Merge tag 'asoc-v3.15-2' into asoc-next

ASoC: Updates for v3.15

This is mostly a few additional fixes from Lars-Peter, a new driver and
cleaning up a git failure with merging the Intel branch (combined with
an xargs failure to pay attention to error codes).  The history lists a
bunch of additional commits for the branch but the content of those
commits is actually present already but not recorded in history due to
git failing.  Unfortunately xargs is used in the merge script and it
doesn't do a good job of noticing errors from the commands it invokes.

# gpg: Signature made Thu 13 Mar 2014 14:25:44 GMT using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"

109 files changed, 1958 insertions, 795 deletions

diff --git a/Documentation/devicetree/bindings/sound/armada-370db-audio.txt b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
new file mode 100644
index 0000000..3893b4d
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/armada-370db-audio.txt
@@ -0,0 +1,24 @@
+Device Tree bindings for the Armada 370 DB audio
+================================================
+
+These Device Tree bindings are used to describe the audio complex
+found on the Armada 370 DB platform.
+
+Mandatory properties:
+
+ * compatible: must be "marvell,a370db-audio"
+
+ * marvell,audio-controller: a phandle that points to the audio
+   controller of the Armada 370 SoC.
+
+ * marvell,audio-codec: a phandle that points to the analog audio
+   codec connected to the Armada 370 SoC.
+
+Example:
+
+	sound {
+	      compatible = "marvell,a370db-audio";
+	      marvell,audio-controller = <&audio_controller>;
+	      marvell,audio-codec = <&audio_codec>;
+	      status = "okay";
+	};
diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
index f0062c5..cb8c07c 100644
--- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt
+++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
@@ -5,6 +5,7 @@ Required properties:
 - compatible:
   "marvell,kirkwood-audio" for Kirkwood platforms
   "marvell,dove-audio" for Dove platforms
+  "marvell,armada370-audio" for Armada 370 platforms
 
 - reg: physical base address of the controller and length of memory mapped
   region.
diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
new file mode 100644
index 0000000..74c66de
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
@@ -0,0 +1,61 @@
+Texas Instruments - tlv320aic31xx Codec module
+
+The tlv320aic31xx serial control bus communicates through I2C protocols
+
+Required properties:
+
+- compatible - "string" - One of:
+    "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp
+    "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp
+    "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP)
+    "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP)
+    "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP)
+    "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP)
+
+- reg - <int> -  I2C slave address
+
+
+Optional properties:
+
+- gpio-reset - gpio pin number used for codec reset
+- ai31xx-micbias-vg - MicBias Voltage setting
+        1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V
+        2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V
+        3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD
+	If this node is not mentioned or if the value is unknown, then
+	micbias	is set to 2.0V.
+- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
+  DVDD-supply : power supplies for the device as covered in
+  Documentation/devicetree/bindings/regulator/regulator.txt
+
+CODEC output pins:
+  * HPL
+  * HPR
+  * SPL, devices with stereo speaker amp
+  * SPR, devices with stereo speaker amp
+  * SPK, devices with mono speaker amp
+  * MICBIAS
+
+CODEC input pins:
+  * MIC1LP
+  * MIC1RP
+  * MIC1LM
+
+The pins can be used in referring sound node's audio-routing property.
+
+Example:
+#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+
+tlv320aic31xx: tlv320aic31xx@18 {
+	compatible = "ti,tlv320aic311x";
+	reg = <0x18>;
+
+	ai31xx-micbias-vg = <MICBIAS_OFF>;
+
+	HPVDD-supply = <&regulator>;
+	SPRVDD-supply = <&regulator>;
+	SPLVDD-supply = <&regulator>;
+	AVDD-supply = <&regulator>;
+	IOVDD-supply = <&regulator>;
+	DVDD-supply = <&regulator>;
+};
diff --git a/MAINTAINERS b/MAINTAINERS
index b3fdb0f..7c42d59 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -2203,6 +2203,13 @@ L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Odd Fixes
 F:	sound/soc/codecs/cs4270*
 
+CIRRUS LOGIC AUDIO CODEC DRIVERS
+M:	Brian Austin <brian.austin@cirrus.com>
+M:	Paul Handrigan <Paul.Handrigan@cirrus.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Maintained
+F:	sound/soc/codecs/cs*
+
 CLEANCACHE API
 M:	Konrad Rzeszutek Wilk <konrad.wilk@oracle.com>
 L:	linux-kernel@vger.kernel.org
diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h
new file mode 100644
index 0000000..f5cb772
--- /dev/null
+++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h
@@ -0,0 +1,8 @@
+#ifndef __DT_TLV320AIC31XX_MICBIAS_H
+#define __DT_TLV320AIC31XX_MICBIAS_H
+
+#define MICBIAS_2_0V		1
+#define MICBIAS_2_5V		2
+#define MICBIAS_AVDDV		3
+
+#endif /* __DT_TLV320AIC31XX_MICBIAS_H */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 959f389..f7de629 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -354,12 +354,6 @@ typedef int (*hw_write_t)(void *,const char* ,int);
 
 extern struct snd_ac97_bus_ops *soc_ac97_ops;
 
-enum snd_soc_control_type {
-	SND_SOC_I2C = 1,
-	SND_SOC_SPI,
-	SND_SOC_REGMAP,
-};
-
 enum snd_soc_pcm_subclass {
 	SND_SOC_PCM_CLASS_PCM	= 0,
 	SND_SOC_PCM_CLASS_BE	= 1,
@@ -406,8 +400,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec,
 int snd_soc_codec_writable_register(struct snd_soc_codec *codec,
 				    unsigned int reg);
 int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
-			       int addr_bits, int data_bits,
-			       enum snd_soc_control_type control);
+			       struct regmap *regmap);
 int snd_soc_cache_sync(struct snd_soc_codec *codec);
 int snd_soc_cache_init(struct snd_soc_codec *codec);
 int snd_soc_cache_exit(struct snd_soc_codec *codec);
@@ -614,7 +607,8 @@ struct snd_soc_jack_gpio {
 	struct snd_soc_jack *jack;
 	struct delayed_work work;
 
-	int (*jack_status_check)(void);
+	void *data;
+	int (*jack_status_check)(void *data);
 };
 
 struct snd_soc_jack {
@@ -717,7 +711,6 @@ struct snd_soc_codec {
 	/* codec IO */
 	void *control_data; /* codec control (i2c/3wire) data */
 	hw_write_t hw_write;
-	unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
 	int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
 	void *reg_cache;
diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c
index 29238a7..5b68b10 100644
--- a/sound/soc/cirrus/snappercl15.c
+++ b/sound/soc/cirrus/snappercl15.c
@@ -65,18 +65,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"MICIN", NULL, "Mic Jack"},
 };
 
-static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
-				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
-
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-	return 0;
-}
-
 static struct snd_soc_dai_link snappercl15_dai = {
 	.name		= "tlv320aic23",
 	.stream_name	= "AIC23",
@@ -84,7 +72,6 @@ static struct snd_soc_dai_link snappercl15_dai = {
 	.codec_dai_name	= "tlv320aic23-hifi",
 	.codec_name	= "tlv320aic23-codec.0-001a",
 	.platform_name	= "ep93xx-i2s",
-	.init		= snappercl15_tlv320aic23_init,
 	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
 			  SND_SOC_DAIFMT_CBS_CFS,
 	.ops		= &snappercl15_ops,
@@ -95,6 +82,11 @@ static struct snd_soc_card snd_soc_snappercl15 = {
 	.owner		= THIS_MODULE,
 	.dai_link	= &snappercl15_dai,
 	.num_links	= 1,
+
+	.dapm_widgets		= tlv320aic23_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(tlv320aic23_dapm_widgets),
+	.dapm_routes		= audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(audio_map),
 };
 
 static int snappercl15_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 8703244..b07e171 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -1327,8 +1327,7 @@ static int pm860x_probe(struct snd_soc_codec *codec)
 
 	pm860x->codec = codec;
 
-	codec->control_data = pm860x->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+	ret = snd_soc_codec_set_cache_io(codec, pm860x->regmap);
 	if (ret)
 		return ret;
 
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 32d7a6f..1a8ff1e 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -85,6 +85,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TLV320AIC23_I2C if I2C
 	select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
+	select SND_SOC_TLV320AIC31XX if I2C
 	select SND_SOC_TLV320AIC32X4 if I2C
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TPA6130A2 if I2C
@@ -449,6 +450,9 @@ config SND_SOC_TLV320AIC26
 	tristate
 	depends on SPI
 
+config SND_SOC_TLV320AIC31XX
+        tristate
+
 config SND_SOC_TLV320AIC32X4
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index cb46c4c..73df822 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -79,6 +79,7 @@ snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
 snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
+snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
 snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
@@ -223,6 +224,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C)	+= snd-soc-tlv320aic23-i2c.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI)	+= snd-soc-tlv320aic23-spi.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
+obj-$(CONFIG_SND_SOC_TLV320AIC31XX)     += snd-soc-tlv320aic31xx.o
 obj-$(CONFIG_SND_SOC_TLV320AIC32X4)     += snd-soc-tlv320aic32x4.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 9381a76..6844d0b 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -322,14 +322,6 @@ static struct snd_soc_dai_driver ad193x_dai = {
 static int ad193x_codec_probe(struct snd_soc_codec *codec)
 {
 	struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	codec->control_data = ad193x->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	/* default setting for ad193x */
 
@@ -347,7 +339,7 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec)
 	regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
 	regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04);
 
-	return ret;
+	return 0;
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 5223800..877f573 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1376,15 +1376,8 @@ static int adau1373_probe(struct snd_soc_codec *codec)
 	struct adau1373_platform_data *pdata = codec->dev->platform_data;
 	bool lineout_differential = false;
 	unsigned int val;
-	int ret;
 	int i;
 
-	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-	if (ret) {
-		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	if (pdata) {
 		if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting))
 			return -EINVAL;
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 7470831..5062e34 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -801,15 +801,8 @@ static struct snd_soc_dai_driver adav80x_dais[] = {
 
 static int adav80x_probe(struct snd_soc_codec *codec)
 {
-	int ret;
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
-	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-	if (ret) {
-		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Force PLLs on for SYSCLK output */
 	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
 	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 684fe91..30e2978 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -388,15 +388,6 @@ static int ak4535_resume(struct snd_soc_codec *codec)
 
 static int ak4535_probe(struct snd_soc_codec *codec)
 {
-	struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	codec->control_data = ak4535->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 	/* power on device */
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
index 684b56f..868c0e2 100644
--- a/sound/soc/codecs/ak4641.c
+++ b/sound/soc/codecs/ak4641.c
@@ -519,14 +519,6 @@ static int ak4641_resume(struct snd_soc_codec *codec)
 
 static int ak4641_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* power on device */
 	ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 1f646c6..92655cc 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -465,14 +465,6 @@ static int ak4642_resume(struct snd_soc_codec *codec)
 
 static int ak4642_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index deb2b44..998fa0c 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -613,17 +613,7 @@ static struct snd_soc_dai_driver ak4671_dai = {
 
 static int ak4671_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
-	ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return ret;
+	return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 }
 
 static int ak4671_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index ed50625..09f7e77 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -904,13 +904,6 @@ static int alc5623_probe(struct snd_soc_codec *codec)
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
-	codec->control_data = alc5623->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	alc5623_reset(codec);
 
 	/* power on device */
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index d885056..ec071a6 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -1063,14 +1063,6 @@ static int alc5632_probe(struct snd_soc_codec *codec)
 	struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	codec->control_data = alc5632->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* power on device  */
 	alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 43737a27..1e25c7a 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -138,9 +138,8 @@ static int cq93vc_probe(struct snd_soc_codec *codec)
 	struct davinci_vc *davinci_vc = codec->dev->platform_data;
 
 	davinci_vc->cq93vc.codec = codec;
-	codec->control_data = davinci_vc->regmap;
 
-	snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP);
+	snd_soc_codec_set_cache_io(codec, davinci_vc->regmap);
 
 	/* Off, with power on */
 	cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 83c835d..3920e62 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -506,15 +506,6 @@ static int cs4270_probe(struct snd_soc_codec *codec)
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	/* Tell ASoC what kind of I/O to use to read the registers.  ASoC will
-	 * then do the I2C transactions itself.
-	 */
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret);
-		return ret;
-	}
-
 	/* Disable auto-mute.  This feature appears to be buggy.  In some
 	 * situations, auto-mute will not deactivate when it should, so we want
 	 * this feature disabled by default.  An application (e.g. alsactl) can
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 8281577..6c0da2b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -487,12 +487,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
 {
 	int ret, reg;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/*
 	 * DAC configuration
 	 * - Use signal processor
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index ea7938d..f0ca6be 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -1109,14 +1109,7 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec)
 static int cs42l52_probe(struct snd_soc_codec *codec)
 {
 	struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec);
-	int ret;
 
-	codec->control_data = cs42l52->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 	regcache_cache_only(cs42l52->regmap, true);
 
 	cs42l52_add_mic_controls(codec);
@@ -1128,7 +1121,7 @@ static int cs42l52_probe(struct snd_soc_codec *codec)
 	cs42l52->sysclk = CS42L52_DEFAULT_CLK;
 	cs42l52->config.format = CS42L52_DEFAULT_FORMAT;
 
-	return ret;
+	return 0;
 }
 
 static int cs42l52_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index e5778c0..0ee60a1 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1345,17 +1345,8 @@ static int cs42l73_resume(struct snd_soc_codec *codec)
 
 static int cs42l73_probe(struct snd_soc_codec *codec)
 {
-	int ret;
 	struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec);
 
-	codec->control_data = cs42l73->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Set Charge Pump Frequency */
@@ -1368,7 +1359,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec)
 	cs42l73->mclksel = CS42L73_CLKID_MCLK1;
 	cs42l73->mclk = 0;
 
-	return ret;
+	return 0;
 }
 
 static int cs42l73_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 01e55fc..137e8eb 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -1071,17 +1071,9 @@ static struct snd_soc_dai_driver da7210_dai = {
 static int da7210_probe(struct snd_soc_codec *codec)
 {
 	struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
-	int ret;
 
 	dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
 
-	codec->control_data = da7210->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	da7210->mclk_rate       = 0;    /* This will be set from set_sysclk() */
 	da7210->master          = 0;    /* This will be set from set_fmt() */
 
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index 439d103..738fa18 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -1393,17 +1393,9 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec,
 
 static int da7213_probe(struct snd_soc_codec *codec)
 {
-	int ret;
 	struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
 	struct da7213_platform_data *pdata = da7213->pdata;
 
-	codec->control_data = da7213->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Default to using ALC auto offset calibration mode. */
 	snd_soc_update_bits(codec, DA7213_ALC_CTRL1,
 			    DA7213_ALC_CALIB_MODE_MAN, 0);
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
index 4d1c302..7d168ec 100644
--- a/sound/soc/codecs/da732x.c
+++ b/sound/soc/codecs/da732x.c
@@ -1297,9 +1297,9 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec)
 	msleep(DA732X_WAIT_FOR_STABILIZATION);
 
 	/* Check DAC offset sign */
-	sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+	sign[DA732X_HPL_DAC] = (snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
 				DA732X_HP_DAC_OFF_CNTL_COMPO);
-	sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+	sign[DA732X_HPR_DAC] = (snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
 				DA732X_HP_DAC_OFF_CNTL_COMPO);
 
 	/* Binary search DAC offset values (both channels at once) */
@@ -1316,10 +1316,10 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec)
 
 		msleep(DA732X_WAIT_FOR_STABILIZATION);
 
-		if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+		if ((snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
 		     DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC])
 			offset[DA732X_HPL_DAC] &= ~step;
-		if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+		if ((snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
 		     DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC])
 			offset[DA732X_HPR_DAC] &= ~step;
 
@@ -1360,9 +1360,9 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec)
 	msleep(DA732X_WAIT_FOR_STABILIZATION);
 
 	/* Check output offset sign */
-	sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) &
+	sign[DA732X_HPL_AMP] = snd_soc_read(codec, DA732X_REG_HPL) &
 			       DA732X_HP_OUT_COMPO;
-	sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) &
+	sign[DA732X_HPR_AMP] = snd_soc_read(codec, DA732X_REG_HPR) &
 			       DA732X_HP_OUT_COMPO;
 
 	snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP |
@@ -1383,10 +1383,10 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec)
 
 		msleep(DA732X_WAIT_FOR_STABILIZATION);
 
-		if ((codec->hw_read(codec, DA732X_REG_HPL) &
+		if ((snd_soc_read(codec, DA732X_REG_HPL) &
 		     DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP])
 			offset[DA732X_HPL_AMP] &= ~step;
-		if ((codec->hw_read(codec, DA732X_REG_HPR) &
+		if ((snd_soc_read(codec, DA732X_REG_HPR) &
 		     DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP])
 			offset[DA732X_HPR_AMP] &= ~step;
 
@@ -1512,23 +1512,14 @@ static int da732x_probe(struct snd_soc_codec *codec)
 {
 	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret = 0;
 
 	da732x->codec = codec;
 
 	dapm->idle_bias_off = false;
 
-	codec->control_data = da732x->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to register codec.\n");
-		goto err;
-	}
-
 	da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-err:
-	return ret;
+
+	return 0;
 }
 
 static int da732x_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c
index f118daa..4ff06b5 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1383,16 +1383,8 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec,
 
 static int da9055_probe(struct snd_soc_codec *codec)
 {
-	int ret;
 	struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec);
 
-	codec->control_data = da9055->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Enable all Gain Ramps */
 	snd_soc_update_bits(codec, DA9055_AUX_L_CTRL,
 			    DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN);
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
index cb736dd..3e264a7 100644
--- a/sound/soc/codecs/isabelle.c
+++ b/sound/soc/codecs/isabelle.c
@@ -1090,23 +1090,7 @@ static struct snd_soc_dai_driver isabelle_dai[] = {
 	},
 };
 
-static int isabelle_probe(struct snd_soc_codec *codec)
-{
-	int ret = 0;
-
-	codec->control_data = dev_get_regmap(codec->dev, NULL);
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_codec_dev_isabelle = {
-	.probe = isabelle_probe,
 	.set_bias_level = isabelle_set_bias_level,
 	.controls = isabelle_snd_controls,
 	.num_controls = ARRAY_SIZE(isabelle_snd_controls),
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 0e5743e..4f048db 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -101,8 +101,7 @@ static const char *lm4857_mode[] = {
 	"Headphone",
 };
 
-static const struct soc_enum lm4857_mode_enum =
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode);
+static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode);
 
 static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("IN"),
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index 6b7fe5e..275b3f7 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -213,15 +213,13 @@ static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" };
 
 static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" };
 
-static const struct soc_enum lm49453_adcl_enum =
-	SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0,
-			ARRAY_SIZE(lm49453_adcl_mux_text),
-			lm49453_adcl_mux_text);
+static SOC_ENUM_SINGLE_DECL(lm49453_adcl_enum,
+			    LM49453_P0_ANALOG_MIXER_ADC_REG, 0,
+			    lm49453_adcl_mux_text);
 
-static const struct soc_enum lm49453_adcr_enum =
-	SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1,
-			ARRAY_SIZE(lm49453_adcr_mux_text),
-			lm49453_adcr_mux_text);
+static SOC_ENUM_SINGLE_DECL(lm49453_adcr_enum,
+			    LM49453_P0_ANALOG_MIXER_ADC_REG, 1,
+			    lm49453_adcr_mux_text);
 
 static const struct snd_kcontrol_new lm49453_adcl_mux_control =
 	SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum);
@@ -1409,22 +1407,6 @@ static int lm49453_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int lm49453_probe(struct snd_soc_codec *codec)
-{
-	struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec);
-	int ret = 0;
-
-	codec->control_data = lm49453->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
-	return 0;
-}
-
 /* power down chip */
 static int lm49453_remove(struct snd_soc_codec *codec)
 {
@@ -1433,7 +1415,6 @@ static int lm49453_remove(struct snd_soc_codec *codec)
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_lm49453 = {
-	.probe = lm49453_probe,
 	.remove = lm49453_remove,
 	.suspend = lm49453_suspend,
 	.resume = lm49453_resume,
diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c
index 31f9156..ec481fc 100644
--- a/sound/soc/codecs/max9768.c
+++ b/sound/soc/codecs/max9768.c
@@ -135,11 +135,6 @@ static int max9768_probe(struct snd_soc_codec *codec)
 	struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	codec->control_data = max9768->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP);
-	if (ret)
-		return ret;
-
 	if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) {
 		ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM);
 		if (ret)
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index bb1ecfc..ef7cf89 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -597,28 +597,27 @@ static const unsigned int max98088_exmode_values[] = {
        0x00, 0x43, 0x10, 0x20, 0x30, 0x40, 0x11, 0x22, 0x32
 };
 
-static const struct soc_enum max98088_exmode_enum =
-       SOC_VALUE_ENUM_SINGLE(M98088_REG_41_SPKDHP, 0, 127,
-                             ARRAY_SIZE(max98088_exmode_texts),
-                             max98088_exmode_texts,
-                             max98088_exmode_values);
+static SOC_VALUE_ENUM_SINGLE_DECL(max98088_exmode_enum,
+				  M98088_REG_41_SPKDHP, 0, 127,
+				  max98088_exmode_texts,
+				  max98088_exmode_values);
 
 static const char *max98088_ex_thresh[] = { /* volts PP */
        "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"};
-static const struct soc_enum max98088_ex_thresh_enum[] = {
-       SOC_ENUM_SINGLE(M98088_REG_42_SPKDHP_THRESH, 0, 8,
-               max98088_ex_thresh),
-};
+static SOC_ENUM_SINGLE_DECL(max98088_ex_thresh_enum,
+			    M98088_REG_42_SPKDHP_THRESH, 0,
+			    max98088_ex_thresh);
 
 static const char *max98088_fltr_mode[] = {"Voice", "Music" };
-static const struct soc_enum max98088_filter_mode_enum[] = {
-       SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 7, 2, max98088_fltr_mode),
-};
+static SOC_ENUM_SINGLE_DECL(max98088_filter_mode_enum,
+			    M98088_REG_18_DAI1_FILTERS, 7,
+			    max98088_fltr_mode);
 
 static const char *max98088_extmic_text[] = { "None", "MIC1", "MIC2" };
 
-static const struct soc_enum max98088_extmic_enum =
-       SOC_ENUM_SINGLE(M98088_REG_48_CFG_MIC, 0, 3, max98088_extmic_text);
+static SOC_ENUM_SINGLE_DECL(max98088_extmic_enum,
+			    M98088_REG_48_CFG_MIC, 0,
+			    max98088_extmic_text);
 
 static const struct snd_kcontrol_new max98088_extmic_mux =
        SOC_DAPM_ENUM("External MIC Mux", max98088_extmic_enum);
@@ -626,12 +625,12 @@ static const struct snd_kcontrol_new max98088_extmic_mux =
 static const char *max98088_dai1_fltr[] = {
        "Off", "fc=258/fs=16k", "fc=500/fs=16k",
        "fc=258/fs=8k", "fc=500/fs=8k", "fc=200"};
-static const struct soc_enum max98088_dai1_dac_filter_enum[] = {
-       SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 0, 6, max98088_dai1_fltr),
-};
-static const struct soc_enum max98088_dai1_adc_filter_enum[] = {
-       SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 4, 6, max98088_dai1_fltr),
-};
+static SOC_ENUM_SINGLE_DECL(max98088_dai1_dac_filter_enum,
+			    M98088_REG_18_DAI1_FILTERS, 0,
+			    max98088_dai1_fltr);
+static SOC_ENUM_SINGLE_DECL(max98088_dai1_adc_filter_enum,
+			    M98088_REG_18_DAI1_FILTERS, 4,
+			    max98088_dai1_fltr);
 
 static int max98088_mic1pre_set(struct snd_kcontrol *kcontrol,
                                struct snd_ctl_elem_value *ucontrol)
@@ -1915,12 +1914,6 @@ static int max98088_probe(struct snd_soc_codec *codec)
 
        regcache_mark_dirty(max98088->regmap);
 
-       ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-       if (ret != 0) {
-               dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-               return ret;
-       }
-
        /* initialize private data */
 
        max98088->sysclk = (unsigned)-1;
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f363de1..96a4745 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2218,14 +2218,6 @@ static int max98090_probe(struct snd_soc_codec *codec)
 
 	max98090->codec = codec;
 
-	codec->control_data = max98090->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Reset the codec, the DSP core, and disable all interrupts */
 	max98090_reset(max98090);
 
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 5bce9cd..03f0536 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -560,25 +560,27 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai,
 }
 
 static const char * const max98095_fltr_mode[] = { "Voice", "Music" };
-static const struct soc_enum max98095_dai1_filter_mode_enum[] = {
-	SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 7, 2, max98095_fltr_mode),
-};
-static const struct soc_enum max98095_dai2_filter_mode_enum[] = {
-	SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 7, 2, max98095_fltr_mode),
-};
+static SOC_ENUM_SINGLE_DECL(max98095_dai1_filter_mode_enum,
+			    M98095_02E_DAI1_FILTERS, 7,
+			    max98095_fltr_mode);
+static SOC_ENUM_SINGLE_DECL(max98095_dai2_filter_mode_enum,
+			    M98095_038_DAI2_FILTERS, 7,
+			    max98095_fltr_mode);
 
 static const char * const max98095_extmic_text[] = { "None", "MIC1", "MIC2" };
 
-static const struct soc_enum max98095_extmic_enum =
-	SOC_ENUM_SINGLE(M98095_087_CFG_MIC, 0, 3, max98095_extmic_text);
+static SOC_ENUM_SINGLE_DECL(max98095_extmic_enum,
+			    M98095_087_CFG_MIC, 0,
+			    max98095_extmic_text);
 
 static const struct snd_kcontrol_new max98095_extmic_mux =
 	SOC_DAPM_ENUM("External MIC Mux", max98095_extmic_enum);
 
 static const char * const max98095_linein_text[] = { "INA", "INB" };
 
-static const struct soc_enum max98095_linein_enum =
-	SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 6, 2, max98095_linein_text);
+static SOC_ENUM_SINGLE_DECL(max98095_linein_enum,
+			    M98095_086_CFG_LINE, 6,
+			    max98095_linein_text);
 
 static const struct snd_kcontrol_new max98095_linein_mux =
 	SOC_DAPM_ENUM("Linein Input Mux", max98095_linein_enum);
@@ -586,24 +588,26 @@ static const struct snd_kcontrol_new max98095_linein_mux =
 static const char * const max98095_line_mode_text[] = {
 	"Stereo", "Differential"};
 
-static const struct soc_enum max98095_linein_mode_enum =
-	SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 7, 2, max98095_line_mode_text);
+static SOC_ENUM_SINGLE_DECL(max98095_linein_mode_enum,
+			    M98095_086_CFG_LINE, 7,
+			    max98095_line_mode_text);
 
-static const struct soc_enum max98095_lineout_mode_enum =
-	SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 4, 2, max98095_line_mode_text);
+static SOC_ENUM_SINGLE_DECL(max98095_lineout_mode_enum,
+			    M98095_086_CFG_LINE, 4,
+			    max98095_line_mode_text);
 
 static const char * const max98095_dai_fltr[] = {
 	"Off", "Elliptical-HPF-16k", "Butterworth-HPF-16k",
 	"Elliptical-HPF-8k", "Butterworth-HPF-8k", "Butterworth-HPF-Fs/240"};
-static const struct soc_enum max98095_dai1_dac_filter_enum[] = {
-	SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 0, 6, max98095_dai_fltr),
-};
-static const struct soc_enum max98095_dai2_dac_filter_enum[] = {
-	SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 0, 6, max98095_dai_fltr),
-};
-static const struct soc_enum max98095_dai3_dac_filter_enum[] = {
-	SOC_ENUM_SINGLE(M98095_042_DAI3_FILTERS, 0, 6, max98095_dai_fltr),
-};
+static SOC_ENUM_SINGLE_DECL(max98095_dai1_dac_filter_enum,
+			    M98095_02E_DAI1_FILTERS, 0,
+			    max98095_dai_fltr);
+static SOC_ENUM_SINGLE_DECL(max98095_dai2_dac_filter_enum,
+			    M98095_038_DAI2_FILTERS, 0,
+			    max98095_dai_fltr);
+static SOC_ENUM_SINGLE_DECL(max98095_dai3_dac_filter_enum,
+			    M98095_042_DAI3_FILTERS, 0,
+			    max98095_dai_fltr);
 
 static int max98095_mic1pre_set(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
@@ -2234,12 +2238,6 @@ static int max98095_probe(struct snd_soc_codec *codec)
 	struct i2c_client *client;
 	int ret = 0;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* reset the codec, the DSP core, and disable all interrupts */
 	max98095_reset(codec);
 
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
index 82757eb..4fdf5aa 100644
--- a/sound/soc/codecs/max9850.c
+++ b/sound/soc/codecs/max9850.c
@@ -312,14 +312,6 @@ static int max9850_resume(struct snd_soc_codec *codec)
 
 static int max9850_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* enable zero-detect */
 	snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1);
 	/* enable slew-rate control */
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index ec89b8f..37d737e 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -612,8 +612,8 @@ static int mc13783_probe(struct snd_soc_codec *codec)
 	struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
-	ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP);
+	ret = snd_soc_codec_set_cache_io(codec,
+			dev_get_regmap(codec->dev->parent, NULL));
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
 		return ret;
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 577fb87..e661e84 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -586,16 +586,6 @@ static int ml26124_resume(struct snd_soc_codec *codec)
 
 static int ml26124_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
-	codec->control_data = priv->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Software Reset */
 	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
 	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index ce199d3..d4c229f 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1570,15 +1570,6 @@ static int rt5631_probe(struct snd_soc_codec *codec)
 {
 	struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec);
 	unsigned int val;
-	int ret;
-
-	codec->control_data = rt5631->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	val = rt5631_read_index(codec, RT5631_ADDA_MIXER_INTL_REG3);
 	if (val & 0x0002)
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 1a1e115..13ccee4 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -1936,16 +1936,8 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 static int rt5640_probe(struct snd_soc_codec *codec)
 {
 	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
-	int ret;
 
 	rt5640->codec = codec;
-	codec->control_data = rt5640->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	codec->dapm.idle_bias_off = 1;
 	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index ab4754a..d3ed1be 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1352,14 +1352,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 	int ret;
 	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
 
-	/* setup i2c data ops */
-	codec->control_data = sgtl5000->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = sgtl5000_enable_regulators(codec);
 	if (ret)
 		return ret;
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index fa2b8e0..244c097 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -21,6 +21,7 @@
 #include <linux/slab.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
+#include <linux/regmap.h>
 #include <sound/soc.h>
 #include <sound/initval.h>
 
@@ -209,8 +210,9 @@ out:
 
 static int si476x_codec_probe(struct snd_soc_codec *codec)
 {
-	codec->control_data = dev_get_regmap(codec->dev->parent, NULL);
-	return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+	struct regmap *regmap = dev_get_regmap(codec->dev->parent, NULL);
+
+	return snd_soc_codec_set_cache_io(codec, regmap);
 }
 
 static struct snd_soc_dai_ops si476x_dai_ops = {
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index bca7d02..42dff26 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -825,8 +825,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
 {
 	pr_debug("codec_probe called\n");
 
-	snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-
 	/* PCM interface config
 	 * This sets the pcm rx slot conguration to max 6 slots
 	 * for max 4 dais (2 stereo and 2 mono)
diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c
index 806f3d8..56adb3e 100644
--- a/sound/soc/codecs/ssm2518.c
+++ b/sound/soc/codecs/ssm2518.c
@@ -648,16 +648,6 @@ static struct snd_soc_dai_driver ssm2518_dai = {
 
 static int ssm2518_probe(struct snd_soc_codec *codec)
 {
-	struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	codec->control_data = ssm2518->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF);
 }
 
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 1294709..97b0454 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -562,13 +562,6 @@ static int ssm260x_codec_probe(struct snd_soc_codec *codec)
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	codec->control_data = ssm2602->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index 2735361..1257774 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -872,16 +872,6 @@ static int sta32x_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	/* Tell ASoC what kind of I/O to use to read the registers.  ASoC will
-	 * then do the I2C transactions itself.
-	 */
-	codec->control_data = sta32x->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret);
-		goto err;
-	}
-
 	/* Chip documentation explicitly requires that the reset values
 	 * of reserved register bits are left untouched.
 	 * Write the register default value to cache for reserved registers,
@@ -946,10 +936,6 @@ static int sta32x_probe(struct snd_soc_codec *codec)
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
 
 	return 0;
-
-err:
-	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
-	return ret;
 }
 
 static int sta32x_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index f15b0e3..a3c61d3 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -322,16 +322,6 @@ static struct snd_soc_dai_driver sta529_dai = {
 
 static int sta529_probe(struct snd_soc_codec *codec)
 {
-	struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	codec->control_data = sta529->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 	sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index dc9a52f..20864ee 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -559,14 +559,6 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec)
 
 static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Reset codec */
 	snd_soc_write(codec, TLV320AIC23_RESET, 0);
 
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index ff5f23d..43069de 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -296,8 +296,6 @@ static int aic26_probe(struct snd_soc_codec *codec)
 	struct aic26 *aic26 = dev_get_drvdata(codec->dev);
 	int ret, reg;
 
-	snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-
 	aic26->codec = codec;
 
 	/* Reset the codec to power on defaults */
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
new file mode 100644
index 0000000..e60e37b
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -0,0 +1,1295 @@
+/*
+ * ALSA SoC TLV320AIC31XX codec driver
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * Based on ground work by: Ajit Kulkarni <x0175765@ti.com>
+ *
+ * This package is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * THIS PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ * The TLV320AIC31xx series of audio codec is a low-power, highly integrated
+ * high performance codec which provides a stereo DAC, a mono ADC,
+ * and mono/stereo Class-D speaker driver.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/regulator/consumer.h>
+#include <linux/of_gpio.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <dt-bindings/sound/tlv320aic31xx-micbias.h>
+
+#include "tlv320aic31xx.h"
+
+static const struct reg_default aic31xx_reg_defaults[] = {
+	{ AIC31XX_CLKMUX, 0x00 },
+	{ AIC31XX_PLLPR, 0x11 },
+	{ AIC31XX_PLLJ, 0x04 },
+	{ AIC31XX_PLLDMSB, 0x00 },
+	{ AIC31XX_PLLDLSB, 0x00 },
+	{ AIC31XX_NDAC, 0x01 },
+	{ AIC31XX_MDAC, 0x01 },
+	{ AIC31XX_DOSRMSB, 0x00 },
+	{ AIC31XX_DOSRLSB, 0x80 },
+	{ AIC31XX_NADC, 0x01 },
+	{ AIC31XX_MADC, 0x01 },
+	{ AIC31XX_AOSR, 0x80 },
+	{ AIC31XX_IFACE1, 0x00 },
+	{ AIC31XX_DATA_OFFSET, 0x00 },
+	{ AIC31XX_IFACE2, 0x00 },
+	{ AIC31XX_BCLKN, 0x01 },
+	{ AIC31XX_DACSETUP, 0x14 },
+	{ AIC31XX_DACMUTE, 0x0c },
+	{ AIC31XX_LDACVOL, 0x00 },
+	{ AIC31XX_RDACVOL, 0x00 },
+	{ AIC31XX_ADCSETUP, 0x00 },
+	{ AIC31XX_ADCFGA, 0x80 },
+	{ AIC31XX_ADCVOL, 0x00 },
+	{ AIC31XX_HPDRIVER, 0x04 },
+	{ AIC31XX_SPKAMP, 0x06 },
+	{ AIC31XX_DACMIXERROUTE, 0x00 },
+	{ AIC31XX_LANALOGHPL, 0x7f },
+	{ AIC31XX_RANALOGHPR, 0x7f },
+	{ AIC31XX_LANALOGSPL, 0x7f },
+	{ AIC31XX_RANALOGSPR, 0x7f },
+	{ AIC31XX_HPLGAIN, 0x02 },
+	{ AIC31XX_HPRGAIN, 0x02 },
+	{ AIC31XX_SPLGAIN, 0x00 },
+	{ AIC31XX_SPRGAIN, 0x00 },
+	{ AIC31XX_MICBIAS, 0x00 },
+	{ AIC31XX_MICPGA, 0x80 },
+	{ AIC31XX_MICPGAPI, 0x00 },
+	{ AIC31XX_MICPGAMI, 0x00 },
+};
+
+static bool aic31xx_volatile(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case AIC31XX_PAGECTL: /* regmap implementation requires this */
+	case AIC31XX_RESET: /* always clears after write */
+	case AIC31XX_OT_FLAG:
+	case AIC31XX_ADCFLAG:
+	case AIC31XX_DACFLAG1:
+	case AIC31XX_DACFLAG2:
+	case AIC31XX_OFFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRDACFLAG2:
+	case AIC31XX_INTRADCFLAG2:
+		return true;
+	}
+	return false;
+}
+
+static bool aic31xx_writeable(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case AIC31XX_OT_FLAG:
+	case AIC31XX_ADCFLAG:
+	case AIC31XX_DACFLAG1:
+	case AIC31XX_DACFLAG2:
+	case AIC31XX_OFFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */
+	case AIC31XX_INTRDACFLAG2:
+	case AIC31XX_INTRADCFLAG2:
+		return false;
+	}
+	return true;
+}
+
+static const struct regmap_range_cfg aic31xx_ranges[] = {
+	{
+		.range_min = 0,
+		.range_max = 12 * 128,
+		.selector_reg = AIC31XX_PAGECTL,
+		.selector_mask = 0xff,
+		.selector_shift = 0,
+		.window_start = 0,
+		.window_len = 128,
+	},
+};
+
+struct regmap_config aic31xx_i2c_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+	.writeable_reg = aic31xx_writeable,
+	.volatile_reg = aic31xx_volatile,
+	.reg_defaults = aic31xx_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+	.ranges = aic31xx_ranges,
+	.num_ranges = ARRAY_SIZE(aic31xx_ranges),
+	.max_register = 12 * 128,
+};
+
+#define AIC31XX_NUM_SUPPLIES	6
+static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = {
+	"HPVDD",
+	"SPRVDD",
+	"SPLVDD",
+	"AVDD",
+	"IOVDD",
+	"DVDD",
+};
+
+struct aic31xx_disable_nb {
+	struct notifier_block nb;
+	struct aic31xx_priv *aic31xx;
+};
+
+struct aic31xx_priv {
+	struct snd_soc_codec *codec;
+	u8 i2c_regs_status;
+	struct device *dev;
+	struct regmap *regmap;
+	struct aic31xx_pdata pdata;
+	struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES];
+	struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES];
+	unsigned int sysclk;
+	int rate_div_line;
+};
+
+struct aic31xx_rate_divs {
+	u32 mclk;
+	u32 rate;
+	u8 p_val;
+	u8 pll_j;
+	u16 pll_d;
+	u16 dosr;
+	u8 ndac;
+	u8 mdac;
+	u8 aosr;
+	u8 nadc;
+	u8 madc;
+};
+
+/* ADC dividers can be disabled by cofiguring them to 0 */
+static const struct aic31xx_rate_divs aic31xx_divs[] = {
+	/* mclk      rate  pll: p  j	 d     dosr ndac mdac  aors nadc madc */
+	/* 8k rate */
+	{12000000,   8000,	1, 8, 1920,	128,  48,  2,	128,  48,  2},
+	{24000000,   8000,	2, 8, 1920,	128,  48,  2,	128,  48,  2},
+	{25000000,   8000,	2, 7, 8643,	128,  48,  2,	128,  48,  2},
+	/* 11.025k rate */
+	{12000000,  11025,	1, 7, 5264,	128,  32,  2,	128,  32,  2},
+	{24000000,  11025,	2, 7, 5264,	128,  32,  2,	128,  32,  2},
+	{25000000,  11025,	2, 7, 2253,	128,  32,  2,	128,  32,  2},
+	/* 16k rate */
+	{12000000,  16000,	1, 8, 1920,	128,  24,  2,	128,  24,  2},
+	{24000000,  16000,	2, 8, 1920,	128,  24,  2,	128,  24,  2},
+	{25000000,  16000,	2, 7, 8643,	128,  24,  2,	128,  24,  2},
+	/* 22.05k rate */
+	{12000000,  22050,	1, 7, 5264,	128,  16,  2,	128,  16,  2},
+	{24000000,  22050,	2, 7, 5264,	128,  16,  2,	128,  16,  2},
+	{25000000,  22050,	2, 7, 2253,	128,  16,  2,	128,  16,  2},
+	/* 32k rate */
+	{12000000,  32000,	1, 8, 1920,	128,  12,  2,	128,  12,  2},
+	{24000000,  32000,	2, 8, 1920,	128,  12,  2,	128,  12,  2},
+	{25000000,  32000,	2, 7, 8643,	128,  12,  2,	128,  12,  2},
+	/* 44.1k rate */
+	{12000000,  44100,	1, 7, 5264,	128,   8,  2,	128,   8,  2},
+	{24000000,  44100,	2, 7, 5264,	128,   8,  2,	128,   8,  2},
+	{25000000,  44100,	2, 7, 2253,	128,   8,  2,	128,   8,  2},
+	/* 48k rate */
+	{12000000,  48000,	1, 8, 1920,	128,   8,  2,	128,   8,  2},
+	{24000000,  48000,	2, 8, 1920,	128,   8,  2,	128,   8,  2},
+	{25000000,  48000,	2, 7, 8643,	128,   8,  2,	128,   8,  2},
+	/* 88.2k rate */
+	{12000000,  88200,	1, 7, 5264,	 64,   8,  2,	 64,   8,  2},
+	{24000000,  88200,	2, 7, 5264,	 64,   8,  2,	 64,   8,  2},
+	{25000000,  88200,	2, 7, 2253,	 64,   8,  2,	 64,   8,  2},
+	/* 96k rate */
+	{12000000,  96000,	1, 8, 1920,	 64,   8,  2,	 64,   8,  2},
+	{24000000,  96000,	2, 8, 1920,	 64,   8,  2,	 64,   8,  2},
+	{25000000,  96000,	2, 7, 8643,	 64,   8,  2,	 64,   8,  2},
+	/* 176.4k rate */
+	{12000000, 176400,	1, 7, 5264,	 32,   8,  2,	 32,   8,  2},
+	{24000000, 176400,	2, 7, 5264,	 32,   8,  2,	 32,   8,  2},
+	{25000000, 176400,	2, 7, 2253,	 32,   8,  2,	 32,   8,  2},
+	/* 192k rate */
+	{12000000, 192000,	1, 8, 1920,	 32,   8,  2,	 32,   8,  2},
+	{24000000, 192000,	2, 8, 1920,	 32,   8,  2,	 32,   8,  2},
+	{25000000, 192000,	2, 7, 8643,	 32,   8,  2,	 32,   8,  2},
+};
+
+static const char * const ldac_in_text[] = {
+	"Off", "Left Data", "Right Data", "Mono"
+};
+
+static const char * const rdac_in_text[] = {
+	"Off", "Right Data", "Left Data", "Mono"
+};
+
+static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text);
+
+static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text);
+
+static const char * const mic_select_text[] = {
+	"Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm"
+};
+
+static const
+SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text);
+
+static const
+SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text);
+static const
+SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text);
+
+static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0);
+static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0);
+static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0);
+static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0);
+
+/*
+ * controls to be exported to the user space
+ */
+static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
+	SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL,
+			   AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv),
+
+	SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1,
+		       adc_fgain_tlv),
+
+	SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1),
+	SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL,
+			   0, -24, 40, 6, 0, adc_cgain_tlv),
+
+	SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0,
+		       119, 0, mic_pga_tlv),
+
+	SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
+		     AIC31XX_HPRGAIN, 2, 1, 0),
+	SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
+			 AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),
+
+	SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
+			 AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new aic311x_snd_controls[] = {
+	SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN,
+		     AIC31XX_SPRGAIN, 2, 1, 0),
+	SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN,
+			 AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv),
+
+	SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL,
+			 AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new aic310x_snd_controls[] = {
+	SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN,
+		   2, 1, 0),
+	SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN,
+		       3, 3, 0, class_D_drv_tlv),
+
+	SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL,
+		       0, 0x7F, 1, sp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new ldac_in_control =
+	SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum);
+
+static const struct snd_kcontrol_new rdac_in_control =
+	SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum);
+
+int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg,
+		      unsigned int mask, unsigned int wbits, int sleep,
+		      int count)
+{
+	unsigned int bits;
+	int counter = count;
+	int ret = regmap_read(aic31xx->regmap, reg, &bits);
+	while ((bits & mask) != wbits && counter && !ret) {
+		usleep_range(sleep, sleep * 2);
+		ret = regmap_read(aic31xx->regmap, reg, &bits);
+		counter--;
+	}
+	if ((bits & mask) != wbits) {
+		dev_err(aic31xx->dev,
+			"%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n",
+			__func__, reg, bits, wbits, ret, mask,
+			(count - counter) * sleep);
+		ret = -1;
+	}
+	return ret;
+}
+
+#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg))
+
+static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
+				    struct snd_kcontrol *kcontrol, int event)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec);
+	unsigned int reg = AIC31XX_DACFLAG1;
+	unsigned int mask;
+
+	switch (WIDGET_BIT(w->reg, w->shift)) {
+	case WIDGET_BIT(AIC31XX_DACSETUP, 7):
+		mask = AIC31XX_LDACPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_DACSETUP, 6):
+		mask = AIC31XX_RDACPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_HPDRIVER, 7):
+		mask = AIC31XX_HPLDRVPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_HPDRIVER, 6):
+		mask = AIC31XX_HPRDRVPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_SPKAMP, 7):
+		mask = AIC31XX_SPLDRVPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_SPKAMP, 6):
+		mask = AIC31XX_SPRDRVPWRSTATUS_MASK;
+		break;
+	case WIDGET_BIT(AIC31XX_ADCSETUP, 7):
+		mask = AIC31XX_ADCPWRSTATUS_MASK;
+		reg = AIC31XX_ADCFLAG;
+		break;
+	default:
+		dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n",
+			w->name, __func__);
+		return -EINVAL;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100);
+	case SND_SOC_DAPM_POST_PMD:
+		return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100);
+	default:
+		dev_dbg(w->codec->dev,
+			"Unhandled dapm widget event %d from %s\n",
+			event, w->name);
+	}
+	return 0;
+}
+
+static const struct snd_kcontrol_new left_output_switches[] = {
+	SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
+	SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0),
+	SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_output_switches[] = {
+	SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
+	SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new p_term_mic1lp =
+	SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum);
+
+static const struct snd_kcontrol_new p_term_mic1rp =
+	SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum);
+
+static const struct snd_kcontrol_new p_term_mic1lm =
+	SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum);
+
+static const struct snd_kcontrol_new m_term_mic1lm =
+	SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum);
+
+static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch =
+	SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch =
+	SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_spl_switch =
+	SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0);
+
+static const struct snd_kcontrol_new aic31xx_dapm_spr_switch =
+	SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0);
+
+static int mic_bias_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* change mic bias voltage to user defined */
+		snd_soc_update_bits(codec, AIC31XX_MICBIAS,
+				    AIC31XX_MICBIAS_MASK,
+				    aic31xx->pdata.micbias_vg <<
+				    AIC31XX_MICBIAS_SHIFT);
+		dev_dbg(codec->dev, "%s: turned on\n", __func__);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		/* turn mic bias off */
+		snd_soc_update_bits(codec, AIC31XX_MICBIAS,
+				    AIC31XX_MICBIAS_MASK, 0);
+		dev_dbg(codec->dev, "%s: turned off\n", __func__);
+		break;
+	}
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_MUX("DAC Left Input",
+			 SND_SOC_NOPM, 0, 0, &ldac_in_control),
+	SND_SOC_DAPM_MUX("DAC Right Input",
+			 SND_SOC_NOPM, 0, 0, &rdac_in_control),
+	/* DACs */
+	SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback",
+			   AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback",
+			   AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* Output Mixers */
+	SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
+			   left_output_switches,
+			   ARRAY_SIZE(left_output_switches)),
+	SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
+			   right_output_switches,
+			   ARRAY_SIZE(right_output_switches)),
+
+	SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_hpl_switch),
+	SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_hpr_switch),
+
+	/* Output drivers */
+	SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0,
+			       NULL, 0, aic31xx_dapm_power_event,
+			       SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0,
+			       NULL, 0, aic31xx_dapm_power_event,
+			       SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),
+
+	/* ADC */
+	SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
+			   aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			   SND_SOC_DAPM_POST_PMD),
+
+	/* Input Selection to MIC_PGA */
+	SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0,
+			 &p_term_mic1lp),
+	SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0,
+			 &p_term_mic1rp),
+	SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0,
+			 &p_term_mic1lm),
+
+	SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0,
+			 &m_term_mic1lm),
+	/* Enabling & Disabling MIC Gain Ctl */
+	SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA,
+			 7, 1, NULL, 0),
+
+	/* Mic Bias */
+	SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
+			    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MIC1LP"),
+	SND_SOC_DAPM_INPUT("MIC1RP"),
+	SND_SOC_DAPM_INPUT("MIC1LM"),
+};
+
+static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = {
+	/* AIC3111 and AIC3110 have stereo class-D amplifier */
+	SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0,
+			       aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0,
+			       aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_spl_switch),
+	SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_spr_switch),
+	SND_SOC_DAPM_OUTPUT("SPL"),
+	SND_SOC_DAPM_OUTPUT("SPR"),
+};
+
+/* AIC3100 and AIC3120 have only mono class-D amplifier */
+static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = {
+	SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0,
+			       aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			       SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0,
+			    &aic31xx_dapm_spl_switch),
+	SND_SOC_DAPM_OUTPUT("SPK"),
+};
+
+static const struct snd_soc_dapm_route
+aic31xx_audio_map[] = {
+	/* DAC Input Routing */
+	{"DAC Left Input", "Left Data", "DAC IN"},
+	{"DAC Left Input", "Right Data", "DAC IN"},
+	{"DAC Left Input", "Mono", "DAC IN"},
+	{"DAC Right Input", "Left Data", "DAC IN"},
+	{"DAC Right Input", "Right Data", "DAC IN"},
+	{"DAC Right Input", "Mono", "DAC IN"},
+	{"DAC Left", NULL, "DAC Left Input"},
+	{"DAC Right", NULL, "DAC Right Input"},
+
+	/* Mic input */
+	{"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"},
+	{"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"},
+	{"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"},
+	{"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"},
+	{"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"},
+	{"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"},
+	{"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"},
+	{"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"},
+	{"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"},
+
+	{"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"},
+	{"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"},
+	{"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"},
+
+	{"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"},
+	{"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"},
+	{"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"},
+	{"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"},
+
+	{"ADC", NULL, "MIC_GAIN_CTL"},
+
+	/* Left Output */
+	{"Output Left", "From Left DAC", "DAC Left"},
+	{"Output Left", "From MIC1LP", "MIC1LP"},
+	{"Output Left", "From MIC1RP", "MIC1RP"},
+
+	/* Right Output */
+	{"Output Right", "From Right DAC", "DAC Right"},
+	{"Output Right", "From MIC1RP", "MIC1RP"},
+
+	/* HPL path */
+	{"HP Left", "Switch", "Output Left"},
+	{"HPL Driver", NULL, "HP Left"},
+	{"HPL", NULL, "HPL Driver"},
+
+	/* HPR path */
+	{"HP Right", "Switch", "Output Right"},
+	{"HPR Driver", NULL, "HP Right"},
+	{"HPR", NULL, "HPR Driver"},
+};
+
+static const struct snd_soc_dapm_route
+aic311x_audio_map[] = {
+	/* SP L path */
+	{"Speaker Left", "Switch", "Output Left"},
+	{"SPL ClassD", NULL, "Speaker Left"},
+	{"SPL", NULL, "SPL ClassD"},
+
+	/* SP R path */
+	{"Speaker Right", "Switch", "Output Right"},
+	{"SPR ClassD", NULL, "Speaker Right"},
+	{"SPR", NULL, "SPR ClassD"},
+};
+
+static const struct snd_soc_dapm_route
+aic310x_audio_map[] = {
+	/* SP L path */
+	{"Speaker", "Switch", "Output Left"},
+	{"SPK ClassD", NULL, "Speaker"},
+	{"SPK", NULL, "SPK ClassD"},
+};
+
+static int aic31xx_add_controls(struct snd_soc_codec *codec)
+{
+	int ret = 0;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+
+	if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT)
+		ret = snd_soc_add_codec_controls(
+			codec, aic311x_snd_controls,
+			ARRAY_SIZE(aic311x_snd_controls));
+	else
+		ret = snd_soc_add_codec_controls(
+			codec, aic310x_snd_controls,
+			ARRAY_SIZE(aic310x_snd_controls));
+
+	return ret;
+}
+
+static int aic31xx_add_widgets(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) {
+		ret = snd_soc_dapm_new_controls(
+			dapm, aic311x_dapm_widgets,
+			ARRAY_SIZE(aic311x_dapm_widgets));
+		if (ret)
+			return ret;
+
+		ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map,
+					      ARRAY_SIZE(aic311x_audio_map));
+		if (ret)
+			return ret;
+	} else {
+		ret = snd_soc_dapm_new_controls(
+			dapm, aic310x_dapm_widgets,
+			ARRAY_SIZE(aic310x_dapm_widgets));
+		if (ret)
+			return ret;
+
+		ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map,
+					      ARRAY_SIZE(aic310x_audio_map));
+		if (ret)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int aic31xx_setup_pll(struct snd_soc_codec *codec,
+			     struct snd_pcm_hw_params *params)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int bclk_n = 0;
+	int i;
+
+	/* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */
+	snd_soc_update_bits(codec, AIC31XX_CLKMUX,
+			    AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL);
+	snd_soc_update_bits(codec, AIC31XX_IFACE2,
+			    AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK);
+
+	for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) {
+		if (aic31xx_divs[i].rate == params_rate(params) &&
+		    aic31xx_divs[i].mclk == aic31xx->sysclk)
+			break;
+	}
+
+	if (i == ARRAY_SIZE(aic31xx_divs)) {
+		dev_err(codec->dev, "%s: Sampling rate %u not supported\n",
+			__func__, params_rate(params));
+		return -EINVAL;
+	}
+
+	/* PLL configuration */
+	snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK,
+			    (aic31xx_divs[i].p_val << 4) | 0x01);
+	snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j);
+
+	snd_soc_write(codec, AIC31XX_PLLDMSB,
+		      aic31xx_divs[i].pll_d >> 8);
+	snd_soc_write(codec, AIC31XX_PLLDLSB,
+		      aic31xx_divs[i].pll_d & 0xff);
+
+	/* DAC dividers configuration */
+	snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK,
+			    aic31xx_divs[i].ndac);
+	snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK,
+			    aic31xx_divs[i].mdac);
+
+	snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8);
+	snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff);
+
+	/* ADC dividers configuration. Write reset value 1 if not used. */
+	snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK,
+			    aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1);
+	snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK,
+			    aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1);
+
+	snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr);
+
+	/* Bit clock divider configuration. */
+	bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac)
+		/ snd_soc_params_to_frame_size(params);
+	if (bclk_n == 0) {
+		dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n",
+			__func__);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AIC31XX_BCLKN,
+			    AIC31XX_PLL_MASK, bclk_n);
+
+	aic31xx->rate_div_line = i;
+
+	dev_dbg(codec->dev,
+		"pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n",
+		aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d,
+		aic31xx_divs[i].p_val, aic31xx_divs[i].dosr,
+		aic31xx_divs[i].ndac, aic31xx_divs[i].mdac,
+		aic31xx_divs[i].aosr, aic31xx_divs[i].nadc,
+		aic31xx_divs[i].madc, bclk_n);
+
+	return 0;
+}
+
+static int aic31xx_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *tmp)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	u8 data = 0;
+
+	dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n",
+		__func__, params_format(params), params_width(params),
+		params_rate(params));
+
+	switch (params_width(params)) {
+	case 16:
+		break;
+	case 20:
+		data = (AIC31XX_WORD_LEN_20BITS <<
+			AIC31XX_IFACE1_DATALEN_SHIFT);
+		break;
+	case 24:
+		data = (AIC31XX_WORD_LEN_24BITS <<
+			AIC31XX_IFACE1_DATALEN_SHIFT);
+		break;
+	case 32:
+		data = (AIC31XX_WORD_LEN_32BITS <<
+			AIC31XX_IFACE1_DATALEN_SHIFT);
+		break;
+	default:
+		dev_err(codec->dev, "%s: Unsupported format %d\n",
+			__func__, params_format(params));
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AIC31XX_IFACE1,
+			    AIC31XX_IFACE1_DATALEN_MASK,
+			    data);
+
+	return aic31xx_setup_pll(codec, params);
+}
+
+static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	if (mute) {
+		snd_soc_update_bits(codec, AIC31XX_DACMUTE,
+				    AIC31XX_DACMUTE_MASK,
+				    AIC31XX_DACMUTE_MASK);
+	} else {
+		snd_soc_update_bits(codec, AIC31XX_DACMUTE,
+				    AIC31XX_DACMUTE_MASK, 0x0);
+	}
+
+	return 0;
+}
+
+static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			       unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 iface_reg1 = 0;
+	u8 iface_reg3 = 0;
+	u8 dsp_a_val = 0;
+
+	dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER;
+		break;
+	default:
+		dev_alert(codec->dev, "Invalid DAI master/slave interface\n");
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		dsp_a_val = 0x1;
+	case SND_SOC_DAIFMT_DSP_B:
+		/* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			iface_reg3 |= AIC31XX_BCLKINV_MASK;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			break;
+		default:
+			return -EINVAL;
+		}
+		iface_reg1 |= (AIC31XX_DSP_MODE <<
+			       AIC31XX_IFACE1_DATATYPE_SHIFT);
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE <<
+			       AIC31XX_IFACE1_DATATYPE_SHIFT);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE <<
+			       AIC31XX_IFACE1_DATATYPE_SHIFT);
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI interface format\n");
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AIC31XX_IFACE1,
+			    AIC31XX_IFACE1_DATATYPE_MASK |
+			    AIC31XX_IFACE1_MASTER_MASK,
+			    iface_reg1);
+	snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET,
+			    AIC31XX_DATA_OFFSET_MASK,
+			    dsp_a_val);
+	snd_soc_update_bits(codec, AIC31XX_IFACE2,
+			    AIC31XX_BCLKINV_MASK,
+			    iface_reg3);
+
+	return 0;
+}
+
+static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n",
+		__func__, clk_id, freq, dir);
+
+	for (i = 0; aic31xx_divs[i].mclk != freq; i++) {
+		if (i == ARRAY_SIZE(aic31xx_divs)) {
+			dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n",
+				__func__, freq);
+			return -EINVAL;
+		}
+	}
+
+	/* set clock on MCLK, BCLK, or GPIO1 as PLL input */
+	snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK,
+			    clk_id << AIC31XX_PLL_CLKIN_SHIFT);
+
+	aic31xx->sysclk = freq;
+	return 0;
+}
+
+static int aic31xx_regulator_event(struct notifier_block *nb,
+				   unsigned long event, void *data)
+{
+	struct aic31xx_disable_nb *disable_nb =
+		container_of(nb, struct aic31xx_disable_nb, nb);
+	struct aic31xx_priv *aic31xx = disable_nb->aic31xx;
+
+	if (event & REGULATOR_EVENT_DISABLE) {
+		/*
+		 * Put codec to reset and as at least one of the
+		 * supplies was disabled.
+		 */
+		if (gpio_is_valid(aic31xx->pdata.gpio_reset))
+			gpio_set_value(aic31xx->pdata.gpio_reset, 0);
+
+		regcache_mark_dirty(aic31xx->regmap);
+		dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__);
+	}
+
+	return 0;
+}
+
+static void aic31xx_clk_on(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	u8 mask = AIC31XX_PM_MASK;
+	u8 on = AIC31XX_PM_MASK;
+
+	dev_dbg(codec->dev, "codec clock -> on (rate %d)\n",
+		aic31xx_divs[aic31xx->rate_div_line].rate);
+	snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on);
+	mdelay(10);
+	snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on);
+	snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on);
+	if (aic31xx_divs[aic31xx->rate_div_line].nadc)
+		snd_soc_update_bits(codec, AIC31XX_NADC, mask, on);
+	if (aic31xx_divs[aic31xx->rate_div_line].madc)
+		snd_soc_update_bits(codec, AIC31XX_MADC, mask, on);
+	snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on);
+}
+
+static void aic31xx_clk_off(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	u8 mask = AIC31XX_PM_MASK;
+	u8 off = 0;
+
+	dev_dbg(codec->dev, "codec clock -> off\n");
+	snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_MADC, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_NADC, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off);
+	snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off);
+}
+
+static int aic31xx_power_on(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies),
+				    aic31xx->supplies);
+	if (ret)
+		return ret;
+
+	if (gpio_is_valid(aic31xx->pdata.gpio_reset)) {
+		gpio_set_value(aic31xx->pdata.gpio_reset, 1);
+		udelay(100);
+	}
+	regcache_cache_only(aic31xx->regmap, false);
+	ret = regcache_sync(aic31xx->regmap);
+	if (ret != 0) {
+		dev_err(codec->dev,
+			"Failed to restore cache: %d\n", ret);
+		regcache_cache_only(aic31xx->regmap, true);
+		regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies),
+				       aic31xx->supplies);
+		return ret;
+	}
+	return 0;
+}
+
+static int aic31xx_power_off(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	regcache_cache_only(aic31xx->regmap, true);
+	ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies),
+				     aic31xx->supplies);
+
+	return ret;
+}
+
+static int aic31xx_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__,
+		codec->dapm.bias_level, level);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
+			aic31xx_clk_on(codec);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		switch (codec->dapm.bias_level) {
+		case SND_SOC_BIAS_OFF:
+			aic31xx_power_on(codec);
+			break;
+		case SND_SOC_BIAS_PREPARE:
+			aic31xx_clk_off(codec);
+			break;
+		default:
+			BUG();
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		aic31xx_power_off(codec);
+		break;
+	}
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static int aic31xx_suspend(struct snd_soc_codec *codec)
+{
+	aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int aic31xx_resume(struct snd_soc_codec *codec)
+{
+	aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int aic31xx_codec_probe(struct snd_soc_codec *codec)
+{
+	int ret = 0;
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	dev_dbg(aic31xx->dev, "## %s\n", __func__);
+
+	aic31xx = snd_soc_codec_get_drvdata(codec);
+	codec->control_data = aic31xx->regmap;
+
+	aic31xx->codec = codec;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+
+	if (ret != 0) {
+		dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n",
+			ret);
+		return ret;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) {
+		aic31xx->disable_nb[i].nb.notifier_call =
+			aic31xx_regulator_event;
+		aic31xx->disable_nb[i].aic31xx = aic31xx;
+		ret = regulator_register_notifier(aic31xx->supplies[i].consumer,
+						  &aic31xx->disable_nb[i].nb);
+		if (ret) {
+			dev_err(codec->dev,
+				"Failed to request regulator notifier: %d\n",
+				ret);
+			return ret;
+		}
+	}
+
+	regcache_cache_only(aic31xx->regmap, true);
+	regcache_mark_dirty(aic31xx->regmap);
+
+	ret = aic31xx_add_controls(codec);
+	if (ret)
+		return ret;
+
+	ret = aic31xx_add_widgets(codec);
+
+	return ret;
+}
+
+static int aic31xx_codec_remove(struct snd_soc_codec *codec)
+{
+	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+	int i;
+	/* power down chip */
+	aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
+		regulator_unregister_notifier(aic31xx->supplies[i].consumer,
+					      &aic31xx->disable_nb[i].nb);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
+	.probe			= aic31xx_codec_probe,
+	.remove			= aic31xx_codec_remove,
+	.suspend		= aic31xx_suspend,
+	.resume			= aic31xx_resume,
+	.set_bias_level		= aic31xx_set_bias_level,
+	.controls		= aic31xx_snd_controls,
+	.num_controls		= ARRAY_SIZE(aic31xx_snd_controls),
+	.dapm_widgets		= aic31xx_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(aic31xx_dapm_widgets),
+	.dapm_routes		= aic31xx_audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(aic31xx_audio_map),
+};
+
+static struct snd_soc_dai_ops aic31xx_dai_ops = {
+	.hw_params	= aic31xx_hw_params,
+	.set_sysclk	= aic31xx_set_dai_sysclk,
+	.set_fmt	= aic31xx_set_dai_fmt,
+	.digital_mute	= aic31xx_dac_mute,
+};
+
+static struct snd_soc_dai_driver aic31xx_dai_driver[] = {
+	{
+		.name = "tlv320aic31xx-hifi",
+		.playback = {
+			.stream_name	 = "Playback",
+			.channels_min	 = 1,
+			.channels_max	 = 2,
+			.rates		 = AIC31XX_RATES,
+			.formats	 = AIC31XX_FORMATS,
+		},
+		.capture = {
+			.stream_name	 = "Capture",
+			.channels_min	 = 1,
+			.channels_max	 = 2,
+			.rates		 = AIC31XX_RATES,
+			.formats	 = AIC31XX_FORMATS,
+		},
+		.ops = &aic31xx_dai_ops,
+		.symmetric_rates = 1,
+	}
+};
+
+#if defined(CONFIG_OF)
+static const struct of_device_id tlv320aic31xx_of_match[] = {
+	{ .compatible = "ti,tlv320aic310x" },
+	{ .compatible = "ti,tlv320aic311x" },
+	{ .compatible = "ti,tlv320aic3100" },
+	{ .compatible = "ti,tlv320aic3110" },
+	{ .compatible = "ti,tlv320aic3120" },
+	{ .compatible = "ti,tlv320aic3111" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match);
+
+static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
+{
+	struct device_node *np = aic31xx->dev->of_node;
+	unsigned int value = MICBIAS_2_0V;
+	int ret;
+
+	of_property_read_u32(np, "ai31xx-micbias-vg", &value);
+	switch (value) {
+	case MICBIAS_2_0V:
+	case MICBIAS_2_5V:
+	case MICBIAS_AVDDV:
+		aic31xx->pdata.micbias_vg = value;
+		break;
+	default:
+		dev_err(aic31xx->dev,
+			"Bad ai31xx-micbias-vg value %d DT\n",
+			value);
+		aic31xx->pdata.micbias_vg = MICBIAS_2_0V;
+	}
+
+	ret = of_get_named_gpio(np, "gpio-reset", 0);
+	if (ret > 0)
+		aic31xx->pdata.gpio_reset = ret;
+}
+#else /* CONFIG_OF */
+static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx)
+{
+}
+#endif /* CONFIG_OF */
+
+void aic31xx_device_init(struct aic31xx_priv *aic31xx)
+{
+	int ret, i;
+
+	dev_set_drvdata(aic31xx->dev, aic31xx);
+
+	if (dev_get_platdata(aic31xx->dev))
+		memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev),
+		       sizeof(aic31xx->pdata));
+	else if (aic31xx->dev->of_node)
+		aic31xx_pdata_from_of(aic31xx);
+
+	if (aic31xx->pdata.gpio_reset) {
+		ret = devm_gpio_request_one(aic31xx->dev,
+					    aic31xx->pdata.gpio_reset,
+					    GPIOF_OUT_INIT_HIGH,
+					    "aic31xx-reset-pin");
+		if (ret < 0) {
+			dev_err(aic31xx->dev, "not able to acquire gpio\n");
+			return;
+		}
+	}
+
+	for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++)
+		aic31xx->supplies[i].supply = aic31xx_supply_names[i];
+
+	ret = devm_regulator_bulk_get(aic31xx->dev,
+				      ARRAY_SIZE(aic31xx->supplies),
+				      aic31xx->supplies);
+	if (ret != 0)
+		dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret);
+
+}
+
+static int aic31xx_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct aic31xx_priv *aic31xx;
+	int ret;
+	const struct regmap_config *regmap_config;
+
+	dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__,
+		id->name, (int) id->driver_data);
+
+	regmap_config = &aic31xx_i2c_regmap;
+
+	aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL);
+	if (aic31xx == NULL)
+		return -ENOMEM;
+
+	aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config);
+
+	if (IS_ERR(aic31xx->regmap)) {
+		ret = PTR_ERR(aic31xx->regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+	aic31xx->dev = &i2c->dev;
+
+	aic31xx->pdata.codec_type = id->driver_data;
+
+	aic31xx_device_init(aic31xx);
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
+				     aic31xx_dai_driver,
+				     ARRAY_SIZE(aic31xx_dai_driver));
+
+	return ret;
+}
+
+static int aic31xx_i2c_remove(struct i2c_client *i2c)
+{
+	struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev);
+
+	kfree(aic31xx);
+	return 0;
+}
+
+static const struct i2c_device_id aic31xx_i2c_id[] = {
+	{ "tlv320aic310x", AIC3100 },
+	{ "tlv320aic311x", AIC3110 },
+	{ "tlv320aic3100", AIC3100 },
+	{ "tlv320aic3110", AIC3110 },
+	{ "tlv320aic3120", AIC3120 },
+	{ "tlv320aic3111", AIC3111 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);
+
+static struct i2c_driver aic31xx_i2c_driver = {
+	.driver = {
+		.name	= "tlv320aic31xx-codec",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(tlv320aic31xx_of_match),
+	},
+	.probe		= aic31xx_i2c_probe,
+	.remove		= (aic31xx_i2c_remove),
+	.id_table	= aic31xx_i2c_id,
+};
+
+module_i2c_driver(aic31xx_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver");
+MODULE_AUTHOR("Jyri Sarha");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
new file mode 100644
index 0000000..52ed57c
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -0,0 +1,258 @@
+/*
+ * ALSA SoC TLV320AIC31XX codec driver
+ *
+ * Copyright (C) 2013 Texas Instruments, Inc.
+ *
+ * This package is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR
+ * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED
+ * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+ *
+ */
+#ifndef _TLV320AIC31XX_H
+#define _TLV320AIC31XX_H
+
+#define AIC31XX_RATES	SNDRV_PCM_RATE_8000_192000
+
+#define AIC31XX_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+
+#define AIC31XX_STEREO_CLASS_D_BIT	0x1
+#define AIC31XX_MINIDSP_BIT		0x2
+
+enum aic31xx_type {
+	AIC3100	= 0,
+	AIC3110 = AIC31XX_STEREO_CLASS_D_BIT,
+	AIC3120 = AIC31XX_MINIDSP_BIT,
+	AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT),
+};
+
+struct aic31xx_pdata {
+	enum aic31xx_type codec_type;
+	unsigned int gpio_reset;
+	int micbias_vg;
+};
+
+/* Page Control Register */
+#define AIC31XX_PAGECTL				0x00
+
+/* Page 0 Registers */
+/* Software reset register */
+#define AIC31XX_RESET				0x01
+/* OT FLAG register */
+#define AIC31XX_OT_FLAG				0x03
+/* Clock clock Gen muxing, Multiplexers*/
+#define AIC31XX_CLKMUX				0x04
+/* PLL P and R-VAL register */
+#define AIC31XX_PLLPR				0x05
+/* PLL J-VAL register */
+#define AIC31XX_PLLJ				0x06
+/* PLL D-VAL MSB register */
+#define AIC31XX_PLLDMSB				0x07
+/* PLL D-VAL LSB register */
+#define AIC31XX_PLLDLSB				0x08
+/* DAC NDAC_VAL register*/
+#define AIC31XX_NDAC				0x0B
+/* DAC MDAC_VAL register */
+#define AIC31XX_MDAC				0x0C
+/* DAC OSR setting register 1, MSB value */
+#define AIC31XX_DOSRMSB				0x0D
+/* DAC OSR setting register 2, LSB value */
+#define AIC31XX_DOSRLSB				0x0E
+#define AIC31XX_MINI_DSP_INPOL			0x10
+/* Clock setting register 8, PLL */
+#define AIC31XX_NADC				0x12
+/* Clock setting register 9, PLL */
+#define AIC31XX_MADC				0x13
+/* ADC Oversampling (AOSR) Register */
+#define AIC31XX_AOSR				0x14
+/* Clock setting register 9, Multiplexers */
+#define AIC31XX_CLKOUTMUX			0x19
+/* Clock setting register 10, CLOCKOUT M divider value */
+#define AIC31XX_CLKOUTMVAL			0x1A
+/* Audio Interface Setting Register 1 */
+#define AIC31XX_IFACE1				0x1B
+/* Audio Data Slot Offset Programming */
+#define AIC31XX_DATA_OFFSET			0x1C
+/* Audio Interface Setting Register 2 */
+#define AIC31XX_IFACE2				0x1D
+/* Clock setting register 11, BCLK N Divider */
+#define AIC31XX_BCLKN				0x1E
+/* Audio Interface Setting Register 3, Secondary Audio Interface */
+#define AIC31XX_IFACESEC1			0x1F
+/* Audio Interface Setting Register 4 */
+#define AIC31XX_IFACESEC2			0x20
+/* Audio Interface Setting Register 5 */
+#define AIC31XX_IFACESEC3			0x21
+/* I2C Bus Condition */
+#define AIC31XX_I2C				0x22
+/* ADC FLAG */
+#define AIC31XX_ADCFLAG				0x24
+/* DAC Flag Registers */
+#define AIC31XX_DACFLAG1			0x25
+#define AIC31XX_DACFLAG2			0x26
+/* Sticky Interrupt flag (overflow) */
+#define AIC31XX_OFFLAG				0x27
+/* Sticy DAC Interrupt flags */
+#define AIC31XX_INTRDACFLAG			0x2C
+/* Sticy ADC Interrupt flags */
+#define AIC31XX_INTRADCFLAG			0x2D
+/* DAC Interrupt flags 2 */
+#define AIC31XX_INTRDACFLAG2			0x2E
+/* ADC Interrupt flags 2 */
+#define AIC31XX_INTRADCFLAG2			0x2F
+/* INT1 interrupt control */
+#define AIC31XX_INT1CTRL			0x30
+/* INT2 interrupt control */
+#define AIC31XX_INT2CTRL			0x31
+/* GPIO1 control */
+#define AIC31XX_GPIO1				0x33
+
+#define AIC31XX_DACPRB				0x3C
+/* ADC Instruction Set Register */
+#define AIC31XX_ADCPRB				0x3D
+/* DAC channel setup register */
+#define AIC31XX_DACSETUP			0x3F
+/* DAC Mute and volume control register */
+#define AIC31XX_DACMUTE				0x40
+/* Left DAC channel digital volume control */
+#define AIC31XX_LDACVOL				0x41
+/* Right DAC channel digital volume control */
+#define AIC31XX_RDACVOL				0x42
+/* Headset detection */
+#define AIC31XX_HSDETECT			0x43
+/* ADC Digital Mic */
+#define AIC31XX_ADCSETUP			0x51
+/* ADC Digital Volume Control Fine Adjust */
+#define AIC31XX_ADCFGA				0x52
+/* ADC Digital Volume Control Coarse Adjust */
+#define AIC31XX_ADCVOL				0x53
+
+
+/* Page 1 Registers */
+/* Headphone drivers */
+#define AIC31XX_HPDRIVER			0x9F
+/* Class-D Speakear Amplifier */
+#define AIC31XX_SPKAMP				0xA0
+/* HP Output Drivers POP Removal Settings */
+#define AIC31XX_HPPOP				0xA1
+/* Output Driver PGA Ramp-Down Period Control */
+#define AIC31XX_SPPGARAMP			0xA2
+/* DAC_L and DAC_R Output Mixer Routing */
+#define AIC31XX_DACMIXERROUTE			0xA3
+/* Left Analog Vol to HPL */
+#define AIC31XX_LANALOGHPL			0xA4
+/* Right Analog Vol to HPR */
+#define AIC31XX_RANALOGHPR			0xA5
+/* Left Analog Vol to SPL */
+#define AIC31XX_LANALOGSPL			0xA6
+/* Right Analog Vol to SPR */
+#define AIC31XX_RANALOGSPR			0xA7
+/* HPL Driver */
+#define AIC31XX_HPLGAIN				0xA8
+/* HPR Driver */
+#define AIC31XX_HPRGAIN				0xA9
+/* SPL Driver */
+#define AIC31XX_SPLGAIN				0xAA
+/* SPR Driver */
+#define AIC31XX_SPRGAIN				0xAB
+/* HP Driver Control */
+#define AIC31XX_HPCONTROL			0xAC
+/* MIC Bias Control */
+#define AIC31XX_MICBIAS				0xAE
+/* MIC PGA*/
+#define AIC31XX_MICPGA				0xAF
+/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */
+#define AIC31XX_MICPGAPI			0xB0
+/* ADC Input Selection for M-Terminal */
+#define AIC31XX_MICPGAMI			0xB1
+/* Input CM Settings */
+#define AIC31XX_MICPGACM			0xB2
+
+/* Bits, masks and shifts */
+
+/* AIC31XX_CLKMUX */
+#define AIC31XX_PLL_CLKIN_MASK			0x0c
+#define AIC31XX_PLL_CLKIN_SHIFT			2
+#define AIC31XX_PLL_CLKIN_MCLK			0
+#define AIC31XX_CODEC_CLKIN_MASK		0x03
+#define AIC31XX_CODEC_CLKIN_SHIFT		0
+#define AIC31XX_CODEC_CLKIN_PLL			3
+#define AIC31XX_CODEC_CLKIN_BCLK		1
+
+/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC,
+   AIC31XX_BCLKN */
+#define AIC31XX_PLL_MASK		0x7f
+#define AIC31XX_PM_MASK			0x80
+
+/* AIC31XX_IFACE1 */
+#define AIC31XX_WORD_LEN_16BITS		0x00
+#define AIC31XX_WORD_LEN_20BITS		0x01
+#define AIC31XX_WORD_LEN_24BITS		0x02
+#define AIC31XX_WORD_LEN_32BITS		0x03
+#define AIC31XX_IFACE1_DATALEN_MASK	0x30
+#define AIC31XX_IFACE1_DATALEN_SHIFT	(4)
+#define AIC31XX_IFACE1_DATATYPE_MASK	0xC0
+#define AIC31XX_IFACE1_DATATYPE_SHIFT	(6)
+#define AIC31XX_I2S_MODE		0x00
+#define AIC31XX_DSP_MODE		0x01
+#define AIC31XX_RIGHT_JUSTIFIED_MODE	0x02
+#define AIC31XX_LEFT_JUSTIFIED_MODE	0x03
+#define AIC31XX_IFACE1_MASTER_MASK	0x0C
+#define AIC31XX_BCLK_MASTER		0x08
+#define AIC31XX_WCLK_MASTER		0x04
+
+/* AIC31XX_DATA_OFFSET */
+#define AIC31XX_DATA_OFFSET_MASK	0xFF
+
+/* AIC31XX_IFACE2 */
+#define AIC31XX_BCLKINV_MASK		0x08
+#define AIC31XX_BDIVCLK_MASK		0x03
+#define AIC31XX_DAC2BCLK		0x00
+#define AIC31XX_DACMOD2BCLK		0x01
+#define AIC31XX_ADC2BCLK		0x02
+#define AIC31XX_ADCMOD2BCLK		0x03
+
+/* AIC31XX_ADCFLAG */
+#define AIC31XX_ADCPWRSTATUS_MASK		0x40
+
+/* AIC31XX_DACFLAG1 */
+#define AIC31XX_LDACPWRSTATUS_MASK		0x80
+#define AIC31XX_RDACPWRSTATUS_MASK		0x08
+#define AIC31XX_HPLDRVPWRSTATUS_MASK		0x20
+#define AIC31XX_HPRDRVPWRSTATUS_MASK		0x02
+#define AIC31XX_SPLDRVPWRSTATUS_MASK		0x10
+#define AIC31XX_SPRDRVPWRSTATUS_MASK		0x01
+
+/* AIC31XX_INTRDACFLAG */
+#define AIC31XX_HPSCDETECT_MASK			0x80
+#define AIC31XX_BUTTONPRESS_MASK		0x20
+#define AIC31XX_HSPLUG_MASK			0x10
+#define AIC31XX_LDRCTHRES_MASK			0x08
+#define AIC31XX_RDRCTHRES_MASK			0x04
+#define AIC31XX_DACSINT_MASK			0x02
+#define AIC31XX_DACAINT_MASK			0x01
+
+/* AIC31XX_INT1CTRL */
+#define AIC31XX_HSPLUGDET_MASK			0x80
+#define AIC31XX_BUTTONPRESSDET_MASK		0x40
+#define AIC31XX_DRCTHRES_MASK			0x20
+#define AIC31XX_AGCNOISE_MASK			0x10
+#define AIC31XX_OC_MASK				0x08
+#define AIC31XX_ENGINE_MASK			0x04
+
+/* AIC31XX_DACSETUP */
+#define AIC31XX_SOFTSTEP_MASK			0x03
+
+/* AIC31XX_DACMUTE */
+#define AIC31XX_DACMUTE_MASK			0x0C
+
+/* AIC31XX_MICBIAS */
+#define AIC31XX_MICBIAS_MASK			0x03
+#define AIC31XX_MICBIAS_SHIFT			0
+
+#endif	/* _TLV320AIC31XX_H */
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index c6bd7e7..1d9b117 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -614,8 +614,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
 	u32 tmp_reg;
 
-	snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-
 	if (gpio_is_valid(aic32x4->rstn_gpio)) {
 		ndelay(10);
 		gpio_set_value(aic32x4->rstn_gpio, 1);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 470fbfb..b183510 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1344,12 +1344,6 @@ static int aic3x_probe(struct snd_soc_codec *codec)
 	INIT_LIST_HEAD(&aic3x->list);
 	aic3x->codec = codec;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) {
 		aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event;
 		aic3x->disable_nb[i].aic3x = aic3x;
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 7935161..6bfc8a1 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -122,7 +122,6 @@ struct tlv320dac33_priv {
 	unsigned int uthr;
 
 	enum dac33_state state;
-	enum snd_soc_control_type control_type;
 	void *control_data;
 };
 
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 8ae5027..83a2c87 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -786,8 +786,6 @@ static int wm2000_probe(struct snd_soc_codec *codec)
 {
 	struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev);
 
-	snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP);
-
 	/* This will trigger a transition to standby mode by default */
 	wm2000_anc_set_mode(wm2000);
 
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index 1e0a083..2e721e0 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1554,15 +1554,8 @@ static int wm2200_probe(struct snd_soc_codec *codec)
 	int ret;
 
 	wm2200->codec = codec;
-	codec->control_data = wm2200->regmap;
 	codec->dapm.bias_level = SND_SOC_BIAS_OFF;
 
-	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2);
 	if (ret != 0)
 		return ret;
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index d3fa65f..eca983f 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2343,13 +2343,6 @@ static int wm5100_probe(struct snd_soc_codec *codec)
 	int ret, i;
 
 	wm5100->codec = codec;
-	codec->control_data = wm5100->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++)
 		snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU,
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 3410905..dcf1d12 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1760,9 +1760,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec)
 	struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	codec->control_data = priv->core.arizona->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+	ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap);
 	if (ret != 0)
 		return ret;
 
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index d7bf884..df5a38d 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1587,10 +1587,9 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 	struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	codec->control_data = priv->core.arizona->regmap;
 	priv->core.arizona->dapm = &codec->dapm;
 
-	ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+	ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap);
 	if (ret != 0)
 		return ret;
 
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index a183dcf..757256b 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1505,9 +1505,7 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;
 
-	codec->control_data = wm8350->regmap;
-
-	snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
+	snd_soc_codec_set_cache_io(codec, wm8350->regmap);
 
 	/* Put the codec into reset if it wasn't already */
 	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 6d684d9..146564f 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1316,10 +1316,9 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec)
 
 	snd_soc_codec_set_drvdata(codec, priv);
 	priv->wm8400 = wm8400;
-	codec->control_data = wm8400->regmap;
 	priv->codec = codec;
 
-	snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
+	snd_soc_codec_set_cache_io(codec, wm8400->regmap);
 
 	ret = devm_regulator_bulk_get(wm8400->dev,
 				 ARRAY_SIZE(power), &power[0]);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 7df7d45..1c1e328 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -589,20 +589,12 @@ static int wm8510_resume(struct snd_soc_codec *codec)
 
 static int wm8510_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	wm8510_reset(codec);
 
 	/* power on device */
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	return ret;
+	return 0;
 }
 
 /* power down chip */
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 5dfd571..601ee81 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -392,18 +392,11 @@ static int wm8523_resume(struct snd_soc_codec *codec)
 static int wm8523_probe(struct snd_soc_codec *codec)
 {
 	struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
-	int ret;
 
 	wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0];
 	wm8523->rate_constraint.count =
 		ARRAY_SIZE(wm8523->rate_constraint_list);
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Change some default settings - latch VU and enable ZC */
 	snd_soc_update_bits(codec, WM8523_DAC_GAINR,
 			    WM8523_DACR_VU, WM8523_DACR_VU);
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 318989a..7558c83 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -869,12 +869,6 @@ static int wm8580_probe(struct snd_soc_codec *codec)
 	struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies),
 				    wm8580->supplies);
 	if (ret != 0) {
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 6efcc40..b0fbcb3 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -367,12 +367,6 @@ static int wm8711_probe(struct snd_soc_codec *codec)
 {
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = wm8711_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index cd89033..bac7fc2 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -228,19 +228,10 @@ static int wm8728_resume(struct snd_soc_codec *codec)
 
 static int wm8728_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n",
-		       ret);
-		return ret;
-	}
-
 	/* power on device */
 	wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	return ret;
+	return 0;
 }
 
 static int wm8728_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index d9655f9..d74f439 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -583,13 +583,6 @@ static int wm8731_probe(struct snd_soc_codec *codec)
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0, i;
 
-	codec->control_data = wm8731->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++)
 		wm8731->supplies[i].supply = wm8731_supply_names[i];
 
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index ecc4e87..b27f26cd 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -570,12 +570,6 @@ static int wm8737_probe(struct snd_soc_codec *codec)
 	struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies),
 				    wm8737->supplies);
 	if (ret != 0) {
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index dd02ebf..b33542a 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -429,12 +429,6 @@ static int wm8741_probe(struct snd_soc_codec *codec)
 		goto err_get;
 	}
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		goto err_enable;
-	}
-
 	ret = wm8741_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 78616a6..33990b6 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -702,12 +702,6 @@ static int wm8750_probe(struct snd_soc_codec *codec)
 {
 	int ret;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = wm8750_reset(codec);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8750: failed to reset: %d\n", ret);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 6a6855d..cbb8d55 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1470,13 +1470,6 @@ static int wm8753_probe(struct snd_soc_codec *codec)
 
 	INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work);
 
-	codec->control_data = wm8753->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = wm8753_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 5bce210..c61aeb3 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -580,12 +580,6 @@ static int wm8770_probe(struct snd_soc_codec *codec)
 	wm8770 = snd_soc_codec_get_drvdata(codec);
 	wm8770->codec = codec;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies),
 				    wm8770->supplies);
 	if (ret) {
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index ef82467..70952ce 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -430,12 +430,6 @@ static int wm8776_probe(struct snd_soc_codec *codec)
 {
 	int ret = 0;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = wm8776_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 72d12bb..ee76f0f 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -546,14 +546,6 @@ static int wm8804_probe(struct snd_soc_codec *codec)
 
 	wm8804 = snd_soc_codec_get_drvdata(codec);
 
-	codec->control_data = wm8804->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
-		return ret;
-	}
-
 	for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++)
 		wm8804->supplies[i].supply = wm8804_supply_names[i];
 
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 43c2201..d09fdce 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -1178,13 +1178,7 @@ static int wm8900_resume(struct snd_soc_codec *codec)
 
 static int wm8900_probe(struct snd_soc_codec *codec)
 {
-	int ret = 0, reg;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
+	int reg;
 
 	reg = snd_soc_read(codec, WM8900_REG_ID);
 	if (reg != 0x8900) {
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index b82b70a..b0084a1 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1897,21 +1897,13 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903)
 static int wm8903_probe(struct snd_soc_codec *codec)
 {
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-	int ret;
 
 	wm8903->codec = codec;
-	codec->control_data = wm8903->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	/* power on device */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	return ret;
+	return 0;
 }
 
 /* power down chip */
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 27299cd..49c35c3 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2048,9 +2048,6 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec)
 static int wm8904_probe(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	codec->control_data = wm8904->regmap;
 
 	switch (wm8904->devtype) {
 	case WM8904:
@@ -2064,12 +2061,6 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 		return -EINVAL;
 	}
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	wm8904_handle_pdata(codec);
 
 	wm8904_add_widgets(codec);
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 87f032d..fc6eec9 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -712,12 +712,6 @@ static int wm8940_probe(struct snd_soc_codec *codec)
 	int ret;
 	u16 reg;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = wm8940_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index d4dcaec..fecd4e4 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -895,14 +895,6 @@ static int wm8955_probe(struct snd_soc_codec *codec)
 	struct wm8955_pdata *pdata = dev_get_platdata(codec->dev);
 	int ret, i;
 
-	codec->control_data = wm8955->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	for (i = 0; i < ARRAY_SIZE(wm8955->supplies); i++)
 		wm8955->supplies[i].supply = wm8955_supply_names[i];
 
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index f156010..d04e9ca 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -976,12 +976,6 @@ static int wm8960_probe(struct snd_soc_codec *codec)
 			wm8960->set_bias_level = wm8960_set_bias_level_capless;
 	}
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = wm8960_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index ce8fa6e..9c88f04 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -836,15 +836,8 @@ static struct snd_soc_dai_driver wm8961_dai = {
 static int wm8961_probe(struct snd_soc_codec *codec)
 {
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret = 0;
 	u16 reg;
 
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Enable class W */
 	reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B);
 	reg |= WM8961_CP_DYN_PWR_MASK;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 62af9dc..5522d25 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3424,13 +3424,6 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	bool dmicclk, dmicdat;
 
 	wm8962->codec = codec;
-	codec->control_data = wm8962->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	wm8962->disable_nb[0].notifier_call = wm8962_regulator_event_0;
 	wm8962->disable_nb[1].notifier_call = wm8962_regulator_event_1;
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 67aba78..09b7b42 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -648,12 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec)
 	int ret = 0;
 	u16 reg;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work);
 	wm8971_workq = create_workqueue("wm8971");
 	if (wm8971_workq == NULL)
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 6e16c43..0627c56 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -593,12 +593,6 @@ static int wm8974_probe(struct snd_soc_codec *codec)
 {
 	int ret = 0;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = wm8974_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index a9e2f46..28ef46c 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -975,19 +975,13 @@ static const int update_reg[] = {
 static int wm8978_probe(struct snd_soc_codec *codec)
 {
 	struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec);
-	int ret = 0, i;
+	int i;
 
 	/*
 	 * Set default system clock to PLL, it is more precise, this is also the
 	 * default hardware setting
 	 */
 	wm8978->sysclk = WM8978_PLL;
-	codec->control_data = wm8978->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	/*
 	 * Set the update bit in all registers, that have one. This way all
diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c
index 58f0551..2b9bfa5 100644
--- a/sound/soc/codecs/wm8983.c
+++ b/sound/soc/codecs/wm8983.c
@@ -995,12 +995,6 @@ static int wm8983_probe(struct snd_soc_codec *codec)
 	int ret;
 	int i;
 
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
-		return ret;
-	}
-
 	ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c
index d786f2b..5473dc9 100644
--- a/sound/soc/codecs/wm8985.c
+++ b/sound/soc/codecs/wm8985.c
@@ -995,13 +995,6 @@ static int wm8985_probe(struct snd_soc_codec *codec)
 	int ret;
 
 	wm8985 = snd_soc_codec_get_drvdata(codec);
-	codec->control_data = wm8985->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
-		return ret;
-	}
 
 	for (i = 0; i < ARRAY_SIZE(wm8985->supplies); i++)
 		wm8985->supplies[i].supply = wm8985_supply_names[i];
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 0277a76..3a1ae4f5 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -810,16 +810,8 @@ static int wm8988_resume(struct snd_soc_codec *codec)
 
 static int wm8988_probe(struct snd_soc_codec *codec)
 {
-	struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
 
-	codec->control_data = wm8988->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	ret = wm8988_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 33f53ab..c413c19 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1289,14 +1289,6 @@ static int wm8990_resume(struct snd_soc_codec *codec)
  */
 static int wm8990_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	wm8990_reset(codec);
 
 	/* charge output caps */
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 32d2195..844cc4a 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -1248,14 +1248,6 @@ static int wm8991_remove(struct snd_soc_codec *codec)
 
 static int wm8991_probe(struct snd_soc_codec *codec)
 {
-	int ret;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
-		return ret;
-	}
-
 	wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 7b0630a..f825dc0 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1493,13 +1493,6 @@ static int wm8993_probe(struct snd_soc_codec *codec)
 	wm8993->hubs_data.dcs_codes_r = -2;
 	wm8993->hubs_data.series_startup = 1;
 
-	codec->control_data = wm8993->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Latch volume update bits and default ZC on */
 	snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME,
 			    WM8993_DAC_VU, WM8993_DAC_VU);
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 79854cb..6303537 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3998,9 +3998,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	int ret, i;
 
 	wm8994->hubs.codec = codec;
-	codec->control_data = control->regmap;
 
-	snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
+	snd_soc_codec_set_cache_io(codec, control->regmap);
 
 	mutex_init(&wm8994->accdet_lock);
 	INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap,
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index ddb197d..d3152cf5 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -2042,13 +2042,6 @@ static int wm8995_probe(struct snd_soc_codec *codec)
 	wm8995 = snd_soc_codec_get_drvdata(codec);
 	wm8995->codec = codec;
 
-	codec->control_data = wm8995->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret);
-		return ret;
-	}
-
 	for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++)
 		wm8995->supplies[i].supply = wm8995_supply_names[i];
 
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index c8244af..c6cbb3b 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2633,14 +2633,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 	init_completion(&wm8996->dcs_done);
 	init_completion(&wm8996->fll_lock);
 
-	codec->control_data = wm8996->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		goto err;
-	}
-
 	if (wm8996->pdata.num_retune_mobile_cfgs)
 		wm8996_retune_mobile_pdata(codec);
 	else
@@ -2679,13 +2671,11 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 		} else {
 			dev_err(codec->dev, "Failed to request IRQ: %d\n",
 				ret);
+			return ret;
 		}
 	}
 
 	return 0;
-
-err:
-	return ret;
 }
 
 static int wm8996_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index e10f44d..004186b 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -1053,9 +1053,7 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec)
 	struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	codec->control_data = priv->core.arizona->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+	ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap);
 	if (ret != 0)
 		return ret;
 
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 721cee7..d18eff3 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1260,15 +1260,6 @@ static struct snd_soc_dai_driver wm9081_dai = {
 static int wm9081_probe(struct snd_soc_codec *codec)
 {
 	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
-	int ret;
-
-	codec->control_data = wm9081->regmap;
-
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
 
 	/* Enable zero cross by default */
 	snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT,
@@ -1283,7 +1274,7 @@ static int wm9081_probe(struct snd_soc_codec *codec)
 				     ARRAY_SIZE(wm9081_eq_controls));
 	}
 
-	return ret;
+	return 0;
 }
 
 static int wm9081_remove(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index a07fe16..8793417 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -522,16 +522,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec,
 
 static int wm9090_probe(struct snd_soc_codec *codec)
 {
-	struct wm9090_priv *wm9090 = dev_get_drvdata(codec->dev);
-	int ret;
-
-	codec->control_data = wm9090->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
 	/* Configure some defaults; they will be written out when we
 	 * bring the bias up.
 	 */
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 4577b69..3c81b38 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -32,7 +32,7 @@ config SND_SOC_INTEL_BAYTRAIL
 
 config SND_SOC_INTEL_HASWELL_MACH
 	tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint"
-	depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS
+	depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C
 	select SND_SOC_INTEL_HASWELL
 	select SND_SOC_RT5640
 	help
@@ -43,7 +43,7 @@ config SND_SOC_INTEL_HASWELL_MACH
 
 config SND_SOC_INTEL_BYT_RT5640_MACH
 	tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec"
-	depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS
+	depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C
 	select SND_SOC_INTEL_BAYTRAIL
 	select SND_SOC_RT5640
 	help
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 78ed4a4..2dc3ecf 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,11 +1,19 @@
 config SND_KIRKWOOD_SOC
 	tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
-	depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
+	depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Kirkwood I2S interface. You will also need to select the
 	  audio interfaces to support below.
 
+config SND_KIRKWOOD_SOC_ARMADA370_DB
+	tristate "SoC Audio support for Armada 370 DB"
+	depends on SND_KIRKWOOD_SOC && (ARCH_MVEBU || COMPILE_TEST) && I2C
+	select SND_SOC_CS42L51
+	help
+	  Say Y if you want to add support for SoC audio on
+	  the Armada 370 Development Board.
+
 config SND_KIRKWOOD_SOC_OPENRD
 	tristate "SoC Audio support for Kirkwood Openrd Client"
 	depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST)
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
index 9e78138..7c1d8fe 100644
--- a/sound/soc/kirkwood/Makefile
+++ b/sound/soc/kirkwood/Makefile
@@ -4,6 +4,8 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
 
 snd-soc-openrd-objs := kirkwood-openrd.o
 snd-soc-t5325-objs := kirkwood-t5325.o
+snd-soc-armada-370-db-objs := armada-370-db.o
 
+obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o
 obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
 obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o
diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c
new file mode 100644
index 0000000..977639b
--- /dev/null
+++ b/sound/soc/kirkwood/armada-370-db.c
@@ -0,0 +1,120 @@
+/*
+ * Copyright (C) 2014 Marvell
+ *
+ * Thomas Petazzoni <thomas.petazzoni@free-electrons.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation; either version 2 of the
+ * License, or (at your option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <linux/of.h>
+#include <linux/platform_data/asoc-kirkwood.h>
+#include "../codecs/cs42l51.h"
+
+static int a370db_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	unsigned int freq;
+
+	switch (params_rate(params)) {
+	default:
+	case 44100:
+		freq = 11289600;
+		break;
+	case 48000:
+		freq = 12288000;
+		break;
+	case 96000:
+		freq = 24576000;
+		break;
+	}
+
+	return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops a370db_ops = {
+	.hw_params = a370db_hw_params,
+};
+
+static const struct snd_soc_dapm_widget a370db_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Out Jack", NULL),
+	SND_SOC_DAPM_LINE("In Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route a370db_route[] = {
+	{ "Out Jack",	NULL,	"HPL" },
+	{ "Out Jack",	NULL,	"HPR" },
+	{ "AIN1L",	NULL,	"In Jack" },
+	{ "AIN1L",	NULL,	"In Jack" },
+};
+
+static struct snd_soc_dai_link a370db_dai[] = {
+{
+	.name = "CS42L51",
+	.stream_name = "analog",
+	.cpu_dai_name = "i2s",
+	.codec_dai_name = "cs42l51-hifi",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS,
+	.ops = &a370db_ops,
+},
+};
+
+static struct snd_soc_card a370db = {
+	.name = "a370db",
+	.owner = THIS_MODULE,
+	.dai_link = a370db_dai,
+	.num_links = ARRAY_SIZE(a370db_dai),
+	.dapm_widgets = a370db_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(a370db_dapm_widgets),
+	.dapm_routes = a370db_route,
+	.num_dapm_routes = ARRAY_SIZE(a370db_route),
+};
+
+static int a370db_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &a370db;
+
+	card->dev = &pdev->dev;
+
+	a370db_dai[0].cpu_of_node =
+		of_parse_phandle(pdev->dev.of_node,
+				 "marvell,audio-controller", 0);
+	a370db_dai[0].platform_of_node = a370db_dai[0].cpu_of_node;
+
+	a370db_dai[0].codec_of_node =
+		of_parse_phandle(pdev->dev.of_node,
+				 "marvell,audio-codec", 0);
+
+	return devm_snd_soc_register_card(card->dev, card);
+}
+
+static const struct of_device_id a370db_dt_ids[] = {
+	{ .compatible = "marvell,a370db-audio" },
+	{ },
+};
+
+static struct platform_driver a370db_driver = {
+	.driver		= {
+		.name	= "a370db-audio",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(a370db_dt_ids),
+	},
+	.probe		= a370db_probe,
+};
+
+module_platform_driver(a370db_driver);
+
+MODULE_AUTHOR("Thomas Petazzoni <thomas.petazzoni@free-electrons.com>");
+MODULE_DESCRIPTION("ALSA SoC a370db audio client");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:a370db-audio");
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index 3920a5e..9f84222 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -633,6 +633,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
 static struct of_device_id mvebu_audio_of_match[] = {
 	{ .compatible = "marvell,kirkwood-audio" },
 	{ .compatible = "marvell,dove-audio" },
+	{ .compatible = "marvell,armada370-audio" },
 	{ }
 };
 MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index f141435..56a5219 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -325,7 +325,7 @@ static void cx81801_close(struct tty_struct *tty)
 
 	snd_soc_dapm_sync_unlocked(dapm);
 
-	snd_soc_dapm_mutex_unlock(codec);
+	snd_soc_dapm_mutex_unlock(dapm);
 }
 
 /* Line discipline .hangup() */
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 41ab6678..259e048 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -41,9 +41,8 @@ static int magician_hp_switch;
 static int magician_spk_switch = 1;
 static int magician_in_sel = MAGICIAN_MIC;
 
-static void magician_ext_control(struct snd_soc_codec *codec)
+static void magician_ext_control(struct snd_soc_dapm_context *dapm)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	snd_soc_dapm_mutex_lock(dapm);
 
@@ -75,10 +74,9 @@ static void magician_ext_control(struct snd_soc_codec *codec)
 static int magician_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
-	magician_ext_control(codec);
+	magician_ext_control(&rtd->card->dapm);
 
 	return 0;
 }
@@ -277,13 +275,13 @@ static int magician_get_hp(struct snd_kcontrol *kcontrol,
 static int magician_set_hp(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 
 	if (magician_hp_switch == ucontrol->value.integer.value[0])
 		return 0;
 
 	magician_hp_switch = ucontrol->value.integer.value[0];
-	magician_ext_control(codec);
+	magician_ext_control(&card->dapm);
 	return 1;
 }
 
@@ -297,13 +295,13 @@ static int magician_get_spk(struct snd_kcontrol *kcontrol,
 static int magician_set_spk(struct snd_kcontrol *kcontrol,
 			    struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 
 	if (magician_spk_switch == ucontrol->value.integer.value[0])
 		return 0;
 
 	magician_spk_switch = ucontrol->value.integer.value[0];
-	magician_ext_control(codec);
+	magician_ext_control(&card->dapm);
 	return 1;
 }
 
@@ -400,7 +398,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
 
 	/* NC codec pins */
 	snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
@@ -410,19 +407,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_dapm_nc_pin(dapm, "VINL");
 	snd_soc_dapm_nc_pin(dapm, "VINR");
 
-	/* Add magician specific controls */
-	err = snd_soc_add_codec_controls(codec, uda1380_magician_controls,
-				ARRAY_SIZE(uda1380_magician_controls));
-	if (err < 0)
-		return err;
-
-	/* Add magician specific widgets */
-	snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
-				  ARRAY_SIZE(uda1380_dapm_widgets));
-
-	/* Set up magician specific audio path interconnects */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
 	return 0;
 }
 
@@ -456,6 +440,12 @@ static struct snd_soc_card snd_soc_card_magician = {
 	.dai_link = magician_dai,
 	.num_links = ARRAY_SIZE(magician_dai),
 
+	.controls = uda1380_magician_controls,
+	.num_controls = ARRAY_SIZE(uda1380_magician_controls),
+	.dapm_widgets = uda1380_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
 };
 
 static struct platform_device *magician_snd_device;
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index cead165..4a956d1 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -44,9 +44,8 @@
 static int tosa_jack_func;
 static int tosa_spk_func;
 
-static void tosa_ext_control(struct snd_soc_codec *codec)
+static void tosa_ext_control(struct snd_soc_dapm_context *dapm)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 
 	snd_soc_dapm_mutex_lock(dapm);
 
@@ -82,10 +81,9 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
-	tosa_ext_control(codec);
+	tosa_ext_control(&rtd->card->dapm);
 
 	return 0;
 }
@@ -104,13 +102,13 @@ static int tosa_get_jack(struct snd_kcontrol *kcontrol,
 static int tosa_set_jack(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 
 	if (tosa_jack_func == ucontrol->value.integer.value[0])
 		return 0;
 
 	tosa_jack_func = ucontrol->value.integer.value[0];
-	tosa_ext_control(codec);
+	tosa_ext_control(&card->dapm);
 	return 1;
 }
 
@@ -124,13 +122,13 @@ static int tosa_get_spk(struct snd_kcontrol *kcontrol,
 static int tosa_set_spk(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
 
 	if (tosa_spk_func == ucontrol->value.integer.value[0])
 		return 0;
 
 	tosa_spk_func = ucontrol->value.integer.value[0];
-	tosa_ext_control(codec);
+	tosa_ext_control(&card->dapm);
 	return 1;
 }
 
@@ -191,24 +189,10 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
 
 	snd_soc_dapm_nc_pin(dapm, "OUT3");
 	snd_soc_dapm_nc_pin(dapm, "MONOOUT");
 
-	/* add tosa specific controls */
-	err = snd_soc_add_codec_controls(codec, tosa_controls,
-				ARRAY_SIZE(tosa_controls));
-	if (err < 0)
-		return err;
-
-	/* add tosa specific widgets */
-	snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets,
-				  ARRAY_SIZE(tosa_dapm_widgets));
-
-	/* set up tosa specific audio path audio_map */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
 	return 0;
 }
 
@@ -239,6 +223,13 @@ static struct snd_soc_card tosa = {
 	.owner = THIS_MODULE,
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
+
+	.controls = tosa_controls,
+	.num_controls = ARRAY_SIZE(tosa_controls),
+	.dapm_widgets = tosa_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
 };
 
 static int tosa_probe(struct platform_device *pdev)
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 945e8ab..0b21d1d 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -104,8 +104,8 @@ static int output_type_get(struct snd_kcontrol *kcontrol,
 static int output_type_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec = kcontrol->private_data;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_card *card = kcontrol->private_data;
+	struct snd_soc_dapm_context *dapm = &card->dapm;
 	unsigned int val = (ucontrol->value.enumerated.item[0] != 0);
 	char *differential = "Audio Out Differential";
 	char *stereo = "Audio Out Stereo";
@@ -137,13 +137,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	/* Add s6105 specific widgets */
-	snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
-				  ARRAY_SIZE(aic3x_dapm_widgets));
-
-	/* Set up s6105 specific audio path audio_map */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+	struct snd_soc_card *card = rtd->card;
 
 	/* not present */
 	snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
@@ -157,17 +151,10 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_dapm_nc_pin(dapm, "RLOUT");
 	snd_soc_dapm_nc_pin(dapm, "HPRCOM");
 
-	/* always connected */
-	snd_soc_dapm_enable_pin(dapm, "Audio In");
-
 	/* must correspond to audio_out_mux.private_value initializer */
-	snd_soc_dapm_disable_pin(dapm, "Audio Out Differential");
-	snd_soc_dapm_sync(dapm);
-	snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo");
-
-	snd_soc_dapm_sync(dapm);
+	snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential");
 
-	snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec));
+	snd_ctl_add(card->snd_card, snd_ctl_new1(&audio_out_mux, card));
 
 	return 0;
 }
@@ -190,6 +177,11 @@ static struct snd_soc_card snd_soc_card_s6105 = {
 	.owner = THIS_MODULE,
 	.dai_link = &s6105_dai,
 	.num_links = 1,
+
+	.dapm_widgets = aic3x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
 };
 
 static struct s6000_snd_platform_data s6105_snd_data __initdata = {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 359c284..b322cf2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1137,6 +1137,16 @@ static int soc_probe_codec(struct snd_soc_card *card,
 
 	codec->dapm.idle_bias_off = driver->idle_bias_off;
 
+	if (!codec->write && dev_get_regmap(codec->dev, NULL)) {
+		/* Set the default I/O up try regmap */
+		ret = snd_soc_codec_set_cache_io(codec, NULL);
+		if (ret < 0) {
+			dev_err(codec->dev,
+				"Failed to set cache I/O: %d\n", ret);
+			goto err_probe;
+		}
+	}
+
 	if (driver->probe) {
 		ret = driver->probe(codec);
 		if (ret < 0) {
@@ -1150,10 +1160,6 @@ static int soc_probe_codec(struct snd_soc_card *card,
 			codec->name);
 	}
 
-	/* If the driver didn't set I/O up try regmap */
-	if (!codec->write && dev_get_regmap(codec->dev, NULL))
-		snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
-
 	if (driver->controls)
 		snd_soc_add_codec_controls(codec, driver->controls,
 				     driver->num_controls);
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index aa886cc..8aa0869 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -69,9 +69,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
  * snd_soc_codec_set_cache_io: Set up standard I/O functions.
  *
  * @codec: CODEC to configure.
- * @addr_bits: Number of bits of register address data.
- * @data_bits: Number of bits of data per register.
- * @control: Control bus used.
+ * @map: Register map to write to
  *
  * Register formats are frequently shared between many I2C and SPI
  * devices.  In order to promote code reuse the ASoC core provides
@@ -85,60 +83,36 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
  * volatile registers.
  */
 int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
-			       int addr_bits, int data_bits,
-			       enum snd_soc_control_type control)
+			       struct regmap *regmap)
 {
-	struct regmap_config config;
 	int ret;
 
-	memset(&config, 0, sizeof(config));
-	codec->write = hw_write;
-	codec->read = hw_read;
+	/* Device has made its own regmap arrangements */
+	if (!regmap)
+		codec->control_data = dev_get_regmap(codec->dev, NULL);
+	else
+		codec->control_data = regmap;
 
-	config.reg_bits = addr_bits;
-	config.val_bits = data_bits;
+	if (IS_ERR(codec->control_data))
+		return PTR_ERR(codec->control_data);
 
-	switch (control) {
-#if IS_ENABLED(CONFIG_REGMAP_I2C)
-	case SND_SOC_I2C:
-		codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev),
-						      &config);
-		break;
-#endif
+	codec->write = hw_write;
+	codec->read = hw_read;
 
-#if IS_ENABLED(CONFIG_REGMAP_SPI)
-	case SND_SOC_SPI:
-		codec->control_data = regmap_init_spi(to_spi_device(codec->dev),
-						      &config);
-		break;
-#endif
+	ret = regmap_get_val_bytes(codec->control_data);
+	/* Errors are legitimate for non-integer byte
+	 * multiples */
+	if (ret > 0)
+		codec->val_bytes = ret;
 
-	case SND_SOC_REGMAP:
-		/* Device has made its own regmap arrangements */
-		codec->using_regmap = true;
-		if (!codec->control_data)
-			codec->control_data = dev_get_regmap(codec->dev, NULL);
-
-		if (codec->control_data) {
-			ret = regmap_get_val_bytes(codec->control_data);
-			/* Errors are legitimate for non-integer byte
-			 * multiples */
-			if (ret > 0)
-				codec->val_bytes = ret;
-		}
-		break;
-
-	default:
-		return -EINVAL;
-	}
+	codec->using_regmap = true;
 
-	return PTR_ERR_OR_ZERO(codec->control_data);
+	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
 #else
 int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
-			       int addr_bits, int data_bits,
-			       enum snd_soc_control_type control)
+			       struct regmap *regmap)
 {
 	return -ENOTSUPP;
 }
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 23d43da..b903f82 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -250,7 +250,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
 		report = 0;
 
 	if (gpio->jack_status_check)
-		report = gpio->jack_status_check();
+		report = gpio->jack_status_check(gpio->data);
 
 	snd_soc_jack_report(jack, report, gpio->report);
 }
@@ -342,7 +342,8 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 		gpio_export(gpios[i].gpio, false);
 
 		/* Update initial jack status */
-		snd_soc_jack_gpio_detect(&gpios[i]);
+		schedule_delayed_work(&gpios[i].work,
+				      msecs_to_jiffies(gpios[i].debounce_time));
 	}
 
 	return 0;