20130320/sound-asoc

41 files changed, 1659 insertions, 477 deletions

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index e13580d6c476..94da62345a27 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -533,6 +533,49 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
 		break;
 
 	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+		/*
+		 * DSP/PCM Mode A format, CODEC supplies BCLK and LRC clocks.
+		 *
+		 * The SSC transmit clock is obtained from the BCLK signal on
+		 * on the TK line, and the SSC receive clock is
+		 * generated from the transmit clock.
+		 *
+		 * Data is transferred on first BCLK after LRC pulse rising
+		 * edge.If stereo, the right channel data is contiguous with
+		 * the left channel data.
+		 */
+		rcmr =	  SSC_BF(RCMR_PERIOD, 0)
+			| SSC_BF(RCMR_STTDLY, START_DELAY)
+			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(RCMR_CKS, SSC_CKS_PIN);
+
+		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
+			| SSC_BF(RFMR_FSLEN, 0)
+			| SSC_BF(RFMR_DATNB, (channels - 1))
+			| SSC_BIT(RFMR_MSBF)
+			| SSC_BF(RFMR_LOOP, 0)
+			| SSC_BF(RFMR_DATLEN, (bits - 1));
+
+		tcmr =	  SSC_BF(TCMR_PERIOD, 0)
+			| SSC_BF(TCMR_STTDLY, START_DELAY)
+			| SSC_BF(TCMR_START, SSC_START_RISING_RF)
+			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+
+		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(TFMR_FSDEN, 0)
+			| SSC_BF(TFMR_FSOS, SSC_FSOS_NONE)
+			| SSC_BF(TFMR_FSLEN, 0)
+			| SSC_BF(TFMR_DATNB, (channels - 1))
+			| SSC_BIT(TFMR_MSBF)
+			| SSC_BF(TFMR_DATDEF, 0)
+			| SSC_BF(TFMR_DATLEN, (bits - 1));
+		break;
+
 	default:
 		printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
 			ssc_p->daifmt);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 45b72561c615..18fea10ce040 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4641 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
+	select SND_SOC_AK5386
 	select SND_SOC_ALC5623 if I2C
 	select SND_SOC_ALC5632 if I2C
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
@@ -63,6 +64,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_STA32X if I2C
 	select SND_SOC_STA529 if I2C
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
+	select SND_SOC_TAS5086 if I2C
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
 	select SND_SOC_TLV320AIC32X4 if I2C
@@ -203,6 +205,9 @@ config SND_SOC_AK4642
 config SND_SOC_AK4671
 	tristate
 
+config SND_SOC_AK5386
+	tristate
+
 config SND_SOC_ALC5623
        tristate
 config SND_SOC_ALC5632
@@ -320,6 +325,9 @@ config SND_SOC_STA529
 config SND_SOC_STAC9766
 	tristate
 
+config SND_SOC_TAS5086
+	tristate
+
 config SND_SOC_TLV320AIC23
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 6a3b3c3b8b41..b9e41c9a1f4c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o
 snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
+snd-soc-ak5386-objs := ak5386.o
 snd-soc-arizona-objs := arizona.o
 snd-soc-cq93vc-objs := cq93vc.o
 snd-soc-cs42l51-objs := cs42l51.o
@@ -55,6 +56,7 @@ snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-sta32x-objs := sta32x.o
 snd-soc-sta529-objs := sta529.o
 snd-soc-stac9766-objs := stac9766.o
+snd-soc-tas5086-objs := tas5086.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -137,6 +139,7 @@ obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_AK4641)	+= snd-soc-ak4641.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
+obj-$(CONFIG_SND_SOC_AK5386)	+= snd-soc-ak5386.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
 obj-$(CONFIG_SND_SOC_ARIZONA)	+= snd-soc-arizona.o
@@ -177,6 +180,7 @@ obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STA32X)   += snd-soc-sta32x.o
 obj-$(CONFIG_SND_SOC_STA529)   += snd-soc-sta529.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
+obj-$(CONFIG_SND_SOC_TAS5086)	+= snd-soc-tas5086.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 068b3ae56a17..1aa10ddf3a61 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -133,6 +133,8 @@ struct adau1373 {
 #define ADAU1373_DAI_FORMAT_DSP		0x3
 
 #define ADAU1373_BCLKDIV_SOURCE		BIT(5)
+#define ADAU1373_BCLKDIV_SR_MASK	(0x07 << 2)
+#define ADAU1373_BCLKDIV_BCLK_MASK	0x03
 #define ADAU1373_BCLKDIV_32		0x03
 #define ADAU1373_BCLKDIV_64		0x02
 #define ADAU1373_BCLKDIV_128		0x01
@@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream,
 	adau1373_dai->enable_src = (div != 0);
 
 	snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id),
-		~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64);
+		ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK,
+		(div << 2) | ADAU1373_BCLKDIV_64);
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index 6f6c335a5baa..c7cfdf957e4d 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -55,6 +55,7 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			      unsigned int format)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
 	int val = 0;
 	int ret;
 
@@ -77,9 +78,9 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
 		return -EINVAL;
 
-	ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
-				  AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1,
-				  val);
+	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+				 AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1,
+				 val);
 	if (ret < 0)
 		return ret;
 
@@ -91,11 +92,12 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	int val = 0;
+	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+	int ret, val = 0;
 
 	/* set the IEC958 bits: consumer mode, no copyright bit */
 	val |= IEC958_AES0_CON_NOT_COPYRIGHT;
-	snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val);
+	regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(0), val);
 
 	val = 0;
 
@@ -132,11 +134,33 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val);
+	ret = regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val);
+	if (ret < 0)
+		return ret;
+
+	/* enable transmitter */
+	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+				 AK4104_TX_TXE, AK4104_TX_TXE);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int ak4104_hw_free(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+
+	/* disable transmitter */
+	return regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+				  AK4104_TX_TXE, 0);
 }
 
 static const struct snd_soc_dai_ops ak4101_dai_ops = {
 	.hw_params = ak4104_hw_params,
+	.hw_free = ak4104_hw_free,
 	.set_fmt = ak4104_set_dai_fmt,
 };
 
@@ -160,20 +184,17 @@ static int ak4104_probe(struct snd_soc_codec *codec)
 	int ret;
 
 	codec->control_data = ak4104->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret != 0)
-		return ret;
 
 	/* set power-up and non-reset bits */
-	ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
-				  AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
-				  AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
+	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+				 AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
+				 AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
 	if (ret < 0)
 		return ret;
 
 	/* enable transmitter */
-	ret = snd_soc_update_bits(codec, AK4104_REG_TX,
-				  AK4104_TX_TXE, AK4104_TX_TXE);
+	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+				 AK4104_TX_TXE, AK4104_TX_TXE);
 	if (ret < 0)
 		return ret;
 
@@ -182,8 +203,10 @@ static int ak4104_probe(struct snd_soc_codec *codec)
 
 static int ak4104_remove(struct snd_soc_codec *codec)
 {
-	snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
-			    AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0);
+	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+
+	regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+			   AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c
new file mode 100644
index 000000000000..1f303983ae02
--- /dev/null
+++ b/sound/soc/codecs/ak5386.c
@@ -0,0 +1,152 @@
+/*
+ * ALSA SoC driver for
+ *    Asahi Kasei AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC
+ *
+ * (c) 2013 Daniel Mack <zonque@gmail.com>
+ *
+ * This program 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.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/of.h>
+#include <linux/of_gpio.h>
+#include <linux/of_device.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+
+struct ak5386_priv {
+	int reset_gpio;
+};
+
+static struct snd_soc_codec_driver soc_codec_ak5386;
+
+static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	format &= SND_SOC_DAIFMT_FORMAT_MASK;
+	if (format != SND_SOC_DAIFMT_LEFT_J &&
+	    format != SND_SOC_DAIFMT_I2S) {
+		dev_err(codec->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ak5386_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	/*
+	 * From the datasheet:
+	 *
+	 * All external clocks (MCLK, SCLK and LRCK) must be present unless
+	 * PDN pin = “L”. If these clocks are not provided, the AK5386 may
+	 * draw excess current due to its use of internal dynamically
+	 * refreshed logic. If the external clocks are not present, place
+	 * the AK5386 in power-down mode (PDN pin = “L”).
+	 */
+
+	if (gpio_is_valid(priv->reset_gpio))
+		gpio_set_value(priv->reset_gpio, 1);
+
+	return 0;
+}
+
+static int ak5386_hw_free(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	if (gpio_is_valid(priv->reset_gpio))
+		gpio_set_value(priv->reset_gpio, 0);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops ak5386_dai_ops = {
+	.set_fmt	= ak5386_set_dai_fmt,
+	.hw_params	= ak5386_hw_params,
+	.hw_free	= ak5386_hw_free,
+};
+
+static struct snd_soc_dai_driver ak5386_dai = {
+	.name		= "ak5386-hifi",
+	.capture	= {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= 2,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
+		.formats	= SNDRV_PCM_FMTBIT_S8     |
+				  SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S24_LE |
+				  SNDRV_PCM_FMTBIT_S24_3LE,
+	},
+	.ops	= &ak5386_dai_ops,
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id ak5386_dt_ids[] = {
+	{ .compatible = "asahi-kasei,ak5386", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, ak5386_dt_ids);
+#endif
+
+static int ak5386_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct ak5386_priv *priv;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->reset_gpio = -EINVAL;
+	dev_set_drvdata(dev, priv);
+
+	if (of_match_device(of_match_ptr(ak5386_dt_ids), dev))
+		priv->reset_gpio = of_get_named_gpio(dev->of_node,
+						      "reset-gpio", 0);
+
+	if (gpio_is_valid(priv->reset_gpio))
+		if (devm_gpio_request_one(dev, priv->reset_gpio,
+					  GPIOF_OUT_INIT_LOW,
+					  "AK5386 Reset"))
+			priv->reset_gpio = -EINVAL;
+
+	return snd_soc_register_codec(dev, &soc_codec_ak5386,
+				      &ak5386_dai, 1);
+}
+
+static int ak5386_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver ak5386_driver = {
+	.probe		= ak5386_probe,
+	.remove		= ak5386_remove,
+	.driver		= {
+		.name	= "ak5386",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(ak5386_dt_ids),
+	},
+};
+
+module_platform_driver(ak5386_driver);
+
+MODULE_DESCRIPTION("ASoC driver for AK5386 ADC");
+MODULE_AUTHOR("Daniel Mack <zonque@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index ac948a671ea6..2b0803ec8234 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -10,6 +10,7 @@
  * published by the Free Software Foundation.
  */
 
+#include <linux/delay.h>
 #include <linux/gcd.h>
 #include <linux/module.h>
 #include <linux/pm_runtime.h>
@@ -332,9 +333,27 @@ const struct soc_enum arizona_ng_hold =
 			4, arizona_ng_hold_text);
 EXPORT_SYMBOL_GPL(arizona_ng_hold);
 
+static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+	int i;
+
+	if (ena)
+		val = ARIZONA_IN_VU;
+	else
+		val = 0;
+
+	for (i = 0; i < priv->num_inputs; i++)
+		snd_soc_update_bits(codec,
+				    ARIZONA_ADC_DIGITAL_VOLUME_1L + (i * 4),
+				    ARIZONA_IN_VU, val);
+}
+
 int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		  int event)
 {
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec);
 	unsigned int reg;
 
 	if (w->shift % 2)
@@ -343,13 +362,29 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		reg = ARIZONA_ADC_DIGITAL_VOLUME_1R + ((w->shift / 2) * 8);
 
 	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		priv->in_pending++;
+		break;
 	case SND_SOC_DAPM_POST_PMU:
 		snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0);
+
+		/* If this is the last input pending then allow VU */
+		priv->in_pending--;
+		if (priv->in_pending == 0) {
+			msleep(1);
+			arizona_in_set_vu(w->codec, 1);
+		}
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
-		snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE,
-				    ARIZONA_IN1L_MUTE);
+		snd_soc_update_bits(w->codec, reg,
+				    ARIZONA_IN1L_MUTE | ARIZONA_IN_VU,
+				    ARIZONA_IN1L_MUTE | ARIZONA_IN_VU);
 		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* Disable volume updates if no inputs are enabled */
+		reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES);
+		if (reg == 0)
+			arizona_in_set_vu(w->codec, 0);
 	}
 
 	return 0;
@@ -469,27 +504,27 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 		break;
 	case 11289600:
 	case 12288000:
-		val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_12MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 22579200:
 	case 24576000:
-		val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_24MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 45158400:
 	case 49152000:
-		val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_49MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 67737600:
 	case 73728000:
-		val |= 4 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_73MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 90316800:
 	case 98304000:
-		val |= 5 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_98MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 135475200:
 	case 147456000:
-		val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_147MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 0:
 		dev_dbg(arizona->dev, "%s cleared\n", name);
@@ -783,7 +818,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	struct arizona *arizona = priv->arizona;
 	int base = dai->driver->base;
 	const int *rates;
-	int i, ret;
+	int i, ret, val;
 	int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1];
 	int bclk, lrclk, wl, frame, bclk_target;
 
@@ -799,6 +834,13 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 		bclk_target *= chan_limit;
 	}
 
+	/* Force stereo for I2S mode */
+	val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
+	if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) {
+		arizona_aif_dbg(dai, "Forcing stereo mode\n");
+		bclk_target *= 2;
+	}
+
 	for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
 		if (rates[i] >= bclk_target &&
 		    rates[i] % params_rate(params) == 0) {
@@ -955,6 +997,16 @@ static struct {
 	{ 1000000, 13500000, 0,  1 },
 };
 
+static struct {
+	unsigned int min;
+	unsigned int max;
+	u16 gain;
+} fll_gains[] = {
+	{       0,   256000, 0 },
+	{  256000,  1000000, 2 },
+	{ 1000000, 13500000, 4 },
+};
+
 struct arizona_fll_cfg {
 	int n;
 	int theta;
@@ -962,6 +1014,7 @@ struct arizona_fll_cfg {
 	int refdiv;
 	int outdiv;
 	int fratio;
+	int gain;
 };
 
 static int arizona_calc_fll(struct arizona_fll *fll,
@@ -1021,6 +1074,18 @@ static int arizona_calc_fll(struct arizona_fll *fll,
 		return -EINVAL;
 	}
 
+	for (i = 0; i < ARRAY_SIZE(fll_gains); i++) {
+		if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) {
+			cfg->gain = fll_gains[i].gain;
+			break;
+		}
+	}
+	if (i == ARRAY_SIZE(fll_gains)) {
+		arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n",
+				Fref);
+		return -EINVAL;
+	}
+
 	cfg->n = target / (ratio * Fref);
 
 	if (target % (ratio * Fref)) {
@@ -1048,13 +1113,15 @@ static int arizona_calc_fll(struct arizona_fll *fll,
 			cfg->n, cfg->theta, cfg->lambda);
 	arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
 			cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv);
+	arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain);
 
 	return 0;
 
 }
 
 static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
-			      struct arizona_fll_cfg *cfg, int source)
+			      struct arizona_fll_cfg *cfg, int source,
+			      bool sync)
 {
 	regmap_update_bits(arizona->regmap, base + 3,
 			   ARIZONA_FLL1_THETA_MASK, cfg->theta);
@@ -1069,87 +1136,84 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
 			   cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
 			   source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
 
+	if (sync)
+		regmap_update_bits(arizona->regmap, base + 0x7,
+				   ARIZONA_FLL1_GAIN_MASK,
+				   cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+	else
+		regmap_update_bits(arizona->regmap, base + 0x9,
+				   ARIZONA_FLL1_GAIN_MASK,
+				   cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+
 	regmap_update_bits(arizona->regmap, base + 2,
 			   ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
 			   ARIZONA_FLL1_CTRL_UPD | cfg->n);
 }
 
-int arizona_set_fll(struct arizona_fll *fll, int source,
-		    unsigned int Fref, unsigned int Fout)
+static bool arizona_is_enabled_fll(struct arizona_fll *fll)
 {
 	struct arizona *arizona = fll->arizona;
-	struct arizona_fll_cfg cfg, sync;
-	unsigned int reg, val;
-	int syncsrc;
-	bool ena;
+	unsigned int reg;
 	int ret;
 
-	if (fll->fref == Fref && fll->fout == Fout)
-		return 0;
-
 	ret = regmap_read(arizona->regmap, fll->base + 1, &reg);
 	if (ret != 0) {
 		arizona_fll_err(fll, "Failed to read current state: %d\n",
 				ret);
 		return ret;
 	}
-	ena = reg & ARIZONA_FLL1_ENA;
 
-	if (Fout) {
-		/* Do we have a 32kHz reference? */
-		regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val);
-		switch (val & ARIZONA_CLK_32K_SRC_MASK) {
-		case ARIZONA_CLK_SRC_MCLK1:
-		case ARIZONA_CLK_SRC_MCLK2:
-			syncsrc = val & ARIZONA_CLK_32K_SRC_MASK;
-			break;
-		default:
-			syncsrc = -1;
-		}
+	return reg & ARIZONA_FLL1_ENA;
+}
 
-		if (source == syncsrc)
-			syncsrc = -1;
+static void arizona_enable_fll(struct arizona_fll *fll,
+			      struct arizona_fll_cfg *ref,
+			      struct arizona_fll_cfg *sync)
+{
+	struct arizona *arizona = fll->arizona;
+	int ret;
 
-		if (syncsrc >= 0) {
-			ret = arizona_calc_fll(fll, &sync, Fref, Fout);
-			if (ret != 0)
-				return ret;
+	/*
+	 * If we have both REFCLK and SYNCCLK then enable both,
+	 * otherwise apply the SYNCCLK settings to REFCLK.
+	 */
+	if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) {
+		regmap_update_bits(arizona->regmap, fll->base + 5,
+				   ARIZONA_FLL1_OUTDIV_MASK,
+				   ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+
+		arizona_apply_fll(arizona, fll->base, ref, fll->ref_src,
+				  false);
+		if (fll->sync_src >= 0)
+			arizona_apply_fll(arizona, fll->base + 0x10, sync,
+					  fll->sync_src, true);
+	} else if (fll->sync_src >= 0) {
+		regmap_update_bits(arizona->regmap, fll->base + 5,
+				   ARIZONA_FLL1_OUTDIV_MASK,
+				   sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+
+		arizona_apply_fll(arizona, fll->base, sync,
+				  fll->sync_src, false);
 
-			ret = arizona_calc_fll(fll, &cfg, 32768, Fout);
-			if (ret != 0)
-				return ret;
-		} else {
-			ret = arizona_calc_fll(fll, &cfg, Fref, Fout);
-			if (ret != 0)
-				return ret;
-		}
-	} else {
-		regmap_update_bits(arizona->regmap, fll->base + 1,
-				   ARIZONA_FLL1_ENA, 0);
 		regmap_update_bits(arizona->regmap, fll->base + 0x11,
 				   ARIZONA_FLL1_SYNC_ENA, 0);
-
-		if (ena)
-			pm_runtime_put_autosuspend(arizona->dev);
-
-		fll->fref = Fref;
-		fll->fout = Fout;
-
-		return 0;
-	}
-
-	regmap_update_bits(arizona->regmap, fll->base + 5,
-			   ARIZONA_FLL1_OUTDIV_MASK,
-			   cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
-
-	if (syncsrc >= 0) {
-		arizona_apply_fll(arizona, fll->base, &cfg, syncsrc);
-		arizona_apply_fll(arizona, fll->base + 0x10, &sync, source);
 	} else {
-		arizona_apply_fll(arizona, fll->base, &cfg, source);
+		arizona_fll_err(fll, "No clocks provided\n");
+		return;
 	}
 
-	if (!ena)
+	/*
+	 * Increase the bandwidth if we're not using a low frequency
+	 * sync source.
+	 */
+	if (fll->sync_src >= 0 && fll->sync_freq > 100000)
+		regmap_update_bits(arizona->regmap, fll->base + 0x17,
+				   ARIZONA_FLL1_SYNC_BW, 0);
+	else
+		regmap_update_bits(arizona->regmap, fll->base + 0x17,
+				   ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW);
+
+	if (!arizona_is_enabled_fll(fll))
 		pm_runtime_get(arizona->dev);
 
 	/* Clear any pending completions */
@@ -1157,7 +1221,8 @@ int arizona_set_fll(struct arizona_fll *fll, int source,
 
 	regmap_update_bits(arizona->regmap, fll->base + 1,
 			   ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
-	if (syncsrc >= 0)
+	if (fll->ref_src >= 0 && fll->sync_src >= 0 &&
+	    fll->ref_src != fll->sync_src)
 		regmap_update_bits(arizona->regmap, fll->base + 0x11,
 				   ARIZONA_FLL1_SYNC_ENA,
 				   ARIZONA_FLL1_SYNC_ENA);
@@ -1166,10 +1231,88 @@ int arizona_set_fll(struct arizona_fll *fll, int source,
 					  msecs_to_jiffies(250));
 	if (ret == 0)
 		arizona_fll_warn(fll, "Timed out waiting for lock\n");
+}
+
+static void arizona_disable_fll(struct arizona_fll *fll)
+{
+	struct arizona *arizona = fll->arizona;
+	bool change;
+
+	regmap_update_bits_check(arizona->regmap, fll->base + 1,
+				 ARIZONA_FLL1_ENA, 0, &change);
+	regmap_update_bits(arizona->regmap, fll->base + 0x11,
+			   ARIZONA_FLL1_SYNC_ENA, 0);
+
+	if (change)
+		pm_runtime_put_autosuspend(arizona->dev);
+}
+
+int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
+			   unsigned int Fref, unsigned int Fout)
+{
+	struct arizona_fll_cfg ref, sync;
+	int ret;
+
+	if (fll->ref_src == source && fll->ref_freq == Fref)
+		return 0;
+
+	if (fll->fout && Fref > 0) {
+		ret = arizona_calc_fll(fll, &ref, Fref, fll->fout);
+		if (ret != 0)
+			return ret;
+
+		if (fll->sync_src >= 0) {
+			ret = arizona_calc_fll(fll, &sync, fll->sync_freq,
+					       fll->fout);
+			if (ret != 0)
+				return ret;
+		}
+	}
+
+	fll->ref_src = source;
+	fll->ref_freq = Fref;
 
-	fll->fref = Fref;
+	if (fll->fout && Fref > 0) {
+		arizona_enable_fll(fll, &ref, &sync);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_set_fll_refclk);
+
+int arizona_set_fll(struct arizona_fll *fll, int source,
+		    unsigned int Fref, unsigned int Fout)
+{
+	struct arizona_fll_cfg ref, sync;
+	int ret;
+
+	if (fll->sync_src == source &&
+	    fll->sync_freq == Fref && fll->fout == Fout)
+		return 0;
+
+	if (Fout) {
+		if (fll->ref_src >= 0) {
+			ret = arizona_calc_fll(fll, &ref, fll->ref_freq,
+					       Fout);
+			if (ret != 0)
+				return ret;
+		}
+
+		ret = arizona_calc_fll(fll, &sync, Fref, Fout);
+		if (ret != 0)
+			return ret;
+	}
+
+	fll->sync_src = source;
+	fll->sync_freq = Fref;
 	fll->fout = Fout;
 
+	if (Fout) {
+		arizona_enable_fll(fll, &ref, &sync);
+	} else {
+		arizona_disable_fll(fll);
+	}
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(arizona_set_fll);
@@ -1178,12 +1321,26 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
 		     int ok_irq, struct arizona_fll *fll)
 {
 	int ret;
+	unsigned int val;
 
 	init_completion(&fll->ok);
 
 	fll->id = id;
 	fll->base = base;
 	fll->arizona = arizona;
+	fll->sync_src = ARIZONA_FLL_SRC_NONE;
+
+	/* Configure default refclk to 32kHz if we have one */
+	regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val);
+	switch (val & ARIZONA_CLK_32K_SRC_MASK) {
+	case ARIZONA_CLK_SRC_MCLK1:
+	case ARIZONA_CLK_SRC_MCLK2:
+		fll->ref_src = val & ARIZONA_CLK_32K_SRC_MASK;
+		break;
+	default:
+		fll->ref_src = ARIZONA_FLL_SRC_NONE;
+	}
+	fll->ref_freq = 32768;
 
 	snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id);
 	snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name),
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 116372c91f5d..572f11bc90b4 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -32,6 +32,7 @@
 #define ARIZONA_CLK_SRC_AIF2BCLK 0x9
 #define ARIZONA_CLK_SRC_AIF3BCLK 0xa
 
+#define ARIZONA_FLL_SRC_NONE      -1
 #define ARIZONA_FLL_SRC_MCLK1      0
 #define ARIZONA_FLL_SRC_MCLK2      1
 #define ARIZONA_FLL_SRC_SLIMCLK    3
@@ -48,6 +49,14 @@
 #define ARIZONA_MIXER_VOL_SHIFT                 1
 #define ARIZONA_MIXER_VOL_WIDTH                 7
 
+#define ARIZONA_CLK_6MHZ   0
+#define ARIZONA_CLK_12MHZ  1
+#define ARIZONA_CLK_24MHZ  2
+#define ARIZONA_CLK_49MHZ  3
+#define ARIZONA_CLK_73MHZ  4
+#define ARIZONA_CLK_98MHZ  5
+#define ARIZONA_CLK_147MHZ 6
+
 #define ARIZONA_MAX_DAI  4
 #define ARIZONA_MAX_ADSP 4
 
@@ -64,6 +73,9 @@ struct arizona_priv {
 	int sysclk;
 	int asyncclk;
 	struct arizona_dai_priv dai[ARIZONA_MAX_DAI];
+
+	int num_inputs;
+	unsigned int in_pending;
 };
 
 #define ARIZONA_NUM_MIXER_INPUTS 99
@@ -198,8 +210,12 @@ struct arizona_fll {
 	unsigned int base;
 	unsigned int vco_mult;
 	struct completion ok;
-	unsigned int fref;
+
 	unsigned int fout;
+	int sync_src;
+	unsigned int sync_freq;
+	int ref_src;
+	unsigned int ref_freq;
 
 	char lock_name[ARIZONA_FLL_NAME_LEN];
 	char clock_ok_name[ARIZONA_FLL_NAME_LEN];
@@ -207,6 +223,8 @@ struct arizona_fll {
 
 extern int arizona_init_fll(struct arizona *arizona, int id, int base,
 			    int lock_irq, int ok_irq, struct arizona_fll *fll);
+extern int arizona_set_fll_refclk(struct arizona_fll *fll, int source,
+				  unsigned int Fref, unsigned int Fout);
 extern int arizona_set_fll(struct arizona_fll *fll, int source,
 			   unsigned int Fref, unsigned int Fout);
 
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 2415a4118dbd..ac0d3b4844a8 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -39,17 +39,15 @@
 
 /*
  * CS4271 registers
- * High byte represents SPI chip address (0x10) + write command (0)
- * Low byte - codec register address
  */
-#define CS4271_MODE1	0x2001	/* Mode Control 1 */
-#define CS4271_DACCTL	0x2002	/* DAC Control */
-#define CS4271_DACVOL	0x2003	/* DAC Volume & Mixing Control */
-#define CS4271_VOLA	0x2004	/* DAC Channel A Volume Control */
-#define CS4271_VOLB	0x2005	/* DAC Channel B Volume Control */
-#define CS4271_ADCCTL	0x2006	/* ADC Control */
-#define CS4271_MODE2	0x2007	/* Mode Control 2 */
-#define CS4271_CHIPID	0x2008	/* Chip ID */
+#define CS4271_MODE1	0x01	/* Mode Control 1 */
+#define CS4271_DACCTL	0x02	/* DAC Control */
+#define CS4271_DACVOL	0x03	/* DAC Volume & Mixing Control */
+#define CS4271_VOLA	0x04	/* DAC Channel A Volume Control */
+#define CS4271_VOLB	0x05	/* DAC Channel B Volume Control */
+#define CS4271_ADCCTL	0x06	/* ADC Control */
+#define CS4271_MODE2	0x07	/* Mode Control 2 */
+#define CS4271_CHIPID	0x08	/* Chip ID */
 
 #define CS4271_FIRSTREG	CS4271_MODE1
 #define CS4271_LASTREG	CS4271_MODE2
@@ -144,23 +142,27 @@
  * Array do not include Chip ID, as codec driver does not use
  * registers read operations at all
  */
-static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = {
-	0,
-	0,
-	CS4271_DACCTL_AMUTE,
-	CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR,
-	0,
-	0,
-	0,
-	0,
+static const struct reg_default cs4271_reg_defaults[] = {
+	{ CS4271_MODE1,		0, },
+	{ CS4271_DACCTL,	CS4271_DACCTL_AMUTE, },
+	{ CS4271_DACVOL,	CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, },
+	{ CS4271_VOLA,		0, },
+	{ CS4271_VOLB,		0, },
+	{ CS4271_ADCCTL,	0, },
+	{ CS4271_MODE2,		0, },
 };
 
+static bool cs4271_volatile_reg(struct device *dev, unsigned int reg)
+{
+	return reg == CS4271_CHIPID;
+}
+
 struct cs4271_private {
 	/* SND_SOC_I2C or SND_SOC_SPI */
-	enum snd_soc_control_type	bus_type;
 	unsigned int			mclk;
 	bool				master;
 	bool				deemph;
+	struct regmap			*regmap;
 	/* Current sample rate for de-emphasis control */
 	int				rate;
 	/* GPIO driving Reset pin, if any */
@@ -210,14 +212,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_LEFT_J:
 		val |= CS4271_MODE1_DAC_DIF_LJ;
-		ret = snd_soc_update_bits(codec, CS4271_ADCCTL,
+		ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL,
 			CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ);
 		if (ret < 0)
 			return ret;
 		break;
 	case SND_SOC_DAIFMT_I2S:
 		val |= CS4271_MODE1_DAC_DIF_I2S;
-		ret = snd_soc_update_bits(codec, CS4271_ADCCTL,
+		ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL,
 			CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S);
 		if (ret < 0)
 			return ret;
@@ -227,7 +229,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	ret = snd_soc_update_bits(codec, CS4271_MODE1,
+	ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1,
 		CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val);
 	if (ret < 0)
 		return ret;
@@ -252,7 +254,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec)
 		val <<= 4;
 	}
 
-	ret = snd_soc_update_bits(codec, CS4271_DACCTL,
+	ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL,
 		CS4271_DACCTL_DEM_MASK, val);
 	if (ret < 0)
 		return ret;
@@ -341,14 +343,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
 		     !dai->capture_active) ||
 		    (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
 		     !dai->playback_active)) {
-			ret = snd_soc_update_bits(codec, CS4271_MODE2,
-						  CS4271_MODE2_PDN,
-						  CS4271_MODE2_PDN);
+			ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+						 CS4271_MODE2_PDN,
+						 CS4271_MODE2_PDN);
 			if (ret < 0)
 				return ret;
 
-			ret = snd_soc_update_bits(codec, CS4271_MODE2,
-						  CS4271_MODE2_PDN, 0);
+			ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+						 CS4271_MODE2_PDN, 0);
 			if (ret < 0)
 				return ret;
 		}
@@ -378,7 +380,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
 
 	val |= cs4271_clk_tab[i].ratio_mask;
 
-	ret = snd_soc_update_bits(codec, CS4271_MODE1,
+	ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1,
 		CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);
 	if (ret < 0)
 		return ret;
@@ -389,6 +391,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream,
 static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 	int val_a = 0;
 	int val_b = 0;
@@ -398,10 +401,13 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute)
 		val_b = CS4271_VOLB_MUTE;
 	}
 
-	ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a);
+	ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA,
+				 CS4271_VOLA_MUTE, val_a);
 	if (ret < 0)
 		return ret;
-	ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b);
+
+	ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB,
+				 CS4271_VOLB_MUTE, val_b);
 	if (ret < 0)
 		return ret;
 
@@ -463,25 +469,33 @@ static struct snd_soc_dai_driver cs4271_dai = {
 static int cs4271_soc_suspend(struct snd_soc_codec *codec)
 {
 	int ret;
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
 	/* Set power-down bit */
-	ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN,
-				  CS4271_MODE2_PDN);
+	ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+				 CS4271_MODE2_PDN, CS4271_MODE2_PDN);
 	if (ret < 0)
 		return ret;
+
 	return 0;
 }
 
 static int cs4271_soc_resume(struct snd_soc_codec *codec)
 {
 	int ret;
+	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
+
 	/* Restore codec state */
-	ret = snd_soc_cache_sync(codec);
+	ret = regcache_sync(cs4271->regmap);
 	if (ret < 0)
 		return ret;
+
 	/* then disable the power-down bit */
-	ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
+	ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+				 CS4271_MODE2_PDN, 0);
 	if (ret < 0)
 		return ret;
+
 	return 0;
 }
 #else
@@ -542,40 +556,22 @@ static int cs4271_probe(struct snd_soc_codec *codec)
 
 	cs4271->gpio_nreset = gpio_nreset;
 
-	/*
-	 * In case of I2C, chip address specified in board data.
-	 * So cache IO operations use 8 bit codec register address.
-	 * In case of SPI, chip address and register address
-	 * passed together as 16 bit value.
-	 * Anyway, register address is masked with 0xFF inside
-	 * soc-cache code.
-	 */
-	if (cs4271->bus_type == SND_SOC_SPI)
-		ret = snd_soc_codec_set_cache_io(codec, 16, 8,
-			cs4271->bus_type);
-	else
-		ret = snd_soc_codec_set_cache_io(codec, 8, 8,
-			cs4271->bus_type);
-	if (ret) {
-		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
-	}
-
-	ret = snd_soc_update_bits(codec, CS4271_MODE2,
-				  CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
-				  CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
+	ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+				 CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
+				 CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
 	if (ret < 0)
 		return ret;
-	ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
+	ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+				 CS4271_MODE2_PDN, 0);
 	if (ret < 0)
 		return ret;
 	/* Power-up sequence requires 85 uS */
 	udelay(85);
 
 	if (amutec_eq_bmutec)
-		snd_soc_update_bits(codec, CS4271_MODE2,
-				    CS4271_MODE2_MUTECAEQUB,
-				    CS4271_MODE2_MUTECAEQUB);
+		regmap_update_bits(cs4271->regmap, CS4271_MODE2,
+				   CS4271_MODE2_MUTECAEQUB,
+				   CS4271_MODE2_MUTECAEQUB);
 
 	return snd_soc_add_codec_controls(codec, cs4271_snd_controls,
 		ARRAY_SIZE(cs4271_snd_controls));
@@ -597,13 +593,24 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = {
 	.remove			= cs4271_remove,
 	.suspend		= cs4271_soc_suspend,
 	.resume			= cs4271_soc_resume,
-	.reg_cache_default	= cs4271_dflt_reg,
-	.reg_cache_size		= ARRAY_SIZE(cs4271_dflt_reg),
-	.reg_word_size		= sizeof(cs4271_dflt_reg[0]),
-	.compress_type		= SND_SOC_FLAT_COMPRESSION,
 };
 
 #if defined(CONFIG_SPI_MASTER)
+
+static const struct regmap_config cs4271_spi_regmap = {
+	.reg_bits = 16,
+	.val_bits = 8,
+	.max_register = CS4271_LASTREG,
+	.read_flag_mask = 0x21,
+	.write_flag_mask = 0x20,
+
+	.reg_defaults = cs4271_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+
+	.volatile_reg = cs4271_volatile_reg,
+};
+
 static int cs4271_spi_probe(struct spi_device *spi)
 {
 	struct cs4271_private *cs4271;
@@ -613,7 +620,9 @@ static int cs4271_spi_probe(struct spi_device *spi)
 		return -ENOMEM;
 
 	spi_set_drvdata(spi, cs4271);
-	cs4271->bus_type = SND_SOC_SPI;
+	cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap);
+	if (IS_ERR(cs4271->regmap))
+		return PTR_ERR(cs4271->regmap);
 
 	return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271,
 		&cs4271_dai, 1);
@@ -643,6 +652,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id);
 
+static const struct regmap_config cs4271_i2c_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+	.max_register = CS4271_LASTREG,
+
+	.reg_defaults = cs4271_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults),
+	.cache_type = REGCACHE_RBTREE,
+
+	.volatile_reg = cs4271_volatile_reg,
+};
+
 static int cs4271_i2c_probe(struct i2c_client *client,
 			    const struct i2c_device_id *id)
 {
@@ -653,7 +674,9 @@ static int cs4271_i2c_probe(struct i2c_client *client,
 		return -ENOMEM;
 
 	i2c_set_clientdata(client, cs4271);
-	cs4271->bus_type = SND_SOC_I2C;
+	cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap);
+	if (IS_ERR(cs4271->regmap))
+		return PTR_ERR(cs4271->regmap);
 
 	return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271,
 		&cs4271_dai, 1);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 6361dab48bd1..3b20c86cdb01 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1180,7 +1180,11 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
 		priv->config[id].mmcc &= 0xC0;
 		priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc;
 		priv->config[id].spc &= 0xFC;
-		priv->config[id].spc |= MCK_SCLK_MCLK;
+		/* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */
+		if (priv->mclk >= 6400000)
+			priv->config[id].spc |= MCK_SCLK_64FS;
+		else
+			priv->config[id].spc |= MCK_SCLK_MCLK;
 	} else {
 		/* CS42L73 Slave */
 		priv->config[id].spc &= 0xFC;
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index a4c16fd70f77..3a7b7fd14e3e 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -739,14 +739,32 @@ static const unsigned int max98088_micboost_tlv[] = {
        2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
 };
 
+static const unsigned int max98088_hp_tlv[] = {
+	TLV_DB_RANGE_HEAD(5),
+	0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0),
+	7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0),
+	15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0),
+	22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0),
+	28, 31, TLV_DB_SCALE_ITEM(150, 50, 0),
+};
+
+static const unsigned int max98088_spk_tlv[] = {
+	TLV_DB_RANGE_HEAD(5),
+	0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0),
+	7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0),
+	15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0),
+	22, 27, TLV_DB_SCALE_ITEM(100, 100, 0),
+	28, 31, TLV_DB_SCALE_ITEM(650, 50, 0),
+};
+
 static const struct snd_kcontrol_new max98088_snd_controls[] = {
 
-       SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L,
-               M98088_REG_3A_LVL_HP_R, 0, 31, 0),
-       SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L,
-               M98088_REG_3E_LVL_SPK_R, 0, 31, 0),
-       SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L,
-               M98088_REG_3C_LVL_REC_R, 0, 31, 0),
+	SOC_DOUBLE_R_TLV("Headphone Volume", M98088_REG_39_LVL_HP_L,
+			 M98088_REG_3A_LVL_HP_R, 0, 31, 0, max98088_hp_tlv),
+	SOC_DOUBLE_R_TLV("Speaker Volume", M98088_REG_3D_LVL_SPK_L,
+			 M98088_REG_3E_LVL_SPK_R, 0, 31, 0, max98088_spk_tlv),
+	SOC_DOUBLE_R_TLV("Receiver Volume", M98088_REG_3B_LVL_REC_L,
+			 M98088_REG_3C_LVL_REC_R, 0, 31, 0, max98088_spk_tlv),
 
        SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L,
                M98088_REG_3A_LVL_HP_R, 7, 1, 1),
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index fc176044994d..ce0d36412c97 100755
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -23,8 +23,6 @@
 #include <sound/max98090.h>
 #include "max98090.h"
 
-#include <linux/version.h>
-
 #define DEBUG
 #define EXTMIC_METHOD
 #define EXTMIC_METHOD_TEST
@@ -509,16 +507,16 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static const char * max98090_perf_pwr_text[] =
+static const char *max98090_perf_pwr_text[] =
 	{ "High Performance", "Low Power" };
-static const char * max98090_pwr_perf_text[] =
+static const char *max98090_pwr_perf_text[] =
 	{ "Low Power", "High Performance" };
 
 static const struct soc_enum max98090_vcmbandgap_enum =
 	SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT,
 		ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text);
 
-static const char * max98090_osr128_text[] = { "64*fs", "128*fs" };
+static const char *max98090_osr128_text[] = { "64*fs", "128*fs" };
 
 static const struct soc_enum max98090_osr128_enum =
 	SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT,
@@ -535,28 +533,28 @@ static const struct soc_enum max98090_filter_dmic34mode_enum =
 		M98090_FLT_DMIC34MODE_SHIFT,
 		ARRAY_SIZE(max98090_mode_text), max98090_mode_text);
 
-static const char * max98090_drcatk_text[] =
+static const char *max98090_drcatk_text[] =
 	{ "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" };
 
 static const struct soc_enum max98090_drcatk_enum =
 	SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT,
 		ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text);
 
-static const char * max98090_drcrls_text[] =
+static const char *max98090_drcrls_text[] =
 	{ "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" };
 
 static const struct soc_enum max98090_drcrls_enum =
 	SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT,
 		ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text);
 
-static const char * max98090_alccmp_text[] =
+static const char *max98090_alccmp_text[] =
 	{ "1:1", "1:1.5", "1:2", "1:4", "1:INF" };
 
 static const struct soc_enum max98090_alccmp_enum =
 	SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT,
 		ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text);
 
-static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" };
+static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" };
 
 static const struct soc_enum max98090_drcexp_enum =
 	SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT,
@@ -859,7 +857,7 @@ static const struct soc_enum mic2_mux_enum =
 static const struct snd_kcontrol_new max98090_mic2_mux =
 	SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum);
 
-static const char * max98090_micpre_text[] = { "Off", "On" };
+static const char *max98090_micpre_text[] = { "Off", "On" };
 
 static const struct soc_enum max98090_pa1en_enum =
 	SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT,
@@ -1703,9 +1701,8 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai,
 		 * seen for the case of TDM mode. The remaining cases have
 		 * normal logic.
 		 */
-		if (max98090->tdm_slots > 1) {
+		if (max98090->tdm_slots > 1)
 			regval ^= M98090_BCI_MASK;
-		}
 
 		snd_soc_write(codec,
 			M98090_REG_INTERFACE_FORMAT, regval);
@@ -2059,17 +2056,14 @@ static irqreturn_t max98090_interrupt(int irq, void *data)
 	if (!active)
 		return IRQ_NONE;
 
-	if (active & M98090_CLD_MASK) {
+	if (active & M98090_CLD_MASK)
 		dev_err(codec->dev, "M98090_CLD_MASK\n");
-	}
 
-	if (active & M98090_SLD_MASK) {
+	if (active & M98090_SLD_MASK)
 		dev_dbg(codec->dev, "M98090_SLD_MASK\n");
-	}
 
-	if (active & M98090_ULK_MASK) {
+	if (active & M98090_ULK_MASK)
 		dev_err(codec->dev, "M98090_ULK_MASK\n");
-	}
 
 	if (active & M98090_JDET_MASK) {
 		dev_dbg(codec->dev, "M98090_JDET_MASK\n");
@@ -2080,13 +2074,11 @@ static irqreturn_t max98090_interrupt(int irq, void *data)
 			msecs_to_jiffies(100));
 	}
 
-	if (active & M98090_DRCACT_MASK) {
+	if (active & M98090_DRCACT_MASK)
 		dev_dbg(codec->dev, "M98090_DRCACT_MASK\n");
-	}
 
-	if (active & M98090_DRCCLP_MASK) {
+	if (active & M98090_DRCCLP_MASK)
 		dev_err(codec->dev, "M98090_DRCCLP_MASK\n");
-	}
 
 	return IRQ_HANDLED;
 }
@@ -2324,7 +2316,7 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
 	max98090->pdata = i2c->dev.platform_data;
 	max98090->irq = i2c->irq;
 
-	max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap);
+	max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap);
 	if (IS_ERR(max98090->regmap)) {
 		ret = PTR_ERR(max98090->regmap);
 		dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
@@ -2334,18 +2326,13 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_max98090, max98090_dai,
 			ARRAY_SIZE(max98090_dai));
-	if (ret < 0)
-		regmap_exit(max98090->regmap);
-
 err_enable:
 	return ret;
 }
 
 static int max98090_i2c_remove(struct i2c_client *client)
 {
-	struct max98090_priv *max98090 = dev_get_drvdata(&client->dev);
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(max98090->regmap);
 	return 0;
 }
 
@@ -2369,7 +2356,7 @@ static int max98090_runtime_suspend(struct device *dev)
 	return 0;
 }
 
-static struct dev_pm_ops max98090_pm = {
+static const struct dev_pm_ops max98090_pm = {
 	SET_RUNTIME_PM_OPS(max98090_runtime_suspend,
 		max98090_runtime_resume, NULL)
 };
diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c
index f2d61a187830..721587c9cd84 100644
--- a/sound/soc/codecs/si476x.c
+++ b/sound/soc/codecs/si476x.c
@@ -1,3 +1,22 @@
+/*
+ * sound/soc/codecs/si476x.c -- Codec driver for SI476X chips
+ *
+ * Copyright (C) 2012 Innovative Converged Devices(ICD)
+ * Copyright (C) 2013 Andrey Smirnov
+ *
+ * Author: Andrey Smirnov <andrew.smirnov@gmail.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; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ */
+
 #include <linux/module.h>
 #include <linux/slab.h>
 #include <sound/pcm.h>
@@ -45,13 +64,23 @@ static unsigned int si476x_codec_read(struct snd_soc_codec *codec,
 				      unsigned int reg)
 {
 	int err;
+	unsigned int val;
 	struct si476x_core *core = codec->control_data;
 
 	si476x_core_lock(core);
-	err = si476x_core_cmd_get_property(core, reg);
+	if (!si476x_core_is_powered_up(core))
+		regcache_cache_only(core->regmap, true);
+
+	err = regmap_read(core->regmap, reg, &val);
+
+	if (!si476x_core_is_powered_up(core))
+		regcache_cache_only(core->regmap, false);
 	si476x_core_unlock(core);
 
-	return err;
+	if (err < 0)
+		return err;
+
+	return val;
 }
 
 static int si476x_codec_write(struct snd_soc_codec *codec,
@@ -61,7 +90,13 @@ static int si476x_codec_write(struct snd_soc_codec *codec,
 	struct si476x_core *core = codec->control_data;
 
 	si476x_core_lock(core);
-	err = si476x_core_cmd_set_property(core, reg, val);
+	if (!si476x_core_is_powered_up(core))
+		regcache_cache_only(core->regmap, true);
+
+	err = regmap_write(core->regmap, reg, val);
+
+	if (!si476x_core_is_powered_up(core))
+		regcache_cache_only(core->regmap, false);
 	si476x_core_unlock(core);
 
 	return err;
@@ -140,7 +175,7 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		dev_err(codec_dai->codec->dev, "Failed to set output format\n");
 		return err;
 	}
-	
+
 	return 0;
 }
 
@@ -159,6 +194,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		width = SI476X_PCM_FORMAT_S8;
+		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		width = SI476X_PCM_FORMAT_S16_LE;
 		break;
@@ -181,7 +217,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream,
 
 	err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT,
 				  SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK,
-				  (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | 
+				  (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) |
 				  (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT));
 	if (err < 0) {
 		dev_err(dai->codec->dev, "Failed to set output width\n");
@@ -250,6 +286,6 @@ static struct platform_driver si476x_platform_driver = {
 };
 module_platform_driver(si476x_platform_driver);
 
-MODULE_AUTHOR("Andrey Smirnov <andrey.smirnov@convergeddevices.net>");
+MODULE_AUTHOR("Andrey Smirnov <andrew.smirnov@gmail.com>");
 MODULE_DESCRIPTION("ASoC Si4761/64 codec driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c
new file mode 100644
index 000000000000..d447c4aa1d5e
--- /dev/null
+++ b/sound/soc/codecs/tas5086.c
@@ -0,0 +1,591 @@
+/*
+ * TAS5086 ASoC codec driver
+ *
+ * Copyright (c) 2013 Daniel Mack <zonque@gmail.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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * TODO:
+ *  - implement DAPM and input muxing
+ *  - implement modulation limit
+ *  - implement non-default PWM start
+ *
+ * Note that this chip has a very unusual register layout, specifically
+ * because the registers are of unequal size, and multi-byte registers
+ * require bulk writes to take effect. Regmap does not support that kind
+ * of devices.
+ *
+ * Currently, the driver does not touch any of the registers >= 0x20, so
+ * it doesn't matter because the entire map can be accessed as 8-bit
+ * array. In case more features will be added in the future
+ * that require access to higher registers, the entire regmap H/W I/O
+ * routines have to be open-coded.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+#include <linux/of_device.h>
+#include <linux/of_gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/tas5086.h>
+
+#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE  |		\
+			     SNDRV_PCM_FMTBIT_S20_3LE |		\
+			     SNDRV_PCM_FMTBIT_S24_3LE)
+
+#define TAS5086_PCM_RATES   (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100  | \
+			     SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200  | \
+			     SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
+			     SNDRV_PCM_RATE_192000)
+
+/*
+ * TAS5086 registers
+ */
+#define TAS5086_CLOCK_CONTROL		0x00	/* Clock control register  */
+#define TAS5086_CLOCK_RATE(val)		(val << 5)
+#define TAS5086_CLOCK_RATE_MASK		(0x7 << 5)
+#define TAS5086_CLOCK_RATIO(val)	(val << 2)
+#define TAS5086_CLOCK_RATIO_MASK	(0x7 << 2)
+#define TAS5086_CLOCK_SCLK_RATIO_48	(1 << 1)
+#define TAS5086_CLOCK_VALID		(1 << 0)
+
+#define TAS5086_DEEMPH_MASK		0x03
+#define TAS5086_SOFT_MUTE_ALL		0x3f
+
+#define TAS5086_DEV_ID			0x01	/* Device ID register */
+#define TAS5086_ERROR_STATUS		0x02	/* Error status register */
+#define TAS5086_SYS_CONTROL_1		0x03	/* System control register 1 */
+#define TAS5086_SERIAL_DATA_IF		0x04	/* Serial data interface register  */
+#define TAS5086_SYS_CONTROL_2		0x05	/* System control register 2 */
+#define TAS5086_SOFT_MUTE		0x06	/* Soft mute register */
+#define TAS5086_MASTER_VOL		0x07	/* Master volume  */
+#define TAS5086_CHANNEL_VOL(X)		(0x08 + (X))	/* Channel 1-6 volume */
+#define TAS5086_VOLUME_CONTROL		0x09	/* Volume control register */
+#define TAS5086_MOD_LIMIT		0x10	/* Modulation limit register */
+#define TAS5086_PWM_START		0x18	/* PWM start register */
+#define TAS5086_SURROUND		0x19	/* Surround register */
+#define TAS5086_SPLIT_CAP_CHARGE	0x1a	/* Split cap charge period register */
+#define TAS5086_OSC_TRIM		0x1b	/* Oscillator trim register */
+#define TAS5086_BKNDERR 		0x1c
+
+/*
+ * Default TAS5086 power-up configuration
+ */
+static const struct reg_default tas5086_reg_defaults[] = {
+	{ 0x00,	0x6c },
+	{ 0x01,	0x03 },
+	{ 0x02,	0x00 },
+	{ 0x03,	0xa0 },
+	{ 0x04,	0x05 },
+	{ 0x05,	0x60 },
+	{ 0x06,	0x00 },
+	{ 0x07,	0xff },
+	{ 0x08,	0x30 },
+	{ 0x09,	0x30 },
+	{ 0x0a,	0x30 },
+	{ 0x0b,	0x30 },
+	{ 0x0c,	0x30 },
+	{ 0x0d,	0x30 },
+	{ 0x0e,	0xb1 },
+	{ 0x0f,	0x00 },
+	{ 0x10,	0x02 },
+	{ 0x11,	0x00 },
+	{ 0x12,	0x00 },
+	{ 0x13,	0x00 },
+	{ 0x14,	0x00 },
+	{ 0x15,	0x00 },
+	{ 0x16,	0x00 },
+	{ 0x17,	0x00 },
+	{ 0x18,	0x3f },
+	{ 0x19,	0x00 },
+	{ 0x1a,	0x18 },
+	{ 0x1b,	0x82 },
+	{ 0x1c,	0x05 },
+};
+
+static bool tas5086_accessible_reg(struct device *dev, unsigned int reg)
+{
+	return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17));
+}
+
+static bool tas5086_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TAS5086_DEV_ID:
+	case TAS5086_ERROR_STATUS:
+		return true;
+	}
+
+	return false;
+}
+
+static bool tas5086_writeable_reg(struct device *dev, unsigned int reg)
+{
+	return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID);
+}
+
+struct tas5086_private {
+	struct regmap	*regmap;
+	unsigned int	mclk, sclk;
+	unsigned int	format;
+	bool		deemph;
+	/* Current sample rate for de-emphasis control */
+	int		rate;
+	/* GPIO driving Reset pin, if any */
+	int		gpio_nreset;
+};
+
+static int tas5086_deemph[] = { 0, 32000, 44100, 48000 };
+
+static int tas5086_set_deemph(struct snd_soc_codec *codec)
+{
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+	int i, val = 0;
+
+	if (priv->deemph)
+		for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++)
+			if (tas5086_deemph[i] == priv->rate)
+				val = i;
+
+	return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1,
+				  TAS5086_DEEMPH_MASK, val);
+}
+
+static int tas5086_get_deemph(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = priv->deemph;
+
+	return 0;
+}
+
+static int tas5086_put_deemph(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	priv->deemph = ucontrol->value.enumerated.item[0];
+
+	return tas5086_set_deemph(codec);
+}
+
+
+static int tas5086_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 tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (clk_id) {
+	case TAS5086_CLK_IDX_MCLK:
+		priv->mclk = freq;
+		break;
+	case TAS5086_CLK_IDX_SCLK:
+		priv->sclk = freq;
+		break;
+	}
+
+	return 0;
+}
+
+static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			       unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	/* The TAS5086 can only be slave to all clocks */
+	if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+		dev_err(codec->dev, "Invalid clocking mode\n");
+		return -EINVAL;
+	}
+
+	/* we need to refer to the data format from hw_params() */
+	priv->format = format;
+
+	return 0;
+}
+
+static const int tas5086_sample_rates[] = {
+	32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000
+};
+
+static const int tas5086_ratios[] = {
+	64, 128, 192, 256, 384, 512
+};
+
+static int index_in_array(const int *array, int len, int needle)
+{
+	int i;
+
+	for (i = 0; i < len; i++)
+		if (array[i] == needle)
+			return i;
+
+	return -ENOENT;
+}
+
+static int tas5086_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+	int val;
+	int ret;
+
+	priv->rate = params_rate(params);
+
+	/* Look up the sample rate and refer to the offset in the list */
+	val = index_in_array(tas5086_sample_rates,
+			     ARRAY_SIZE(tas5086_sample_rates), priv->rate);
+
+	if (val < 0) {
+		dev_err(codec->dev, "Invalid sample rate\n");
+		return -EINVAL;
+	}
+
+	ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL,
+				 TAS5086_CLOCK_RATE_MASK,
+				 TAS5086_CLOCK_RATE(val));
+	if (ret < 0)
+		return ret;
+
+	/* MCLK / Fs ratio */
+	val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios),
+			     priv->mclk / priv->rate);
+	if (val < 0) {
+		dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n");
+		return -EINVAL;
+	}
+
+	ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL,
+				 TAS5086_CLOCK_RATIO_MASK,
+				 TAS5086_CLOCK_RATIO(val));
+	if (ret < 0)
+		return ret;
+
+
+	ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL,
+				 TAS5086_CLOCK_SCLK_RATIO_48,
+				 (priv->sclk == 48 * priv->rate) ? 
+					TAS5086_CLOCK_SCLK_RATIO_48 : 0);
+	if (ret < 0)
+		return ret;
+
+	/*
+	 * The chip has a very unituitive register mapping and muxes information
+	 * about data format and sample depth into the same register, but not on
+	 * a logical bit-boundary. Hence, we have to refer to the format passed
+	 * in the set_dai_fmt() callback and set up everything from here.
+	 *
+	 * First, determine the 'base' value, using the format ...
+	 */
+	switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+		val = 0x00;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		val = 0x03;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val = 0x06;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	/* ... then add the offset for the sample bit depth. */
+	switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+		val += 0;
+                break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val += 1;
+		break;
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		val += 2;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid bit width\n");
+		return -EINVAL;
+	};
+
+	ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val);
+	if (ret < 0)
+		return ret;
+
+	/* clock is considered valid now */
+	ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL,
+				 TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID);
+	if (ret < 0)
+		return ret;
+
+	return tas5086_set_deemph(codec);
+}
+
+static int tas5086_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+	unsigned int val = 0;
+
+	if (mute)
+		val = TAS5086_SOFT_MUTE_ALL;
+
+	return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val);
+}
+
+/* TAS5086 controls */
+static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1);
+
+static const struct snd_kcontrol_new tas5086_controls[] = {
+	SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL,
+		       0, 0xff, 1, tas5086_dac_tlv),
+	SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume",
+			 TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1),
+			 0, 0xff, 1, tas5086_dac_tlv),
+	SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume",
+			 TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3),
+			 0, 0xff, 1, tas5086_dac_tlv),
+	SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume",
+			 TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5),
+			 0, 0xff, 1, tas5086_dac_tlv),
+	SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0,
+			    tas5086_get_deemph, tas5086_put_deemph),
+};
+
+static const struct snd_soc_dai_ops tas5086_dai_ops = {
+	.hw_params	= tas5086_hw_params,
+	.set_sysclk	= tas5086_set_dai_sysclk,
+	.set_fmt	= tas5086_set_dai_fmt,
+	.mute_stream	= tas5086_mute_stream,
+};
+
+static struct snd_soc_dai_driver tas5086_dai = {
+	.name = "tas5086-hifi",
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 2,
+		.channels_max	= 6,
+		.rates		= TAS5086_PCM_RATES,
+		.formats	= TAS5086_PCM_FORMATS,
+	},
+	.ops = &tas5086_dai_ops,
+};
+
+#ifdef CONFIG_PM
+static int tas5086_soc_resume(struct snd_soc_codec *codec)
+{
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	/* Restore codec state */
+	return regcache_sync(priv->regmap);
+}
+#else
+#define tas5086_soc_resume	NULL
+#endif /* CONFIG_PM */
+
+#ifdef CONFIG_OF
+static const struct of_device_id tas5086_dt_ids[] = {
+	{ .compatible = "ti,tas5086", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, tas5086_dt_ids);
+#endif
+
+/* charge period values in microseconds */
+static const int tas5086_charge_period[] = {
+	  13000,  16900,   23400,   31200,   41600,   54600,   72800,   96200,
+	 130000, 156000,  234000,  312000,  416000,  546000,  728000,  962000,
+	1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000,
+};
+
+static int tas5086_probe(struct snd_soc_codec *codec)
+{
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+	int charge_period = 1300000; /* hardware default is 1300 ms */
+	int i, ret;
+
+	if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) {
+		struct device_node *of_node = codec->dev->of_node;
+		of_property_read_u32(of_node, "ti,charge-period", &charge_period);
+	}
+
+	/* lookup and set split-capacitor charge period */
+	if (charge_period == 0) {
+		regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0);
+	} else {
+		i = index_in_array(tas5086_charge_period,
+				   ARRAY_SIZE(tas5086_charge_period),
+				   charge_period);
+		if (i >= 0)
+			regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE,
+				     i + 0x08);
+		else
+			dev_warn(codec->dev,
+				 "Invalid split-cap charge period of %d ns.\n",
+				 charge_period);
+	}
+
+	/* enable factory trim */
+	ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00);
+	if (ret < 0)
+		return ret;
+
+	/* start all channels */
+	ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20);
+	if (ret < 0)
+		return ret;
+
+	/* set master volume to 0 dB */
+	ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30);
+	if (ret < 0)
+		return ret;
+
+	/* mute all channels for now */
+	ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE,
+			   TAS5086_SOFT_MUTE_ALL);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int tas5086_remove(struct snd_soc_codec *codec)
+{
+	struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec);
+
+	if (gpio_is_valid(priv->gpio_nreset))
+		/* Set codec to the reset state */
+		gpio_set_value(priv->gpio_nreset, 0);
+
+	return 0;
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_tas5086 = {
+	.probe			= tas5086_probe,
+	.remove			= tas5086_remove,
+	.resume			= tas5086_soc_resume,
+	.controls		= tas5086_controls,
+	.num_controls		= ARRAY_SIZE(tas5086_controls),
+};
+
+static const struct i2c_device_id tas5086_i2c_id[] = {
+	{ "tas5086", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id);
+
+static const struct regmap_config tas5086_regmap = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+	.max_register		= ARRAY_SIZE(tas5086_reg_defaults),
+	.reg_defaults		= tas5086_reg_defaults,
+	.num_reg_defaults	= ARRAY_SIZE(tas5086_reg_defaults),
+	.cache_type		= REGCACHE_RBTREE,
+	.volatile_reg		= tas5086_volatile_reg,
+	.writeable_reg		= tas5086_writeable_reg,
+	.readable_reg		= tas5086_accessible_reg,
+};
+
+static int tas5086_i2c_probe(struct i2c_client *i2c,
+			     const struct i2c_device_id *id)
+{
+	struct tas5086_private *priv;
+	struct device *dev = &i2c->dev;
+	int gpio_nreset = -EINVAL;
+	int i, ret;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap);
+	if (IS_ERR(priv->regmap)) {
+		ret = PTR_ERR(priv->regmap);
+		dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret);
+		return ret;
+	}
+
+	i2c_set_clientdata(i2c, priv);
+
+	if (of_match_device(of_match_ptr(tas5086_dt_ids), dev)) {
+		struct device_node *of_node = dev->of_node;
+		gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0);
+	}
+
+	if (gpio_is_valid(gpio_nreset))
+		if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset"))
+			gpio_nreset = -EINVAL;
+
+	if (gpio_is_valid(gpio_nreset)) {
+		/* Reset codec - minimum assertion time is 400ns */
+		gpio_direction_output(gpio_nreset, 0);
+		udelay(1);
+		gpio_set_value(gpio_nreset, 1);
+
+		/* Codec needs ~15ms to wake up */
+		msleep(15);
+	}
+
+	priv->gpio_nreset = gpio_nreset;
+
+	/* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */
+	ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i);
+	if (ret < 0)
+		return ret;
+
+	if (i != 0x3) {
+		dev_err(dev,
+			"Failed to identify TAS5086 codec (got %02x)\n", i);
+		return -ENODEV;
+	}
+
+	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086,
+		&tas5086_dai, 1);
+}
+
+static int tas5086_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_codec(&i2c->dev);
+	return 0;
+}
+
+static struct i2c_driver tas5086_i2c_driver = {
+	.driver = {
+		.name	= "tas5086",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(tas5086_dt_ids),
+	},
+	.id_table	= tas5086_i2c_id,
+	.probe		= tas5086_i2c_probe,
+	.remove		= tas5086_i2c_remove,
+};
+
+module_i2c_driver(tas5086_i2c_driver);
+
+MODULE_AUTHOR("Daniel Mack <zonque@gmail.com>");
+MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index b82bbf584146..f0b98bc9ebb4 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -984,22 +984,28 @@ SND_SOC_DAPM_INPUT("IN3R"),
 
 SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 
 SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
 		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
@@ -1494,6 +1500,12 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 		return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout);
 	case WM5102_FLL2:
 		return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout);
+	case WM5102_FLL1_REFCLK:
+		return arizona_set_fll_refclk(&wm5102->fll[0], source, Fref,
+					      Fout);
+	case WM5102_FLL2_REFCLK:
+		return arizona_set_fll_refclk(&wm5102->fll[1], source, Fref,
+					      Fout);
 	default:
 		return -EINVAL;
 	}
@@ -1604,13 +1616,6 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec)
 #define WM5102_DIG_VU 0x0200
 
 static unsigned int wm5102_digital_vu[] = {
-	ARIZONA_ADC_DIGITAL_VOLUME_1L,
-	ARIZONA_ADC_DIGITAL_VOLUME_1R,
-	ARIZONA_ADC_DIGITAL_VOLUME_2L,
-	ARIZONA_ADC_DIGITAL_VOLUME_2R,
-	ARIZONA_ADC_DIGITAL_VOLUME_3L,
-	ARIZONA_ADC_DIGITAL_VOLUME_3R,
-
 	ARIZONA_DAC_DIGITAL_VOLUME_1L,
 	ARIZONA_DAC_DIGITAL_VOLUME_1R,
 	ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -1653,6 +1658,7 @@ static int wm5102_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, wm5102);
 
 	wm5102->core.arizona = arizona;
+	wm5102->core.num_inputs = 6;
 
 	wm5102->core.adsp[0].part = "wm5102";
 	wm5102->core.adsp[0].num = 1;
diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h
index d30477f3070c..adb38040f661 100644
--- a/sound/soc/codecs/wm5102.h
+++ b/sound/soc/codecs/wm5102.h
@@ -15,7 +15,9 @@
 
 #include "arizona.h"
 
-#define WM5102_FLL1 1
-#define WM5102_FLL2 2
+#define WM5102_FLL1        1
+#define WM5102_FLL2        2
+#define WM5102_FLL1_REFCLK 3
+#define WM5102_FLL2_REFCLK 4
 
 #endif
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index cdeb301da1f6..b3ba6b2f9fc7 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -416,28 +416,36 @@ SND_SOC_DAPM_INPUT("IN4R"),
 
 SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
-		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
 
 SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
 		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
@@ -880,6 +888,12 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 		return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout);
 	case WM5110_FLL2:
 		return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout);
+	case WM5110_FLL1_REFCLK:
+		return arizona_set_fll_refclk(&wm5110->fll[0], source, Fref,
+					      Fout);
+	case WM5110_FLL2_REFCLK:
+		return arizona_set_fll_refclk(&wm5110->fll[1], source, Fref,
+					      Fout);
 	default:
 		return -EINVAL;
 	}
@@ -987,15 +1001,6 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec)
 #define WM5110_DIG_VU 0x0200
 
 static unsigned int wm5110_digital_vu[] = {
-	ARIZONA_ADC_DIGITAL_VOLUME_1L,
-	ARIZONA_ADC_DIGITAL_VOLUME_1R,
-	ARIZONA_ADC_DIGITAL_VOLUME_2L,
-	ARIZONA_ADC_DIGITAL_VOLUME_2R,
-	ARIZONA_ADC_DIGITAL_VOLUME_3L,
-	ARIZONA_ADC_DIGITAL_VOLUME_3R,
-	ARIZONA_ADC_DIGITAL_VOLUME_4L,
-	ARIZONA_ADC_DIGITAL_VOLUME_4R,
-
 	ARIZONA_DAC_DIGITAL_VOLUME_1L,
 	ARIZONA_DAC_DIGITAL_VOLUME_1R,
 	ARIZONA_DAC_DIGITAL_VOLUME_2L,
@@ -1040,6 +1045,7 @@ static int wm5110_probe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, wm5110);
 
 	wm5110->core.arizona = arizona;
+	wm5110->core.num_inputs = 8;
 
 	for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++)
 		wm5110->fll[i].vco_mult = 3;
diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h
index 75e9351ccab0..e6c0cd4235c5 100644
--- a/sound/soc/codecs/wm5110.h
+++ b/sound/soc/codecs/wm5110.h
@@ -15,7 +15,9 @@
 
 #include "arizona.h"
 
-#define WM5110_FLL1 1
-#define WM5110_FLL2 2
+#define WM5110_FLL1        1
+#define WM5110_FLL2        2
+#define WM5110_FLL1_REFCLK 3
+#define WM5110_FLL2_REFCLK 4
 
 #endif
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index a64b93425ae3..0a4ffdd1d2a7 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -204,6 +204,7 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
 static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1);
 
 static const struct snd_kcontrol_new wm8960_snd_controls[] = {
 SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
@@ -213,6 +214,15 @@ SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
 SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
 	7, 1, 0),
 
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
+	       WM8960_INBMIX1, 4, 7, 0, boost_tlv),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
+	       WM8960_INBMIX1, 1, 7, 0, boost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
+	       WM8960_INBMIX2, 4, 7, 0, boost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
+	       WM8960_INBMIX2, 1, 7, 0, boost_tlv),
+
 SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
 		 0, 255, 0, dac_tlv),
 
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index f3f7e75f8628..b0ef39eb623d 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -549,8 +549,9 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
 		buf_size = sizeof(adsp1_id);
 
 		algs = be32_to_cpu(adsp1_id.algs);
+		dsp->fw_id = be32_to_cpu(adsp1_id.fw.id);
 		adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n",
-			  be32_to_cpu(adsp1_id.fw.id),
+			  dsp->fw_id,
 			  (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16,
 			  (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8,
 			  be32_to_cpu(adsp1_id.fw.ver) & 0xff,
@@ -573,8 +574,9 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp)
 		buf_size = sizeof(adsp2_id);
 
 		algs = be32_to_cpu(adsp2_id.algs);
+		dsp->fw_id = be32_to_cpu(adsp2_id.fw.id);
 		adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n",
-			  be32_to_cpu(adsp2_id.fw.id),
+			  dsp->fw_id,
 			  (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16,
 			  (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8,
 			  be32_to_cpu(adsp2_id.fw.ver) & 0xff,
@@ -781,8 +783,24 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 		case (WMFW_INFO_TEXT << 8):
 			break;
 		case (WMFW_ABSOLUTE << 8):
-			region_name = "register";
-			reg = offset;
+			/*
+			 * Old files may use this for global
+			 * coefficients.
+			 */
+			if (le32_to_cpu(blk->id) == dsp->fw_id &&
+			    offset == 0) {
+				region_name = "global coefficients";
+				mem = wm_adsp_find_region(dsp, type);
+				if (!mem) {
+					adsp_err(dsp, "No ZM\n");
+					break;
+				}
+				reg = wm_adsp_region_to_reg(mem, 0);
+
+			} else {
+				region_name = "register";
+				reg = offset;
+			}
 			break;
 
 		case WMFW_ADSP1_DM:
@@ -828,7 +846,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 						&buf_list);
 			if (!buf) {
 				adsp_err(dsp, "Out of memory\n");
-				return -ENOMEM;
+				ret = -ENOMEM;
+				goto out_fw;
 			}
 
 			adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n",
@@ -865,7 +884,7 @@ out_fw:
 	wm_adsp_buf_free(&buf_list);
 out:
 	kfree(file);
-	return 0;
+	return ret;
 }
 
 int wm_adsp1_init(struct wm_adsp *adsp)
diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h
index cb8871a3ec00..d6fd8af53b5d 100644
--- a/sound/soc/codecs/wm_adsp.h
+++ b/sound/soc/codecs/wm_adsp.h
@@ -46,6 +46,8 @@ struct wm_adsp {
 
 	struct list_head alg_regions;
 
+	int fw_id;
+
 	const struct wm_adsp_region *mem;
 	int num_mems;
 
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 867ae97ddcec..f5d81b948759 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -199,11 +199,12 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg)
 	list_add_tail(&cache->list, &hubs->dcs_cache);
 }
 
-static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec,
+static int wm_hubs_read_dc_servo(struct snd_soc_codec *codec,
 				  u16 *reg_l, u16 *reg_r)
 {
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 	u16 dcs_reg, reg;
+	int ret = 0;
 
 	switch (hubs->dcs_readback_mode) {
 	case 2:
@@ -236,8 +237,9 @@ static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec,
 		break;
 	default:
 		WARN(1, "Unknown DCS readback method\n");
-		return;
+		ret = -1;
 	}
+	return ret;
 }
 
 /*
@@ -286,7 +288,8 @@ static void enable_dc_servo(struct snd_soc_codec *codec)
 				  WM8993_DCS_TRIG_STARTUP_1);
 	}
 
-	wm_hubs_read_dc_servo(codec, &reg_l, &reg_r);
+	if (wm_hubs_read_dc_servo(codec, &reg_l, &reg_r) < 0)
+		return;
 
 	dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
 
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 9321e5c9d8c1..46c9705cec09 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -235,6 +235,8 @@
 #define DISMOD		(val)(val<<2)
 #define TXSTATE		BIT(4)
 #define RXSTATE		BIT(5)
+#define SRMOD_MASK	3
+#define SRMOD_INACTIVE	0
 
 /*
  * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits
@@ -657,12 +659,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 	return 0;
 }
 
-static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
+static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream,
+				    int channels)
 {
 	int i;
 	u8 tx_ser = 0;
 	u8 rx_ser = 0;
-
+	u8 ser;
+	u8 slots = dev->tdm_slots;
+	u8 max_active_serializers = (channels + slots - 1) / slots;
 	/* Default configuration */
 	mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
 
@@ -682,17 +687,33 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
 	for (i = 0; i < dev->num_serializer; i++) {
 		mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i),
 					dev->serial_dir[i]);
-		if (dev->serial_dir[i] == TX_MODE) {
+		if (dev->serial_dir[i] == TX_MODE &&
+					tx_ser < max_active_serializers) {
 			mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
 					AXR(i));
 			tx_ser++;
-		} else if (dev->serial_dir[i] == RX_MODE) {
+		} else if (dev->serial_dir[i] == RX_MODE &&
+					rx_ser < max_active_serializers) {
 			mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG,
 					AXR(i));
 			rx_ser++;
+		} else {
+			mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i),
+					SRMOD_INACTIVE, SRMOD_MASK);
 		}
 	}
 
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		ser = tx_ser;
+	else
+		ser = rx_ser;
+
+	if (ser < max_active_serializers) {
+		dev_warn(dev->dev, "stream has more channels (%d) than are "
+			"enabled in mcasp (%d)\n", channels, ser * slots);
+		return -EINVAL;
+	}
+
 	if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (dev->txnumevt * tx_ser > 64)
 			dev->txnumevt = 1;
@@ -729,6 +750,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
 				((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
 		}
 	}
+
+	return 0;
 }
 
 static void davinci_hw_param(struct davinci_audio_dev *dev, int stream)
@@ -812,8 +835,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 					&dev->dma_params[substream->stream];
 	int word_length;
 	u8 fifo_level;
+	u8 slots = dev->tdm_slots;
+	int channels;
+	struct snd_interval *pcm_channels = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_CHANNELS);
+	channels = pcm_channels->min;
 
-	davinci_hw_common_param(dev, substream->stream);
+	if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL)
+		return -EINVAL;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		fifo_level = dev->txnumevt;
 	else
@@ -862,6 +891,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		dma_params->acnt = dma_params->data_type;
 
 	dma_params->fifo_level = fifo_level;
+	dma_params->active_serializers = (channels + slots - 1) / slots;
 	davinci_config_channel_size(dev, word_length);
 
 	return 0;
@@ -936,13 +966,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 		.name		= "davinci-mcasp.0",
 		.playback	= {
 			.channels_min	= 2,
-			.channels_max 	= 2,
+			.channels_max	= 32 * 16,
 			.rates 		= DAVINCI_MCASP_RATES,
 			.formats	= DAVINCI_MCASP_PCM_FMTS,
 		},
 		.capture 	= {
 			.channels_min 	= 2,
-			.channels_max 	= 2,
+			.channels_max	= 32 * 16,
 			.rates 		= DAVINCI_MCASP_RATES,
 			.formats	= DAVINCI_MCASP_PCM_FMTS,
 		},
@@ -1015,8 +1045,16 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of(
 		pdata->op_mode = val;
 
 	ret = of_property_read_u32(np, "tdm-slots", &val);
-	if (ret >= 0)
+	if (ret >= 0) {
+		if (val < 2 || val > 32) {
+			dev_err(&pdev->dev,
+				"tdm-slots must be in rage [2-32]\n");
+			ret = -EINVAL;
+			goto nodata;
+		}
+
 		pdata->tdm_slots = val;
+	}
 
 	ret = of_property_read_u32(np, "num-serializer", &val);
 	if (ret >= 0)
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index afab81f844ae..078031d61167 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -181,6 +181,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	unsigned short acnt;
 	unsigned int count;
 	unsigned int fifo_level;
+	unsigned char serializers = prtd->params->active_serializers;
 
 	period_size = snd_pcm_lib_period_bytes(substream);
 	dma_offset = prtd->period * period_size;
@@ -194,14 +195,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	data_type = prtd->params->data_type;
 	count = period_size / data_type;
 	if (fifo_level)
-		count /= fifo_level;
+		count /= fifo_level * serializers;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		src = dma_pos;
 		dst = prtd->params->dma_addr;
 		src_bidx = data_type;
-		dst_bidx = 0;
-		src_cidx = data_type * fifo_level;
+		dst_bidx = 4;
+		src_cidx = data_type * fifo_level * serializers;
 		dst_cidx = 0;
 	} else {
 		src = prtd->params->dma_addr;
@@ -209,7 +210,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 		src_bidx = 0;
 		dst_bidx = data_type;
 		src_cidx = 0;
-		dst_cidx = data_type * fifo_level;
+		dst_cidx = data_type * fifo_level * serializers;
 	}
 
 	acnt = prtd->params->acnt;
@@ -223,9 +224,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 		edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0,
 							ASYNC);
 	else
-		edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level,
-							count, fifo_level,
-							ABSYNC);
+		edma_set_transfer_params(prtd->asp_link[0], acnt,
+						fifo_level * serializers,
+						count, fifo_level * serializers,
+						ABSYNC);
 }
 
 static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index b6ef7039dd09..32d7634d7b26 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -27,6 +27,7 @@ struct davinci_pcm_dma_params {
 	unsigned char data_type;	/* xfer data type */
 	unsigned char convert_mono_stereo;
 	unsigned int fifo_level;
+	unsigned char active_serializers; /* num. of active audio serializers */
 };
 
 int davinci_soc_platform_register(struct device *dev);
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 3f333e5b4673..47f046a8fdab 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -262,7 +262,7 @@ static int imx_audmux_probe(struct platform_device *pdev)
 		return PTR_ERR(pinctrl);
 	}
 
-	audmux_clk = clk_get(&pdev->dev, "audmux");
+	audmux_clk = devm_clk_get(&pdev->dev, "audmux");
 	if (IS_ERR(audmux_clk)) {
 		dev_dbg(&pdev->dev, "cannot get clock: %ld\n",
 				PTR_ERR(audmux_clk));
@@ -282,7 +282,6 @@ static int imx_audmux_remove(struct platform_device *pdev)
 {
 	if (audmux_type == IMX31_AUDMUX)
 		audmux_debugfs_remove();
-	clk_put(audmux_clk);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 55464a5b0706..0e3fc8d8e0a3 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -400,7 +400,7 @@ static struct snd_soc_dai_driver imx_ac97_dai = {
 		.stream_name = "AC97 Playback",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_48000,
+		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
@@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97)
 
 	if (imx_ssi->ac97_reset)
 		imx_ssi->ac97_reset(ac97);
+	/* First read sometimes fails, do a dummy read */
+	imx_ssi_ac97_read(ac97, 0);
 }
 
 static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
@@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97)
 
 	if (imx_ssi->ac97_warm_reset)
 		imx_ssi->ac97_warm_reset(ac97);
+
+	/* First read sometimes fails, do a dummy read */
+	imx_ssi_ac97_read(ac97, 0);
 }
 
 struct snd_ac97_bus_ops soc_ac97_ops = {
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 8e52c1485df3..eb4373840bb6 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = {
 	.num_links = ARRAY_SIZE(pcm030_fabric_dai),
 };
 
-static int __init pcm030_fabric_probe(struct platform_device *op)
+static int pcm030_fabric_probe(struct platform_device *op)
 {
 	struct device_node *np = op->dev.of_node;
 	struct device_node *platform_np;
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index ba49ccd9eed9..8ebaf117d81f 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -493,19 +493,9 @@ static int asoc_dmic_probe(struct platform_device *pdev)
 		goto err_put_clk;
 	}
 
-	if (!devm_request_mem_region(&pdev->dev, res->start,
-				     resource_size(res), pdev->name)) {
-		dev_err(dmic->dev, "memory region already claimed\n");
-		ret = -ENODEV;
-		goto err_put_clk;
-	}
-
-	dmic->io_base = devm_ioremap(&pdev->dev, res->start,
-				     resource_size(res));
-	if (!dmic->io_base) {
-		ret = -ENOMEM;
-		goto err_put_clk;
-	}
+	dmic->io_base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(dmic->io_base))
+		return PTR_ERR(dmic->io_base);
 
 	ret = snd_soc_register_dai(&pdev->dev, &omap_dmic_dai);
 	if (ret)
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 5ca11bdac21e..ddfcc1834ff0 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -369,7 +369,7 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai)
 	pm_runtime_get_sync(mcpdm->dev);
 	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00);
 
-	ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler,
+	ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler,
 				0, "McPDM", (void *)mcpdm);
 
 	pm_runtime_put_sync(mcpdm->dev);
@@ -389,7 +389,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
 
-	free_irq(mcpdm->irq, (void *)mcpdm);
 	pm_runtime_disable(mcpdm->dev);
 
 	return 0;
@@ -465,14 +464,9 @@ static int asoc_mcpdm_probe(struct platform_device *pdev)
 	if (res == NULL)
 		return -ENOMEM;
 
-	if (!devm_request_mem_region(&pdev->dev, res->start,
-				     resource_size(res), "McPDM"))
-		return -EBUSY;
-
-	mcpdm->io_base = devm_ioremap(&pdev->dev, res->start,
-				      resource_size(res));
-	if (!mcpdm->io_base)
-		return -ENOMEM;
+	mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(mcpdm->io_base))
+		return PTR_ERR(mcpdm->io_base);
 
 	mcpdm->irq = platform_get_irq(pdev, 0);
 	if (mcpdm->irq < 0)
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 805512f2555a..9e46e1d8cb1b 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -80,12 +80,18 @@ static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
 static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *k, int event)
 {
+	int ret;
+
 	/*
 	 * The PCM1773 DAC datasheet requires 1ms delay between switching
 	 * VCC power on/off and /PD pin high/low
 	 */
 	if (SND_SOC_DAPM_EVENT_ON(event)) {
-		regulator_enable(omap3pandora_dac_reg);
+		ret = regulator_enable(omap3pandora_dac_reg);
+		if (ret) {
+			dev_err(w->dapm->dev, "Failed to power DAC: %d\n", ret);
+			return ret;
+		}
 		mdelay(1);
 		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
 	} else {
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 90e7e6653233..475fb0d8b3c6 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -35,11 +35,10 @@ config SND_SAMSUNG_I2S
 	tristate
 
 config SND_SOC_SAMSUNG_NEO1973_WM8753
-	tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)"
-	depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02)
+	tristate "Audio support for Openmoko Neo1973 Smartphones (GTA02)"
+	depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02
 	select SND_S3C24XX_I2S
 	select SND_SOC_WM8753
-	select SND_SOC_LM4857 if MACH_NEO1973_GTA01
 	select SND_SOC_DFBMCS320
 	help
 	  Say Y here to enable audio support for the Openmoko Neo1973
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index c724026a246f..f830c41f97dd 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -296,7 +296,6 @@ struct fsi_core {
 
 struct fsi_master {
 	void __iomem *base;
-	int irq;
 	struct fsi_priv fsia;
 	struct fsi_priv fsib;
 	const struct fsi_core *core;
@@ -1886,6 +1885,10 @@ static struct snd_soc_platform_driver fsi_soc_platform = {
 	.pcm_free	= fsi_pcm_free,
 };
 
+static const struct snd_soc_component_driver fsi_soc_component = {
+	.name		= "fsi",
+};
+
 /*
  *		platform function
  */
@@ -2002,7 +2005,6 @@ static int fsi_probe(struct platform_device *pdev)
 	}
 
 	/* master setting */
-	master->irq		= irq;
 	master->core		= core;
 	spin_lock_init(&master->lock);
 
@@ -2046,10 +2048,10 @@ static int fsi_probe(struct platform_device *pdev)
 		goto exit_fsib;
 	}
 
-	ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai,
-				    ARRAY_SIZE(fsi_soc_dai));
+	ret = snd_soc_register_component(&pdev->dev, &fsi_soc_component,
+				    fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
 	if (ret < 0) {
-		dev_err(&pdev->dev, "cannot snd dai register\n");
+		dev_err(&pdev->dev, "cannot snd component register\n");
 		goto exit_snd_soc;
 	}
 
@@ -2074,7 +2076,7 @@ static int fsi_remove(struct platform_device *pdev)
 
 	pm_runtime_disable(&pdev->dev);
 
-	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
+	snd_soc_unregister_component(&pdev->dev);
 	snd_soc_unregister_platform(&pdev->dev);
 
 	fsi_stream_remove(&master->fsia);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b7e84a7cd9ee..046d0ec44e7d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -58,6 +58,7 @@ static DEFINE_MUTEX(client_mutex);
 static LIST_HEAD(dai_list);
 static LIST_HEAD(platform_list);
 static LIST_HEAD(codec_list);
+static LIST_HEAD(component_list);
 
 /*
  * This is a timeout to do a DAPM powerdown after a stream is closed().
@@ -3140,7 +3141,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 	if (params->mask) {
 		ret = regmap_read(codec->control_data, params->base, &val);
 		if (ret != 0)
-			return ret;
+			goto out;
 
 		val &= params->mask;
 
@@ -3158,13 +3159,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 			((u32 *)data)[0] |= cpu_to_be32(val);
 			break;
 		default:
-			return -EINVAL;
+			ret = -EINVAL;
+			goto out;
 		}
 	}
 
 	ret = regmap_raw_write(codec->control_data, params->base,
 			       data, len);
 
+out:
 	kfree(data);
 
 	return ret;
@@ -4022,8 +4025,8 @@ int snd_soc_register_codec(struct device *dev,
 	/* create CODEC component name */
 	codec->name = fmt_single_name(dev, &codec->id);
 	if (codec->name == NULL) {
-		kfree(codec);
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto fail_codec;
 	}
 
 	if (codec_drv->compress_type)
@@ -4062,7 +4065,7 @@ int snd_soc_register_codec(struct device *dev,
 						      reg_size, GFP_KERNEL);
 			if (!codec->reg_def_copy) {
 				ret = -ENOMEM;
-				goto fail;
+				goto fail_codec_name;
 			}
 		}
 	}
@@ -4086,18 +4089,22 @@ int snd_soc_register_codec(struct device *dev,
 	mutex_unlock(&client_mutex);
 
 	/* register any DAIs */
-	if (num_dai) {
-		ret = snd_soc_register_dais(dev, dai_drv, num_dai);
-		if (ret < 0)
-			dev_err(codec->dev, "ASoC: Failed to regster"
-				" DAIs: %d\n", ret);
+	ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+	if (ret < 0) {
+		dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret);
+		goto fail_codec_name;
 	}
 
 	dev_dbg(codec->dev, "ASoC: Registered codec '%s'\n", codec->name);
 	return 0;
 
-fail:
+fail_codec_name:
+	mutex_lock(&client_mutex);
+	list_del(&codec->list);
+	mutex_unlock(&client_mutex);
+
 	kfree(codec->name);
+fail_codec:
 	kfree(codec);
 	return ret;
 }
@@ -4111,7 +4118,6 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec);
 void snd_soc_unregister_codec(struct device *dev)
 {
 	struct snd_soc_codec *codec;
-	int i;
 
 	list_for_each_entry(codec, &codec_list, list) {
 		if (dev == codec->dev)
@@ -4120,9 +4126,7 @@ void snd_soc_unregister_codec(struct device *dev)
 	return;
 
 found:
-	if (codec->num_dai)
-		for (i = 0; i < codec->num_dai; i++)
-			snd_soc_unregister_dai(dev);
+	snd_soc_unregister_dais(dev, codec->num_dai);
 
 	mutex_lock(&client_mutex);
 	list_del(&codec->list);
@@ -4137,6 +4141,82 @@ found:
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
 
+
+/**
+ * snd_soc_register_component - Register a component with the ASoC core
+ *
+ */
+int snd_soc_register_component(struct device *dev,
+			 const struct snd_soc_component_driver *cmpnt_drv,
+			 struct snd_soc_dai_driver *dai_drv,
+			 int num_dai)
+{
+	struct snd_soc_component *cmpnt;
+	int ret;
+
+	dev_dbg(dev, "component register %s\n", dev_name(dev));
+
+	cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL);
+	if (!cmpnt) {
+		dev_err(dev, "ASoC: Failed to allocate memory\n");
+		return -ENOMEM;
+	}
+
+	cmpnt->name = fmt_single_name(dev, &cmpnt->id);
+	if (!cmpnt->name) {
+		dev_err(dev, "ASoC: Failed to simplifying name\n");
+		return -ENOMEM;
+	}
+
+	cmpnt->dev	= dev;
+	cmpnt->driver	= cmpnt_drv;
+	cmpnt->num_dai	= num_dai;
+
+	ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+	if (ret < 0) {
+		dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret);
+		goto error_component_name;
+	}
+
+	mutex_lock(&client_mutex);
+	list_add(&cmpnt->list, &component_list);
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name);
+
+	return ret;
+
+error_component_name:
+	kfree(cmpnt->name);
+
+	return ret;
+}
+
+/**
+ * snd_soc_unregister_component - Unregister a component from the ASoC core
+ *
+ */
+void snd_soc_unregister_component(struct device *dev)
+{
+	struct snd_soc_component *cmpnt;
+
+	list_for_each_entry(cmpnt, &component_list, list) {
+		if (dev == cmpnt->dev)
+			goto found;
+	}
+	return;
+
+found:
+	snd_soc_unregister_dais(dev, cmpnt->num_dai);
+
+	mutex_lock(&client_mutex);
+	list_del(&cmpnt->list);
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name);
+	kfree(cmpnt->name);
+}
+
 /* Retrieve a card's name from device tree */
 int snd_soc_of_parse_card_name(struct snd_soc_card *card,
 			       const char *propname)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 1d6a9b3ceb27..33acd8b892dc 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -831,6 +831,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 		if (path->weak)
 			continue;
 
+		if (path->walking)
+			return 1;
+
 		if (path->walked)
 			continue;
 
@@ -838,6 +841,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 
 		if (path->sink && path->connect) {
 			path->walked = 1;
+			path->walking = 1;
 
 			/* do we need to add this widget to the list ? */
 			if (list) {
@@ -847,11 +851,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 					dev_err(widget->dapm->dev,
 						"ASoC: could not add widget %s\n",
 						widget->name);
+					path->walking = 0;
 					return con;
 				}
 			}
 
 			con += is_connected_output_ep(path->sink, list);
+
+			path->walking = 0;
 		}
 	}
 
@@ -931,6 +938,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 		if (path->weak)
 			continue;
 
+		if (path->walking)
+			return 1;
+
 		if (path->walked)
 			continue;
 
@@ -938,6 +948,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 
 		if (path->source && path->connect) {
 			path->walked = 1;
+			path->walking = 1;
 
 			/* do we need to add this widget to the list ? */
 			if (list) {
@@ -947,11 +958,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 					dev_err(widget->dapm->dev,
 						"ASoC: could not add widget %s\n",
 						widget->name);
+					path->walking = 0;
 					return con;
 				}
 			}
 
 			con += is_connected_input_ep(path->source, list);
+
+			path->walking = 0;
 		}
 	}
 
@@ -3123,7 +3137,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		break;
 	}
 
-	dapm->n_widgets++;
 	w->dapm = dapm;
 	w->codec = dapm->codec;
 	w->platform = dapm->platform;
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index c80adb9da472..48d05d9e1002 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -161,20 +161,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
 			sizeof(struct tegra_alc5632), GFP_KERNEL);
 	if (!alc5632) {
 		dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n");
-		ret = -ENOMEM;
-		goto err;
+		return -ENOMEM;
 	}
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, alc5632);
 
-	if (!(pdev->dev.of_node)) {
-		dev_err(&pdev->dev, "Must be instantiated using device tree\n");
-		ret = -EINVAL;
-		goto err;
-	}
-
 	alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
 	if (alc5632->gpio_hp_det == -EPROBE_DEFER)
 		return -EPROBE_DEFER;
@@ -197,11 +190,11 @@ static int tegra_alc5632_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	tegra_alc5632_dai.cpu_of_node = of_parse_phandle(
-			pdev->dev.of_node, "nvidia,i2s-controller", 0);
+	tegra_alc5632_dai.cpu_of_node = of_parse_phandle(np,
+			"nvidia,i2s-controller", 0);
 	if (!tegra_alc5632_dai.cpu_of_node) {
 		dev_err(&pdev->dev,
-		"Property 'nvidia,i2s-controller' missing or invalid\n");
+			"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
 		goto err;
 	}
diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c
index ba419f86384d..49861c6ed874 100644
--- a/sound/soc/tegra/tegra_asoc_utils.c
+++ b/sound/soc/tegra/tegra_asoc_utils.c
@@ -176,11 +176,7 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
 		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20;
 	else if (of_machine_is_compatible("nvidia,tegra30"))
 		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30;
-	else if (!dev->of_node)
-		/* non-DT is always Tegra20 */
-		data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA20;
 	else
-		/* DT boot, but unknown SoC */
 		return -EINVAL;
 
 	data->clk_pll_a = clk_get_sys(NULL, "pll_a");
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index c8ef88a67c59..f87fc53e9b8c 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -124,6 +124,7 @@ static struct snd_soc_card snd_soc_tegra_wm8753 = {
 
 static int tegra_wm8753_driver_probe(struct platform_device *pdev)
 {
+	struct device_node *np = pdev->dev.of_node;
 	struct snd_soc_card *card = &snd_soc_tegra_wm8753;
 	struct tegra_wm8753 *machine;
 	int ret;
@@ -132,8 +133,7 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev)
 			       GFP_KERNEL);
 	if (!machine) {
 		dev_err(&pdev->dev, "Can't allocate tegra_wm8753 struct\n");
-		ret = -ENOMEM;
-		goto err;
+		return -ENOMEM;
 	}
 
 	card->dev = &pdev->dev;
@@ -148,8 +148,8 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev)
 	if (ret)
 		goto err;
 
-	tegra_wm8753_dai.codec_of_node = of_parse_phandle(
-			pdev->dev.of_node, "nvidia,audio-codec", 0);
+	tegra_wm8753_dai.codec_of_node = of_parse_phandle(np,
+			"nvidia,audio-codec", 0);
 	if (!tegra_wm8753_dai.codec_of_node) {
 		dev_err(&pdev->dev,
 			"Property 'nvidia,audio-codec' missing or invalid\n");
@@ -157,8 +157,8 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	tegra_wm8753_dai.cpu_of_node = of_parse_phandle(
-			pdev->dev.of_node, "nvidia,i2s-controller", 0);
+	tegra_wm8753_dai.cpu_of_node = of_parse_phandle(np,
+			"nvidia,i2s-controller", 0);
 	if (!tegra_wm8753_dai.cpu_of_node) {
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing or invalid\n");
@@ -166,8 +166,7 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	tegra_wm8753_dai.platform_of_node =
-				tegra_wm8753_dai.cpu_of_node;
+	tegra_wm8753_dai.platform_of_node = tegra_wm8753_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index bbd79bf56303..4ac73730d79a 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -39,7 +39,6 @@
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
-#include <sound/tegra_wm8903.h>
 
 #include "../codecs/wm8903.h"
 
@@ -48,7 +47,11 @@
 #define DRV_NAME "tegra-snd-wm8903"
 
 struct tegra_wm8903 {
-	struct tegra_wm8903_platform_data pdata;
+	int gpio_spkr_en;
+	int gpio_hp_det;
+	int gpio_hp_mute;
+	int gpio_int_mic_en;
+	int gpio_ext_mic_en;
 	struct tegra_asoc_utils_data util_data;
 };
 
@@ -129,12 +132,11 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w,
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_soc_card *card = dapm->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
-	if (!gpio_is_valid(pdata->gpio_spkr_en))
+	if (!gpio_is_valid(machine->gpio_spkr_en))
 		return 0;
 
-	gpio_set_value_cansleep(pdata->gpio_spkr_en,
+	gpio_set_value_cansleep(machine->gpio_spkr_en,
 				SND_SOC_DAPM_EVENT_ON(event));
 
 	return 0;
@@ -146,12 +148,11 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w,
 	struct snd_soc_dapm_context *dapm = w->dapm;
 	struct snd_soc_card *card = dapm->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
-	if (!gpio_is_valid(pdata->gpio_hp_mute))
+	if (!gpio_is_valid(machine->gpio_hp_mute))
 		return 0;
 
-	gpio_set_value_cansleep(pdata->gpio_hp_mute,
+	gpio_set_value_cansleep(machine->gpio_hp_mute,
 				!SND_SOC_DAPM_EVENT_ON(event));
 
 	return 0;
@@ -163,17 +164,6 @@ static const struct snd_soc_dapm_widget tegra_wm8903_dapm_widgets[] = {
 	SND_SOC_DAPM_MIC("Mic Jack", NULL),
 };
 
-static const struct snd_soc_dapm_route harmony_audio_map[] = {
-	{"Headphone Jack", NULL, "HPOUTR"},
-	{"Headphone Jack", NULL, "HPOUTL"},
-	{"Int Spk", NULL, "ROP"},
-	{"Int Spk", NULL, "RON"},
-	{"Int Spk", NULL, "LOP"},
-	{"Int Spk", NULL, "LON"},
-	{"Mic Jack", NULL, "MICBIAS"},
-	{"IN1L", NULL, "Mic Jack"},
-};
-
 static const struct snd_kcontrol_new tegra_wm8903_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Int Spk"),
 };
@@ -185,10 +175,9 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
-	if (gpio_is_valid(pdata->gpio_hp_det)) {
-		tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det;
+	if (gpio_is_valid(machine->gpio_hp_det)) {
+		tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det;
 		snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
 				&tegra_wm8903_hp_jack);
 		snd_soc_jack_add_pins(&tegra_wm8903_hp_jack,
@@ -226,9 +215,6 @@ static int tegra_wm8903_remove(struct snd_soc_card *card)
 static struct snd_soc_dai_link tegra_wm8903_dai = {
 	.name = "WM8903",
 	.stream_name = "WM8903 PCM",
-	.codec_name = "wm8903.0-001a",
-	.platform_name = "tegra20-i2s.0",
-	.cpu_dai_name = "tegra20-i2s.0",
 	.codec_dai_name = "wm8903-hifi",
 	.init = tegra_wm8903_init,
 	.ops = &tegra_wm8903_ops,
@@ -257,96 +243,25 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev)
 	struct device_node *np = pdev->dev.of_node;
 	struct snd_soc_card *card = &snd_soc_tegra_wm8903;
 	struct tegra_wm8903 *machine;
-	struct tegra_wm8903_platform_data *pdata;
 	int ret;
 
-	if (!pdev->dev.platform_data && !pdev->dev.of_node) {
-		dev_err(&pdev->dev, "No platform data supplied\n");
-		return -EINVAL;
-	}
-
 	machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm8903),
 			       GFP_KERNEL);
 	if (!machine) {
 		dev_err(&pdev->dev, "Can't allocate tegra_wm8903 struct\n");
-		ret = -ENOMEM;
-		goto err;
+		return -ENOMEM;
 	}
-	pdata = &machine->pdata;
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
-	if (pdev->dev.platform_data) {
-		memcpy(pdata, card->dev->platform_data, sizeof(*pdata));
-	} else if (np) {
-		pdata->gpio_spkr_en = of_get_named_gpio(np,
-						"nvidia,spkr-en-gpios", 0);
-		if (pdata->gpio_spkr_en == -EPROBE_DEFER)
-			return -EPROBE_DEFER;
-
-		pdata->gpio_hp_mute = of_get_named_gpio(np,
-						"nvidia,hp-mute-gpios", 0);
-		if (pdata->gpio_hp_mute == -EPROBE_DEFER)
-			return -EPROBE_DEFER;
-
-		pdata->gpio_hp_det = of_get_named_gpio(np,
-						"nvidia,hp-det-gpios", 0);
-		if (pdata->gpio_hp_det == -EPROBE_DEFER)
-			return -EPROBE_DEFER;
-
-		pdata->gpio_int_mic_en = of_get_named_gpio(np,
-						"nvidia,int-mic-en-gpios", 0);
-		if (pdata->gpio_int_mic_en == -EPROBE_DEFER)
-			return -EPROBE_DEFER;
-
-		pdata->gpio_ext_mic_en = of_get_named_gpio(np,
-						"nvidia,ext-mic-en-gpios", 0);
-		if (pdata->gpio_ext_mic_en == -EPROBE_DEFER)
-			return -EPROBE_DEFER;
-	}
-
-	if (np) {
-		ret = snd_soc_of_parse_card_name(card, "nvidia,model");
-		if (ret)
-			goto err;
-
-		ret = snd_soc_of_parse_audio_routing(card,
-						     "nvidia,audio-routing");
-		if (ret)
-			goto err;
-
-		tegra_wm8903_dai.codec_name = NULL;
-		tegra_wm8903_dai.codec_of_node = of_parse_phandle(np,
-				"nvidia,audio-codec", 0);
-		if (!tegra_wm8903_dai.codec_of_node) {
-			dev_err(&pdev->dev,
-				"Property 'nvidia,audio-codec' missing or invalid\n");
-			ret = -EINVAL;
-			goto err;
-		}
-
-		tegra_wm8903_dai.cpu_dai_name = NULL;
-		tegra_wm8903_dai.cpu_of_node = of_parse_phandle(np,
-				"nvidia,i2s-controller", 0);
-		if (!tegra_wm8903_dai.cpu_of_node) {
-			dev_err(&pdev->dev,
-				"Property 'nvidia,i2s-controller' missing or invalid\n");
-			ret = -EINVAL;
-			goto err;
-		}
-
-		tegra_wm8903_dai.platform_name = NULL;
-		tegra_wm8903_dai.platform_of_node =
-					tegra_wm8903_dai.cpu_of_node;
-	} else {
-		card->dapm_routes = harmony_audio_map;
-		card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
-	}
-
-	if (gpio_is_valid(pdata->gpio_spkr_en)) {
-		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_spkr_en,
+	machine->gpio_spkr_en = of_get_named_gpio(np, "nvidia,spkr-en-gpios",
+						  0);
+	if (machine->gpio_spkr_en == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+	if (gpio_is_valid(machine->gpio_spkr_en)) {
+		ret = devm_gpio_request_one(&pdev->dev, machine->gpio_spkr_en,
 					    GPIOF_OUT_INIT_LOW, "spkr_en");
 		if (ret) {
 			dev_err(card->dev, "cannot get spkr_en gpio\n");
@@ -354,8 +269,12 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		}
 	}
 
-	if (gpio_is_valid(pdata->gpio_hp_mute)) {
-		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_hp_mute,
+	machine->gpio_hp_mute = of_get_named_gpio(np, "nvidia,hp-mute-gpios",
+						  0);
+	if (machine->gpio_hp_mute == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+	if (gpio_is_valid(machine->gpio_hp_mute)) {
+		ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_mute,
 					    GPIOF_OUT_INIT_HIGH, "hp_mute");
 		if (ret) {
 			dev_err(card->dev, "cannot get hp_mute gpio\n");
@@ -363,9 +282,18 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		}
 	}
 
-	if (gpio_is_valid(pdata->gpio_int_mic_en)) {
+	machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
+	if (machine->gpio_hp_det == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+
+	machine->gpio_int_mic_en = of_get_named_gpio(np,
+						"nvidia,int-mic-en-gpios", 0);
+	if (machine->gpio_int_mic_en == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+	if (gpio_is_valid(machine->gpio_int_mic_en)) {
 		/* Disable int mic; enable signal is active-high */
-		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_int_mic_en,
+		ret = devm_gpio_request_one(&pdev->dev,
+					    machine->gpio_int_mic_en,
 					    GPIOF_OUT_INIT_LOW, "int_mic_en");
 		if (ret) {
 			dev_err(card->dev, "cannot get int_mic_en gpio\n");
@@ -373,9 +301,14 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		}
 	}
 
-	if (gpio_is_valid(pdata->gpio_ext_mic_en)) {
+	machine->gpio_ext_mic_en = of_get_named_gpio(np,
+						"nvidia,ext-mic-en-gpios", 0);
+	if (machine->gpio_ext_mic_en == -EPROBE_DEFER)
+		return -EPROBE_DEFER;
+	if (gpio_is_valid(machine->gpio_ext_mic_en)) {
 		/* Enable ext mic; enable signal is active-low */
-		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_ext_mic_en,
+		ret = devm_gpio_request_one(&pdev->dev,
+					    machine->gpio_ext_mic_en,
 					    GPIOF_OUT_INIT_LOW, "ext_mic_en");
 		if (ret) {
 			dev_err(card->dev, "cannot get ext_mic_en gpio\n");
@@ -383,6 +316,34 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev)
 		}
 	}
 
+	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
+	if (ret)
+		goto err;
+
+	ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing");
+	if (ret)
+		goto err;
+
+	tegra_wm8903_dai.codec_of_node = of_parse_phandle(np,
+						"nvidia,audio-codec", 0);
+	if (!tegra_wm8903_dai.codec_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,audio-codec' missing or invalid\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	tegra_wm8903_dai.cpu_of_node = of_parse_phandle(np,
+			"nvidia,i2s-controller", 0);
+	if (!tegra_wm8903_dai.cpu_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,i2s-controller' missing or invalid\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	tegra_wm8903_dai.platform_of_node = tegra_wm8903_dai.cpu_of_node;
+
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
 		goto err;
diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c
index 68d42403d9b5..ce98e5b28360 100644
--- a/sound/soc/tegra/tegra_wm9712.c
+++ b/sound/soc/tegra/tegra_wm9712.c
@@ -79,11 +79,6 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev)
 	struct tegra_wm9712 *machine;
 	int ret;
 
-	if (!pdev->dev.of_node) {
-		dev_err(&pdev->dev, "No platform data supplied\n");
-		return -EINVAL;
-	}
-
 	machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_wm9712),
 			       GFP_KERNEL);
 	if (!machine) {
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 7fcf6c2297db..05c68aab5cf0 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -97,9 +97,6 @@ static const struct snd_soc_dapm_route trimslice_audio_map[] = {
 static struct snd_soc_dai_link trimslice_tlv320aic23_dai = {
 	.name = "TLV320AIC23",
 	.stream_name = "AIC23",
-	.codec_name = "tlv320aic23-codec.2-001a",
-	.platform_name = "tegra20-i2s.0",
-	.cpu_dai_name = "tegra20-i2s.0",
 	.codec_dai_name = "tlv320aic23-hifi",
 	.ops = &trimslice_asoc_ops,
 	.dai_fmt = SND_SOC_DAIFMT_I2S |
@@ -122,6 +119,7 @@ static struct snd_soc_card snd_soc_trimslice = {
 
 static int tegra_snd_trimslice_probe(struct platform_device *pdev)
 {
+	struct device_node *np = pdev->dev.of_node;
 	struct snd_soc_card *card = &snd_soc_trimslice;
 	struct tegra_trimslice *trimslice;
 	int ret;
@@ -130,44 +128,38 @@ static int tegra_snd_trimslice_probe(struct platform_device *pdev)
 				 GFP_KERNEL);
 	if (!trimslice) {
 		dev_err(&pdev->dev, "Can't allocate tegra_trimslice\n");
-		ret = -ENOMEM;
+		return -ENOMEM;
+	}
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, trimslice);
+
+	trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle(np,
+			"nvidia,audio-codec", 0);
+	if (!trimslice_tlv320aic23_dai.codec_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,audio-codec' missing or invalid\n");
+		ret = -EINVAL;
 		goto err;
 	}
 
-	if (pdev->dev.of_node) {
-		trimslice_tlv320aic23_dai.codec_name = NULL;
-		trimslice_tlv320aic23_dai.codec_of_node = of_parse_phandle(
-				pdev->dev.of_node, "nvidia,audio-codec", 0);
-		if (!trimslice_tlv320aic23_dai.codec_of_node) {
-			dev_err(&pdev->dev,
-				"Property 'nvidia,audio-codec' missing or invalid\n");
-			ret = -EINVAL;
-			goto err;
-		}
-
-		trimslice_tlv320aic23_dai.cpu_dai_name = NULL;
-		trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle(
-				pdev->dev.of_node, "nvidia,i2s-controller", 0);
-		if (!trimslice_tlv320aic23_dai.cpu_of_node) {
-			dev_err(&pdev->dev,
-				"Property 'nvidia,i2s-controller' missing or invalid\n");
-			ret = -EINVAL;
-			goto err;
-		}
-
-		trimslice_tlv320aic23_dai.platform_name = NULL;
-		trimslice_tlv320aic23_dai.platform_of_node =
-				trimslice_tlv320aic23_dai.cpu_of_node;
+	trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle(np,
+			"nvidia,i2s-controller", 0);
+	if (!trimslice_tlv320aic23_dai.cpu_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,i2s-controller' missing or invalid\n");
+		ret = -EINVAL;
+		goto err;
 	}
 
+	trimslice_tlv320aic23_dai.platform_of_node =
+			trimslice_tlv320aic23_dai.cpu_of_node;
+
 	ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev);
 	if (ret)
 		goto err;
 
-	card->dev = &pdev->dev;
-	platform_set_drvdata(pdev, card);
-	snd_soc_card_set_drvdata(card, trimslice);
-
 	ret = snd_soc_register_card(card);
 	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",