Merge branch 'linus' into x86/gart

30 files changed, 347 insertions, 275 deletions

diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c
index fee5f8e5895..3f326219f1e 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/at32/playpaq_wm8510.c
@@ -36,8 +36,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/arch/at32ap700x.h>
-#include <asm/arch/portmux.h>
+#include <mach/at32ap700x.h>
+#include <mach/portmux.h>
 
 #include "../codecs/wm8510.h"
 #include "at32-pcm.h"
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
index d47492b2b6e..7ab48bd25e4 100644
--- a/sound/soc/at91/at91-pcm.c
+++ b/sound/soc/at91/at91-pcm.c
@@ -28,8 +28,8 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <asm/arch/hardware.h>
-#include <asm/arch/at91_ssc.h>
+#include <mach/hardware.h>
+#include <mach/at91_ssc.h>
 
 #include "at91-pcm.h"
 
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
index 58d0f00a07b..e5aada2cb10 100644
--- a/sound/soc/at91/at91-pcm.h
+++ b/sound/soc/at91/at91-pcm.h
@@ -19,7 +19,7 @@
 #ifndef _AT91_PCM_H
 #define _AT91_PCM_H
 
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 
 struct at91_ssc_periph {
 	void __iomem	*base;
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
index 090e607f869..5d44515e62e 100644
--- a/sound/soc/at91/at91-ssc.c
+++ b/sound/soc/at91/at91-ssc.c
@@ -28,9 +28,9 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/arch/hardware.h>
-#include <asm/arch/at91_pmc.h>
-#include <asm/arch/at91_ssc.h>
+#include <mach/hardware.h>
+#include <mach/at91_pmc.h>
+#include <mach/at91_ssc.h>
 
 #include "at91-pcm.h"
 #include "at91-ssc.h"
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
index d532de95424..b081e83766b 100644
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -33,8 +33,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/gpio.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
 
 #include "../codecs/wm8731.h"
 #include "at91-pcm.h"
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index ba4b5c199f2..9384702c7eb 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -231,7 +231,7 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
 
 	/* if both TX and RX are idle, disable PSC */
 	stat = au_readl(I2S_STAT(pscdata));
-	if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) {
+	if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
 		au_writel(0, I2S_CFG(pscdata));
 		au_sync();
 		au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index e23cb09f0d1..c6a8edf302a 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -348,8 +348,9 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("ROUT1"),
 	SND_SOC_DAPM_OUTPUT("LOUT2"),
 	SND_SOC_DAPM_OUTPUT("ROUT2"),
-	SND_SOC_DAPM_OUTPUT("MONO"),
+	SND_SOC_DAPM_OUTPUT("MONO1"),
 	SND_SOC_DAPM_OUTPUT("OUT3"),
+	SND_SOC_DAPM_OUTPUT("VREF"),
 
 	SND_SOC_DAPM_INPUT("LINPUT1"),
 	SND_SOC_DAPM_INPUT("LINPUT2"),
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 9fc8edd8222..1fb7f9a7aec 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -427,20 +427,20 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"HPOUTR", NULL, "Headphone PGA"},
 	{"Headphone PGA", NULL, "Right HP Mixer"},
 
-	/* mono hp mixer */
-	{"Mono HP Mixer", NULL, "Left HP Mixer"},
-	{"Mono HP Mixer", NULL, "Right HP Mixer"},
+	/* mono mixer */
+	{"Mono Mixer", NULL, "Left HP Mixer"},
+	{"Mono Mixer", NULL, "Right HP Mixer"},
 
 	/* Out3 Mux */
 	{"Out3 Mux", "Left", "Left HP Mixer"},
 	{"Out3 Mux", "Mono", "Phone Mixer"},
-	{"Out3 Mux", "Left + Right", "Mono HP Mixer"},
+	{"Out3 Mux", "Left + Right", "Mono Mixer"},
 	{"Out 3 PGA", NULL, "Out3 Mux"},
 	{"OUT3", NULL, "Out 3 PGA"},
 
 	/* speaker Mux */
 	{"Speaker Mux", "Speaker Mix", "Speaker Mixer"},
-	{"Speaker Mux", "Headphone Mix", "Mono HP Mixer"},
+	{"Speaker Mux", "Headphone Mix", "Mono Mixer"},
 	{"Speaker PGA", NULL, "Speaker Mux"},
 	{"LOUT2", NULL, "Speaker PGA"},
 	{"ROUT2", NULL, "Speaker PGA"},
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 5e2c306399e..65fdbd81a37 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -19,9 +19,8 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <asm/mach-types.h>
 #include <asm/dma.h>
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 
 #include "../codecs/tlv320aic3x.h"
 #include "davinci-pcm.h"
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index da2bc590286..d2d3da9729f 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -132,12 +132,17 @@ struct fsl_dma_private {
  * Since each link descriptor has a 32-bit byte count field, we set
  * period_bytes_max to the largest 32-bit number.  We also have no maximum
  * number of periods.
+ *
+ * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a
+ * limitation in the SSI driver requires the sample rates for playback and
+ * capture to be the same.
  */
 static const struct snd_pcm_hardware fsl_dma_hardware = {
 
 	.info   		= SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_MMAP |
-				  SNDRV_PCM_INFO_MMAP_VALID,
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_JOINT_DUPLEX,
 	.formats		= FSLDMA_PCM_FORMATS,
 	.rates  		= FSLDMA_PCM_RATES,
 	.rate_min       	= 5512,
@@ -322,14 +327,75 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
  * fsl_dma_open: open a new substream.
  *
  * Each substream has its own DMA buffer.
+ *
+ * ALSA divides the DMA buffer into N periods.  We create NUM_DMA_LINKS link
+ * descriptors that ping-pong from one period to the next.  For example, if
+ * there are six periods and two link descriptors, this is how they look
+ * before playback starts:
+ *
+ *      	   The last link descriptor
+ *   ____________  points back to the first
+ *  |   	 |
+ *  V   	 |
+ *  ___    ___   |
+ * |   |->|   |->|
+ * |___|  |___|
+ *   |      |
+ *   |      |
+ *   V      V
+ *  _________________________________________
+ * |      |      |      |      |      |      |  The DMA buffer is
+ * |      |      |      |      |      |      |    divided into 6 parts
+ * |______|______|______|______|______|______|
+ *
+ * and here's how they look after the first period is finished playing:
+ *
+ *   ____________
+ *  |   	 |
+ *  V   	 |
+ *  ___    ___   |
+ * |   |->|   |->|
+ * |___|  |___|
+ *   |      |
+ *   |______________
+ *          |       |
+ *          V       V
+ *  _________________________________________
+ * |      |      |      |      |      |      |
+ * |      |      |      |      |      |      |
+ * |______|______|______|______|______|______|
+ *
+ * The first link descriptor now points to the third period.  The DMA
+ * controller is currently playing the second period.  When it finishes, it
+ * will jump back to the first descriptor and play the third period.
+ *
+ * There are four reasons we do this:
+ *
+ * 1. The only way to get the DMA controller to automatically restart the
+ *    transfer when it gets to the end of the buffer is to use chaining
+ *    mode.  Basic direct mode doesn't offer that feature.
+ * 2. We need to receive an interrupt at the end of every period.  The DMA
+ *    controller can generate an interrupt at the end of every link transfer
+ *    (aka segment).  Making each period into a DMA segment will give us the
+ *    interrupts we need.
+ * 3. By creating only two link descriptors, regardless of the number of
+ *    periods, we do not need to reallocate the link descriptors if the
+ *    number of periods changes.
+ * 4. All of the audio data is still stored in a single, contiguous DMA
+ *    buffer, which is what ALSA expects.  We're just dividing it into
+ *    contiguous parts, and creating a link descriptor for each one.
  */
 static int fsl_dma_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_dma_private *dma_private;
+	struct ccsr_dma_channel __iomem *dma_channel;
 	dma_addr_t ld_buf_phys;
+	u64 temp_link;  	/* Pointer to next link descriptor */
+	u32 mr;
 	unsigned int channel;
 	int ret = 0;
+	unsigned int i;
 
 	/*
 	 * Reject any DMA buffer whose size is not a multiple of the period
@@ -390,68 +456,74 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
 	snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware);
 	runtime->private_data = dma_private;
 
+	/* Program the fixed DMA controller parameters */
+
+	dma_channel = dma_private->dma_channel;
+
+	temp_link = dma_private->ld_buf_phys +
+		sizeof(struct fsl_dma_link_descriptor);
+
+	for (i = 0; i < NUM_DMA_LINKS; i++) {
+		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
+
+		link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+		link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+		link->next = cpu_to_be64(temp_link);
+
+		temp_link += sizeof(struct fsl_dma_link_descriptor);
+	}
+	/* The last link descriptor points to the first */
+	dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
+
+	/* Tell the DMA controller where the first link descriptor is */
+	out_be32(&dma_channel->clndar,
+		CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
+	out_be32(&dma_channel->eclndar,
+		CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
+
+	/* The manual says the BCR must be clear before enabling EMP */
+	out_be32(&dma_channel->bcr, 0);
+
+	/*
+	 * Program the mode register for interrupts, external master control,
+	 * and source/destination hold.  Also clear the Channel Abort bit.
+	 */
+	mr = in_be32(&dma_channel->mr) &
+		~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
+
+	/*
+	 * We want External Master Start and External Master Pause enabled,
+	 * because the SSI is controlling the DMA controller.  We want the DMA
+	 * controller to be set up in advance, and then we signal only the SSI
+	 * to start transferring.
+	 *
+	 * We want End-Of-Segment Interrupts enabled, because this will generate
+	 * an interrupt at the end of each segment (each link descriptor
+	 * represents one segment).  Each DMA segment is the same thing as an
+	 * ALSA period, so this is how we get an interrupt at the end of every
+	 * period.
+	 *
+	 * We want Error Interrupt enabled, so that we can get an error if
+	 * the DMA controller is mis-programmed somehow.
+	 */
+	mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
+		CCSR_DMA_MR_EMS_EN;
+
+	/* For playback, we want the destination address to be held.  For
+	   capture, set the source address to be held. */
+	mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
+
+	out_be32(&dma_channel->mr, mr);
+
 	return 0;
 }
 
 /**
- * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors.
+ * fsl_dma_hw_params: continue initializing the DMA links
  *
- * ALSA divides the DMA buffer into N periods.  We create NUM_DMA_LINKS link
- * descriptors that ping-pong from one period to the next.  For example, if
- * there are six periods and two link descriptors, this is how they look
- * before playback starts:
- *
- *      	   The last link descriptor
- *   ____________  points back to the first
- *  |   	 |
- *  V   	 |
- *  ___    ___   |
- * |   |->|   |->|
- * |___|  |___|
- *   |      |
- *   |      |
- *   V      V
- *  _________________________________________
- * |      |      |      |      |      |      |  The DMA buffer is
- * |      |      |      |      |      |      |    divided into 6 parts
- * |______|______|______|______|______|______|
- *
- * and here's how they look after the first period is finished playing:
- *
- *   ____________
- *  |   	 |
- *  V   	 |
- *  ___    ___   |
- * |   |->|   |->|
- * |___|  |___|
- *   |      |
- *   |______________
- *          |       |
- *          V       V
- *  _________________________________________
- * |      |      |      |      |      |      |
- * |      |      |      |      |      |      |
- * |______|______|______|______|______|______|
- *
- * The first link descriptor now points to the third period.  The DMA
- * controller is currently playing the second period.  When it finishes, it
- * will jump back to the first descriptor and play the third period.
- *
- * There are four reasons we do this:
- *
- * 1. The only way to get the DMA controller to automatically restart the
- *    transfer when it gets to the end of the buffer is to use chaining
- *    mode.  Basic direct mode doesn't offer that feature.
- * 2. We need to receive an interrupt at the end of every period.  The DMA
- *    controller can generate an interrupt at the end of every link transfer
- *    (aka segment).  Making each period into a DMA segment will give us the
- *    interrupts we need.
- * 3. By creating only two link descriptors, regardless of the number of
- *    periods, we do not need to reallocate the link descriptors if the
- *    number of periods changes.
- * 4. All of the audio data is still stored in a single, contiguous DMA
- *    buffer, which is what ALSA expects.  We're just dividing it into
- *    contiguous parts, and creating a link descriptor for each one.
+ * This function obtains hardware parameters about the opened stream and
+ * programs the DMA controller accordingly.
  *
  * Note that due to a quirk of the SSI's STX register, the target address
  * for the DMA operations depends on the sample size.  So we don't program
@@ -463,11 +535,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_dma_private *dma_private = runtime->private_data;
-	struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
 
 	dma_addr_t temp_addr;   /* Pointer to next period */
-	u64 temp_link;  	/* Pointer to next link descriptor */
-	u32 mr; 		/* Temporary variable for MR register */
 
 	unsigned int i;
 
@@ -485,8 +554,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 		dma_private->dma_buf_next = dma_private->dma_buf_phys;
 
 	/*
-	 * Initialize each link descriptor.
-	 *
 	 * The actual address in STX0 (destination for playback, source for
 	 * capture) is based on the sample size, but we don't know the sample
 	 * size in this function, so we'll have to adjust that later.  See
@@ -502,16 +569,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 	 * buffer itself.
 	 */
 	temp_addr = substream->dma_buffer.addr;
-	temp_link = dma_private->ld_buf_phys +
-		sizeof(struct fsl_dma_link_descriptor);
 
 	for (i = 0; i < NUM_DMA_LINKS; i++) {
 		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
 
 		link->count = cpu_to_be32(period_size);
-		link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-		link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-		link->next = cpu_to_be64(temp_link);
 
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			link->source_addr = cpu_to_be32(temp_addr);
@@ -519,51 +581,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 			link->dest_addr = cpu_to_be32(temp_addr);
 
 		temp_addr += period_size;
-		temp_link += sizeof(struct fsl_dma_link_descriptor);
 	}
-	/* The last link descriptor points to the first */
-	dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys);
-
-	/* Tell the DMA controller where the first link descriptor is */
-	out_be32(&dma_channel->clndar,
-		CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys));
-	out_be32(&dma_channel->eclndar,
-		CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys));
-
-	/* The manual says the BCR must be clear before enabling EMP */
-	out_be32(&dma_channel->bcr, 0);
-
-	/*
-	 * Program the mode register for interrupts, external master control,
-	 * and source/destination hold.  Also clear the Channel Abort bit.
-	 */
-	mr = in_be32(&dma_channel->mr) &
-		~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE);
-
-	/*
-	 * We want External Master Start and External Master Pause enabled,
-	 * because the SSI is controlling the DMA controller.  We want the DMA
-	 * controller to be set up in advance, and then we signal only the SSI
-	 * to start transfering.
-	 *
-	 * We want End-Of-Segment Interrupts enabled, because this will generate
-	 * an interrupt at the end of each segment (each link descriptor
-	 * represents one segment).  Each DMA segment is the same thing as an
-	 * ALSA period, so this is how we get an interrupt at the end of every
-	 * period.
-	 *
-	 * We want Error Interrupt enabled, so that we can get an error if
-	 * the DMA controller is mis-programmed somehow.
-	 */
-	mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN |
-		CCSR_DMA_MR_EMS_EN;
-
-	/* For playback, we want the destination address to be held.  For
-	   capture, set the source address to be held. */
-	mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-		CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
-
-	out_be32(&dma_channel->mr, mr);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 71bff33f552..157a7895ffa 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -67,6 +67,8 @@
  * @ssi: pointer to the SSI's registers
  * @ssi_phys: physical address of the SSI registers
  * @irq: IRQ of this SSI
+ * @first_stream: pointer to the stream that was opened first
+ * @second_stream: pointer to second stream
  * @dev: struct device pointer
  * @playback: the number of playback streams opened
  * @capture: the number of capture streams opened
@@ -79,6 +81,8 @@ struct fsl_ssi_private {
 	struct ccsr_ssi __iomem *ssi;
 	dma_addr_t ssi_phys;
 	unsigned int irq;
+	struct snd_pcm_substream *first_stream;
+	struct snd_pcm_substream *second_stream;
 	struct device *dev;
 	unsigned int playback;
 	unsigned int capture;
@@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
 		 */
 	}
 
+	if (!ssi_private->first_stream)
+		ssi_private->first_stream = substream;
+	else {
+		/* This is the second stream open, so we need to impose sample
+		 * rate and maybe sample size constraints.  Note that this can
+		 * cause a race condition if the second stream is opened before
+		 * the first stream is fully initialized.
+		 *
+		 * We provide some protection by checking to make sure the first
+		 * stream is initialized, but it's not perfect.  ALSA sometimes
+		 * re-initializes the driver with a different sample rate or
+		 * size.  If the second stream is opened before the first stream
+		 * has received its final parameters, then the second stream may
+		 * be constrained to the wrong sample rate or size.
+		 *
+		 * FIXME: This code does not handle opening and closing streams
+		 * repeatedly.  If you open two streams and then close the first
+		 * one, you may not be able to open another stream until you
+		 * close the second one as well.
+		 */
+		struct snd_pcm_runtime *first_runtime =
+			ssi_private->first_stream->runtime;
+
+		if (!first_runtime->rate || !first_runtime->sample_bits) {
+			dev_err(substream->pcm->card->dev,
+				"set sample rate and size in %s stream first\n",
+				substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+				? "capture" : "playback");
+			return -EAGAIN;
+		}
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE,
+			first_runtime->rate, first_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+			first_runtime->sample_bits,
+			first_runtime->sample_bits);
+
+		ssi_private->second_stream = substream;
+	}
+
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		ssi_private->playback++;
 
@@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
 
 	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-	u32 wl;
 
-	wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
+	if (substream == ssi_private->first_stream) {
+		u32 wl;
 
-	clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+		/* The SSI should always be disabled at this points (SSIEN=0) */
+		wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		/* In synchronous mode, the SSI uses STCCR for capture */
 		clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-	else
-		clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
-
-	setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+	}
 
 	return 0;
 }
@@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			setbits32(&ssi->scr, CCSR_SSI_SCR_TE);
+			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+			setbits32(&ssi->scr,
+				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
 		} else {
-			setbits32(&ssi->scr, CCSR_SSI_SCR_RE);
+			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+			setbits32(&ssi->scr,
+				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
 
 			/*
 			 * I think we need this delay to allow time for the SSI
@@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
 		ssi_private->capture--;
 
+	if (ssi_private->first_stream == substream)
+		ssi_private->first_stream = ssi_private->second_stream;
+
+	ssi_private->second_stream = NULL;
+
 	/*
 	 * If this is the last active substream, disable the SSI and release
 	 * the IRQ.
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 02cec96859b..7694621ec40 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -29,9 +29,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/hardware.h>
+#include <mach/hardware.h>
 #include <linux/gpio.h>
-#include <asm/arch/mcbsp.h>
+#include <mach/mcbsp.h>
 
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 00b0c9d73cd..35310e16d7f 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -30,9 +30,9 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/arch/control.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/mcbsp.h>
+#include <mach/control.h>
+#include <mach/dma.h>
+#include <mach/mcbsp.h>
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index e092f3d836d..690bfeaec4a 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -27,7 +27,7 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 #include "omap-pcm.h"
 
 static const struct snd_pcm_hardware omap_pcm_hardware = {
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 12f6ac99b04..9212c37a33b 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -48,6 +48,7 @@ config SND_PXA2XX_SOC_POODLE
 config SND_PXA2XX_SOC_TOSA
 	tristate "SoC AC97 Audio support for Tosa"
 	depends on SND_PXA2XX_SOC && MACH_TOSA
+	depends on MFD_TC6393XB
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index c0294464a23..0a53f72077f 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -25,10 +25,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/corgi.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/corgi.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm8731.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 06e8afb2527..6781c5be242 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -21,9 +21,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index 02dcac39cdf..d9c3f7b28be 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -30,9 +30,9 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 65a4e9a8c39..a4697f7e292 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -26,10 +26,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/locomo.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/poodle.h>
-#include <asm/arch/audio.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/poodle.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm8731.h"
 #include "pxa2xx-pcm.h"
@@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream)
 }
 
 /* we need to unmute the HP at shutdown as the mute burns power on poodle */
-static int poodle_shutdown(struct snd_pcm_substream *substream)
+static void poodle_shutdown(struct snd_pcm_substream *substream)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
-
 	/* set = unmute headphone */
 	locomo_gpio_write(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_MUTE_L, 1);
 	locomo_gpio_write(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_MUTE_R, 1);
-	return 0;
 }
 
 static int poodle_hw_params(struct snd_pcm_substream *substream,
@@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = {
 	SOC_ENUM_SINGLE_EXT(2, spk_function),
 };
 
-static const snd_kcontrol_new_t wm8731_poodle_controls[] = {
+static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
 	SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
 		poodle_set_jack),
 	SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 059af815ea0..d94a495bd6b 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -26,10 +26,10 @@
 
 #include <asm/irq.h>
 #include <linux/mutex.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 8f96d87f7b4..8548818eea0 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -21,10 +21,10 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/pxa2xx-gpio.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/pxa2xx-gpio.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 2df03ee5819..4345f387fe4 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -22,9 +22,9 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 64385797da5..37cb768fc93 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -26,10 +26,10 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/akita.h>
-#include <asm/arch/spitz.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/akita.h>
+#include <mach/spitz.h>
 #include "../codecs/wm8750.h"
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
@@ -297,7 +297,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_disable_pin(codec, "LINPUT3");
 	snd_soc_dapm_disable_pin(codec, "RINPUT3");
 	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "MONO");
+	snd_soc_dapm_disable_pin(codec, "MONO1");
 
 	/* Add spitz specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index b6edb61a3a3..2baaa750f12 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -21,6 +21,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/device.h>
+#include <linux/gpio.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -28,11 +29,10 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <asm/hardware/tmio.h>
-#include <asm/arch/pxa-regs.h>
-#include <asm/arch/hardware.h>
-#include <asm/arch/audio.h>
-#include <asm/arch/tosa.h>
+#include <mach/tosa.h>
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
@@ -137,10 +137,7 @@ static int tosa_set_spk(struct snd_kcontrol *kcontrol,
 static int tosa_hp_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *k, int event)
 {
-	if (SND_SOC_DAPM_EVENT_ON(event))
-		set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
-	else
-		reset_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
+	gpio_set_value(TOSA_GPIO_L_MUTE, SND_SOC_DAPM_EVENT_ON(event) ? 1 :0);
 	return 0;
 }
 
@@ -254,16 +251,28 @@ static int __init tosa_init(void)
 	if (!machine_is_tosa())
 		return -ENODEV;
 
+	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
+	if (ret)
+		return ret;
+	gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
+
 	tosa_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!tosa_snd_device)
-		return -ENOMEM;
+	if (!tosa_snd_device) {
+		ret = -ENOMEM;
+		goto err_alloc;
+	}
 
 	platform_set_drvdata(tosa_snd_device, &tosa_snd_devdata);
 	tosa_snd_devdata.dev = &tosa_snd_device->dev;
 	ret = platform_device_add(tosa_snd_device);
 
-	if (ret)
-		platform_device_put(tosa_snd_device);
+	if (!ret)
+		return 0;
+
+	platform_device_put(tosa_snd_device);
+
+err_alloc:
+	gpio_free(TOSA_GPIO_L_MUTE);
 
 	return ret;
 }
@@ -271,6 +280,7 @@ static int __init tosa_init(void)
 static void __exit tosa_exit(void)
 {
 	platform_device_unregister(tosa_snd_device);
+	gpio_free(TOSA_GPIO_L_MUTE);
 }
 
 module_init(tosa_init);
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 4d7a9aa15f1..8089f8ee05c 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -24,14 +24,13 @@
 #include <sound/soc-dapm.h>
 #include <sound/tlv.h>
 
-#include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/hardware.h>
-#include <asm/arch/audio.h>
+#include <mach/regs-clock.h>
+#include <mach/regs-gpio.h>
+#include <mach/hardware.h>
+#include <mach/audio.h>
 #include <linux/io.h>
-#include <asm/arch/spi-gpio.h>
+#include <mach/spi-gpio.h>
 
 #include <asm/plat-s3c24xx/regs-iis.h>
 
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ee4676ed128..ded7d995a92 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -28,16 +28,16 @@
 #include <sound/pcm_params.h>
 #include <sound/initval.h>
 #include <sound/soc.h>
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 
 #include <linux/io.h>
 #include <asm/dma.h>
 
 #include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
 
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/audio.h>
-#include <asm/arch/dma.h>
+#include <mach/regs-gpio.h>
+#include <mach/audio.h>
+#include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c2412-i2s.h"
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 783349b7fed..19c5c3cf5d8 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -27,13 +27,13 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
+#include <mach/hardware.h>
 #include <asm/plat-s3c/regs-ac97.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/audio.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
 #include <asm/dma.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 397524282b5..ba4476b55fb 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -27,12 +27,12 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include <asm/hardware.h>
-#include <asm/arch/regs-gpio.h>
-#include <asm/arch/regs-clock.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/regs-gpio.h>
+#include <mach/regs-clock.h>
+#include <mach/audio.h>
 #include <asm/dma.h>
-#include <asm/arch/dma.h>
+#include <mach/dma.h>
 
 #include <asm/plat-s3c24xx/regs-iis.h>
 
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index cef79b34dc6..e13e614bada 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -27,9 +27,9 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
-#include <asm/hardware.h>
-#include <asm/arch/dma.h>
-#include <asm/arch/audio.h>
+#include <mach/hardware.h>
+#include <mach/dma.h>
+#include <mach/audio.h>
 
 #include "s3c24xx-pcm.h"
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 2c87061c2a6..f9d100bc847 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -470,6 +470,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(dapm_reg_event);
 
 /*
  * Scan each dapm widget for complete audio path.
@@ -523,24 +524,6 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 				continue;
 			}
 
-			/* programmable gain/attenuation */
-			if (w->id == snd_soc_dapm_pga) {
-				int on;
-				in = is_connected_input_ep(w);
-				dapm_clear_walk(w->codec);
-				out = is_connected_output_ep(w);
-				dapm_clear_walk(w->codec);
-				w->power = on = (out != 0 && in != 0) ? 1 : 0;
-
-				if (!on)
-					dapm_set_pga(w, on); /* lower volume to reduce pops */
-				dapm_update_bits(w);
-				if (on)
-					dapm_set_pga(w, on); /* restore volume from zero */
-
-				continue;
-			}
-
 			/* pre and post event widgets */
 			if (w->id == snd_soc_dapm_pre) {
 				if (!w->event)
@@ -586,45 +569,56 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 			power_change = (w->power == power) ? 0: 1;
 			w->power = power;
 
+			if (!power_change)
+				continue;
+
 			/* call any power change event handlers */
-			if (power_change) {
-				if (w->event) {
-					pr_debug("power %s event for %s flags %x\n",
-						 w->power ? "on" : "off", w->name, w->event_flags);
-					if (power) {
-						/* power up event */
-						if (w->event_flags & SND_SOC_DAPM_PRE_PMU) {
-							ret = w->event(w,
-								NULL, SND_SOC_DAPM_PRE_PMU);
-							if (ret < 0)
-								return ret;
-						}
-						dapm_update_bits(w);
-						if (w->event_flags & SND_SOC_DAPM_POST_PMU){
-							ret = w->event(w,
-								NULL, SND_SOC_DAPM_POST_PMU);
-							if (ret < 0)
-								return ret;
-						}
-					} else {
-						/* power down event */
-						if (w->event_flags & SND_SOC_DAPM_PRE_PMD) {
-							ret = w->event(w,
-								NULL, SND_SOC_DAPM_PRE_PMD);
-							if (ret < 0)
-								return ret;
-						}
-						dapm_update_bits(w);
-						if (w->event_flags & SND_SOC_DAPM_POST_PMD) {
-							ret = w->event(w,
-								NULL, SND_SOC_DAPM_POST_PMD);
-							if (ret < 0)
-								return ret;
-						}
-					}
-				} else
-					/* no event handler */
-					dapm_update_bits(w);
+			if (w->event)
+				pr_debug("power %s event for %s flags %x\n",
+					 w->power ? "on" : "off",
+					 w->name, w->event_flags);
+
+			/* power up pre event */
+			if (power && w->event &&
+			    (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
+				ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
+				if (ret < 0)
+					return ret;
+			}
+
+			/* power down pre event */
+			if (!power && w->event &&
+			    (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
+				ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
+				if (ret < 0)
+					return ret;
+			}
+
+			/* Lower PGA volume to reduce pops */
+			if (w->id == snd_soc_dapm_pga && !power)
+				dapm_set_pga(w, power);
+
+			dapm_update_bits(w);
+
+			/* Raise PGA volume to reduce pops */
+			if (w->id == snd_soc_dapm_pga && power)
+				dapm_set_pga(w, power);
+
+			/* power up post event */
+			if (power && w->event &&
+			    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
+				ret = w->event(w,
+					       NULL, SND_SOC_DAPM_POST_PMU);
+				if (ret < 0)
+					return ret;
+			}
+
+			/* power down post event */
+			if (!power && w->event &&
+			    (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
+				ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
+				if (ret < 0)
+					return ret;
 			}
 		}
 	}