26 files changed, 125 insertions, 66 deletions

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" },