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 @@
 - 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 @@
 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 @@
 
 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_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 *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 @@
 	/* 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 @@
 	{"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 @@
 	.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 @@
 	.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 @@
 
 	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 @@
 	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 @@
 	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-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-$(CONFIG_SND_SOC_TLV320AIC23_I2C)	+= snd-soc-tlv320aic23-i2c.o
 obj-$(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 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 @@
 	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 @@
 	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 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_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_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_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 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 @@
 	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 @@
 	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 @@
 	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 @@
 	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 @@
 {
 	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 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 @@
 	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_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 @@
 	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 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_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 @@
 	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 @@
 
 		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 @@
 	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 @@
 
 		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 @@
 {
 	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 f118daa9..4ff06b5 100644
--- a/sound/soc/codecs/da9055.c
+++ b/sound/soc/codecs/da9055.c
@@ -1383,16 +1383,8 @@
 
 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 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 @@
 	"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_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 @@
 	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 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 @@
 	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 @@
        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 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 @@
 
        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 @@
 
 	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 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 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 @@
 	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_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 @@
 	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_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 @@
 {
 	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_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 @@
 	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 @@
 
 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 @@
 {
 	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 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 @@
 	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 @@
 		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 @@
 	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 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_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 @@
 	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 @@
 	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 @@
 	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 @@
 	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 @@
 {
 	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 @@
 	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 @@
 	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 @@
 	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 @@
 	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 @@
 	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 @@
 
 	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_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_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 @@
 	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 @@
 {
 	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_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 @@
 	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 @@
 	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 @@
 		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 @@
 {
 	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 @@
 
 	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 @@
 	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 @@
 {
 	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 @@
 
 	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_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 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 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 @@
 		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 @@
 	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 @@
 	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 @@
 			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 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 @@
 	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 @@
 	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 @@
 {
 	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 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 @@
 	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 @@
 	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_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_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_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 @@
 	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 @@
 	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 @@
 	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 @@
 	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 @@
 		} 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 @@
 	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 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 @@
 				     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_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_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_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 @@
 
 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 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 @@
 
 	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_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 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_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_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 @@
 {
 	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 @@
 	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 @@
 	.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 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_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_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 @@
 {
 	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 @@
 	.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_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 @@
 {
 	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 @@
 	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_disable_pin(&card->dapm, "Audio Out Differential");
 
-	snd_soc_dapm_sync(dapm);
-
-	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 @@
 	.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 @@
 
 	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 @@
 			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 @@
  * 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 @@
  * 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));
+	/* Device has made its own regmap arrangements */
+	if (!regmap)
+		codec->control_data = dev_get_regmap(codec->dev, NULL);
+	else
+		codec->control_data = regmap;
+
+	if (IS_ERR(codec->control_data))
+		return PTR_ERR(codec->control_data);
+
 	codec->write = hw_write;
 	codec->read = hw_read;
 
-	config.reg_bits = addr_bits;
-	config.val_bits = data_bits;
+	ret = regmap_get_val_bytes(codec->control_data);
+	/* Errors are legitimate for non-integer byte
+	 * multiples */
+	if (ret > 0)
+		codec->val_bytes = ret;
 
-	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->using_regmap = true;
 
-#if IS_ENABLED(CONFIG_REGMAP_SPI)
-	case SND_SOC_SPI:
-		codec->control_data = regmap_init_spi(to_spi_device(codec->dev),
-						      &config);
-		break;
-#endif
-
-	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;
-	}
-
-	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 @@
 		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 @@
 		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;