * @pin: pin name
  *
  * Disables input/output pin and its parents or children widgets.
+ *
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
 int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm,
 			     const char *pin)
 {
-	return snd_soc_dapm_set_pin(dapm, pin, 0);
+	int ret;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+	ret = snd_soc_dapm_set_pin(dapm, pin, 0);
+
+	mutex_unlock(&dapm->card->dapm_mutex);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
 
 /**
+ * snd_soc_dapm_nc_pin_unlocked - permanently disable pin.
+ * @dapm: DAPM context
+ * @pin: pin name
+ *
+ * Marks the specified pin as being not connected, disabling it along
+ * any parent or child widgets.  At present this is identical to
+ * snd_soc_dapm_disable_pin() but in future it will be extended to do
+ * additional things such as disabling controls which only affect
+ * paths through the pin.
+ *
+ * Requires external locking.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_nc_pin_unlocked(struct snd_soc_dapm_context *dapm,
+			       const char *pin)
+{
+	return snd_soc_dapm_set_pin(dapm, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin_unlocked);
+
+/**
  * snd_soc_dapm_nc_pin - permanently disable pin.
  * @dapm: DAPM context
  * @pin: pin name
@@ -3845,7 +3950,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
  */
 int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin)
 {
-	return snd_soc_dapm_set_pin(dapm, pin, 0);
+	int ret;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+	ret = snd_soc_dapm_set_pin(dapm, pin, 0);
+
+	mutex_unlock(&dapm->card->dapm_mutex);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
 
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index e0305a148568..9edd68db9f48 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -183,14 +183,16 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev)
 	irq = platform_get_irq(pdev, 0);
 	if (irq < 0)
 		return irq;
+
+	drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
+	if (!drvdata)
+		return -ENOMEM;
+
 	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	drvdata->base = devm_ioremap_resource(&pdev->dev, r);
 	if (IS_ERR(drvdata->base))
 		return PTR_ERR(drvdata->base);
 
-	drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
-	if (!drvdata)
-		return -ENOMEM;
 	platform_set_drvdata(pdev, drvdata);
 	drvdata->physbase = r->start;
 	if (sizeof(drvdata->physbase) > sizeof(r->start) &&