Merge branch 'master' into for-next

Fast-forwarded to current state of Linus' tree as there are patches to be
applied for files that didn't exist on the old branch.

44 files changed, 185 insertions, 108 deletions

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 5d230cee3fa7..7fbfa051f6e1 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -672,7 +672,7 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
 	/* re-enable interrupts */
 	ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
 
-	/* Re-enable recieve and transmit as appropriate */
+	/* Re-enable receive and transmit as appropriate */
 	cr = 0;
 	cr |=
 	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d63c1754e05f..6943e24a74a1 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -51,7 +51,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TWL6040 if TWL4030_CORE
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
-	select SND_SOC_WL1273 if RADIO_WL1273
+	select SND_SOC_WL1273 if MFD_WL1273_CORE
 	select SND_SOC_WM2000 if I2C
 	select SND_SOC_WM8350 if MFD_WM8350
 	select SND_SOC_WM8400 if MFD_WM8400
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 4f377c9e868d..eecffb548947 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -481,7 +481,7 @@ struct _pll_div {
 };
 
 /* Note : pll code from original alc5623 driver. Not sure of how good it is */
-/* usefull only for master mode */
+/* useful only for master mode */
 static const struct _pll_div codec_master_pll_div[] = {
 
 	{  2048000,  8192000,	0x0ea0},
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 347a567b01e1..b8066ef10bb0 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -153,7 +153,8 @@ static int cq93vc_resume(struct snd_soc_codec *codec)
 
 static int cq93vc_probe(struct snd_soc_codec *codec)
 {
-	struct davinci_vc *davinci_vc = snd_soc_codec_get_drvdata(codec);
+	struct davinci_vc *davinci_vc =
+			mfd_get_data(to_platform_device(codec->dev));
 
 	davinci_vc->cq93vc.codec = codec;
 	codec->control_data = davinci_vc;
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index f7cd346fd727..f5ccdbf7ebc6 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec)
 	snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes,
 		ARRAY_SIZE(jz4740_codec_dapm_routes));
 
-	snd_soc_dapm_new_widgets(codec);
-
 	jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 72de47e5d040..2c2a681da0d7 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -161,7 +161,7 @@ static const struct snd_kcontrol_new lm4857_controls[] = {
 		lm4857_get_mode, lm4857_set_mode),
 };
 
-/* There is a demux inbetween the the input signal and the output signals.
+/* There is a demux between the input signal and the output signals.
  * Currently there is no easy way to model it in ASoC and since it does not make
  * much of a difference in practice simply connect the input direclty to the
  * outputs. */
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 1f7217f703ee..ff29380c9ed3 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -772,6 +772,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+#ifdef CONFIG_REGULATOR
 static int ldo_regulator_is_enabled(struct regulator_dev *dev)
 {
 	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
@@ -901,6 +902,19 @@ static int ldo_regulator_remove(struct snd_soc_codec *codec)
 
 	return 0;
 }
+#else
+static int ldo_regulator_register(struct snd_soc_codec *codec,
+				struct regulator_init_data *init_data,
+				int voltage)
+{
+	return -EINVAL;
+}
+
+static int ldo_regulator_remove(struct snd_soc_codec *codec)
+{
+	return 0;
+}
+#endif
 
 /*
  * set dac bias
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 2a30eae1881c..4d9fb279e146 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -26,7 +26,9 @@
 #define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
 
 #include <linux/platform_device.h>
+#include <linux/delay.h>
 #include <linux/slab.h>
+
 #include <asm/intel_scu_ipc.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -925,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = {
 		.owner		= THIS_MODULE,
 	},
 	.probe		= sn95031_device_probe,
-	.remove		= sn95031_device_remove,
+	.remove		= __devexit_p(sn95031_device_remove),
 };
 
 static int __init sn95031_init(void)
diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h
index 62b1f2261429..67f19c3bebe6 100644
--- a/sound/soc/codecs/tlv320aic26.h
+++ b/sound/soc/codecs/tlv320aic26.h
@@ -14,14 +14,14 @@
 #define AIC26_PAGE_ADDR(page, offset)	((page << 6) | offset)
 #define AIC26_NUM_REGS			AIC26_PAGE_ADDR(3, 0)
 
-/* Page 0: Auxillary data registers */
+/* Page 0: Auxiliary data registers */
 #define AIC26_REG_BAT1			AIC26_PAGE_ADDR(0, 0x05)
 #define AIC26_REG_BAT2			AIC26_PAGE_ADDR(0, 0x06)
 #define AIC26_REG_AUX			AIC26_PAGE_ADDR(0, 0x07)
 #define AIC26_REG_TEMP1			AIC26_PAGE_ADDR(0, 0x09)
 #define AIC26_REG_TEMP2			AIC26_PAGE_ADDR(0, 0x0A)
 
-/* Page 1: Auxillary control registers */
+/* Page 1: Auxiliary control registers */
 #define AIC26_REG_AUX_ADC		AIC26_PAGE_ADDR(1, 0x00)
 #define AIC26_REG_STATUS		AIC26_PAGE_ADDR(1, 0x01)
 #define AIC26_REG_REFERENCE		AIC26_PAGE_ADDR(1, 0x03)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 3bedab26892f..6c43c13f0430 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -884,7 +884,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	if (bypass_pll)
 		return 0;
 
-	/* Use PLL, compute apropriate setup for j, d, r and p, the closest
+	/* Use PLL, compute appropriate setup for j, d, r and p, the closest
 	 * one wins the game. Try with d==0 first, next with d!=0.
 	 * Constraints for j are according to the datasheet.
 	 * The sysclk is divided by 1000 to prevent integer overflows.
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 00b6d87e7bdb..082e9d51963f 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -324,6 +324,10 @@ static void dac33_init_chip(struct snd_soc_codec *codec)
 	dac33_write(codec, DAC33_OUT_AMP_CTRL,
 		    dac33_read_reg_cache(codec, DAC33_OUT_AMP_CTRL));
 
+	dac33_write(codec, DAC33_LDAC_PWR_CTRL,
+		    dac33_read_reg_cache(codec, DAC33_LDAC_PWR_CTRL));
+	dac33_write(codec, DAC33_RDAC_PWR_CTRL,
+		    dac33_read_reg_cache(codec, DAC33_RDAC_PWR_CTRL));
 }
 
 static inline int dac33_read_id(struct snd_soc_codec *codec)
@@ -670,6 +674,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 {
 	struct snd_soc_codec *codec = dac33->codec;
 	unsigned int delay;
+	unsigned long flags;
 
 	switch (dac33->fifo_mode) {
 	case DAC33_FIFO_MODE1:
@@ -677,10 +682,10 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 			DAC33_THRREG(dac33->nsample));
 
 		/* Take the timestamps */
-		spin_lock_irq(&dac33->lock);
+		spin_lock_irqsave(&dac33->lock, flags);
 		dac33->t_stamp2 = ktime_to_us(ktime_get());
 		dac33->t_stamp1 = dac33->t_stamp2;
-		spin_unlock_irq(&dac33->lock);
+		spin_unlock_irqrestore(&dac33->lock, flags);
 
 		dac33_write16(codec, DAC33_PREFILL_MSB,
 				DAC33_THRREG(dac33->alarm_threshold));
@@ -692,11 +697,11 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 		break;
 	case DAC33_FIFO_MODE7:
 		/* Take the timestamp */
-		spin_lock_irq(&dac33->lock);
+		spin_lock_irqsave(&dac33->lock, flags);
 		dac33->t_stamp1 = ktime_to_us(ktime_get());
 		/* Move back the timestamp with drain time */
 		dac33->t_stamp1 -= dac33->mode7_us_to_lthr;
-		spin_unlock_irq(&dac33->lock);
+		spin_unlock_irqrestore(&dac33->lock, flags);
 
 		dac33_write16(codec, DAC33_PREFILL_MSB,
 				DAC33_THRREG(DAC33_MODE7_MARGIN));
@@ -714,13 +719,14 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
 static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33)
 {
 	struct snd_soc_codec *codec = dac33->codec;
+	unsigned long flags;
 
 	switch (dac33->fifo_mode) {
 	case DAC33_FIFO_MODE1:
 		/* Take the timestamp */
-		spin_lock_irq(&dac33->lock);
+		spin_lock_irqsave(&dac33->lock, flags);
 		dac33->t_stamp2 = ktime_to_us(ktime_get());
-		spin_unlock_irq(&dac33->lock);
+		spin_unlock_irqrestore(&dac33->lock, flags);
 
 		dac33_write16(codec, DAC33_NSAMPLE_MSB,
 				DAC33_THRREG(dac33->nsample));
@@ -773,10 +779,11 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev)
 {
 	struct snd_soc_codec *codec = dev;
 	struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+	unsigned long flags;
 
-	spin_lock(&dac33->lock);
+	spin_lock_irqsave(&dac33->lock, flags);
 	dac33->t_stamp1 = ktime_to_us(ktime_get());
-	spin_unlock(&dac33->lock);
+	spin_unlock_irqrestore(&dac33->lock, flags);
 
 	/* Do not schedule the workqueue in Mode7 */
 	if (dac33->fifo_mode != DAC33_FIFO_MODE7)
@@ -1020,7 +1027,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 		/*
 		 * For FIFO bypass mode:
 		 * Enable the FIFO bypass (Disable the FIFO use)
-		 * Set the BCLK as continous
+		 * Set the BCLK as continuous
 		 */
 		fifoctrl_a |= DAC33_FBYPAS;
 		aictrl_b |= DAC33_BCLKON;
@@ -1173,15 +1180,16 @@ static snd_pcm_sframes_t dac33_dai_delay(
 	unsigned int time_delta, uthr;
 	int samples_out, samples_in, samples;
 	snd_pcm_sframes_t delay = 0;
+	unsigned long flags;
 
 	switch (dac33->fifo_mode) {
 	case DAC33_FIFO_BYPASS:
 		break;
 	case DAC33_FIFO_MODE1:
-		spin_lock(&dac33->lock);
+		spin_lock_irqsave(&dac33->lock, flags);
 		t0 = dac33->t_stamp1;
 		t1 = dac33->t_stamp2;
-		spin_unlock(&dac33->lock);
+		spin_unlock_irqrestore(&dac33->lock, flags);
 		t_now = ktime_to_us(ktime_get());
 
 		/* We have not started to fill the FIFO yet, delay is 0 */
@@ -1246,10 +1254,10 @@ static snd_pcm_sframes_t dac33_dai_delay(
 		}
 		break;
 	case DAC33_FIFO_MODE7:
-		spin_lock(&dac33->lock);
+		spin_lock_irqsave(&dac33->lock, flags);
 		t0 = dac33->t_stamp1;
 		uthr = dac33->uthr;
-		spin_unlock(&dac33->lock);
+		spin_unlock_irqrestore(&dac33->lock, flags);
 		t_now = ktime_to_us(ktime_get());
 
 		/* We have not started to fill the FIFO yet, delay is 0 */
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index e4d464b937d6..575238d68e5e 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -26,6 +26,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/mfd/core.h>
 #include <linux/i2c/twl.h>
 #include <linux/slab.h>
 #include <sound/core.h>
@@ -280,7 +281,7 @@ static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
 				 i, val, twl4030_reg[i]);
 		}
 	}
-	dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
+	dev_dbg(codec->dev, "Found %d non-matching registers. %s\n",
 		 difference, difference ? "Not OK" : "OK");
 }
 
@@ -732,7 +733,8 @@ static int aif_event(struct snd_soc_dapm_widget *w,
 
 static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 {
-	struct twl4030_codec_audio_data *pdata = codec->dev->platform_data;
+	struct twl4030_codec_audio_data *pdata =
+			mfd_get_data(to_platform_device(codec->dev));
 	unsigned char hs_gain, hs_pop;
 	struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
 	/* Base values for ramp delay calculation: 2^19 - 2^26 */
@@ -2016,7 +2018,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 	u8 mode;
 
 	/* If the system master clock is not 26MHz, the voice PCM interface is
-	 * not avilable.
+	 * not available.
 	 */
 	if (twl4030->sysclk != 26000) {
 		dev_err(codec->dev, "The board is configured for %u Hz, while"
@@ -2026,7 +2028,7 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 	}
 
 	/* If the codec mode is not option2, the voice PCM interface is not
-	 * avilable.
+	 * available.
 	 */
 	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
 		& TWL4030_OPT_MODE;
@@ -2297,7 +2299,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
 
 static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 {
-	struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data;
+	struct twl4030_codec_audio_data *pdata = mfd_get_data(pdev);
 
 	if (!pdata) {
 		dev_err(&pdev->dev, "platform_data is missing\n");
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 482fcdb59bfa..255901c4460d 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1629,8 +1629,10 @@ static int twl6040_probe(struct snd_soc_codec *codec)
 	priv->naudint = naudint;
 	priv->workqueue = create_singlethread_workqueue("twl6040-codec");
 
-	if (!priv->workqueue)
+	if (!priv->workqueue) {
+		ret = -ENOMEM;
 		goto work_err;
+	}
 
 	INIT_DELAYED_WORK(&priv->delayed_work, twl6040_accessory_work);
 
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index e76847a9438b..48ffd406a71d 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -486,7 +486,8 @@ static struct snd_soc_dai_driver uda134x_dai = {
 static int uda134x_soc_probe(struct snd_soc_codec *codec)
 {
 	struct uda134x_priv *uda134x;
-	struct uda134x_platform_data *pd = dev_get_drvdata(codec->card->dev);
+	struct uda134x_platform_data *pd = codec->card->dev->platform_data;
+
 	int ret;
 
 	printk(KERN_INFO "UDA134X SoC Audio Codec\n");
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 861b28f543d2..c8a874d0d4ca 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -3,7 +3,7 @@
  *
  * Author:      Matti Aaltonen, <matti.j.aaltonen@nokia.com>
  *
- * Copyright:   (C) 2010 Nokia Corporation
+ * Copyright:   (C) 2010, 2011 Nokia Corporation
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -179,7 +179,12 @@ static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" };
+/*
+ * TODO: Implement the audio routing in the driver. Now this control
+ * only indicates the setting that has been done elsewhere (in the user
+ * space).
+ */
+static const char * const wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" };
 
 static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol)
@@ -239,7 +244,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static const char *wl1273_audio_strings[] = { "Digital", "Analog" };
+static const char * const wl1273_audio_strings[] = { "Digital", "Analog" };
 
 static const struct soc_enum wl1273_audio_enum =
 	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings),
@@ -436,7 +441,8 @@ EXPORT_SYMBOL_GPL(wl1273_get_format);
 
 static int wl1273_probe(struct snd_soc_codec *codec)
 {
-	struct wl1273_core **core = codec->dev->platform_data;
+	struct wl1273_core **core =
+			mfd_get_data(to_platform_device(codec->dev));
 	struct wl1273_priv *wl1273;
 	int r;
 
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 3c3bc079167e..736b785e3756 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -22,6 +22,7 @@
 #include <linux/regulator/consumer.h>
 #include <linux/mfd/wm8400-audio.h>
 #include <linux/mfd/wm8400-private.h>
+#include <linux/mfd/core.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -1377,7 +1378,7 @@ static void wm8400_probe_deferred(struct work_struct *work)
 
 static int wm8400_codec_probe(struct snd_soc_codec *codec)
 {
-	struct wm8400 *wm8400 = dev_get_platdata(codec->dev);
+	struct wm8400 *wm8400 = mfd_get_data(to_platform_device(codec->dev));
 	struct wm8400_priv *priv;
 	int ret;
 	u16 reg;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 8f6b5ee6645b..4bbc0a79f01e 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -772,7 +772,7 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 			reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
 			snd_soc_write(codec, WM8580_PWRDN1, reg);
 
-			/* Make VMID high impedence */
+			/* Make VMID high impedance */
 			reg = snd_soc_read(codec,  WM8580_ADC_CONTROL1);
 			reg &= ~0x100;
 			snd_soc_write(codec, WM8580_ADC_CONTROL1, reg);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 3f09deea8d9d..ffa2ffe5ec11 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1312,7 +1312,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
 	SNDRV_PCM_FMTBIT_S24_LE)
 
 /*
- * The WM8753 supports upto 4 different and mutually exclusive DAI
+ * The WM8753 supports up to 4 different and mutually exclusive DAI
  * configurations. This gives 2 PCM's available for use, hifi and voice.
  * NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
  * is connected between the wm8753 and a BT codec or GSM modem.
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ae1cadfae84c..f52b623bb692 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re
 	case WM8903_REVISION_NUMBER:
 	case WM8903_INTERRUPT_STATUS_1:
 	case WM8903_WRITE_SEQUENCER_4:
-	case WM8903_POWER_MANAGEMENT_3:
-	case WM8903_POWER_MANAGEMENT_2:
 	case WM8903_DC_SERVO_READBACK_1:
 	case WM8903_DC_SERVO_READBACK_2:
 	case WM8903_DC_SERVO_READBACK_3:
@@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0,
 SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
 		   right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
 
-SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
-		   4, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
+		   1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2,
 		   0, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0,
 		   NULL, 0),
-SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0,
 		   NULL, 0),
 
 SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
 SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0),
 
 SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
 		   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
 		   NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
 		   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
 		   NULL, 0),
 SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
 		   NULL, 0),
-SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
 		   NULL, 0),
 
 SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
@@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Left Speaker PGA", NULL, "Left Speaker Mixer" },
 	{ "Right Speaker PGA", NULL, "Right Speaker Mixer" },
 
-	{ "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
-	{ "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
-	{ "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
-	{ "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
+	{ "HPL_ENA", NULL, "Left Headphone Output PGA" },
+	{ "HPR_ENA", NULL, "Right Headphone Output PGA" },
+	{ "HPL_ENA_DLY", NULL, "HPL_ENA" },
+	{ "HPR_ENA_DLY", NULL, "HPR_ENA" },
+	{ "LINEOUTL_ENA", NULL, "Left Line Output PGA" },
+	{ "LINEOUTR_ENA", NULL, "Right Line Output PGA" },
+	{ "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" },
+	{ "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" },
 
 	{ "HPL_DCS", NULL, "DCS Master" },
 	{ "HPR_DCS", NULL, "DCS Master" },
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 443ae580445c..9b3bba4df5b3 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1895,7 +1895,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index 5e0214d6293e..3c7198779c31 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -176,7 +176,7 @@ static int wm8995_pll_factors(struct device *dev,
 	return 0;
 }
 
-/* Lookup table specifiying SRATE (table 25 in datasheet); some of the
+/* Lookup table specifying SRATE (table 25 in datasheet); some of the
  * output frequencies have been rounded to the standard frequencies
  * they are intended to match where the error is slight. */
 static struct {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3b71dd65c966..500011eb8b2b 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3137,7 +3137,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("FLL Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c
index 28fdfd66661d..3c2ee1bb73cd 100644
--- a/sound/soc/codecs/wm8991.c
+++ b/sound/soc/codecs/wm8991.c
@@ -981,7 +981,7 @@ static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai,
 		reg = snd_soc_read(codec, WM8991_CLOCKING_2);
 		snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC);
 
-		/* set up N , fractional mode and pre-divisor if neccessary */
+		/* set up N , fractional mode and pre-divisor if necessary */
 		snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM |
 			      (pll_div.div2 ? WM8991_PRESCALE : 0));
 		snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8));
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 379fa22c5b6c..056aef904347 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -324,7 +324,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3dc64c8b6a5c..84e1bd1d2822 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -82,18 +82,18 @@ struct wm8994_priv {
 
 	int mbc_ena[3];
 
-	/* Platform dependant DRC configuration */
+	/* Platform dependent DRC configuration */
 	const char **drc_texts;
 	int drc_cfg[WM8994_NUM_DRC];
 	struct soc_enum drc_enum;
 
-	/* Platform dependant ReTune mobile configuration */
+	/* Platform dependent ReTune mobile configuration */
 	int num_retune_mobile_texts;
 	const char **retune_mobile_texts;
 	int retune_mobile_cfg[WM8994_NUM_EQ];
 	struct soc_enum retune_mobile_enum;
 
-	/* Platform dependant MBC configuration */
+	/* Platform dependent MBC configuration */
 	int mbc_cfg;
 	const char **mbc_texts;
 	struct soc_enum mbc_enum;
@@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Latch volume updates (right only; we always do left then right). */
+	snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME,
+			    WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
 	snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME,
 			    WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU);
+	snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME,
+			    WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
 	snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME,
 			    WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU);
+	snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME,
+			    WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
 	snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME,
 			    WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU);
+	snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME,
+			    WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
 	snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME,
 			    WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU);
+	snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME,
+			    WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
 	snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME,
 			    WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU);
+	snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME,
+			    WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
 	snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME,
 			    WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU);
+	snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME,
+			    WM8994_DAC1_VU, WM8994_DAC1_VU);
 	snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME,
 			    WM8994_DAC1_VU, WM8994_DAC1_VU);
+	snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME,
+			    WM8994_DAC2_VU, WM8994_DAC2_VU);
 	snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME,
 			    WM8994_DAC2_VU, WM8994_DAC2_VU);
 
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 55cdf2982020..91c6b39de50c 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -305,7 +305,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol,
 /*
  * Stop any attempts to change speaker mode while the speaker is enabled.
  *
- * We also have some special anti-pop controls dependant on speaker
+ * We also have some special anti-pop controls dependent on speaker
  * mode which must be changed along with the mode.
  */
 static int speaker_mode_put(struct snd_kcontrol *kcontrol,
@@ -456,7 +456,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	pr_debug("Fvco=%dHz\n", target);
 
-	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	/* Find an appropriate FLL_FRATIO and factor it out of the target */
 	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
 		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
 			fll_div->fll_fratio = fll_fratios[i].fll_fratio;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 7b6b3c18e299..4005e9af5d61 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 
 	{ "SPKL", "Input Switch", "MIXINL" },
 	{ "SPKL", "IN1LP Switch", "IN1LP" },
-	{ "SPKL", "Output Switch", "Left Output Mixer" },
+	{ "SPKL", "Output Switch", "Left Output PGA" },
 	{ "SPKL", NULL, "TOCLK" },
 
 	{ "SPKR", "Input Switch", "MIXINR" },
 	{ "SPKR", "IN1RP Switch", "IN1RP" },
-	{ "SPKR", "Output Switch", "Right Output Mixer" },
+	{ "SPKR", "Output Switch", "Right Output PGA" },
 	{ "SPKR", NULL, "TOCLK" },
 
 	{ "SPKL Boost", "Direct Voice Switch", "Direct Voice" },
@@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "SPKOUTRP", NULL, "SPKR Driver" },
 	{ "SPKOUTRN", NULL, "SPKR Driver" },
 
-	{ "Left Headphone Mux", "Mixer", "Left Output Mixer" },
-	{ "Right Headphone Mux", "Mixer", "Right Output Mixer" },
+	{ "Left Headphone Mux", "Mixer", "Left Output PGA" },
+	{ "Right Headphone Mux", "Mixer", "Right Output PGA" },
 
 	{ "Headphone PGA", NULL, "Left Headphone Mux" },
 	{ "Headphone PGA", NULL, "Right Headphone Mux" },
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 9d2afccc3a2d..13e05a302a92 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -205,7 +205,7 @@ static struct snd_soc_dai_driver davinci_vcif_dai = {
 
 static int davinci_vcif_probe(struct platform_device *pdev)
 {
-	struct davinci_vc *davinci_vc = platform_get_drvdata(pdev);
+	struct davinci_vc *davinci_vc = mfd_get_data(pdev);
 	struct davinci_vcif_dev *davinci_vcif_dev;
 	int ret;
 
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 671ef8dd524c..aab7765f401a 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
 		slave_config.direction = DMA_TO_DEVICE;
 		slave_config.dst_addr = dma_params->dma_addr;
 		slave_config.dst_addr_width = buswidth;
-		slave_config.dst_maxburst = dma_params->burstsize;
+		slave_config.dst_maxburst = dma_params->burstsize * buswidth;
 	} else {
 		slave_config.direction = DMA_FROM_DEVICE;
 		slave_config.src_addr = dma_params->dma_addr;
 		slave_config.src_addr_width = buswidth;
-		slave_config.src_maxburst = dma_params->burstsize;
+		slave_config.src_maxburst = dma_params->burstsize * buswidth;
 	}
 
 	ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config);
@@ -303,6 +303,11 @@ static struct snd_soc_platform_driver imx_soc_platform_mx2 = {
 
 static int __devinit imx_soc_platform_probe(struct platform_device *pdev)
 {
+	struct imx_ssi *ssi = platform_get_drvdata(pdev);
+
+	ssi->dma_params_tx.burstsize = 6;
+	ssi->dma_params_rx.burstsize = 4;
+
 	return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2);
 }
 
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index bc92ec620004..ac2ded969253 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -16,7 +16,7 @@
  * sane processor vendors have a FIFO per AC97 slot, the i.MX has only
  * one FIFO which combines all valid receive slots. We cannot even select
  * which slots we want to receive. The WM9712 with which this driver
- * was developped with always sends GPIO status data in slot 12 which
+ * was developed with always sends GPIO status data in slot 12 which
  * we receive in our (PCM-) data stream. The only chance we have is to
  * manually skip this data in the FIQ handler. With sampling rates different
  * from 48000Hz not every frame has valid receive data, so the ratio
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index a4406a134892..dc8a87530e3e 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -234,7 +234,4 @@ void imx_pcm_free(struct snd_pcm *pcm);
  */
 #define IMX_SSI_DMABUF_SIZE	(64 * 1024)
 
-#define DMA_RXFIFO_BURST      0x4
-#define DMA_TXFIFO_BURST      0x6
-
 #endif /* _IMX_SSI_H */
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index 0fd6a630db01..e13c6ce46328 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 	priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
 	snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
 
-	/* Ensure that all constraints linked to dma burst are fullfilled */
+	/* Ensure that all constraints linked to dma burst are fulfilled */
 	err = snd_pcm_hw_constraint_minmax(runtime,
 			SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
 			priv->burst * 2,
@@ -170,7 +170,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream)
 
 		/*
 		 * Enable Error interrupts. We're only ack'ing them but
-		 * it's usefull for diagnostics
+		 * it's useful for diagnostics
 		 */
 		writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK);
 	}
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index ee2c22475a76..d567c322a2fb 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -116,18 +116,20 @@ struct snd_soc_dai_driver sst_platform_dai[] = {
 static inline void sst_set_stream_status(struct sst_runtime_stream *stream,
 					int state)
 {
-	spin_lock(&stream->status_lock);
+	unsigned long flags;
+	spin_lock_irqsave(&stream->status_lock, flags);
 	stream->stream_status = state;
-	spin_unlock(&stream->status_lock);
+	spin_unlock_irqrestore(&stream->status_lock, flags);
 }
 
 static inline int sst_get_stream_status(struct sst_runtime_stream *stream)
 {
 	int state;
+	unsigned long flags;
 
-	spin_lock(&stream->status_lock);
+	spin_lock_irqsave(&stream->status_lock, flags);
 	state = stream->stream_status;
-	spin_unlock(&stream->status_lock);
+	spin_unlock_irqrestore(&stream->status_lock, flags);
 	return state;
 }
 
@@ -440,7 +442,7 @@ static int sst_platform_remove(struct platform_device *pdev)
 
 	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai));
 	snd_soc_unregister_platform(&pdev->dev);
-	pr_debug("sst_platform_remove sucess\n");
+	pr_debug("sst_platform_remove success\n");
 	return 0;
 }
 
@@ -463,7 +465,7 @@ module_init(sst_soc_platform_init);
 static void __exit sst_soc_platform_exit(void)
 {
 	platform_driver_unregister(&sst_platform_driver);
-	pr_debug("sst_soc_platform_exit sucess\n");
+	pr_debug("sst_soc_platform_exit success\n");
 }
 module_exit(sst_soc_platform_exit);
 
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 3167be689621..462cbcbea74a 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -248,7 +248,7 @@ static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
  */
 
 /* To actually apply any modem controlled configuration changes to the codec,
- * we must connect codec DAI pins to the modem for a moment.  Be carefull not
+ * we must connect codec DAI pins to the modem for a moment.  Be careful not
  * to interfere with our digital mute function that shares the same hardware. */
 static struct timer_list cx81801_timer;
 static bool cx81801_cmd_pending;
@@ -402,9 +402,9 @@ static struct tty_ldisc_ops cx81801_ops = {
 
 
 /*
- * Even if not very usefull, the sound card can still work without any of the
+ * Even if not very useful, the sound card can still work without any of the
  * above functonality activated.  You can still control its audio input/output
- * constellation and speakerphone gain from userspace by issueing AT commands
+ * constellation and speakerphone gain from userspace by issuing AT commands
  * over the modem port.
  */
 
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 784cff5f67e8..9027da466cae 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -310,7 +310,7 @@ static struct snd_soc_dai_link corgi_dai = {
 	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8731-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8731-codec-0.001b",
+	.codec_name = "wm8731-codec.0-001b",
 	.init = corgi_wm8731_init,
 	.ops = &corgi_ops,
 };
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 02fb66416ddc..2ce0b2d891d5 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -65,6 +65,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (prtd->dma_ch >= 0) {
 		pxa_free_dma(prtd->dma_ch);
 		prtd->dma_ch = -1;
+		prtd->params = NULL;
 	}
 
 	return 0;
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index ac577263b3e3..b6445757fc54 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -167,7 +167,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 	.codec_name = "wm9713-codec",
 	.platform_name = "pxa-pcm-audio",
 	.cpu_dai_name = "pxa2xx-ac97",
-	.codec_name = "wm9713-hifi",
+	.codec_dai_name = "wm9713-hifi",
 	.init = zylonite_wm9713_init,
 },
 {
@@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 	.codec_name = "wm9713-codec",
 	.platform_name = "pxa-pcm-audio",
 	.cpu_dai_name = "pxa2xx-ac97-aux",
-	.codec_name = "wm9713-aux",
+	.codec_dai_name = "wm9713-aux",
 },
 {
 	.name = "WM9713 Voice",
@@ -184,7 +184,7 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 	.codec_name = "wm9713-codec",
 	.platform_name = "pxa-pcm-audio",
 	.cpu_dai_name = "pxa-ssp-dai.2",
-	.codec_name = "wm9713-voice",
+	.codec_dai_name = "wm9713-voice",
 	.ops = &zylonite_voice_ops,
 },
 };
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 78bfdb3f5d7e..452230975632 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -228,7 +228,7 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Handset Mic"),
 };
 
-/* GTA02 specific routes and controlls */
+/* GTA02 specific routes and controls */
 
 #ifdef CONFIG_MACH_NEO1973_GTA02
 
@@ -372,7 +372,7 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-/* GTA01 specific controlls */
+/* GTA01 specific controls */
 
 #ifdef CONFIG_MACH_NEO1973_GTA01
 
diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c
index 38aac7d57a59..9c7e8b48aed6 100644
--- a/sound/soc/samsung/pcm.c
+++ b/sound/soc/samsung/pcm.c
@@ -350,8 +350,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai,
 	ctl = readl(regs + S3C_PCM_CTL);
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_NB_NF:
-		/* Nothing to do, NB_NF by default */
+	case SND_SOC_DAIFMT_IB_NF:
+		/* Nothing to do, IB_NF by default */
 		break;
 	default:
 		dev_err(pcm->dev, "Unsupported clock inversion!\n");
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index 3cb700751078..dc9d551f6788 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -219,7 +219,7 @@ static struct snd_soc_ops s3c24xx_uda134x_ops = {
 static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
 	.name = "UDA134X",
 	.stream_name = "UDA134X",
-	.codec_name = "uda134x-hifi",
+	.codec_name = "uda134x-codec",
 	.codec_dai_name = "uda134x-hifi",
 	.cpu_dai_name = "s3c24xx-iis",
 	.ops = &s3c24xx_uda134x_ops,
@@ -314,6 +314,7 @@ static int s3c24xx_uda134x_probe(struct platform_device *pdev)
 
 	platform_set_drvdata(s3c24xx_uda134x_snd_device,
 			     &snd_soc_s3c24xx_uda134x);
+	platform_device_add_data(s3c24xx_uda134x_snd_device, &s3c24xx_uda134x, sizeof(s3c24xx_uda134x));
 	ret = platform_device_add(s3c24xx_uda134x_snd_device);
 	if (ret) {
 		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 0c9997e2d8c0..23c0e83d4c19 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1200,10 +1200,11 @@ static int fsi_probe(struct platform_device *pdev)
 	master->fsib.master	= master;
 
 	pm_runtime_enable(&pdev->dev);
-	pm_runtime_resume(&pdev->dev);
 	dev_set_drvdata(&pdev->dev, master);
 
+	pm_runtime_get_sync(&pdev->dev);
 	fsi_soft_all_reset(master);
+	pm_runtime_put_sync(&pdev->dev);
 
 	ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
 			  id_entry->name, master);
@@ -1218,8 +1219,17 @@ static int fsi_probe(struct platform_device *pdev)
 		goto exit_free_irq;
 	}
 
-	return snd_soc_register_dais(&pdev->dev, fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
+	ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai,
+				    ARRAY_SIZE(fsi_soc_dai));
+	if (ret < 0) {
+		dev_err(&pdev->dev, "cannot snd dai register\n");
+		goto exit_snd_soc;
+	}
+
+	return ret;
 
+exit_snd_soc:
+	snd_soc_unregister_platform(&pdev->dev);
 exit_free_irq:
 	free_irq(irq, master);
 exit_iounmap:
@@ -1238,12 +1248,11 @@ static int fsi_remove(struct platform_device *pdev)
 
 	master = dev_get_drvdata(&pdev->dev);
 
-	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
-	snd_soc_unregister_platform(&pdev->dev);
-
+	free_irq(master->irq, master);
 	pm_runtime_disable(&pdev->dev);
 
-	free_irq(master->irq, master);
+	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai));
+	snd_soc_unregister_platform(&pdev->dev);
 
 	iounmap(master->base);
 	kfree(master);
@@ -1321,3 +1330,4 @@ module_exit(fsi_mobile_exit);
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip FSI audio driver");
 MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
+MODULE_ALIAS("platform:fsi-pcm-audio");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 17efacdb248a..d8562ce4de7a 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -92,8 +92,8 @@ static int min_bytes_needed(unsigned long val)
 static int format_register_str(struct snd_soc_codec *codec,
 			       unsigned int reg, char *buf, size_t len)
 {
-	int wordsize = codec->driver->reg_word_size * 2;
-	int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+	int wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+	int regsize = codec->driver->reg_word_size * 2;
 	int ret;
 	char tmpbuf[len + 1];
 	char regbuf[regsize + 1];
@@ -132,8 +132,8 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf,
 	size_t total = 0;
 	loff_t p = 0;
 
-	wordsize = codec->driver->reg_word_size * 2;
-	regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+	wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+	regsize = codec->driver->reg_word_size * 2;
 
 	len = wordsize + regsize + 2 + 1;
 
@@ -259,8 +259,6 @@ static ssize_t codec_reg_write_file(struct file *file,
 	while (*start == ' ')
 		start++;
 	reg = simple_strtoul(start, &start, 16);
-	if ((reg >= codec->driver->reg_cache_size) || (reg % step))
-		return -EINVAL;
 	while (*start == ' ')
 		start++;
 	if (strict_strtoul(start, 16, &value))
@@ -631,6 +629,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			runtime->hw.rates |= codec_dai_drv->capture.rates;
 	}
 
+	ret = -EINVAL;
 	snd_pcm_limit_hw_rates(runtime);
 	if (!runtime->hw.rates) {
 		printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
@@ -642,7 +641,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			codec_dai->name, cpu_dai->name);
 		goto config_err;
 	}
-	if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
+	if (!runtime->hw.channels_min || !runtime->hw.channels_max ||
+	    runtime->hw.channels_min > runtime->hw.channels_max) {
 		printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
 				codec_dai->name, cpu_dai->name);
 		goto config_err;
@@ -2062,6 +2062,7 @@ const struct dev_pm_ops snd_soc_pm_ops = {
 	.resume = snd_soc_resume,
 	.poweroff = snd_soc_poweroff,
 };
+EXPORT_SYMBOL_GPL(snd_soc_pm_ops);
 
 /* ASoC platform driver */
 static struct platform_driver soc_driver = {
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index fcab80b36a37..fc017c0a7b5d 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -331,7 +331,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 			goto err;
 
 		if (gpios[i].wake) {
-			ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
+			ret = irq_set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
 			if (ret != 0)
 				printk(KERN_ERR
 				  "Failed to mark GPIO %d as wake source: %d\n",
diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
index 8585957477eb..556a57133925 100644
--- a/sound/soc/tegra/harmony.c
+++ b/sound/soc/tegra/harmony.c
@@ -370,6 +370,7 @@ static struct platform_driver tegra_snd_harmony_driver = {
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
 	},
 	.probe = tegra_snd_harmony_probe,
 	.remove = __devexit_p(tegra_snd_harmony_remove),