Merge branch 'linus'

Conflicts:

	drivers/scsi/ipr.c

Signed-off-by: James Bottomley <James.Bottomley@SteelEye.com>

33 files changed, 10211 insertions, 0 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
new file mode 100644
index 00000000000..ec821a57f84
--- /dev/null
+++ b/sound/soc/Kconfig
@@ -0,0 +1,32 @@
+#
+# SoC audio configuration
+#
+
+menu "SoC audio support"
+	depends on SND!=n
+
+config SND_SOC_AC97_BUS
+	bool
+
+config SND_SOC
+	tristate "SoC audio support"
+	---help---
+
+	  If you want SoC support, you should say Y here and also to the
+	  specific driver for your SoC below. You will also need to select the
+	  specific codec(s) attached to the SoC
+
+	  This SoC audio support can also be built as a module.  If so, the module
+	  will be called snd-soc-core.
+
+# All the supported Soc's
+menu "SoC Platforms"
+depends on SND_SOC
+source "sound/soc/at91/Kconfig"
+source "sound/soc/pxa/Kconfig"
+endmenu
+
+# Supported codecs
+source "sound/soc/codecs/Kconfig"
+
+endmenu
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
new file mode 100644
index 00000000000..98e6f49dafc
--- /dev/null
+++ b/sound/soc/Makefile
@@ -0,0 +1,4 @@
+snd-soc-core-objs := soc-core.o soc-dapm.o
+
+obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
+obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
new file mode 100644
index 00000000000..5bcf08b728b
--- /dev/null
+++ b/sound/soc/at91/Kconfig
@@ -0,0 +1,32 @@
+menu "SoC Audio for the Atmel AT91"
+
+config SND_AT91_SOC
+	tristate "SoC Audio for the Atmel AT91 System-on-Chip"
+	depends on ARCH_AT91 && SND
+	select SND_PCM
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the AT91 SSC interface. You will also need
+	  to select the audio interfaces to support below.
+
+config SND_AT91_SOC_I2S
+	tristate
+
+config SND_AT91_SOC_ETI_B1_WM8731
+	tristate "SoC I2S Audio support for WM8731-based Endrelia ETI-B1 boards"
+	depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
+	select SND_AT91_SOC_I2S
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on WM8731-based
+	  Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.
+
+config SND_AT91_SOC_ETI_SLAVE
+	bool "Run codec in slave Mode on Endrelia boards"
+	depends on SND_AT91_SOC_ETI_B1_WM8731
+	default n
+	help
+	  Say Y if you want to run with the AT91 SSC generating the BCLK
+	  and LRC signals on Endrelia boards.
+
+endmenu
diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile
new file mode 100644
index 00000000000..b77b01ab202
--- /dev/null
+++ b/sound/soc/at91/Makefile
@@ -0,0 +1,11 @@
+# AT91 Platform Support
+snd-soc-at91-objs := at91-pcm.o
+snd-soc-at91-i2s-objs := at91-i2s.o
+
+obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
+obj-$(CONFIG_SND_AT91_SOC_I2S) += snd-soc-at91-i2s.o
+
+# AT91 Machine Support
+snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
+
+obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o
diff --git a/sound/soc/at91/at91-i2s.c b/sound/soc/at91/at91-i2s.c
new file mode 100644
index 00000000000..fcc544a96ba
--- /dev/null
+++ b/sound/soc/at91/at91-i2s.c
@@ -0,0 +1,720 @@
+/*
+ * at91-i2s.c  --  ALSA SoC I2S Audio Layer Platform driver
+ *
+ * Author: Frank Mandarino <fmandarino@endrelia.com>
+ *         Endrelia Technologies Inc.
+ *
+ * Based on pxa2xx Platform drivers by
+ * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#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 <asm/arch/at91_pdc.h>
+
+#include "at91-pcm.h"
+#include "at91-i2s.h"
+
+#if 0
+#define	DBG(x...)	printk(KERN_DEBUG "at91-i2s:" x)
+#else
+#define	DBG(x...)
+#endif
+
+#if defined(CONFIG_ARCH_AT91SAM9260)
+#define NUM_SSC_DEVICES		1
+#else
+#define NUM_SSC_DEVICES		3
+#endif
+
+
+/*
+ * SSC PDC registers required by the PCM DMA engine.
+ */
+static struct at91_pdc_regs pdc_tx_reg = {
+	.xpr		= AT91_PDC_TPR,
+	.xcr		= AT91_PDC_TCR,
+	.xnpr		= AT91_PDC_TNPR,
+	.xncr		= AT91_PDC_TNCR,
+};
+
+static struct at91_pdc_regs pdc_rx_reg = {
+	.xpr		= AT91_PDC_RPR,
+	.xcr		= AT91_PDC_RCR,
+	.xnpr		= AT91_PDC_RNPR,
+	.xncr		= AT91_PDC_RNCR,
+};
+
+/*
+ * SSC & PDC status bits for transmit and receive.
+ */
+static struct at91_ssc_mask ssc_tx_mask = {
+	.ssc_enable	= AT91_SSC_TXEN,
+	.ssc_disable	= AT91_SSC_TXDIS,
+	.ssc_endx	= AT91_SSC_ENDTX,
+	.ssc_endbuf	= AT91_SSC_TXBUFE,
+	.pdc_enable	= AT91_PDC_TXTEN,
+	.pdc_disable	= AT91_PDC_TXTDIS,
+};
+
+static struct at91_ssc_mask ssc_rx_mask = {
+	.ssc_enable	= AT91_SSC_RXEN,
+	.ssc_disable	= AT91_SSC_RXDIS,
+	.ssc_endx	= AT91_SSC_ENDRX,
+	.ssc_endbuf	= AT91_SSC_RXBUFF,
+	.pdc_enable	= AT91_PDC_RXTEN,
+	.pdc_disable	= AT91_PDC_RXTDIS,
+};
+
+
+/*
+ * DMA parameters.
+ */
+static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
+	{{
+	.name		= "SSC0/I2S PCM Stereo out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC0/I2S PCM Stereo in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	}},
+#if NUM_SSC_DEVICES == 3
+	{{
+	.name		= "SSC1/I2S PCM Stereo out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC1/I2S PCM Stereo in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	}},
+	{{
+	.name		= "SSC2/I2S PCM Stereo out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC1/I2S PCM Stereo in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	}},
+#endif
+};
+
+struct at91_ssc_state {
+	u32	ssc_cmr;
+	u32	ssc_rcmr;
+	u32	ssc_rfmr;
+	u32	ssc_tcmr;
+	u32	ssc_tfmr;
+	u32	ssc_sr;
+	u32	ssc_imr;
+};
+
+static struct at91_ssc_info {
+	char		*name;
+	struct at91_ssc_periph ssc;
+	spinlock_t 	lock;		/* lock for dir_mask */
+	unsigned short	dir_mask;	/* 0=unused, 1=playback, 2=capture */
+	unsigned short	initialized;	/* 1=SSC has been initialized */
+	unsigned short	daifmt;
+	unsigned short	cmr_div;
+	unsigned short	tcmr_period;
+	unsigned short	rcmr_period;
+	struct at91_pcm_dma_params *dma_params[2];
+	struct at91_ssc_state ssc_state;
+
+} ssc_info[NUM_SSC_DEVICES] = {
+	{
+	.name		= "ssc0",
+	.lock		= SPIN_LOCK_UNLOCKED,
+	.dir_mask	= 0,
+	.initialized	= 0,
+	},
+#if NUM_SSC_DEVICES == 3
+	{
+	.name		= "ssc1",
+	.lock		= SPIN_LOCK_UNLOCKED,
+	.dir_mask	= 0,
+	.initialized	= 0,
+	},
+	{
+	.name		= "ssc2",
+	.lock		= SPIN_LOCK_UNLOCKED,
+	.dir_mask	= 0,
+	.initialized	= 0,
+	},
+#endif
+};
+
+static unsigned int at91_i2s_sysclk;
+
+/*
+ * SSC interrupt handler.  Passes PDC interrupts to the DMA
+ * interrupt handler in the PCM driver.
+ */
+static irqreturn_t at91_i2s_interrupt(int irq, void *dev_id)
+{
+	struct at91_ssc_info *ssc_p = dev_id;
+	struct at91_pcm_dma_params *dma_params;
+	u32 ssc_sr;
+	int i;
+
+	ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)
+			& at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
+
+	/*
+	 * Loop through the substreams attached to this SSC.  If
+	 * a DMA-related interrupt occurred on that substream, call
+	 * the DMA interrupt handler function, if one has been
+	 * registered in the dma_params structure by the PCM driver.
+	 */
+	for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) {
+		dma_params = ssc_p->dma_params[i];
+
+		if (dma_params != NULL && dma_params->dma_intr_handler != NULL &&
+			(ssc_sr &
+			(dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf)))
+
+			dma_params->dma_intr_handler(ssc_sr, dma_params->substream);
+	}
+
+	return IRQ_HANDLED;
+}
+
+/*
+ * Startup.  Only that one substream allowed in each direction.
+ */
+static int at91_i2s_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	int dir_mask;
+
+	DBG("i2s_startup: SSC_SR=0x%08lx\n",
+			at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
+	dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2;
+
+	spin_lock_irq(&ssc_p->lock);
+	if (ssc_p->dir_mask & dir_mask) {
+		spin_unlock_irq(&ssc_p->lock);
+		return -EBUSY;
+	}
+	ssc_p->dir_mask |= dir_mask;
+	spin_unlock_irq(&ssc_p->lock);
+
+	return 0;
+}
+
+/*
+ * Shutdown.  Clear DMA parameters and shutdown the SSC if there
+ * are no other substreams open.
+ */
+static void at91_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	struct at91_pcm_dma_params *dma_params;
+	int dir, dir_mask;
+
+	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+	dma_params = ssc_p->dma_params[dir];
+
+	if (dma_params != NULL) {
+		at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
+				dma_params->mask->ssc_disable);
+		DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"),
+			at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR));
+
+		dma_params->ssc_base = NULL;
+		dma_params->substream = NULL;
+		ssc_p->dma_params[dir] = NULL;
+	}
+
+	dir_mask = 1 << dir;
+
+	spin_lock_irq(&ssc_p->lock);
+	ssc_p->dir_mask &= ~dir_mask;
+	if (!ssc_p->dir_mask) {
+		/* Shutdown the SSC clock. */
+		DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
+		at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid);
+
+		if (ssc_p->initialized) {
+			free_irq(ssc_p->ssc.pid, ssc_p);
+			ssc_p->initialized = 0;
+		}
+
+		/* Reset the SSC */
+		at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
+
+		/* Clear the SSC dividers */
+		ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0;
+	}
+	spin_unlock_irq(&ssc_p->lock);
+}
+
+/*
+ * Record the SSC system clock rate.
+ */
+static int at91_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	/*
+	 * The only clock supplied to the SSC is the AT91 master clock,
+	 * which is only used if the SSC is generating BCLK and/or
+	 * LRC clocks.
+	 */
+	switch (clk_id) {
+	case AT91_SYSCLK_MCK:
+		at91_i2s_sysclk = freq;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Record the DAI format for use in hw_params().
+ */
+static int at91_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S)
+		return -EINVAL;
+
+	ssc_p->daifmt = fmt;
+	return 0;
+}
+
+/*
+ * Record SSC clock dividers for use in hw_params().
+ */
+static int at91_i2s_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+	int div_id, int div)
+{
+	struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+	switch (div_id) {
+	case AT91SSC_CMR_DIV:
+		/*
+		 * The same master clock divider is used for both
+		 * transmit and receive, so if a value has already
+		 * been set, it must match this value.
+		 */
+		if (ssc_p->cmr_div == 0)
+			ssc_p->cmr_div = div;
+		else
+			if (div != ssc_p->cmr_div)
+				return -EBUSY;
+		break;
+
+	case AT91SSC_TCMR_PERIOD:
+		ssc_p->tcmr_period = div;
+		break;
+
+	case AT91SSC_RCMR_PERIOD:
+		ssc_p->rcmr_period = div;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Configure the SSC.
+ */
+static int at91_i2s_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int id = rtd->dai->cpu_dai->id;
+	struct at91_ssc_info *ssc_p = &ssc_info[id];
+	struct at91_pcm_dma_params *dma_params;
+	int dir, channels, bits;
+	u32 tfmr, rfmr, tcmr, rcmr;
+	int start_event;
+	int ret;
+
+	/*
+	 * Currently, there is only one set of dma params for
+	 * each direction.  If more are added, this code will
+	 * have to be changed to select the proper set.
+	 */
+	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+
+	dma_params = &ssc_dma_params[id][dir];
+	dma_params->ssc_base = ssc_p->ssc.base;
+	dma_params->substream = substream;
+
+	ssc_p->dma_params[dir] = dma_params;
+
+	/*
+	 * The cpu_dai->dma_data field is only used to communicate the
+	 * appropriate DMA parameters to the pcm driver hw_params()
+	 * function.  It should not be used for other purposes
+	 * as it is common to all substreams.
+	 */
+	rtd->dai->cpu_dai->dma_data = dma_params;
+
+	channels = params_channels(params);
+
+	/*
+	 * The SSC only supports up to 16-bit samples in I2S format, due
+	 * to the size of the Frame Mode Register FSLEN field.  Also, I2S
+	 * implies signed data.
+	 */
+	bits = 16;
+	dma_params->pdc_xfer_size = 2;
+
+	/*
+	 * Compute SSC register settings.
+	 */
+	switch (ssc_p->daifmt) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		/*
+		 * SSC provides BCLK and LRC clocks.
+		 *
+		 * The SSC transmit and receive clocks are generated from the
+		 * MCK divider, and the BCLK signal is output on the SSC TK line.
+		 */
+		rcmr =	  (( ssc_p->rcmr_period		<< 24) & AT91_SSC_PERIOD)
+			| (( 1				<< 16) & AT91_SSC_STTDLY)
+			| (( AT91_SSC_START_FALLING_RF	     ) & AT91_SSC_START)
+			| (( AT91_SSC_CK_RISING		     ) & AT91_SSC_CKI)
+			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
+			| (( AT91_SSC_CKS_DIV		     ) & AT91_SSC_CKS);
+
+		rfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
+			| (( AT91_SSC_FSOS_NEGATIVE	     ) & AT91_SSC_FSOS)
+			| (((bits - 1)			<< 16) & AT91_SSC_FSLEN)
+			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
+			| (( 1				<<  7) & AT91_SSC_MSBF)
+			| (( 0				<<  5) & AT91_SSC_LOOP)
+			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
+
+		tcmr =	  (( ssc_p->tcmr_period		<< 24) & AT91_SSC_PERIOD)
+			| (( 1				<< 16) & AT91_SSC_STTDLY)
+			| (( AT91_SSC_START_FALLING_RF       ) & AT91_SSC_START)
+			| (( AT91_SSC_CKI_FALLING	     ) & AT91_SSC_CKI)
+			| (( AT91_SSC_CKO_CONTINUOUS	     ) & AT91_SSC_CKO)
+			| (( AT91_SSC_CKS_DIV		     ) & AT91_SSC_CKS);
+
+		tfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
+			| (( 0				<< 23) & AT91_SSC_FSDEN)
+			| (( AT91_SSC_FSOS_NEGATIVE	     ) & AT91_SSC_FSOS)
+			| (((bits - 1)			<< 16) & AT91_SSC_FSLEN)
+			| (((channels - 1)		<<  8) & AT91_SSC_DATNB)
+			| (( 1				<<  7) & AT91_SSC_MSBF)
+			| (( 0				<<  5) & AT91_SSC_DATDEF)
+			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFM:
+
+		/*
+		 * CODEC supplies BCLK and LRC clocks.
+		 *
+		 * The SSC transmit clock is obtained from the BCLK signal on
+		 * on the TK line, and the SSC receive clock is generated from the
+		 * transmit clock.
+		 *
+		 * For single channel data, one sample is transferred on the falling
+		 * edge of the LRC clock.  For two channel data, one sample is
+		 * transferred on both edges of the LRC clock.
+		 */
+		start_event = channels == 1
+				? AT91_SSC_START_FALLING_RF
+				: AT91_SSC_START_EDGE_RF;
+
+		rcmr =	  (( 0				<< 24) & AT91_SSC_PERIOD)
+			| (( 1				<< 16) & AT91_SSC_STTDLY)
+			| (( start_event		     ) & AT91_SSC_START)
+			| (( AT91_SSC_CK_RISING		     ) & AT91_SSC_CKI)
+			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
+			| (( AT91_SSC_CKS_CLOCK		     ) & AT91_SSC_CKS);
+
+		rfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
+			| (( AT91_SSC_FSOS_NONE		     ) & AT91_SSC_FSOS)
+			| (( 0				<< 16) & AT91_SSC_FSLEN)
+			| (( 0				<<  8) & AT91_SSC_DATNB)
+			| (( 1				<<  7) & AT91_SSC_MSBF)
+			| (( 0				<<  5) & AT91_SSC_LOOP)
+			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
+
+		tcmr =	  (( 0				<< 24) & AT91_SSC_PERIOD)
+			| (( 1				<< 16) & AT91_SSC_STTDLY)
+			| (( start_event		     ) & AT91_SSC_START)
+			| (( AT91_SSC_CKI_FALLING	     ) & AT91_SSC_CKI)
+			| (( AT91_SSC_CKO_NONE		     ) & AT91_SSC_CKO)
+			| (( AT91_SSC_CKS_PIN		     ) & AT91_SSC_CKS);
+
+		tfmr =	  (( AT91_SSC_FSEDGE_POSITIVE	     ) & AT91_SSC_FSEDGE)
+			| (( 0				<< 23) & AT91_SSC_FSDEN)
+			| (( AT91_SSC_FSOS_NONE		     ) & AT91_SSC_FSOS)
+			| (( 0				<< 16) & AT91_SSC_FSLEN)
+			| (( 0				<<  8) & AT91_SSC_DATNB)
+			| (( 1				<<  7) & AT91_SSC_MSBF)
+			| (( 0				<<  5) & AT91_SSC_DATDEF)
+			| (((bits - 1)			<<  0) & AT91_SSC_DATALEN);
+		break;
+
+	case SND_SOC_DAIFMT_CBS_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
+	default:
+		printk(KERN_WARNING "at91-i2s: unsupported DAI format 0x%x.\n",
+			ssc_p->daifmt);
+		return -EINVAL;
+		break;
+	}
+	DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr);
+
+	if (!ssc_p->initialized) {
+
+		/* Enable PMC peripheral clock for this SSC */
+		DBG("Starting pid %d clock\n", ssc_p->ssc.pid);
+		at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
+
+		/* Reset the SSC and its PDC registers */
+		at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST);
+
+		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RPR, 0);
+		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RCR, 0);
+		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RNPR, 0);
+		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_RNCR, 0);
+		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TPR, 0);
+		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TCR, 0);
+		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TNPR, 0);
+		at91_ssc_write(ssc_p->ssc.base + AT91_PDC_TNCR, 0);
+
+		if ((ret = request_irq(ssc_p->ssc.pid, at91_i2s_interrupt,
+					0, ssc_p->name, ssc_p)) < 0) {
+			printk(KERN_WARNING "at91-i2s: request_irq failure\n");
+
+			DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
+			at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid);
+			return ret;
+		}
+
+		ssc_p->initialized = 1;
+	}
+
+	/* set SSC clock mode register */
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div);
+
+	/* set receive clock mode and format */
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr);
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr);
+
+	/* set transmit clock mode and format */
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr);
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr);
+
+	DBG("hw_params: SSC initialized\n");
+	return 0;
+}
+
+
+static int at91_i2s_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	struct at91_pcm_dma_params *dma_params;
+	int dir;
+
+	dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1;
+	dma_params = ssc_p->dma_params[dir];
+
+	at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR,
+			dma_params->mask->ssc_enable);
+
+	DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit",
+		at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR));
+	return 0;
+}
+
+
+#ifdef CONFIG_PM
+static int at91_i2s_suspend(struct platform_device *pdev,
+	struct snd_soc_cpu_dai *cpu_dai)
+{
+	struct at91_ssc_info *ssc_p;
+
+	if(!cpu_dai->active)
+		return 0;
+
+	ssc_p = &ssc_info[cpu_dai->id];
+
+	/* Save the status register before disabling transmit and receive. */
+	ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR);
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
+			AT91_SSC_TXDIS | AT91_SSC_RXDIS);
+
+	/* Save the current interrupt mask, then disable unmasked interrupts. */
+	ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR);
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr);
+
+	ssc_p->ssc_state.ssc_cmr  = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR);
+	ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR);
+	ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR);
+	ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR);
+	ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR);
+
+	return 0;
+}
+
+static int at91_i2s_resume(struct platform_device *pdev,
+	struct snd_soc_cpu_dai *cpu_dai)
+{
+	struct at91_ssc_info *ssc_p;
+
+	if(!cpu_dai->active)
+		return 0;
+
+	ssc_p = &ssc_info[cpu_dai->id];
+
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr);
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr);
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr);
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr);
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR,  ssc_p->ssc_state.ssc_cmr);
+
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER,  ssc_p->ssc_state.ssc_imr);
+
+	at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR,
+		((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) |
+		((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0));
+
+	return 0;
+}
+
+#else
+#define at91_i2s_suspend	NULL
+#define at91_i2s_resume		NULL
+#endif
+
+#define AT91_I2S_RATES (SNDRV_PCM_RATE_8000  | SNDRV_PCM_RATE_11025 |\
+			SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+			SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+			SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+			SNDRV_PCM_RATE_96000)
+
+struct snd_soc_cpu_dai at91_i2s_dai[NUM_SSC_DEVICES] = {
+	{	.name = "at91_ssc0/i2s",
+		.id = 0,
+		.type = SND_SOC_DAI_I2S,
+		.suspend = at91_i2s_suspend,
+		.resume = at91_i2s_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = AT91_I2S_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = AT91_I2S_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.ops = {
+			.startup = at91_i2s_startup,
+			.shutdown = at91_i2s_shutdown,
+			.prepare = at91_i2s_prepare,
+			.hw_params = at91_i2s_hw_params,},
+		.dai_ops = {
+			.set_sysclk = at91_i2s_set_dai_sysclk,
+			.set_fmt = at91_i2s_set_dai_fmt,
+			.set_clkdiv = at91_i2s_set_dai_clkdiv,},
+		.private_data = &ssc_info[0].ssc,
+	},
+#if NUM_SSC_DEVICES == 3
+	{	.name = "at91_ssc1/i2s",
+		.id = 1,
+		.type = SND_SOC_DAI_I2S,
+		.suspend = at91_i2s_suspend,
+		.resume = at91_i2s_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = AT91_I2S_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = AT91_I2S_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.ops = {
+			.startup = at91_i2s_startup,
+			.shutdown = at91_i2s_shutdown,
+			.prepare = at91_i2s_prepare,
+			.hw_params = at91_i2s_hw_params,},
+		.dai_ops = {
+			.set_sysclk = at91_i2s_set_dai_sysclk,
+			.set_fmt = at91_i2s_set_dai_fmt,
+			.set_clkdiv = at91_i2s_set_dai_clkdiv,},
+		.private_data = &ssc_info[1].ssc,
+	},
+	{	.name = "at91_ssc2/i2s",
+		.id = 2,
+		.type = SND_SOC_DAI_I2S,
+		.suspend = at91_i2s_suspend,
+		.resume = at91_i2s_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = AT91_I2S_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = AT91_I2S_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.ops = {
+			.startup = at91_i2s_startup,
+			.shutdown = at91_i2s_shutdown,
+			.prepare = at91_i2s_prepare,
+			.hw_params = at91_i2s_hw_params,},
+		.dai_ops = {
+			.set_sysclk = at91_i2s_set_dai_sysclk,
+			.set_fmt = at91_i2s_set_dai_fmt,
+			.set_clkdiv = at91_i2s_set_dai_clkdiv,},
+		.private_data = &ssc_info[2].ssc,
+	},
+#endif
+};
+
+EXPORT_SYMBOL_GPL(at91_i2s_dai);
+
+/* Module information */
+MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com");
+MODULE_DESCRIPTION("AT91 I2S ASoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-i2s.h b/sound/soc/at91/at91-i2s.h
new file mode 100644
index 00000000000..f8a875ba0cc
--- /dev/null
+++ b/sound/soc/at91/at91-i2s.h
@@ -0,0 +1,27 @@
+/*
+ * at91-i2s.h - ALSA I2S interface for the Atmel AT91 SoC
+ *
+ * Author:	Frank Mandarino <fmandarino@endrelia.com>
+ *		Endrelia Technologies Inc.
+ * Created:	Jan 9, 2007
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AT91_I2S_H
+#define _AT91_I2S_H
+
+/* I2S system clock ids */
+#define AT91_SYSCLK_MCK		0 /* SSC uses AT91 MCK as system clock */
+
+/* I2S divider ids */
+#define AT91SSC_CMR_DIV		0 /* MCK divider for BCLK */
+#define AT91SSC_TCMR_PERIOD	1 /* BCLK divider for transmit FS */
+#define AT91SSC_RCMR_PERIOD	2 /* BCLK divider for receive FS */
+
+extern struct snd_soc_cpu_dai at91_i2s_dai[];
+
+#endif /* _AT91_I2S_H */
+
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
new file mode 100644
index 00000000000..e88b12e7cc4
--- /dev/null
+++ b/sound/soc/at91/at91-pcm.c
@@ -0,0 +1,432 @@
+/*
+ * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC
+ *
+ * Author:	Frank Mandarino <fmandarino@endrelia.com>
+ *		Endrelia Technologies Inc.
+ * Created:	Mar 3, 2006
+ *
+ * Based on pxa2xx-pcm.c by:
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Nov 30, 2004
+ * Copyright:	(C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/arch/hardware.h>
+#include <asm/arch/at91_ssc.h>
+#include <asm/arch/at91_pdc.h>
+
+#include "at91-pcm.h"
+
+#if 0
+#define	DBG(x...)	printk(KERN_INFO "at91-pcm: " x)
+#else
+#define	DBG(x...)
+#endif
+
+static const struct snd_pcm_hardware at91_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_INTERLEAVED |
+				  SNDRV_PCM_INFO_PAUSE,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192,
+	.periods_min		= 2,
+	.periods_max		= 1024,
+	.buffer_bytes_max	= 32 * 1024,
+};
+
+struct at91_runtime_data {
+	struct at91_pcm_dma_params *params;
+	dma_addr_t dma_buffer;			/* physical address of dma buffer */
+	dma_addr_t dma_buffer_end;		/* first address beyond DMA buffer */
+	size_t period_size;
+	dma_addr_t period_ptr;			/* physical address of next period */
+	u32 pdc_xpr_save;			/* PDC register save */
+	u32 pdc_xcr_save;
+	u32 pdc_xnpr_save;
+	u32 pdc_xncr_save;
+};
+
+static void at91_pcm_dma_irq(u32 ssc_sr,
+	struct snd_pcm_substream *substream)
+{
+	struct at91_runtime_data *prtd = substream->runtime->private_data;
+	struct at91_pcm_dma_params *params = prtd->params;
+	static int count = 0;
+
+	count++;
+
+	if (ssc_sr & params->mask->ssc_endbuf) {
+
+		printk(KERN_WARNING
+			"at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n",
+			substream->stream == SNDRV_PCM_STREAM_PLAYBACK
+				? "underrun" : "overrun",
+			params->name, ssc_sr, count);
+
+		/* re-start the PDC */
+		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+
+		prtd->period_ptr += prtd->period_size;
+		if (prtd->period_ptr >= prtd->dma_buffer_end) {
+			prtd->period_ptr = prtd->dma_buffer;
+		}
+
+		at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
+		at91_ssc_write(params->ssc_base + params->pdc->xcr,
+				prtd->period_size / params->pdc_xfer_size);
+
+		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
+	}
+
+	if (ssc_sr & params->mask->ssc_endx) {
+
+		/* Load the PDC next pointer and counter registers */
+		prtd->period_ptr += prtd->period_size;
+		if (prtd->period_ptr >= prtd->dma_buffer_end) {
+			prtd->period_ptr = prtd->dma_buffer;
+		}
+		at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
+		at91_ssc_write(params->ssc_base + params->pdc->xncr,
+				prtd->period_size / params->pdc_xfer_size);
+	}
+
+	snd_pcm_period_elapsed(substream);
+}
+
+static int at91_pcm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct at91_runtime_data *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* this may get called several times by oss emulation
+	 * with different params */
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	runtime->dma_bytes = params_buffer_bytes(params);
+
+	prtd->params = rtd->dai->cpu_dai->dma_data;
+	prtd->params->dma_intr_handler = at91_pcm_dma_irq;
+
+	prtd->dma_buffer = runtime->dma_addr;
+	prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes;
+	prtd->period_size = params_period_bytes(params);
+
+	DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n",
+		prtd->params->name, runtime->dma_bytes, prtd->period_size);
+	return 0;
+}
+
+static int at91_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct at91_runtime_data *prtd = substream->runtime->private_data;
+	struct at91_pcm_dma_params *params = prtd->params;
+
+	if (params != NULL) {
+		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+		prtd->params->dma_intr_handler = NULL;
+	}
+
+	return 0;
+}
+
+static int at91_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct at91_runtime_data *prtd = substream->runtime->private_data;
+	struct at91_pcm_dma_params *params = prtd->params;
+
+	at91_ssc_write(params->ssc_base + AT91_SSC_IDR,
+			params->mask->ssc_endx | params->mask->ssc_endbuf);
+
+	at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+	return 0;
+}
+
+static int at91_pcm_trigger(struct snd_pcm_substream *substream,
+	int cmd)
+{
+	struct at91_runtime_data *prtd = substream->runtime->private_data;
+	struct at91_pcm_dma_params *params = prtd->params;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		prtd->period_ptr = prtd->dma_buffer;
+
+		at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr);
+		at91_ssc_write(params->ssc_base + params->pdc->xcr,
+				prtd->period_size / params->pdc_xfer_size);
+
+		prtd->period_ptr += prtd->period_size;
+		at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr);
+		at91_ssc_write(params->ssc_base + params->pdc->xncr,
+				prtd->period_size / params->pdc_xfer_size);
+
+		DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n",
+			(unsigned long) prtd->period_ptr,
+			at91_ssc_read(params->ssc_base + params->pdc->xpr),
+			at91_ssc_read(params->ssc_base + params->pdc->xcr),
+			at91_ssc_read(params->ssc_base + params->pdc->xnpr),
+			at91_ssc_read(params->ssc_base + params->pdc->xncr));
+
+		at91_ssc_write(params->ssc_base + AT91_SSC_IER,
+			params->mask->ssc_endx | params->mask->ssc_endbuf);
+
+		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
+
+		DBG("sr=%lx imr=%lx\n", at91_ssc_read(params->ssc_base + AT91_SSC_SR),
+					at91_ssc_read(params->ssc_base + AT91_SSC_IER));
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+		break;
+
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static snd_pcm_uframes_t at91_pcm_pointer(
+	struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct at91_runtime_data *prtd = runtime->private_data;
+	struct at91_pcm_dma_params *params = prtd->params;
+	dma_addr_t ptr;
+	snd_pcm_uframes_t x;
+
+	ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr);
+	x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
+
+	if (x == runtime->buffer_size)
+		x = 0;
+	return x;
+}
+
+static int at91_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct at91_runtime_data *prtd;
+	int ret = 0;
+
+	snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware);
+
+	/* ensure that buffer size is a multiple of period size */
+	ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		goto out;
+
+	prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL);
+	if (prtd == NULL) {
+		ret = -ENOMEM;
+		goto out;
+	}
+	runtime->private_data = prtd;
+
+ out:
+	return ret;
+}
+
+static int at91_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct at91_runtime_data *prtd = substream->runtime->private_data;
+
+	kfree(prtd);
+	return 0;
+}
+
+static int at91_pcm_mmap(struct snd_pcm_substream *substream,
+	struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+				     runtime->dma_area,
+				     runtime->dma_addr,
+				     runtime->dma_bytes);
+}
+
+struct snd_pcm_ops at91_pcm_ops = {
+	.open		= at91_pcm_open,
+	.close		= at91_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= at91_pcm_hw_params,
+	.hw_free	= at91_pcm_hw_free,
+	.prepare	= at91_pcm_prepare,
+	.trigger	= at91_pcm_trigger,
+	.pointer	= at91_pcm_pointer,
+	.mmap		= at91_pcm_mmap,
+};
+
+static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+	int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = at91_pcm_hardware.buffer_bytes_max;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+					   &buf->addr, GFP_KERNEL);
+
+	DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+		(void *) buf->area,
+		(void *) buf->addr,
+		size);
+
+	if (!buf->area)
+		return -ENOMEM;
+
+	buf->bytes = size;
+	return 0;
+}
+
+static u64 at91_pcm_dmamask = 0xffffffff;
+
+static int at91_pcm_new(struct snd_card *card,
+	struct snd_soc_codec_dai *dai, struct snd_pcm *pcm)
+{
+	int ret = 0;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &at91_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = 0xffffffff;
+
+	if (dai->playback.channels_min) {
+		ret = at91_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (dai->capture.channels_min) {
+		ret = at91_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+ out:
+	return ret;
+}
+
+static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+
+		dma_free_writecombine(pcm->card->dev, buf->bytes,
+				      buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+#ifdef CONFIG_PM
+static int at91_pcm_suspend(struct platform_device *pdev,
+	struct snd_soc_cpu_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = dai->runtime;
+	struct at91_runtime_data *prtd;
+	struct at91_pcm_dma_params *params;
+
+	if (!runtime)
+		return 0;
+
+	prtd = runtime->private_data;
+	params = prtd->params;
+
+	/* disable the PDC and save the PDC registers */
+
+	at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_disable);
+
+	prtd->pdc_xpr_save  = at91_ssc_read(params->ssc_base + params->pdc->xpr);
+	prtd->pdc_xcr_save  = at91_ssc_read(params->ssc_base + params->pdc->xcr);
+	prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr);
+	prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr);
+
+	return 0;
+}
+
+static int at91_pcm_resume(struct platform_device *pdev,
+	struct snd_soc_cpu_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = dai->runtime;
+	struct at91_runtime_data *prtd;
+	struct at91_pcm_dma_params *params;
+
+	if (!runtime)
+		return 0;
+
+	prtd = runtime->private_data;
+	params = prtd->params;
+
+	/* restore the PDC registers and enable the PDC */
+	at91_ssc_write(params->ssc_base + params->pdc->xpr,  prtd->pdc_xpr_save);
+	at91_ssc_write(params->ssc_base + params->pdc->xcr,  prtd->pdc_xcr_save);
+	at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save);
+	at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save);
+
+	at91_ssc_write(params->ssc_base + AT91_PDC_PTCR, params->mask->pdc_enable);
+	return 0;
+}
+#else
+#define at91_pcm_suspend	NULL
+#define at91_pcm_resume		NULL
+#endif
+
+struct snd_soc_platform at91_soc_platform = {
+	.name		= "at91-audio",
+	.pcm_ops 	= &at91_pcm_ops,
+	.pcm_new	= at91_pcm_new,
+	.pcm_free	= at91_pcm_free_dma_buffers,
+	.suspend	= at91_pcm_suspend,
+	.resume		= at91_pcm_resume,
+};
+
+EXPORT_SYMBOL_GPL(at91_soc_platform);
+
+MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
+MODULE_DESCRIPTION("Atmel AT91 PCM module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h
new file mode 100644
index 00000000000..58d0f00a07b
--- /dev/null
+++ b/sound/soc/at91/at91-pcm.h
@@ -0,0 +1,72 @@
+/*
+ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC
+ *
+ * Author:	Frank Mandarino <fmandarino@endrelia.com>
+ *		Endrelia Technologies Inc.
+ * Created:	Mar 3, 2006
+ *
+ * Based on pxa2xx-pcm.h by:
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Nov 30, 2004
+ * Copyright:	MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AT91_PCM_H
+#define _AT91_PCM_H
+
+#include <asm/arch/hardware.h>
+
+struct at91_ssc_periph {
+	void __iomem	*base;
+	u32		pid;
+};
+
+/*
+ * Registers and status bits that are required by the PCM driver.
+ */
+struct at91_pdc_regs {
+	unsigned int	xpr;		/* PDC recv/trans pointer */
+	unsigned int	xcr;		/* PDC recv/trans counter */
+	unsigned int	xnpr;		/* PDC next recv/trans pointer */
+	unsigned int	xncr;		/* PDC next recv/trans counter */
+	unsigned int	ptcr;		/* PDC transfer control */
+};
+
+struct at91_ssc_mask {
+	u32	ssc_enable;		/* SSC recv/trans enable */
+	u32	ssc_disable;		/* SSC recv/trans disable */
+	u32	ssc_endx;		/* SSC ENDTX or ENDRX */
+	u32	ssc_endbuf;		/* SSC TXBUFE or RXBUFF */
+	u32	pdc_enable;		/* PDC recv/trans enable */
+	u32	pdc_disable;		/* PDC recv/trans disable */
+};
+
+/*
+ * This structure, shared between the PCM driver and the interface,
+ * contains all information required by the PCM driver to perform the
+ * PDC DMA operation.  All fields except dma_intr_handler() are initialized
+ * by the interface.  The dms_intr_handler() pointer is set by the PCM
+ * driver and called by the interface SSC interrupt handler if it is
+ * non-NULL.
+ */
+struct at91_pcm_dma_params {
+	char *name;			/* stream identifier */
+	int pdc_xfer_size;		/* PDC counter increment in bytes */
+	void __iomem *ssc_base;		/* SSC base address */
+	struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */
+	struct at91_ssc_mask *mask;/* SSC & PDC status bits */
+	struct snd_pcm_substream *substream;
+	void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
+};
+
+extern struct snd_soc_platform at91_soc_platform;
+
+#define at91_ssc_read(a)	((unsigned long) __raw_readl(a))
+#define at91_ssc_write(a,v)	__raw_writel((v),(a))
+
+#endif /* _AT91_PCM_H */
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
new file mode 100644
index 00000000000..8179df3bb2f
--- /dev/null
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -0,0 +1,375 @@
+/*
+ * eti_b1_wm8731  --  SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
+ *
+ * Author:	Frank Mandarino <fmandarino@endrelia.com>
+ *		Endrelia Technologies Inc.
+ * Created:	Mar 29, 2006
+ *
+ * Based on corgi.c by:
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/arch/hardware.h>
+#include <asm/arch/at91_pio.h>
+#include <asm/arch/gpio.h>
+
+#include "../codecs/wm8731.h"
+#include "at91-pcm.h"
+#include "at91-i2s.h"
+
+#if 0
+#define	DBG(x...)	printk(KERN_INFO "eti_b1_wm8731: " x)
+#else
+#define	DBG(x...)
+#endif
+
+#define AT91_PIO_TF1	(1 << (AT91_PIN_PB6 - PIN_BASE) % 32)
+#define AT91_PIO_TK1	(1 << (AT91_PIN_PB7 - PIN_BASE) % 32)
+#define AT91_PIO_TD1	(1 << (AT91_PIN_PB8 - PIN_BASE) % 32)
+#define AT91_PIO_RD1	(1 << (AT91_PIN_PB9 - PIN_BASE) % 32)
+#define AT91_PIO_RK1	(1 << (AT91_PIN_PB10 - PIN_BASE) % 32)
+#define AT91_PIO_RF1	(1 << (AT91_PIN_PB11 - PIN_BASE) % 32)
+
+static struct clk *pck1_clk;
+static struct clk *pllb_clk;
+
+
+static int eti_b1_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* cpu clock is the AT91 master clock sent to the SSC */
+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
+		60000000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* codec system clock is supplied by PCK1, set to 12MHz */
+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK,
+		12000000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* Start PCK1 clock. */
+	clk_enable(pck1_clk);
+	DBG("pck1 started\n");
+
+	return 0;
+}
+
+static void eti_b1_shutdown(struct snd_pcm_substream *substream)
+{
+	/* Stop PCK1 clock. */
+	clk_disable(pck1_clk);
+	DBG("pck1 stopped\n");
+}
+
+static int eti_b1_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
+	unsigned int rate;
+	int cmr_div, period;
+
+	/* set codec DAI configuration */
+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/*
+	 * The SSC clock dividers depend on the sample rate.  The CMR.DIV
+	 * field divides the system master clock MCK to drive the SSC TK
+	 * signal which provides the codec BCLK.  The TCMR.PERIOD and
+	 * RCMR.PERIOD fields further divide the BCLK signal to drive
+	 * the SSC TF and RF signals which provide the codec DACLRC and
+	 * ADCLRC clocks.
+	 *
+	 * The dividers were determined through trial and error, where a
+	 * CMR.DIV value is chosen such that the resulting BCLK value is
+	 * divisible, or almost divisible, by (2 * sample rate), and then
+	 * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
+	 */
+	rate = params_rate(params);
+
+	switch (rate) {
+	case 8000:
+		cmr_div = 25;	/* BCLK = 60MHz/(2*25) = 1.2MHz */
+		period = 74;	/* LRC = BCLK/(2*(74+1)) = 8000Hz */
+		break;
+	case 32000:
+		cmr_div = 7;	/* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
+		period = 66;	/* LRC = BCLK/(2*(66+1)) = 31982.942Hz */
+		break;
+	case 48000:
+		cmr_div = 13;	/* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */
+		period = 23;	/* LRC = BCLK/(2*(23+1)) = 48076.923Hz */
+		break;
+	default:
+		printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate);
+		return -EINVAL;
+	}
+
+	/* set the MCK divider for BCLK */
+	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
+	if (ret < 0)
+		return ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* set the BCLK divider for DACLRC */
+		ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
+						AT91SSC_TCMR_PERIOD, period);
+	} else {
+		/* set the BCLK divider for ADCLRC */
+		ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
+						AT91SSC_RCMR_PERIOD, period);
+	}
+	if (ret < 0)
+		return ret;
+
+#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
+	/*
+	 * Codec in Master Mode.
+	 */
+
+	/* set codec DAI configuration */
+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
+
+	return 0;
+}
+
+static struct snd_soc_ops eti_b1_ops = {
+	.startup = eti_b1_startup,
+	.hw_params = eti_b1_hw_params,
+	.shutdown = eti_b1_shutdown,
+};
+
+
+static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const char *intercon[][3] = {
+
+	/* speaker connected to LHPOUT */
+	{"Ext Spk", NULL, "LHPOUT"},
+
+	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
+	{"MICIN", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Int Mic"},
+
+	/* terminator */
+	{NULL, NULL, NULL},
+};
+
+/*
+ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
+ */
+static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
+{
+	int i;
+
+	DBG("eti_b1_wm8731_init() called\n");
+
+	/* Add specific widgets */
+	for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]);
+	}
+
+	/* Set up specific audio path interconnects */
+	for(i = 0; intercon[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, intercon[i][0],
+			intercon[i][1], intercon[i][2]);
+	}
+
+	/* not connected */
+	snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+	snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
+
+	/* always connected */
+	snd_soc_dapm_set_endpoint(codec, "Int Mic", 1);
+	snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+
+	snd_soc_dapm_sync_endpoints(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link eti_b1_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai = &at91_i2s_dai[1],
+	.codec_dai = &wm8731_dai,
+	.init = eti_b1_wm8731_init,
+	.ops = &eti_b1_ops,
+};
+
+static struct snd_soc_machine snd_soc_machine_eti_b1 = {
+	.name = "ETI_B1",
+	.dai_link = &eti_b1_dai,
+	.num_links = 1,
+};
+
+static struct wm8731_setup_data eti_b1_wm8731_setup = {
+	.i2c_address = 0x1a,
+};
+
+static struct snd_soc_device eti_b1_snd_devdata = {
+	.machine = &snd_soc_machine_eti_b1,
+	.platform = &at91_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &eti_b1_wm8731_setup,
+};
+
+static struct platform_device *eti_b1_snd_device;
+
+static int __init eti_b1_init(void)
+{
+	int ret;
+	u32 ssc_pio_lines;
+	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
+
+	if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
+		DBG("SSC1 memory region is busy\n");
+		return -EBUSY;
+	}
+
+	ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K);
+	if (!ssc->base) {
+		DBG("SSC1 memory ioremap failed\n");
+		ret = -ENOMEM;
+		goto fail_release_mem;
+	}
+
+	ssc->pid = AT91RM9200_ID_SSC1;
+
+	eti_b1_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!eti_b1_snd_device) {
+		DBG("platform device allocation failed\n");
+		ret = -ENOMEM;
+		goto fail_io_unmap;
+	}
+
+	platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata);
+	eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev;
+
+	ret = platform_device_add(eti_b1_snd_device);
+	if (ret) {
+		DBG("platform device add failed\n");
+		platform_device_put(eti_b1_snd_device);
+		goto fail_io_unmap;
+	}
+
+ 	ssc_pio_lines = AT91_PIO_TF1 | AT91_PIO_TK1 | AT91_PIO_TD1
+			| AT91_PIO_RD1 /* | AT91_PIO_RK1 */ | AT91_PIO_RF1;
+
+	/* Reset all PIO registers and assign lines to peripheral A */
+ 	at91_sys_write(AT91_PIOB + PIO_PDR,  ssc_pio_lines);
+ 	at91_sys_write(AT91_PIOB + PIO_ODR,  ssc_pio_lines);
+ 	at91_sys_write(AT91_PIOB + PIO_IFDR, ssc_pio_lines);
+ 	at91_sys_write(AT91_PIOB + PIO_CODR, ssc_pio_lines);
+ 	at91_sys_write(AT91_PIOB + PIO_IDR,  ssc_pio_lines);
+ 	at91_sys_write(AT91_PIOB + PIO_MDDR, ssc_pio_lines);
+ 	at91_sys_write(AT91_PIOB + PIO_PUDR, ssc_pio_lines);
+ 	at91_sys_write(AT91_PIOB + PIO_ASR,  ssc_pio_lines);
+ 	at91_sys_write(AT91_PIOB + PIO_OWDR, ssc_pio_lines);
+
+	/*
+	 * Set PCK1 parent to PLLB and its rate to 12 Mhz.
+	 */
+	pllb_clk = clk_get(NULL, "pllb");
+	pck1_clk = clk_get(NULL, "pck1");
+
+	clk_set_parent(pck1_clk, pllb_clk);
+	clk_set_rate(pck1_clk, 12000000);
+
+	DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk));
+
+	/* assign the GPIO pin to PCK1 */
+	at91_set_B_periph(AT91_PIN_PA24, 0);
+
+#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
+	printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n");
+#else
+	printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n");
+#endif
+	return ret;
+
+fail_io_unmap:
+	iounmap(ssc->base);
+fail_release_mem:
+	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
+	return ret;
+}
+
+static void __exit eti_b1_exit(void)
+{
+	struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
+
+	clk_put(pck1_clk);
+	clk_put(pllb_clk);
+
+	platform_device_unregister(eti_b1_snd_device);
+
+	iounmap(ssc->base);
+	release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
+}
+
+module_init(eti_b1_init);
+module_exit(eti_b1_exit);
+
+/* Module information */
+MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
+MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
new file mode 100644
index 00000000000..78ac2688e12
--- /dev/null
+++ b/sound/soc/codecs/Kconfig
@@ -0,0 +1,15 @@
+config SND_SOC_AC97_CODEC
+	tristate
+	depends SND_SOC
+
+config SND_SOC_WM8731
+	tristate
+	depends SND_SOC
+
+config SND_SOC_WM8750
+	tristate
+	depends SND_SOC
+
+config SND_SOC_WM9712
+	tristate
+	depends SND_SOC
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
new file mode 100644
index 00000000000..3249a6e4f1d
--- /dev/null
+++ b/sound/soc/codecs/Makefile
@@ -0,0 +1,9 @@
+snd-soc-ac97-objs := ac97.o
+snd-soc-wm8731-objs := wm8731.o
+snd-soc-wm8750-objs := wm8750.o
+snd-soc-wm9712-objs := wm9712.o
+
+obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
+obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
+obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
+obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
new file mode 100644
index 00000000000..55bc55eb6e2
--- /dev/null
+++ b/sound/soc/codecs/ac97.c
@@ -0,0 +1,156 @@
+/*
+ * ac97.c  --  ALSA Soc AC97 codec support
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    17th Oct 2005   Initial version.
+ *
+ * Generic AC97 support.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#define AC97_VERSION "0.6"
+
+static int ac97_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+		  AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
+	return snd_ac97_set_rate(codec->ac97, reg, runtime->rate);
+}
+
+#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+static struct snd_soc_codec_dai ac97_dai = {
+	.name = "AC97 HiFi",
+	.playback = {
+		.stream_name = "AC97 Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = STD_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "AC97 Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = STD_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = {
+		.prepare = ac97_prepare,},
+};
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	return soc_ac97_ops.read(codec->ac97, reg);
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int val)
+{
+	soc_ac97_ops.write(codec->ac97, reg, val);
+	return 0;
+}
+
+static int ac97_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct snd_ac97_bus *ac97_bus;
+	struct snd_ac97_template ac97_template;
+	int ret = 0;
+
+	printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->codec == NULL)
+		return -ENOMEM;
+	codec = socdev->codec;
+	mutex_init(&codec->mutex);
+
+	codec->name = "AC97";
+	codec->owner = THIS_MODULE;
+	codec->dai = &ac97_dai;
+	codec->num_dai = 1;
+	codec->write = ac97_write;
+	codec->read = ac97_read;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if(ret < 0)
+		goto err;
+
+	/* add codec as bus device for standard ac97 */
+	ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus);
+	if(ret < 0)
+		goto bus_err;
+
+	memset(&ac97_template, 0, sizeof(struct snd_ac97_template));
+	ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97);
+	if(ret < 0)
+		goto bus_err;
+
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0)
+		goto bus_err;
+	return 0;
+
+bus_err:
+	snd_soc_free_pcms(socdev);
+
+err:
+	kfree(socdev->codec->reg_cache);
+	kfree(socdev->codec);
+	socdev->codec = NULL;
+	return ret;
+}
+
+static int ac97_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if(codec == NULL)
+		return 0;
+
+	snd_soc_free_pcms(socdev);
+	kfree(socdev->codec->reg_cache);
+	kfree(socdev->codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ac97= {
+	.probe = 	ac97_soc_probe,
+	.remove = 	ac97_soc_remove,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_ac97);
+
+MODULE_DESCRIPTION("Soc Generic AC97 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h
new file mode 100644
index 00000000000..930ddfc2321
--- /dev/null
+++ b/sound/soc/codecs/ac97.h
@@ -0,0 +1,18 @@
+/*
+ * linux/sound/codecs/ac97.h -- ALSA SoC Layer
+ *
+ * Author:		Liam Girdwood
+ * Created:		Dec 1st 2005
+ * Copyright:	Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_AC97_H
+#define __LINUX_SND_SOC_AC97_H
+
+extern struct snd_soc_codec_device soc_codec_dev_ac97;
+
+#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
new file mode 100644
index 00000000000..7ca0b526828
--- /dev/null
+++ b/sound/soc/codecs/wm8731.c
@@ -0,0 +1,758 @@
+/*
+ * wm8731.c  --  WM8731 ALSA SoC Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8731.h"
+
+#define AUDIO_NAME "wm8731"
+#define WM8731_VERSION "0.13"
+
+/*
+ * Debug
+ */
+
+#define WM8731_DEBUG 0
+
+#ifdef WM8731_DEBUG
+#define dbg(format, arg...) \
+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
+#else
+#define dbg(format, arg...) do {} while (0)
+#endif
+#define err(format, arg...) \
+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
+#define info(format, arg...) \
+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
+#define warn(format, arg...) \
+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
+
+struct snd_soc_codec_device soc_codec_dev_wm8731;
+
+/* codec private data */
+struct wm8731_priv {
+	unsigned int sysclk;
+};
+
+/*
+ * wm8731 register cache
+ * We can't read the WM8731 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u16 wm8731_reg[WM8731_CACHEREGNUM] = {
+    0x0097, 0x0097, 0x0079, 0x0079,
+    0x000a, 0x0008, 0x009f, 0x000a,
+    0x0000, 0x0000
+};
+
+/*
+ * read wm8731 register cache
+ */
+static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg == WM8731_RESET)
+		return 0;
+	if (reg >= WM8731_CACHEREGNUM)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write wm8731 register cache
+ */
+static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec,
+	u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= WM8731_CACHEREGNUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the WM8731 register space
+ */
+static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[2];
+
+	/* data is
+	 *   D15..D9 WM8731 register offset
+	 *   D8...D0 register data
+	 */
+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+	data[1] = value & 0x00ff;
+
+	wm8731_write_reg_cache (codec, reg, value);
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+#define wm8731_reset(c)	wm8731_write(c, WM8731_RESET, 0)
+
+static const char *wm8731_input_select[] = {"Line In", "Mic"};
+static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum wm8731_enum[] = {
+	SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select),
+	SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph),
+};
+
+static const struct snd_kcontrol_new wm8731_snd_controls[] = {
+
+SOC_DOUBLE_R("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V,
+	0, 127, 0),
+SOC_DOUBLE_R("Master Playback ZC Switch", WM8731_LOUT1V, WM8731_ROUT1V,
+	7, 1, 0),
+
+SOC_DOUBLE_R("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0),
+SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1),
+
+SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0),
+SOC_SINGLE("Capture Mic Switch", WM8731_APANA, 1, 1, 1),
+
+SOC_SINGLE("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1),
+
+SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1),
+SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
+
+SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
+};
+
+/* add non dapm controls */
+static int wm8731_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
+		if ((err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&wm8731_snd_controls[i],codec, NULL))) < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/* Output Mixer */
+static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new wm8731_input_mux_controls =
+SOC_DAPM_ENUM("Input Select", wm8731_enum[0]);
+
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
+	&wm8731_output_mixer_controls[0],
+	ARRAY_SIZE(wm8731_output_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8731_PWR, 3, 1),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("LHPOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("RHPOUT"),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8731_PWR, 2, 1),
+SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &wm8731_input_mux_controls),
+SND_SOC_DAPM_PGA("Line Input", WM8731_PWR, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8731_PWR, 1, 1),
+SND_SOC_DAPM_INPUT("MICIN"),
+SND_SOC_DAPM_INPUT("RLINEIN"),
+SND_SOC_DAPM_INPUT("LLINEIN"),
+};
+
+static const char *intercon[][3] = {
+	/* output mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "HiFi Playback Switch", "DAC"},
+	{"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+	/* outputs */
+	{"RHPOUT", NULL, "Output Mixer"},
+	{"ROUT", NULL, "Output Mixer"},
+	{"LHPOUT", NULL, "Output Mixer"},
+	{"LOUT", NULL, "Output Mixer"},
+
+	/* input mux */
+	{"Input Mux", "Line In", "Line Input"},
+	{"Input Mux", "Mic", "Mic Bias"},
+	{"ADC", NULL, "Input Mux"},
+
+	/* inputs */
+	{"Line Input", NULL, "LLINEIN"},
+	{"Line Input", NULL, "RLINEIN"},
+	{"Mic Bias", NULL, "MICIN"},
+
+	/* terminator */
+	{NULL, NULL, NULL},
+};
+
+static int wm8731_add_widgets(struct snd_soc_codec *codec)
+{
+	int i;
+
+	for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	}
+
+	/* set up audio path interconnects */
+	for(i = 0; intercon[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, intercon[i][0],
+			intercon[i][1], intercon[i][2]);
+	}
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+struct _coeff_div {
+	u32 mclk;
+	u32 rate;
+	u16 fs;
+	u8 sr:4;
+	u8 bosr:1;
+	u8 usb:1;
+};
+
+/* codec mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+	/* 48k */
+	{12288000, 48000, 256, 0x0, 0x0, 0x0},
+	{18432000, 48000, 384, 0x0, 0x1, 0x0},
+	{12000000, 48000, 250, 0x0, 0x0, 0x1},
+
+	/* 32k */
+	{12288000, 32000, 384, 0x6, 0x0, 0x0},
+	{18432000, 32000, 576, 0x6, 0x1, 0x0},
+	{12000000, 32000, 375, 0x6, 0x0, 0x1},
+
+	/* 8k */
+	{12288000, 8000, 1536, 0x3, 0x0, 0x0},
+	{18432000, 8000, 2304, 0x3, 0x1, 0x0},
+	{11289600, 8000, 1408, 0xb, 0x0, 0x0},
+	{16934400, 8000, 2112, 0xb, 0x1, 0x0},
+	{12000000, 8000, 1500, 0x3, 0x0, 0x1},
+
+	/* 96k */
+	{12288000, 96000, 128, 0x7, 0x0, 0x0},
+	{18432000, 96000, 192, 0x7, 0x1, 0x0},
+	{12000000, 96000, 125, 0x7, 0x0, 0x1},
+
+	/* 44.1k */
+	{11289600, 44100, 256, 0x8, 0x0, 0x0},
+	{16934400, 44100, 384, 0x8, 0x1, 0x0},
+	{12000000, 44100, 272, 0x8, 0x1, 0x1},
+
+	/* 88.2k */
+	{11289600, 88200, 128, 0xf, 0x0, 0x0},
+	{16934400, 88200, 192, 0xf, 0x1, 0x0},
+	{12000000, 88200, 136, 0xf, 0x1, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+			return i;
+	}
+	return 0;
+}
+
+static int wm8731_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct wm8731_priv *wm8731 = codec->private_data;
+	u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
+	int i = get_coeff(wm8731->sysclk, params_rate(params));
+	u16 srate = (coeff_div[i].sr << 2) |
+		(coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
+	wm8731_write(codec, WM8731_SRATE, srate);
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x0004;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x0008;
+		break;
+	}
+
+	wm8731_write(codec, WM8731_IFACE, iface);
+	return 0;
+}
+
+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	/* set active */
+	wm8731_write(codec, WM8731_ACTIVE, 0x0001);
+
+	return 0;
+}
+
+static void wm8731_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	/* deactivate */
+	if (!codec->active) {
+		udelay(50);
+		wm8731_write(codec, WM8731_ACTIVE, 0x0);
+	}
+}
+
+static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
+
+	if (mute)
+		wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8);
+	else
+		wm8731_write(codec, WM8731_APDIGI, mute_reg);
+	return 0;
+}
+
+static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8731_priv *wm8731 = codec->private_data;
+
+	switch (freq) {
+	case 11289600:
+	case 12000000:
+	case 12288000:
+	case 16934400:
+	case 18432000:
+		wm8731->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+
+static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = 0;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface |= 0x0040;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 0x0002;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x0001;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x0003;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= 0x0013;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= 0x0090;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= 0x0080;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x0010;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set iface */
+	wm8731_write(codec, WM8731_IFACE, iface);
+	return 0;
+}
+
+static int wm8731_dapm_event(struct snd_soc_codec *codec, int event)
+{
+	u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
+
+	switch (event) {
+	case SNDRV_CTL_POWER_D0: /* full On */
+		/* vref/mid, osc on, dac unmute */
+		wm8731_write(codec, WM8731_PWR, reg);
+		break;
+	case SNDRV_CTL_POWER_D1: /* partial On */
+	case SNDRV_CTL_POWER_D2: /* partial On */
+		break;
+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+		/* everything off except vref/vmid, */
+		wm8731_write(codec, WM8731_PWR, reg | 0x0040);
+		break;
+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+		/* everything off, dac mute, inactive */
+		wm8731_write(codec, WM8731_ACTIVE, 0x0);
+		wm8731_write(codec, WM8731_PWR, 0xffff);
+		break;
+	}
+	codec->dapm_state = event;
+	return 0;
+}
+
+#define WM8731_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+		SNDRV_PCM_RATE_96000)
+
+#define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_codec_dai wm8731_dai = {
+	.name = "WM8731",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	.ops = {
+		.prepare = wm8731_pcm_prepare,
+		.hw_params = wm8731_hw_params,
+		.shutdown = wm8731_shutdown,
+	},
+	.dai_ops = {
+		.digital_mute = wm8731_mute,
+		.set_sysclk = wm8731_set_dai_sysclk,
+		.set_fmt = wm8731_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL_GPL(wm8731_dai);
+
+static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8731_write(codec, WM8731_ACTIVE, 0x0);
+	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	return 0;
+}
+
+static int wm8731_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+	u8 data[2];
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) {
+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+		data[1] = cache[i] & 0x00ff;
+		codec->hw_write(codec->control_data, data, 2);
+	}
+	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm8731_dapm_event(codec, codec->suspend_dapm_state);
+	return 0;
+}
+
+/*
+ * initialise the WM8731 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8731_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int reg, ret = 0;
+
+	codec->name = "WM8731";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8731_read_reg_cache;
+	codec->write = wm8731_write;
+	codec->dapm_event = wm8731_dapm_event;
+	codec->dai = &wm8731_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = sizeof(wm8731_reg);
+	codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	wm8731_reset(codec);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8731: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+	/* set the update bits */
+	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
+	wm8731_write(codec, WM8731_LOUT1V, reg | 0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
+	wm8731_write(codec, WM8731_ROUT1V, reg | 0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
+	wm8731_write(codec, WM8731_LINVOL, reg | 0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
+	wm8731_write(codec, WM8731_RINVOL, reg | 0x0100);
+
+	wm8731_add_controls(codec);
+	wm8731_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8731: failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+	return ret;
+}
+
+static struct snd_soc_device *wm8731_socdev;
+
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+
+/*
+ * WM8731 2 wire address is determined by GPIO5
+ * state during powerup.
+ *    low  = 0x1a
+ *    high = 0x1b
+ */
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8731_i2c_driver;
+static struct i2c_client client_template;
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+   around */
+
+static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+	struct snd_soc_device *socdev = wm8731_socdev;
+	struct wm8731_setup_data *setup = socdev->codec_data;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct i2c_client *i2c;
+	int ret;
+
+	if (addr != setup->i2c_address)
+		return -ENODEV;
+
+	client_template.adapter = adap;
+	client_template.addr = addr;
+
+	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+	if (i2c == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = i2c_attach_client(i2c);
+	if (ret < 0) {
+		err("failed to attach codec at addr %x\n", addr);
+		goto err;
+	}
+
+	ret = wm8731_init(socdev);
+	if (ret < 0) {
+		err("failed to initialise WM8731\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+
+static int wm8731_i2c_detach(struct i2c_client *client)
+{
+	struct snd_soc_codec* codec = i2c_get_clientdata(client);
+	i2c_detach_client(client);
+	kfree(codec->reg_cache);
+	kfree(client);
+	return 0;
+}
+
+static int wm8731_i2c_attach(struct i2c_adapter *adap)
+{
+	return i2c_probe(adap, &addr_data, wm8731_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver wm8731_i2c_driver = {
+	.driver = {
+		.name = "WM8731 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.id =             I2C_DRIVERID_WM8731,
+	.attach_adapter = wm8731_i2c_attach,
+	.detach_client =  wm8731_i2c_detach,
+	.command =        NULL,
+};
+
+static struct i2c_client client_template = {
+	.name =   "WM8731",
+	.driver = &wm8731_i2c_driver,
+};
+#endif
+
+static int wm8731_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct wm8731_setup_data *setup;
+	struct snd_soc_codec *codec;
+	struct wm8731_priv *wm8731;
+	int ret = 0;
+
+	info("WM8731 Audio Codec %s", WM8731_VERSION);
+
+	setup = socdev->codec_data;
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+	if (wm8731 == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+
+	codec->private_data = wm8731;
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	wm8731_socdev = socdev;
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+	if (setup->i2c_address) {
+		normal_i2c[0] = setup->i2c_address;
+		codec->hw_write = (hw_write_t)i2c_master_send;
+		ret = i2c_add_driver(&wm8731_i2c_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add i2c driver");
+	}
+#else
+	/* Add other interfaces here */
+#endif
+	return ret;
+}
+
+/* power down chip */
+static int wm8731_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec->control_data)
+		wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8731_i2c_driver);
+#endif
+	kfree(codec->private_data);
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8731 = {
+	.probe = 	wm8731_probe,
+	.remove = 	wm8731_remove,
+	.suspend = 	wm8731_suspend,
+	.resume =	wm8731_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
+
+MODULE_DESCRIPTION("ASoC WM8731 driver");
+MODULE_AUTHOR("Richard Purdie");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
new file mode 100644
index 00000000000..5bcab6a7afb
--- /dev/null
+++ b/sound/soc/codecs/wm8731.h
@@ -0,0 +1,44 @@
+/*
+ * wm8731.h  --  WM8731 Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8731_H
+#define _WM8731_H
+
+/* WM8731 register space */
+
+#define WM8731_LINVOL   0x00
+#define WM8731_RINVOL   0x01
+#define WM8731_LOUT1V   0x02
+#define WM8731_ROUT1V   0x03
+#define WM8731_APANA    0x04
+#define WM8731_APDIGI   0x05
+#define WM8731_PWR      0x06
+#define WM8731_IFACE    0x07
+#define WM8731_SRATE    0x08
+#define WM8731_ACTIVE   0x09
+#define WM8731_RESET	0x0f
+
+#define WM8731_CACHEREGNUM 	10
+
+#define WM8731_SYSCLK	0
+#define WM8731_DAI		0
+
+struct wm8731_setup_data {
+	unsigned short i2c_address;
+};
+
+extern struct snd_soc_codec_dai wm8731_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8731;
+
+#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
new file mode 100644
index 00000000000..7073e8e294f
--- /dev/null
+++ b/sound/soc/codecs/wm8750.c
@@ -0,0 +1,1049 @@
+/*
+ * wm8750.c -- WM8750 ALSA SoC audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8750.h"
+
+#define AUDIO_NAME "WM8750"
+#define WM8750_VERSION "0.12"
+
+/*
+ * Debug
+ */
+
+#define WM8750_DEBUG 0
+
+#ifdef WM8750_DEBUG
+#define dbg(format, arg...) \
+	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
+#else
+#define dbg(format, arg...) do {} while (0)
+#endif
+#define err(format, arg...) \
+	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
+#define info(format, arg...) \
+	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
+#define warn(format, arg...) \
+	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
+
+/* codec private data */
+struct wm8750_priv {
+	unsigned int sysclk;
+};
+
+/*
+ * wm8750 register cache
+ * We can't read the WM8750 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8750_reg[] = {
+	0x0097, 0x0097, 0x0079, 0x0079,  /*  0 */
+	0x0000, 0x0008, 0x0000, 0x000a,  /*  4 */
+	0x0000, 0x0000, 0x00ff, 0x00ff,  /*  8 */
+	0x000f, 0x000f, 0x0000, 0x0000,  /* 12 */
+	0x0000, 0x007b, 0x0000, 0x0032,  /* 16 */
+	0x0000, 0x00c3, 0x00c3, 0x00c0,  /* 20 */
+	0x0000, 0x0000, 0x0000, 0x0000,  /* 24 */
+	0x0000, 0x0000, 0x0000, 0x0000,  /* 28 */
+	0x0000, 0x0000, 0x0050, 0x0050,  /* 32 */
+	0x0050, 0x0050, 0x0050, 0x0050,  /* 36 */
+	0x0079, 0x0079, 0x0079,          /* 40 */
+};
+
+/*
+ * read wm8750 register cache
+ */
+static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg > WM8750_CACHE_REGNUM)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write wm8750 register cache
+ */
+static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg > WM8750_CACHE_REGNUM)
+		return;
+	cache[reg] = value;
+}
+
+static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[2];
+
+	/* data is
+	 *   D15..D9 WM8753 register offset
+	 *   D8...D0 register data
+	 */
+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+	data[1] = value & 0x00ff;
+
+	wm8750_write_reg_cache (codec, reg, value);
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+#define wm8750_reset(c)	wm8750_write(c, WM8750_RESET, 0)
+
+/*
+ * WM8750 Controls
+ */
+static const char *wm8750_bass[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm8750_bass_filter[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
+static const char *wm8750_treble[] = {"8kHz", "4kHz"};
+static const char *wm8750_3d_lc[] = {"200Hz", "500Hz"};
+static const char *wm8750_3d_uc[] = {"2.2kHz", "1.5kHz"};
+static const char *wm8750_3d_func[] = {"Capture", "Playback"};
+static const char *wm8750_alc_func[] = {"Off", "Right", "Left", "Stereo"};
+static const char *wm8750_ng_type[] = {"Constant PGA Gain",
+	"Mute ADC Output"};
+static const char *wm8750_line_mux[] = {"Line 1", "Line 2", "Line 3", "PGA",
+	"Differential"};
+static const char *wm8750_pga_sel[] = {"Line 1", "Line 2", "Line 3",
+	"Differential"};
+static const char *wm8750_out3[] = {"VREF", "ROUT1 + Vol", "MonoOut",
+	"ROUT1"};
+static const char *wm8750_diff_sel[] = {"Line 1", "Line 2"};
+static const char *wm8750_adcpol[] = {"Normal", "L Invert", "R Invert",
+	"L + R Invert"};
+static const char *wm8750_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *wm8750_mono_mux[] = {"Stereo", "Mono (Left)",
+	"Mono (Right)", "Digital Mono"};
+
+static const struct soc_enum wm8750_enum[] = {
+SOC_ENUM_SINGLE(WM8750_BASS, 7, 2, wm8750_bass),
+SOC_ENUM_SINGLE(WM8750_BASS, 6, 2, wm8750_bass_filter),
+SOC_ENUM_SINGLE(WM8750_TREBLE, 6, 2, wm8750_treble),
+SOC_ENUM_SINGLE(WM8750_3D, 5, 2, wm8750_3d_lc),
+SOC_ENUM_SINGLE(WM8750_3D, 6, 2, wm8750_3d_uc),
+SOC_ENUM_SINGLE(WM8750_3D, 7, 2, wm8750_3d_func),
+SOC_ENUM_SINGLE(WM8750_ALC1, 7, 4, wm8750_alc_func),
+SOC_ENUM_SINGLE(WM8750_NGATE, 1, 2, wm8750_ng_type),
+SOC_ENUM_SINGLE(WM8750_LOUTM1, 0, 5, wm8750_line_mux),
+SOC_ENUM_SINGLE(WM8750_ROUTM1, 0, 5, wm8750_line_mux),
+SOC_ENUM_SINGLE(WM8750_LADCIN, 6, 4, wm8750_pga_sel), /* 10 */
+SOC_ENUM_SINGLE(WM8750_RADCIN, 6, 4, wm8750_pga_sel),
+SOC_ENUM_SINGLE(WM8750_ADCTL2, 7, 4, wm8750_out3),
+SOC_ENUM_SINGLE(WM8750_ADCIN, 8, 2, wm8750_diff_sel),
+SOC_ENUM_SINGLE(WM8750_ADCDAC, 5, 4, wm8750_adcpol),
+SOC_ENUM_SINGLE(WM8750_ADCDAC, 1, 4, wm8750_deemph),
+SOC_ENUM_SINGLE(WM8750_ADCIN, 6, 4, wm8750_mono_mux), /* 16 */
+
+};
+
+static const struct snd_kcontrol_new wm8750_snd_controls[] = {
+
+SOC_DOUBLE_R("Capture Volume", WM8750_LINVOL, WM8750_RINVOL, 0, 63, 0),
+SOC_DOUBLE_R("Capture ZC Switch", WM8750_LINVOL, WM8750_RINVOL, 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8750_LINVOL, WM8750_RINVOL, 7, 1, 1),
+
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8750_LOUT1V,
+	WM8750_ROUT1V, 7, 1, 0),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8750_LOUT2V,
+	WM8750_ROUT2V, 7, 1, 0),
+
+SOC_ENUM("Playback De-emphasis", wm8750_enum[15]),
+
+SOC_ENUM("Capture Polarity", wm8750_enum[14]),
+SOC_SINGLE("Playback 6dB Attenuate", WM8750_ADCDAC, 7, 1, 0),
+SOC_SINGLE("Capture 6dB Attenuate", WM8750_ADCDAC, 8, 1, 0),
+
+SOC_DOUBLE_R("PCM Volume", WM8750_LDAC, WM8750_RDAC, 0, 255, 0),
+
+SOC_ENUM("Bass Boost", wm8750_enum[0]),
+SOC_ENUM("Bass Filter", wm8750_enum[1]),
+SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1),
+
+SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 0),
+SOC_ENUM("Treble Cut-off", wm8750_enum[2]),
+
+SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0),
+SOC_SINGLE("3D Volume", WM8750_3D, 1, 15, 0),
+SOC_ENUM("3D Lower Cut-off", wm8750_enum[3]),
+SOC_ENUM("3D Upper Cut-off", wm8750_enum[4]),
+SOC_ENUM("3D Mode", wm8750_enum[5]),
+
+SOC_SINGLE("ALC Capture Target Volume", WM8750_ALC1, 0, 7, 0),
+SOC_SINGLE("ALC Capture Max Volume", WM8750_ALC1, 4, 7, 0),
+SOC_ENUM("ALC Capture Function", wm8750_enum[6]),
+SOC_SINGLE("ALC Capture ZC Switch", WM8750_ALC2, 7, 1, 0),
+SOC_SINGLE("ALC Capture Hold Time", WM8750_ALC2, 0, 15, 0),
+SOC_SINGLE("ALC Capture Decay Time", WM8750_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack Time", WM8750_ALC3, 0, 15, 0),
+SOC_SINGLE("ALC Capture NG Threshold", WM8750_NGATE, 3, 31, 0),
+SOC_ENUM("ALC Capture NG Type", wm8750_enum[4]),
+SOC_SINGLE("ALC Capture NG Switch", WM8750_NGATE, 0, 1, 0),
+
+SOC_SINGLE("Left ADC Capture Volume", WM8750_LADC, 0, 255, 0),
+SOC_SINGLE("Right ADC Capture Volume", WM8750_RADC, 0, 255, 0),
+
+SOC_SINGLE("ZC Timeout Switch", WM8750_ADCTL1, 0, 1, 0),
+SOC_SINGLE("Playback Invert Switch", WM8750_ADCTL1, 1, 1, 0),
+
+SOC_SINGLE("Right Speaker Playback Invert Switch", WM8750_ADCTL2, 4, 1, 0),
+
+/* Unimplemented */
+/* ADCDAC Bit 0 - ADCHPD */
+/* ADCDAC Bit 4 - HPOR */
+/* ADCTL1 Bit 2,3 - DATSEL */
+/* ADCTL1 Bit 4,5 - DMONOMIX */
+/* ADCTL1 Bit 6,7 - VSEL */
+/* ADCTL2 Bit 2 - LRCM */
+/* ADCTL2 Bit 3 - TRI */
+/* ADCTL3 Bit 5 - HPFLREN */
+/* ADCTL3 Bit 6 - VROI */
+/* ADCTL3 Bit 7,8 - ADCLRM */
+/* ADCIN Bit 4 - LDCM */
+/* ADCIN Bit 5 - RDCM */
+
+SOC_DOUBLE_R("Mic Boost", WM8750_LADCIN, WM8750_RADCIN, 4, 3, 0),
+
+SOC_DOUBLE_R("Bypass Left Playback Volume", WM8750_LOUTM1,
+	WM8750_LOUTM2, 4, 7, 1),
+SOC_DOUBLE_R("Bypass Right Playback Volume", WM8750_ROUTM1,
+	WM8750_ROUTM2, 4, 7, 1),
+SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8750_MOUTM1,
+	WM8750_MOUTM2, 4, 7, 1),
+
+SOC_SINGLE("Mono Playback ZC Switch", WM8750_MOUTV, 7, 1, 0),
+
+SOC_DOUBLE_R("Headphone Playback Volume", WM8750_LOUT1V, WM8750_ROUT1V,
+	0, 127, 0),
+SOC_DOUBLE_R("Speaker Playback Volume", WM8750_LOUT2V, WM8750_ROUT2V,
+	0, 127, 0),
+
+SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8750_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&wm8750_snd_controls[i],codec, NULL));
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8750_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("Playback Switch", WM8750_LOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_LOUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8750_LOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_LOUTM2, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new wm8750_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8750_ROUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_ROUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Playback Switch", WM8750_ROUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_ROUTM2, 7, 1, 0),
+};
+
+/* Mono Mixer */
+static const struct snd_kcontrol_new wm8750_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Left Playback Switch", WM8750_MOUTM1, 8, 1, 0),
+SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_MOUTM1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Playback Switch", WM8750_MOUTM2, 8, 1, 0),
+SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_MOUTM2, 7, 1, 0),
+};
+
+/* Left Line Mux */
+static const struct snd_kcontrol_new wm8750_left_line_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[8]);
+
+/* Right Line Mux */
+static const struct snd_kcontrol_new wm8750_right_line_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[9]);
+
+/* Left PGA Mux */
+static const struct snd_kcontrol_new wm8750_left_pga_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[10]);
+
+/* Right PGA Mux */
+static const struct snd_kcontrol_new wm8750_right_pga_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[11]);
+
+/* Out 3 Mux */
+static const struct snd_kcontrol_new wm8750_out3_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[12]);
+
+/* Differential Mux */
+static const struct snd_kcontrol_new wm8750_diffmux_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[13]);
+
+/* Mono ADC Mux */
+static const struct snd_kcontrol_new wm8750_monomux_controls =
+SOC_DAPM_ENUM("Route", wm8750_enum[16]);
+
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+		&wm8750_left_mixer_controls[0],
+		ARRAY_SIZE(wm8750_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+		&wm8750_right_mixer_controls[0],
+		ARRAY_SIZE(wm8750_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Mono Mixer", WM8750_PWR2, 2, 0,
+		&wm8750_mono_mixer_controls[0],
+		ARRAY_SIZE(wm8750_mono_mixer_controls)),
+
+	SND_SOC_DAPM_PGA("Right Out 2", WM8750_PWR2, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 2", WM8750_PWR2, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Out 1", WM8750_PWR2, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 1", WM8750_PWR2, 6, 0, NULL, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8750_PWR2, 7, 0),
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8750_PWR2, 8, 0),
+
+	SND_SOC_DAPM_MICBIAS("Mic Bias", WM8750_PWR1, 1, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8750_PWR1, 2, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8750_PWR1, 3, 0),
+
+	SND_SOC_DAPM_MUX("Left PGA Mux", WM8750_PWR1, 5, 0,
+		&wm8750_left_pga_controls),
+	SND_SOC_DAPM_MUX("Right PGA Mux", WM8750_PWR1, 4, 0,
+		&wm8750_right_pga_controls),
+	SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+		&wm8750_left_line_controls),
+	SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+		&wm8750_right_line_controls),
+
+	SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8750_out3_controls),
+	SND_SOC_DAPM_PGA("Out 3", WM8750_PWR2, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mono Out 1", WM8750_PWR2, 2, 0, NULL, 0),
+
+	SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+		&wm8750_diffmux_controls),
+	SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+		&wm8750_monomux_controls),
+	SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+		&wm8750_monomux_controls),
+
+	SND_SOC_DAPM_OUTPUT("LOUT1"),
+	SND_SOC_DAPM_OUTPUT("ROUT1"),
+	SND_SOC_DAPM_OUTPUT("LOUT2"),
+	SND_SOC_DAPM_OUTPUT("ROUT2"),
+	SND_SOC_DAPM_OUTPUT("MONO"),
+	SND_SOC_DAPM_OUTPUT("OUT3"),
+
+	SND_SOC_DAPM_INPUT("LINPUT1"),
+	SND_SOC_DAPM_INPUT("LINPUT2"),
+	SND_SOC_DAPM_INPUT("LINPUT3"),
+	SND_SOC_DAPM_INPUT("RINPUT1"),
+	SND_SOC_DAPM_INPUT("RINPUT2"),
+	SND_SOC_DAPM_INPUT("RINPUT3"),
+};
+
+static const char *audio_map[][3] = {
+	/* left mixer */
+	{"Left Mixer", "Playback Switch", "Left DAC"},
+	{"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
+	{"Left Mixer", "Right Playback Switch", "Right DAC"},
+	{"Left Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+	/* right mixer */
+	{"Right Mixer", "Left Playback Switch", "Left DAC"},
+	{"Right Mixer", "Left Bypass Switch", "Left Line Mux"},
+	{"Right Mixer", "Playback Switch", "Right DAC"},
+	{"Right Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+	/* left out 1 */
+	{"Left Out 1", NULL, "Left Mixer"},
+	{"LOUT1", NULL, "Left Out 1"},
+
+	/* left out 2 */
+	{"Left Out 2", NULL, "Left Mixer"},
+	{"LOUT2", NULL, "Left Out 2"},
+
+	/* right out 1 */
+	{"Right Out 1", NULL, "Right Mixer"},
+	{"ROUT1", NULL, "Right Out 1"},
+
+	/* right out 2 */
+	{"Right Out 2", NULL, "Right Mixer"},
+	{"ROUT2", NULL, "Right Out 2"},
+
+	/* mono mixer */
+	{"Mono Mixer", "Left Playback Switch", "Left DAC"},
+	{"Mono Mixer", "Left Bypass Switch", "Left Line Mux"},
+	{"Mono Mixer", "Right Playback Switch", "Right DAC"},
+	{"Mono Mixer", "Right Bypass Switch", "Right Line Mux"},
+
+	/* mono out */
+	{"Mono Out 1", NULL, "Mono Mixer"},
+	{"MONO1", NULL, "Mono Out 1"},
+
+	/* out 3 */
+	{"Out3 Mux", "VREF", "VREF"},
+	{"Out3 Mux", "ROUT1 + Vol", "ROUT1"},
+	{"Out3 Mux", "ROUT1", "Right Mixer"},
+	{"Out3 Mux", "MonoOut", "MONO1"},
+	{"Out 3", NULL, "Out3 Mux"},
+	{"OUT3", NULL, "Out 3"},
+
+	/* Left Line Mux */
+	{"Left Line Mux", "Line 1", "LINPUT1"},
+	{"Left Line Mux", "Line 2", "LINPUT2"},
+	{"Left Line Mux", "Line 3", "LINPUT3"},
+	{"Left Line Mux", "PGA", "Left PGA Mux"},
+	{"Left Line Mux", "Differential", "Differential Mux"},
+
+	/* Right Line Mux */
+	{"Right Line Mux", "Line 1", "RINPUT1"},
+	{"Right Line Mux", "Line 2", "RINPUT2"},
+	{"Right Line Mux", "Line 3", "RINPUT3"},
+	{"Right Line Mux", "PGA", "Right PGA Mux"},
+	{"Right Line Mux", "Differential", "Differential Mux"},
+
+	/* Left PGA Mux */
+	{"Left PGA Mux", "Line 1", "LINPUT1"},
+	{"Left PGA Mux", "Line 2", "LINPUT2"},
+	{"Left PGA Mux", "Line 3", "LINPUT3"},
+	{"Left PGA Mux", "Differential", "Differential Mux"},
+
+	/* Right PGA Mux */
+	{"Right PGA Mux", "Line 1", "RINPUT1"},
+	{"Right PGA Mux", "Line 2", "RINPUT2"},
+	{"Right PGA Mux", "Line 3", "RINPUT3"},
+	{"Right PGA Mux", "Differential", "Differential Mux"},
+
+	/* Differential Mux */
+	{"Differential Mux", "Line 1", "LINPUT1"},
+	{"Differential Mux", "Line 1", "RINPUT1"},
+	{"Differential Mux", "Line 2", "LINPUT2"},
+	{"Differential Mux", "Line 2", "RINPUT2"},
+
+	/* Left ADC Mux */
+	{"Left ADC Mux", "Stereo", "Left PGA Mux"},
+	{"Left ADC Mux", "Mono (Left)", "Left PGA Mux"},
+	{"Left ADC Mux", "Digital Mono", "Left PGA Mux"},
+
+	/* Right ADC Mux */
+	{"Right ADC Mux", "Stereo", "Right PGA Mux"},
+	{"Right ADC Mux", "Mono (Right)", "Right PGA Mux"},
+	{"Right ADC Mux", "Digital Mono", "Right PGA Mux"},
+
+	/* ADC */
+	{"Left ADC", NULL, "Left ADC Mux"},
+	{"Right ADC", NULL, "Right ADC Mux"},
+
+	/* terminator */
+	{NULL, NULL, NULL},
+};
+
+static int wm8750_add_widgets(struct snd_soc_codec *codec)
+{
+	int i;
+
+	for(i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+	}
+
+	/* set up audio path audio_mapnects */
+	for(i = 0; audio_map[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+			audio_map[i][1], audio_map[i][2]);
+	}
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+struct _coeff_div {
+	u32 mclk;
+	u32 rate;
+	u16 fs;
+	u8 sr:5;
+	u8 usb:1;
+};
+
+/* codec hifi mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+	/* 8k */
+	{12288000, 8000, 1536, 0x6, 0x0},
+	{11289600, 8000, 1408, 0x16, 0x0},
+	{18432000, 8000, 2304, 0x7, 0x0},
+	{16934400, 8000, 2112, 0x17, 0x0},
+	{12000000, 8000, 1500, 0x6, 0x1},
+
+	/* 11.025k */
+	{11289600, 11025, 1024, 0x18, 0x0},
+	{16934400, 11025, 1536, 0x19, 0x0},
+	{12000000, 11025, 1088, 0x19, 0x1},
+
+	/* 16k */
+	{12288000, 16000, 768, 0xa, 0x0},
+	{18432000, 16000, 1152, 0xb, 0x0},
+	{12000000, 16000, 750, 0xa, 0x1},
+
+	/* 22.05k */
+	{11289600, 22050, 512, 0x1a, 0x0},
+	{16934400, 22050, 768, 0x1b, 0x0},
+	{12000000, 22050, 544, 0x1b, 0x1},
+
+	/* 32k */
+	{12288000, 32000, 384, 0xc, 0x0},
+	{18432000, 32000, 576, 0xd, 0x0},
+	{12000000, 32000, 375, 0xa, 0x1},
+
+	/* 44.1k */
+	{11289600, 44100, 256, 0x10, 0x0},
+	{16934400, 44100, 384, 0x11, 0x0},
+	{12000000, 44100, 272, 0x11, 0x1},
+
+	/* 48k */
+	{12288000, 48000, 256, 0x0, 0x0},
+	{18432000, 48000, 384, 0x1, 0x0},
+	{12000000, 48000, 250, 0x0, 0x1},
+
+	/* 88.2k */
+	{11289600, 88200, 128, 0x1e, 0x0},
+	{16934400, 88200, 192, 0x1f, 0x0},
+	{12000000, 88200, 136, 0x1f, 0x1},
+
+	/* 96k */
+	{12288000, 96000, 128, 0xe, 0x0},
+	{18432000, 96000, 192, 0xf, 0x0},
+	{12000000, 96000, 125, 0xe, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+			return i;
+	}
+
+	printk(KERN_ERR "wm8750: could not get coeff for mclk %d @ rate %d\n",
+		mclk, rate);
+	return -EINVAL;
+}
+
+static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8750_priv *wm8750 = codec->private_data;
+
+	switch (freq) {
+	case 11289600:
+	case 12000000:
+	case 12288000:
+	case 16934400:
+	case 18432000:
+		wm8750->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = 0;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface = 0x0040;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 0x0002;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x0001;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x0003;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= 0x0013;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= 0x0090;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= 0x0080;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x0010;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8750_write(codec, WM8750_IFACE, iface);
+	return 0;
+}
+
+static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct wm8750_priv *wm8750 = codec->private_data;
+	u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3;
+	u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0;
+	int coeff = get_coeff(wm8750->sysclk, params_rate(params));
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x0004;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x0008;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= 0x000c;
+		break;
+	}
+
+	/* set iface & srate */
+	wm8750_write(codec, WM8750_IFACE, iface);
+	if (coeff >= 0)
+		wm8750_write(codec, WM8750_SRATE, srate |
+			(coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
+
+	return 0;
+}
+
+static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
+
+	if (mute)
+		wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8);
+	else
+		wm8750_write(codec, WM8750_ADCDAC, mute_reg);
+	return 0;
+}
+
+static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
+{
+	u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e;
+
+	switch (event) {
+	case SNDRV_CTL_POWER_D0: /* full On */
+		/* set vmid to 50k and unmute dac */
+		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
+		break;
+	case SNDRV_CTL_POWER_D1: /* partial On */
+	case SNDRV_CTL_POWER_D2: /* partial On */
+		/* set vmid to 5k for quick power up */
+		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
+		break;
+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+		/* mute dac and set vmid to 500k, enable VREF */
+		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
+		break;
+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+		wm8750_write(codec, WM8750_PWR1, 0x0001);
+		break;
+	}
+	codec->dapm_state = event;
+	return 0;
+}
+
+#define WM8750_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_codec_dai wm8750_dai = {
+	.name = "WM8750",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8750_RATES,
+		.formats = WM8750_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8750_RATES,
+		.formats = WM8750_FORMATS,},
+	.ops = {
+		.hw_params = wm8750_pcm_hw_params,
+	},
+	.dai_ops = {
+		.digital_mute = wm8750_mute,
+		.set_fmt = wm8750_set_dai_fmt,
+		.set_sysclk = wm8750_set_dai_sysclk,
+	},
+};
+EXPORT_SYMBOL_GPL(wm8750_dai);
+
+static void wm8750_work(struct work_struct *work)
+{
+	struct snd_soc_codec *codec =
+		container_of(work, struct snd_soc_codec, delayed_work.work);
+	wm8750_dapm_event(codec, codec->dapm_state);
+}
+
+static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	return 0;
+}
+
+static int wm8750_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+	u8 data[2];
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) {
+		if (i == WM8750_RESET)
+			continue;
+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+		data[1] = cache[i] & 0x00ff;
+		codec->hw_write(codec->control_data, data, 2);
+	}
+
+	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+	/* charge wm8750 caps */
+	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
+		wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
+		codec->dapm_state = SNDRV_CTL_POWER_D0;
+		schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+	}
+
+	return 0;
+}
+
+/*
+ * initialise the WM8750 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8750_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int reg, ret = 0;
+
+	codec->name = "WM8750";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8750_read_reg_cache;
+	codec->write = wm8750_write;
+	codec->dapm_event = wm8750_dapm_event;
+	codec->dai = &wm8750_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = sizeof(wm8750_reg);
+	codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KRENEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	wm8750_reset(codec);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8750: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* charge output caps */
+	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
+	codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+	schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+
+	/* set the update bits */
+	reg = wm8750_read_reg_cache(codec, WM8750_LDAC);
+	wm8750_write(codec, WM8750_LDAC, reg | 0x0100);
+	reg = wm8750_read_reg_cache(codec, WM8750_RDAC);
+	wm8750_write(codec, WM8750_RDAC, reg | 0x0100);
+	reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V);
+	wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100);
+	reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V);
+	wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100);
+	reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V);
+	wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100);
+	reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V);
+	wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100);
+	reg = wm8750_read_reg_cache(codec, WM8750_LINVOL);
+	wm8750_write(codec, WM8750_LINVOL, reg | 0x0100);
+	reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
+	wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
+
+	wm8750_add_controls(codec);
+	wm8750_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8750: failed to register card\n");
+		goto card_err;
+	}
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+	return ret;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+   around */
+static struct snd_soc_device *wm8750_socdev;
+
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+
+/*
+ * WM8731 2 wire address is determined by GPIO5
+ * state during powerup.
+ *    low  = 0x1a
+ *    high = 0x1b
+ */
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8750_i2c_driver;
+static struct i2c_client client_template;
+
+static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+	struct snd_soc_device *socdev = wm8750_socdev;
+	struct wm8750_setup_data *setup = socdev->codec_data;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct i2c_client *i2c;
+	int ret;
+
+	if (addr != setup->i2c_address)
+		return -ENODEV;
+
+	client_template.adapter = adap;
+	client_template.addr = addr;
+
+	i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL);
+	if (i2c == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = i2c_attach_client(i2c);
+	if (ret < 0) {
+		err("failed to attach codec at addr %x\n", addr);
+		goto err;
+	}
+
+	ret = wm8750_init(socdev);
+	if (ret < 0) {
+	err("failed to initialise WM8750\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+
+static int wm8750_i2c_detach(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	i2c_detach_client(client);
+	kfree(codec->reg_cache);
+	kfree(client);
+	return 0;
+}
+
+static int wm8750_i2c_attach(struct i2c_adapter *adap)
+{
+	return i2c_probe(adap, &addr_data, wm8750_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver wm8750_i2c_driver = {
+	.driver = {
+		.name = "WM8750 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.id =             I2C_DRIVERID_WM8750,
+	.attach_adapter = wm8750_i2c_attach,
+	.detach_client =  wm8750_i2c_detach,
+	.command =        NULL,
+};
+
+static struct i2c_client client_template = {
+	.name =   "WM8750",
+	.driver = &wm8750_i2c_driver,
+};
+#endif
+
+static int wm8750_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct wm8750_setup_data *setup = socdev->codec_data;
+	struct snd_soc_codec *codec;
+	struct wm8750_priv *wm8750;
+	int ret = 0;
+
+	info("WM8750 Audio Codec %s", WM8750_VERSION);
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL);
+	if (wm8750 == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+
+	codec->private_data = wm8750;
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	wm8750_socdev = socdev;
+	INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work);
+	
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+	if (setup->i2c_address) {
+		normal_i2c[0] = setup->i2c_address;
+		codec->hw_write = (hw_write_t)i2c_master_send;
+		ret = i2c_add_driver(&wm8750_i2c_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add i2c driver");
+	}
+#else
+		/* Add other interfaces here */
+#endif
+
+	return ret;
+}
+
+/*
+ * This function forces any delayed work to be queued and run.
+ */
+static int run_delayed_work(struct delayed_work *dwork)
+{
+	int ret;
+
+	/* cancel any work waiting to be queued. */
+	ret = cancel_delayed_work(dwork);
+
+	/* if there was any work waiting then we run it now and
+	 * wait for it's completion */
+	if (ret) {
+		schedule_delayed_work(dwork, 0);
+		flush_scheduled_work();
+	}
+	return ret;
+}
+
+/* power down chip */
+static int wm8750_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec->control_data)
+		wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	run_delayed_work(&codec->delayed_work);
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8750_i2c_driver);
+#endif
+	kfree(codec->private_data);
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8750 = {
+	.probe = 	wm8750_probe,
+	.remove = 	wm8750_remove,
+	.suspend = 	wm8750_suspend,
+	.resume =	wm8750_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
+
+MODULE_DESCRIPTION("ASoC WM8750 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h
new file mode 100644
index 00000000000..a97a54a6348
--- /dev/null
+++ b/sound/soc/codecs/wm8750.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _WM8750_H
+#define _WM8750_H
+
+/* WM8750 register space */
+
+#define WM8750_LINVOL    0x00
+#define WM8750_RINVOL    0x01
+#define WM8750_LOUT1V    0x02
+#define WM8750_ROUT1V    0x03
+#define WM8750_ADCDAC    0x05
+#define WM8750_IFACE     0x07
+#define WM8750_SRATE     0x08
+#define WM8750_LDAC      0x0a
+#define WM8750_RDAC      0x0b
+#define WM8750_BASS      0x0c
+#define WM8750_TREBLE    0x0d
+#define WM8750_RESET     0x0f
+#define WM8750_3D        0x10
+#define WM8750_ALC1      0x11
+#define WM8750_ALC2      0x12
+#define WM8750_ALC3      0x13
+#define WM8750_NGATE     0x14
+#define WM8750_LADC      0x15
+#define WM8750_RADC      0x16
+#define WM8750_ADCTL1    0x17
+#define WM8750_ADCTL2    0x18
+#define WM8750_PWR1      0x19
+#define WM8750_PWR2      0x1a
+#define WM8750_ADCTL3    0x1b
+#define WM8750_ADCIN     0x1f
+#define WM8750_LADCIN    0x20
+#define WM8750_RADCIN    0x21
+#define WM8750_LOUTM1    0x22
+#define WM8750_LOUTM2    0x23
+#define WM8750_ROUTM1    0x24
+#define WM8750_ROUTM2    0x25
+#define WM8750_MOUTM1    0x26
+#define WM8750_MOUTM2    0x27
+#define WM8750_LOUT2V    0x28
+#define WM8750_ROUT2V    0x29
+#define WM8750_MOUTV     0x2a
+
+#define WM8750_CACHE_REGNUM 0x2a
+
+#define WM8750_SYSCLK	0
+
+struct wm8750_setup_data {
+	unsigned short i2c_address;
+};
+
+extern struct snd_soc_codec_dai wm8750_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8750;
+
+#endif
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
new file mode 100644
index 00000000000..92a64871bcd
--- /dev/null
+++ b/sound/soc/codecs/wm9712.c
@@ -0,0 +1,771 @@
+/*
+ * wm9712.c  --  ALSA Soc WM9712 codec support
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    4th Feb 2006   Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#define WM9712_VERSION "0.4"
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+	unsigned int reg);
+static int ac97_write(struct snd_soc_codec *codec,
+	unsigned int reg, unsigned int val);
+
+/*
+ * WM9712 register cache
+ */
+static const u16 wm9712_reg[] = {
+	0x6174, 0x8000, 0x8000, 0x8000, // 6
+	0xf0f0, 0xaaa0, 0xc008, 0x6808, // e
+	0xe808, 0xaaa0, 0xad00, 0x8000, // 16
+	0xe808, 0x3000, 0x8000, 0x0000, // 1e
+	0x0000, 0x0000, 0x0000, 0x000f, // 26
+	0x0405, 0x0410, 0xbb80, 0xbb80, // 2e
+	0x0000, 0xbb80, 0x0000, 0x0000, // 36
+	0x0000, 0x2000, 0x0000, 0x0000, // 3e
+	0x0000, 0x0000, 0x0000, 0x0000, // 46
+	0x0000, 0x0000, 0xf83e, 0xffff, // 4e
+	0x0000, 0x0000, 0x0000, 0xf83e, // 56
+	0x0008, 0x0000, 0x0000, 0x0000, // 5e
+	0xb032, 0x3e00, 0x0000, 0x0000, // 66
+	0x0000, 0x0000, 0x0000, 0x0000, // 6e
+	0x0000, 0x0000, 0x0000, 0x0006, // 76
+	0x0001, 0x0000, 0x574d, 0x4c12, // 7e
+	0x0000, 0x0000 // virtual hp mixers
+};
+
+/* virtual HP mixers regs */
+#define HPL_MIXER	0x80
+#define HPR_MIXER	0x82
+
+static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"};
+static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"};
+static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right",
+	"Mono"};
+static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"};
+static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"};
+static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"};
+static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
+static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2",
+	"Stereo"};
+static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer",
+	"Line", "Headphone Mixer", "Phone Mixer", "Phone"};
+static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"};
+static const char *wm9712_diff_sel[] = {"Mic", "Line"};
+
+static const struct soc_enum wm9712_enum[] = {
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select),
+SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux),
+SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src),
+SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc),
+SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base),
+SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain),
+SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel),
+SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel),
+SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type),
+SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel),
+};
+
+static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
+SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
+SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1),
+
+SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
+SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0),
+SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0),
+SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
+SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0),
+
+SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
+SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
+SOC_ENUM("ALC Function", wm9712_enum[0]),
+SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
+SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1),
+SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
+SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
+SOC_ENUM("ALC NG Type", wm9712_enum[10]),
+SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1),
+
+SOC_SINGLE("Mic Headphone  Volume", AC97_VIDEO, 12, 7, 1),
+SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1),
+
+SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1),
+SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1),
+SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1),
+
+SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1),
+SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1),
+SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1),
+
+SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1),
+SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1),
+SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1),
+
+SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 0),
+SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1),
+
+SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
+SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
+
+SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
+SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
+SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0),
+
+SOC_ENUM("Bass Control", wm9712_enum[5]),
+SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1),
+SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1),
+SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
+SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0),
+SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0),
+
+SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1),
+SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
+SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1),
+SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
+
+SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
+SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
+};
+
+/* add non dapm controls */
+static int wm9712_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL));
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+/* We have to create a fake left and right HP mixers because
+ * the codec only has a single control that is shared by both channels.
+ * This makes it impossible to determine the audio path.
+ */
+static int mixer_event (struct snd_soc_dapm_widget *w, int event)
+{
+	u16 l, r, beep, line, phone, mic, pcm, aux;
+
+	l = ac97_read(w->codec, HPL_MIXER);
+	r = ac97_read(w->codec, HPR_MIXER);
+	beep = ac97_read(w->codec, AC97_PC_BEEP);
+	mic = ac97_read(w->codec, AC97_VIDEO);
+	phone = ac97_read(w->codec, AC97_PHONE);
+	line = ac97_read(w->codec, AC97_LINE);
+	pcm = ac97_read(w->codec, AC97_PCM);
+	aux = ac97_read(w->codec, AC97_CD);
+
+	if (l & 0x1 || r & 0x1)
+		ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff);
+	else
+		ac97_write(w->codec, AC97_VIDEO, mic | 0x8000);
+
+	if (l & 0x2 || r & 0x2)
+		ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
+	else
+		ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
+
+	if (l & 0x4 || r & 0x4)
+		ac97_write(w->codec, AC97_LINE, line & 0x7fff);
+	else
+		ac97_write(w->codec, AC97_LINE, line | 0x8000);
+
+	if (l & 0x8 || r & 0x8)
+		ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
+	else
+		ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
+
+	if (l & 0x10 || r & 0x10)
+		ac97_write(w->codec, AC97_CD, aux & 0x7fff);
+	else
+		ac97_write(w->codec, AC97_CD, aux | 0x8000);
+
+	if (l & 0x20 || r & 0x20)
+		ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
+	else
+		ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
+
+	return 0;
+}
+
+/* Left Headphone Mixers */
+static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
+	SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0),
+	SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0),
+	SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0),
+	SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0),
+	SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0),
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0),
+};
+
+/* Right Headphone Mixers */
+static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
+	SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0),
+	SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0),
+	SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0),
+	SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0),
+	SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0),
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0),
+};
+
+/* Speaker Mixer */
+static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = {
+	SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1),
+	SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1),
+	SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1),
+	SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1),
+	SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1),
+};
+
+/* Phone Mixer */
+static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = {
+	SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1),
+	SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1),
+	SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1),
+	SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1),
+	SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1),
+	SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1),
+};
+
+/* ALC headphone mux */
+static const struct snd_kcontrol_new wm9712_alc_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[1]);
+
+/* out 3 mux */
+static const struct snd_kcontrol_new wm9712_out3_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[2]);
+
+/* spk mux */
+static const struct snd_kcontrol_new wm9712_spk_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[3]);
+
+/* Capture to Phone mux */
+static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[4]);
+
+/* Capture left select */
+static const struct snd_kcontrol_new wm9712_capture_selectl_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[8]);
+
+/* Capture right select */
+static const struct snd_kcontrol_new wm9712_capture_selectr_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[9]);
+
+/* Mic select */
+static const struct snd_kcontrol_new wm9712_mic_src_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[7]);
+
+/* diff select */
+static const struct snd_kcontrol_new wm9712_diff_sel_controls =
+SOC_DAPM_ENUM("Route", wm9712_enum[11]);
+
+static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = {
+SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0,
+	&wm9712_alc_mux_controls),
+SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0,
+	&wm9712_out3_mux_controls),
+SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0,
+	&wm9712_spk_mux_controls),
+SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0,
+	&wm9712_capture_phone_mux_controls),
+SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
+	&wm9712_capture_selectl_controls),
+SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
+	&wm9712_capture_selectr_controls),
+SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0,
+	&wm9712_mic_src_controls),
+SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
+	&wm9712_diff_sel_controls),
+SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1,
+	&wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls),
+	mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1,
+	&wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls),
+	 mixer_event, SND_SOC_DAPM_POST_REG),
+SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1,
+	&wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)),
+SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1,
+	&wm9712_speaker_mixer_controls[0],
+	ARRAY_SIZE(wm9712_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1),
+SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1),
+SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1),
+SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1),
+SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
+SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
+SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
+SND_SOC_DAPM_OUTPUT("MONOOUT"),
+SND_SOC_DAPM_OUTPUT("HPOUTL"),
+SND_SOC_DAPM_OUTPUT("HPOUTR"),
+SND_SOC_DAPM_OUTPUT("LOUT2"),
+SND_SOC_DAPM_OUTPUT("ROUT2"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_INPUT("LINEINL"),
+SND_SOC_DAPM_INPUT("LINEINR"),
+SND_SOC_DAPM_INPUT("PHONE"),
+SND_SOC_DAPM_INPUT("PCBEEP"),
+SND_SOC_DAPM_INPUT("MIC1"),
+SND_SOC_DAPM_INPUT("MIC2"),
+};
+
+static const char *audio_map[][3] = {
+	/* virtual mixer - mixes left & right channels for spk and mono */
+	{"AC97 Mixer", NULL, "Left DAC"},
+	{"AC97 Mixer", NULL, "Right DAC"},
+
+	/* Left HP mixer */
+	{"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+	{"Left HP Mixer", "Aux Playback Switch",  "Aux DAC"},
+	{"Left HP Mixer", "Phone Bypass Switch",  "Phone PGA"},
+	{"Left HP Mixer", "Line Bypass Switch",   "Line PGA"},
+	{"Left HP Mixer", "PCM Playback Switch",  "Left DAC"},
+	{"Left HP Mixer", "Mic Sidetone Switch",  "Mic PGA"},
+	{"Left HP Mixer", NULL,  "ALC Sidetone Mux"},
+	//{"Right HP Mixer", NULL, "HP Mixer"},
+
+	/* Right HP mixer */
+	{"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+	{"Right HP Mixer", "Aux Playback Switch",  "Aux DAC"},
+	{"Right HP Mixer", "Phone Bypass Switch",  "Phone PGA"},
+	{"Right HP Mixer", "Line Bypass Switch",   "Line PGA"},
+	{"Right HP Mixer", "PCM Playback Switch",  "Right DAC"},
+	{"Right HP Mixer", "Mic Sidetone Switch",  "Mic PGA"},
+	{"Right HP Mixer", NULL,  "ALC Sidetone Mux"},
+
+	/* speaker mixer */
+	{"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"},
+	{"Speaker Mixer", "Line Bypass Switch",   "Line PGA"},
+	{"Speaker Mixer", "PCM Playback Switch",  "AC97 Mixer"},
+	{"Speaker Mixer", "Phone Bypass Switch",  "Phone PGA"},
+	{"Speaker Mixer", "Aux Playback Switch",  "Aux DAC"},
+
+	/* Phone mixer */
+	{"Phone Mixer", "PCBeep Bypass Switch",  "PCBEEP"},
+	{"Phone Mixer", "Line Bypass Switch",    "Line PGA"},
+	{"Phone Mixer", "Aux Playback Switch",   "Aux DAC"},
+	{"Phone Mixer", "PCM Playback Switch",   "AC97 Mixer"},
+	{"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"},
+	{"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"},
+
+	/* inputs */
+	{"Line PGA", NULL, "LINEINL"},
+	{"Line PGA", NULL, "LINEINR"},
+	{"Phone PGA", NULL, "PHONE"},
+	{"Mic PGA", NULL, "MIC1"},
+	{"Mic PGA", NULL, "MIC2"},
+
+	/* left capture selector */
+	{"Left Capture Select", "Mic", "MIC1"},
+	{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
+	{"Left Capture Select", "Line", "LINEINL"},
+	{"Left Capture Select", "Headphone Mixer", "Left HP Mixer"},
+	{"Left Capture Select", "Phone Mixer", "Phone Mixer"},
+	{"Left Capture Select", "Phone", "PHONE"},
+
+	/* right capture selector */
+	{"Right Capture Select", "Mic", "MIC2"},
+	{"Right Capture Select", "Speaker Mixer", "Speaker Mixer"},
+	{"Right Capture Select", "Line", "LINEINR"},
+	{"Right Capture Select", "Headphone Mixer", "Right HP Mixer"},
+	{"Right Capture Select", "Phone Mixer", "Phone Mixer"},
+	{"Right Capture Select", "Phone", "PHONE"},
+
+	/* ALC Sidetone */
+	{"ALC Sidetone Mux", "Stereo", "Left Capture Select"},
+	{"ALC Sidetone Mux", "Stereo", "Right Capture Select"},
+	{"ALC Sidetone Mux", "Left", "Left Capture Select"},
+	{"ALC Sidetone Mux", "Right", "Right Capture Select"},
+
+	/* ADC's */
+	{"Left ADC", NULL, "Left Capture Select"},
+	{"Right ADC", NULL, "Right Capture Select"},
+
+	/* outputs */
+	{"MONOOUT", NULL, "Phone Mixer"},
+	{"HPOUTL", NULL, "Headphone PGA"},
+	{"Headphone PGA", NULL, "Left HP Mixer"},
+	{"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"},
+
+	/* Out3 Mux */
+	{"Out3 Mux", "Left", "Left HP Mixer"},
+	{"Out3 Mux", "Mono", "Phone Mixer"},
+	{"Out3 Mux", "Left + Right", "Mono HP 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 PGA", NULL, "Speaker Mux"},
+	{"LOUT2", NULL, "Speaker PGA"},
+	{"ROUT2", NULL, "Speaker PGA"},
+
+	{NULL, NULL, NULL},
+};
+
+static int wm9712_add_widgets(struct snd_soc_codec *codec)
+{
+	int i;
+
+	for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
+	}
+
+	/* set up audio path audio_mapnects */
+	for(i = 0; audio_map[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+			audio_map[i][1], audio_map[i][2]);
+	}
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static unsigned int ac97_read(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+		reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
+		reg == AC97_REC_GAIN)
+		return soc_ac97_ops.read(codec->ac97, reg);
+	else {
+		reg = reg >> 1;
+
+		if (reg > (ARRAY_SIZE(wm9712_reg)))
+			return -EIO;
+
+		return cache[reg];
+	}
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int val)
+{
+	u16 *cache = codec->reg_cache;
+
+	soc_ac97_ops.write(codec->ac97, reg, val);
+	reg = reg >> 1;
+	if (reg <= (ARRAY_SIZE(wm9712_reg)))
+		cache[reg] = val;
+
+	return 0;
+}
+
+static int ac97_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int reg;
+	u16 vra;
+
+	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = AC97_PCM_FRONT_DAC_RATE;
+	else
+		reg = AC97_PCM_LR_ADC_RATE;
+
+	return ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 vra, xsle;
+
+	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+	xsle = ac97_read(codec, AC97_PCI_SID);
+	ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -ENODEV;
+
+	return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
+}
+
+#define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_codec_dai wm9712_dai[] = {
+{
+	.name = "AC97 HiFi",
+	.playback = {
+		.stream_name = "HiFi Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM9712_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "HiFi Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM9712_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = {
+		.prepare = ac97_prepare,},
+},
+{
+	.name = "AC97 Aux",
+	.playback = {
+		.stream_name = "Aux Playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = WM9712_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = {
+		.prepare = ac97_aux_prepare,},
+}
+};
+EXPORT_SYMBOL_GPL(wm9712_dai);
+
+static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
+{
+	u16 reg;
+
+	switch (event) {
+	case SNDRV_CTL_POWER_D0: /* full On */
+		/* liam - maybe enable thermal shutdown */
+		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff;
+		ac97_write(codec, AC97_EXTENDED_MID, reg);
+		break;
+	case SNDRV_CTL_POWER_D1: /* partial On */
+	case SNDRV_CTL_POWER_D2: /* partial On */
+		break;
+	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+		/* enable master bias and vmid */
+		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff;
+		ac97_write(codec, AC97_EXTENDED_MID, reg);
+		ac97_write(codec, AC97_POWERDOWN, 0x0000);
+		break;
+	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+		/* disable everything including AC link */
+		ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
+		ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
+		ac97_write(codec, AC97_POWERDOWN, 0xffff);
+		break;
+	}
+	codec->dapm_state = event;
+	return 0;
+}
+
+static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
+{
+	if (try_warm && soc_ac97_ops.warm_reset) {
+		soc_ac97_ops.warm_reset(codec->ac97);
+		if (!(ac97_read(codec, 0) & 0x8000))
+			return 1;
+	}
+
+	soc_ac97_ops.reset(codec->ac97);
+	if (ac97_read(codec, 0) & 0x8000)
+		goto err;
+	return 0;
+
+err:
+	printk(KERN_ERR "WM9712 AC97 reset failed\n");
+	return -EIO;
+}
+
+static int wm9712_soc_suspend(struct platform_device *pdev,
+	pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	return 0;
+}
+
+static int wm9712_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i, ret;
+	u16 *cache = codec->reg_cache;
+
+	ret = wm9712_reset(codec, 1);
+	if (ret < 0){
+		printk(KERN_ERR "could not reset AC97 codec\n");
+		return ret;
+	}
+
+	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+
+	if (ret == 0) {
+		/* Sync reg_cache with the hardware after cold reset */
+		for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) {
+			if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
+				(i > 0x58 && i != 0x5c))
+				continue;
+			soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+		}
+	}
+
+	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
+		wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0);
+
+	return ret;
+}
+
+static int wm9712_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->codec == NULL)
+		return -ENOMEM;
+	codec = socdev->codec;
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache =
+		kzalloc(sizeof(u16) * ARRAY_SIZE(wm9712_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL) {
+		ret = -ENOMEM;
+		goto cache_err;
+	}
+	memcpy(codec->reg_cache, wm9712_reg, sizeof(u16) * ARRAY_SIZE(wm9712_reg));
+	codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9712_reg);
+	codec->reg_cache_step = 2;
+
+	codec->name = "WM9712";
+	codec->owner = THIS_MODULE;
+	codec->dai = wm9712_dai;
+	codec->num_dai = ARRAY_SIZE(wm9712_dai);
+	codec->write = ac97_write;
+	codec->read = ac97_read;
+	codec->dapm_event = wm9712_dapm_event;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+	if (ret < 0) {
+		printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
+		goto codec_err;
+	}
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0)
+		goto pcm_err;
+
+	ret = wm9712_reset(codec, 0);
+	if (ret < 0) {
+		printk(KERN_ERR "AC97 link error\n");
+		goto reset_err;
+	}
+
+	/* set alc mux to none */
+	ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
+
+	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9712_add_controls(codec);
+	wm9712_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm9712: failed to register card\n");
+		goto reset_err;
+	}
+
+	return 0;
+
+reset_err:
+	snd_soc_free_pcms(socdev);
+
+pcm_err:
+	snd_soc_free_ac97_codec(codec);
+
+codec_err:
+	kfree(codec->reg_cache);
+
+cache_err:
+	kfree(socdev->codec);
+	socdev->codec = NULL;
+	return ret;
+}
+
+static int wm9712_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec == NULL)
+		return 0;
+
+	snd_soc_dapm_free(socdev);
+	snd_soc_free_pcms(socdev);
+	snd_soc_free_ac97_codec(codec);
+	kfree(codec->reg_cache);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9712 = {
+	.probe = 	wm9712_soc_probe,
+	.remove = 	wm9712_soc_remove,
+	.suspend =	wm9712_soc_suspend,
+	.resume =	wm9712_soc_resume,
+};
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
+
+MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h
new file mode 100644
index 00000000000..719105d61e6
--- /dev/null
+++ b/sound/soc/codecs/wm9712.h
@@ -0,0 +1,14 @@
+/*
+ * wm9712.h  --  WM9712 Soc Audio driver
+ */
+
+#ifndef _WM9712_H
+#define _WM9712_H
+
+#define WM9712_DAI_AC97_HIFI	0
+#define WM9712_DAI_AC97_AUX		1
+
+extern struct snd_soc_codec_dai wm9712_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm9712;
+
+#endif
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
new file mode 100644
index 00000000000..579e1c8d2b2
--- /dev/null
+++ b/sound/soc/pxa/Kconfig
@@ -0,0 +1,60 @@
+menu "SoC Audio for the Intel PXA2xx"
+
+config SND_PXA2XX_SOC
+	tristate "SoC Audio for the Intel PXA2xx chip"
+	depends on ARCH_PXA && SND
+	select SND_PCM
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the PXA2xx AC97, I2S or SSP interface. You will also need
+	  to select the audio interfaces to support below.
+
+config SND_PXA2XX_AC97
+	tristate
+	select SND_AC97_CODEC
+
+config SND_PXA2XX_SOC_AC97
+	tristate
+	select AC97_BUS
+	select SND_SOC_AC97_BUS
+
+config SND_PXA2XX_SOC_I2S
+	tristate
+
+config SND_PXA2XX_SOC_CORGI
+	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
+	depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on Sharp
+	  Zaurus SL-C7x0 models (Corgi, Shepherd, Husky).
+
+config SND_PXA2XX_SOC_SPITZ
+	tristate "SoC Audio support for Sharp Zaurus SL-Cxx00"
+	depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_WM8750
+	help
+	  Say Y if you want to add support for SoC audio on Sharp
+	  Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita).
+
+config SND_PXA2XX_SOC_POODLE
+	tristate "SoC Audio support for Poodle"
+	depends on SND_PXA2XX_SOC && MACH_POODLE
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on Sharp
+	  Zaurus SL-5600 model (Poodle).
+
+config SND_PXA2XX_SOC_TOSA
+	tristate "SoC AC97 Audio support for Tosa"
+	depends on SND_PXA2XX_SOC && MACH_TOSA
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on Sharp
+	  Zaurus SL-C6000x models (Tosa).
+
+endmenu
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
new file mode 100644
index 00000000000..78e0d6b07d1
--- /dev/null
+++ b/sound/soc/pxa/Makefile
@@ -0,0 +1,20 @@
+# PXA Platform Support
+snd-soc-pxa2xx-objs := pxa2xx-pcm.o
+snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
+snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
+obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
+obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+
+# PXA Machine Support
+snd-soc-corgi-objs := corgi.o
+snd-soc-poodle-objs := poodle.o
+snd-soc-tosa-objs := tosa.o
+snd-soc-spitz-objs := spitz.o
+
+obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
+obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
+obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
+
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
new file mode 100644
index 00000000000..5ee51a994ac
--- /dev/null
+++ b/sound/soc/pxa/corgi.c
@@ -0,0 +1,383 @@
+/*
+ * corgi.c  --  SoC audio for Corgi
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    30th Nov 2005   Initial version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#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 "../codecs/wm8731.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define CORGI_HP        0
+#define CORGI_MIC       1
+#define CORGI_LINE      2
+#define CORGI_HEADSET   3
+#define CORGI_HP_OFF    4
+#define CORGI_SPK_ON    0
+#define CORGI_SPK_OFF   1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define CORGI_AUDIO_CLOCK 12288000
+
+static int corgi_jack_func;
+static int corgi_spk_func;
+
+static void corgi_ext_control(struct snd_soc_codec *codec)
+{
+	int spk = 0, mic = 0, line = 0, hp = 0, hs = 0;
+
+	/* set up jack connection */
+	switch (corgi_jack_func) {
+	case CORGI_HP:
+		hp = 1;
+		/* set = unmute headphone */
+		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		break;
+	case CORGI_MIC:
+		mic = 1;
+		/* reset = mute headphone */
+		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		break;
+	case CORGI_LINE:
+		line = 1;
+		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		break;
+	case CORGI_HEADSET:
+		hs = 1;
+		mic = 1;
+		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		break;
+	}
+
+	if (corgi_spk_func == CORGI_SPK_ON)
+		spk = 1;
+
+	/* set the enpoints to their new connetion states */
+	snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
+	snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic);
+	snd_soc_dapm_set_endpoint(codec, "Line Jack", line);
+	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
+	snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
+
+	/* signal a DAPM event */
+	snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int corgi_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	corgi_ext_control(codec);
+	return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on corgi */
+static int corgi_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 */
+	set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
+	set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+	return 0;
+}
+
+static int corgi_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops corgi_ops = {
+	.startup = corgi_startup,
+	.hw_params = corgi_hw_params,
+	.shutdown = corgi_shutdown,
+};
+
+static int corgi_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = corgi_jack_func;
+	return 0;
+}
+
+static int corgi_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (corgi_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	corgi_jack_func = ucontrol->value.integer.value[0];
+	corgi_ext_control(codec);
+	return 1;
+}
+
+static int corgi_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = corgi_spk_func;
+	return 0;
+}
+
+static int corgi_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (corgi_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	corgi_spk_func = ucontrol->value.integer.value[0];
+	corgi_ext_control(codec);
+	return 1;
+}
+
+static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
+	else
+		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
+
+	return 0;
+}
+
+static int corgi_mic_event(struct snd_soc_dapm_widget *w, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
+	else
+		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
+
+	return 0;
+}
+
+/* corgi machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event),
+SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event),
+SND_SOC_DAPM_LINE("Line Jack", NULL),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Corgi machine audio map (connections to the codec pins) */
+static const char *audio_map[][3] = {
+
+	/* headset Jack  - in = micin, out = LHPOUT*/
+	{"Headset Jack", NULL, "LHPOUT"},
+
+	/* headphone connected to LHPOUT1, RHPOUT1 */
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	/* speaker connected to LOUT, ROUT */
+	{"Ext Spk", NULL, "ROUT"},
+	{"Ext Spk", NULL, "LOUT"},
+
+	/* mic is connected to MICIN (via right channel of headphone jack) */
+	{"MICIN", NULL, "Mic Jack"},
+
+	/* Same as the above but no mic bias for line signals */
+	{"MICIN", NULL, "Line Jack"},
+
+	{NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+	"Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum corgi_enum[] = {
+	SOC_ENUM_SINGLE_EXT(5, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
+	SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack,
+		corgi_set_jack),
+	SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk,
+		corgi_set_spk),
+};
+
+/*
+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
+ */
+static int corgi_wm8731_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
+	snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+
+	/* Add corgi specific controls */
+	for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
+		err = snd_ctl_add(codec->card,
+			snd_soc_cnew(&wm8731_corgi_controls[i],codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* Add corgi specific widgets */
+	for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	}
+
+	/* Set up corgi specific audio path audio_map */
+	for(i = 0; audio_map[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+			audio_map[i][1], audio_map[i][2]);
+	}
+
+	snd_soc_dapm_sync_endpoints(codec);
+	return 0;
+}
+
+/* corgi digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link corgi_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai = &pxa_i2s_dai,
+	.codec_dai = &wm8731_dai,
+	.init = corgi_wm8731_init,
+	.ops = &corgi_ops,
+};
+
+/* corgi audio machine driver */
+static struct snd_soc_machine snd_soc_machine_corgi = {
+	.name = "Corgi",
+	.dai_link = &corgi_dai,
+	.num_links = 1,
+};
+
+/* corgi audio private data */
+static struct wm8731_setup_data corgi_wm8731_setup = {
+	.i2c_address = 0x1b,
+};
+
+/* corgi audio subsystem */
+static struct snd_soc_device corgi_snd_devdata = {
+	.machine = &snd_soc_machine_corgi,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &corgi_wm8731_setup,
+};
+
+static struct platform_device *corgi_snd_device;
+
+static int __init corgi_init(void)
+{
+	int ret;
+
+	if (!(machine_is_corgi() || machine_is_shepherd() || machine_is_husky()))
+		return -ENODEV;
+
+	corgi_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!corgi_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(corgi_snd_device, &corgi_snd_devdata);
+	corgi_snd_devdata.dev = &corgi_snd_device->dev;
+	ret = platform_device_add(corgi_snd_device);
+
+	if (ret)
+		platform_device_put(corgi_snd_device);
+
+	return ret;
+}
+
+static void __exit corgi_exit(void)
+{
+	platform_device_unregister(corgi_snd_device);
+}
+
+module_init(corgi_init);
+module_exit(corgi_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Corgi");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
new file mode 100644
index 00000000000..0915cf74042
--- /dev/null
+++ b/sound/soc/pxa/poodle.c
@@ -0,0 +1,352 @@
+/*
+ * poodle.c  --  SoC audio for Poodle
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#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 "../codecs/wm8731.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define POODLE_HP        1
+#define POODLE_HP_OFF    0
+#define POODLE_SPK_ON    1
+#define POODLE_SPK_OFF   0
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define POODLE_AUDIO_CLOCK 12288000
+
+static int poodle_jack_func;
+static int poodle_spk_func;
+
+static void poodle_ext_control(struct snd_soc_codec *codec)
+{
+	int spk = 0;
+
+	/* set up jack connection */
+	if (poodle_jack_func == POODLE_HP) {
+		/* 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);
+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+	} else {
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_MUTE_L, 0);
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_MUTE_R, 0);
+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+	}
+
+	if (poodle_spk_func == POODLE_SPK_ON)
+		spk = 1;
+
+	/* set the enpoints to their new connetion states */
+	snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
+
+	/* signal a DAPM event */
+	snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int poodle_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	poodle_ext_control(codec);
+	return 0;
+}
+
+/* we need to unmute the HP at shutdown as the mute burns power on poodle */
+static int 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,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops poodle_ops = {
+	.startup = poodle_startup,
+	.hw_params = poodle_hw_params,
+	.shutdown = poodle_shutdown,
+};
+
+static int poodle_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = poodle_jack_func;
+	return 0;
+}
+
+static int poodle_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (poodle_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	poodle_jack_func = ucontrol->value.integer.value[0];
+	poodle_ext_control(codec);
+	return 1;
+}
+
+static int poodle_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = poodle_spk_func;
+	return 0;
+}
+
+static int poodle_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (poodle_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	poodle_spk_func = ucontrol->value.integer.value[0];
+	poodle_ext_control(codec);
+	return 1;
+}
+
+static int poodle_amp_event(struct snd_soc_dapm_widget *w, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_AMP_ON, 0);
+	else
+		locomo_gpio_write(&poodle_locomo_device.dev,
+			POODLE_LOCOMO_GPIO_AMP_ON, 1);
+
+	return 0;
+}
+
+/* poodle machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", NULL),
+SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
+};
+
+/* Corgi machine audio_mapnections to the codec pins */
+static const char *audio_map[][3] = {
+
+	/* headphone connected to LHPOUT1, RHPOUT1 */
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	/* speaker connected to LOUT, ROUT */
+	{"Ext Spk", NULL, "ROUT"},
+	{"Ext Spk", NULL, "LOUT"},
+
+	{NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Off", "Headphone"};
+static const char *spk_function[] = {"Off", "On"};
+static const struct soc_enum poodle_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const snd_kcontrol_new_t 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,
+		poodle_set_spk),
+};
+
+/*
+ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
+ */
+static int poodle_wm8731_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
+	snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+	snd_soc_dapm_set_endpoint(codec, "MICIN", 1);
+
+	/* Add poodle specific controls */
+	for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
+		err = snd_ctl_add(codec->card,
+			snd_soc_cnew(&wm8731_poodle_controls[i],codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* Add poodle specific widgets */
+	for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	}
+
+	/* Set up poodle specific audio path audio_map */
+	for (i = 0; audio_map[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+			audio_map[i][1], audio_map[i][2]);
+	}
+
+	snd_soc_dapm_sync_endpoints(codec);
+	return 0;
+}
+
+/* poodle digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link poodle_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731",
+	.cpu_dai = &pxa_i2s_dai,
+	.codec_dai = &wm8731_dai,
+	.init = poodle_wm8731_init,
+	.ops = &poodle_ops,
+};
+
+/* poodle audio machine driver */
+static struct snd_soc_machine snd_soc_machine_poodle = {
+	.name = "Poodle",
+	.dai_link = &poodle_dai,
+	.num_links = 1,
+};
+
+/* poodle audio private data */
+static struct wm8731_setup_data poodle_wm8731_setup = {
+	.i2c_address = 0x1b,
+};
+
+/* poodle audio subsystem */
+static struct snd_soc_device poodle_snd_devdata = {
+	.machine = &snd_soc_machine_poodle,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &poodle_wm8731_setup,
+};
+
+static struct platform_device *poodle_snd_device;
+
+static int __init poodle_init(void)
+{
+	int ret;
+
+	if (!machine_is_poodle())
+		return -ENODEV;
+
+	locomo_gpio_set_dir(&poodle_locomo_device.dev,
+		POODLE_LOCOMO_GPIO_AMP_ON, 0);
+	/* should we mute HP at startup - burning power ?*/
+	locomo_gpio_set_dir(&poodle_locomo_device.dev,
+		POODLE_LOCOMO_GPIO_MUTE_L, 0);
+	locomo_gpio_set_dir(&poodle_locomo_device.dev,
+		POODLE_LOCOMO_GPIO_MUTE_R, 0);
+
+	poodle_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!poodle_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(poodle_snd_device, &poodle_snd_devdata);
+	poodle_snd_devdata.dev = &poodle_snd_device->dev;
+	ret = platform_device_add(poodle_snd_device);
+
+	if (ret)
+		platform_device_put(poodle_snd_device);
+
+	return ret;
+}
+
+static void __exit poodle_exit(void)
+{
+	platform_device_unregister(poodle_snd_device);
+}
+
+module_init(poodle_init);
+module_exit(poodle_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Poodle");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
new file mode 100644
index 00000000000..1bbbeff84ef
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -0,0 +1,431 @@
+/*
+ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip.
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Dec 02, 2004
+ * Copyright:	MontaVista Software Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/wait.h>
+#include <linux/delay.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/irq.h>
+#include <linux/mutex.h>
+#include <asm/hardware.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/audio.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static DEFINE_MUTEX(car_mutex);
+static DECLARE_WAIT_QUEUE_HEAD(gsr_wq);
+static volatile long gsr_bits;
+
+/*
+ * Beware PXA27x bugs:
+ *
+ *   o Slot 12 read from modem space will hang controller.
+ *   o CDONE, SDONE interrupt fails after any slot 12 IO.
+ *
+ * We therefore have an hybrid approach for waiting on SDONE (interrupt or
+ * 1 jiffy timeout if interrupt never comes).
+ */
+
+static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97,
+	unsigned short reg)
+{
+	unsigned short val = -1;
+	volatile u32 *reg_addr;
+
+	mutex_lock(&car_mutex);
+
+	/* set up primary or secondary codec/modem space */
+#ifdef CONFIG_PXA27x
+	reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+#else
+	if (reg == AC97_GPIO_STATUS)
+		reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
+	else
+		reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+#endif
+	reg_addr += (reg >> 1);
+
+#ifndef CONFIG_PXA27x
+	if (reg == AC97_GPIO_STATUS) {
+		/* read from controller cache */
+		val = *reg_addr;
+		goto out;
+	}
+#endif
+
+	/* start read access across the ac97 link */
+	GSR = GSR_CDONE | GSR_SDONE;
+	gsr_bits = 0;
+	val = *reg_addr;
+
+	wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1);
+	if (!((GSR | gsr_bits) & GSR_SDONE)) {
+		printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n",
+				__FUNCTION__, reg, GSR | gsr_bits);
+		val = -1;
+		goto out;
+	}
+
+	/* valid data now */
+	GSR = GSR_CDONE | GSR_SDONE;
+	gsr_bits = 0;
+	val = *reg_addr;
+	/* but we've just started another cycle... */
+	wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1);
+
+out:	mutex_unlock(&car_mutex);
+	return val;
+}
+
+static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+	unsigned short val)
+{
+	volatile u32 *reg_addr;
+
+	mutex_lock(&car_mutex);
+
+	/* set up primary or secondary codec/modem space */
+#ifdef CONFIG_PXA27x
+	reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+#else
+	if (reg == AC97_GPIO_STATUS)
+		reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE;
+	else
+		reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE;
+#endif
+	reg_addr += (reg >> 1);
+
+	GSR = GSR_CDONE | GSR_SDONE;
+	gsr_bits = 0;
+	*reg_addr = val;
+	wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1);
+	if (!((GSR | gsr_bits) & GSR_CDONE))
+		printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n",
+				__FUNCTION__, reg, GSR | gsr_bits);
+
+	mutex_unlock(&car_mutex);
+}
+
+static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+	gsr_bits = 0;
+
+#ifdef CONFIG_PXA27x
+	/* warm reset broken on Bulverde,
+	   so manually keep AC97 reset high */
+	pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH);
+	udelay(10);
+	GCR |= GCR_WARM_RST;
+	pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+	udelay(500);
+#else
+	GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN;
+	wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
+#endif
+
+	if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)))
+		printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
+				 __FUNCTION__, gsr_bits);
+
+	GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
+	GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
+}
+
+static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+	GCR &=  GCR_COLD_RST;  /* clear everything but nCRST */
+	GCR &= ~GCR_COLD_RST;  /* then assert nCRST */
+
+	gsr_bits = 0;
+#ifdef CONFIG_PXA27x
+	/* PXA27x Developers Manual section 13.5.2.2.1 */
+	pxa_set_cken(1 << 31, 1);
+	udelay(5);
+	pxa_set_cken(1 << 31, 0);
+	GCR = GCR_COLD_RST;
+	udelay(50);
+#else
+	GCR = GCR_COLD_RST;
+	GCR |= GCR_CDONE_IE|GCR_SDONE_IE;
+	wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1);
+#endif
+
+	if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)))
+		printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
+				 __FUNCTION__, gsr_bits);
+
+	GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN);
+	GCR |= GCR_SDONE_IE|GCR_CDONE_IE;
+}
+
+static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id)
+{
+	long status;
+
+	status = GSR;
+	if (status) {
+		GSR = status;
+		gsr_bits |= status;
+		wake_up(&gsr_wq);
+
+#ifdef CONFIG_PXA27x
+		/* Although we don't use those we still need to clear them
+		   since they tend to spuriously trigger when MMC is used
+		   (hardware bug? go figure)... */
+		MISR = MISR_EOC;
+		PISR = PISR_EOC;
+		MCSR = MCSR_EOC;
+#endif
+
+		return IRQ_HANDLED;
+	}
+
+	return IRQ_NONE;
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+	.read	= pxa2xx_ac97_read,
+	.write	= pxa2xx_ac97_write,
+	.warm_reset	= pxa2xx_ac97_warm_reset,
+	.reset	= pxa2xx_ac97_cold_reset,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
+	.name			= "AC97 PCM Stereo out",
+	.dev_addr		= __PREG(PCDR),
+	.drcmr			= &DRCMRTXPCDR,
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
+	.name			= "AC97 PCM Stereo in",
+	.dev_addr		= __PREG(PCDR),
+	.drcmr			= &DRCMRRXPCDR,
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
+	.name			= "AC97 Aux PCM (Slot 5) Mono out",
+	.dev_addr		= __PREG(MODR),
+	.drcmr			= &DRCMRTXMODR,
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
+	.name			= "AC97 Aux PCM (Slot 5) Mono in",
+	.dev_addr		= __PREG(MODR),
+	.drcmr			= &DRCMRRXMODR,
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
+	.name			= "AC97 Mic PCM (Slot 6) Mono in",
+	.dev_addr		= __PREG(MCDR),
+	.drcmr			= &DRCMRRXMCDR,
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+#ifdef CONFIG_PM
+static int pxa2xx_ac97_suspend(struct platform_device *pdev,
+	struct snd_soc_cpu_dai *dai)
+{
+	GCR |= GCR_ACLINK_OFF;
+	pxa_set_cken(CKEN2_AC97, 0);
+	return 0;
+}
+
+static int pxa2xx_ac97_resume(struct platform_device *pdev,
+	struct snd_soc_cpu_dai *dai)
+{
+	pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
+	pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
+	pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
+	pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
+#ifdef CONFIG_PXA27x
+	/* Use GPIO 113 as AC97 Reset on Bulverde */
+	pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+#endif
+	pxa_set_cken(CKEN2_AC97, 1);
+	return 0;
+}
+
+#else
+#define pxa2xx_ac97_suspend	NULL
+#define pxa2xx_ac97_resume	NULL
+#endif
+
+static int pxa2xx_ac97_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL);
+	if (ret < 0)
+		goto err;
+
+	pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
+	pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
+	pxa_gpio_mode(GPIO28_BITCLK_AC97_MD);
+	pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
+#ifdef CONFIG_PXA27x
+	/* Use GPIO 113 as AC97 Reset on Bulverde */
+	pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+#endif
+	pxa_set_cken(CKEN2_AC97, 1);
+	return 0;
+
+ err:
+	if (CKEN & CKEN2_AC97) {
+		GCR |= GCR_ACLINK_OFF;
+		free_irq(IRQ_AC97, NULL);
+		pxa_set_cken(CKEN2_AC97, 0);
+	}
+	return ret;
+}
+
+static void pxa2xx_ac97_remove(struct platform_device *pdev)
+{
+	GCR |= GCR_ACLINK_OFF;
+	free_irq(IRQ_AC97, NULL);
+	pxa_set_cken(CKEN2_AC97, 0);
+}
+
+static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
+	else
+		cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in;
+
+	return 0;
+}
+
+static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
+	else
+		cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in;
+
+	return 0;
+}
+
+static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return -ENODEV;
+	else
+		cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in;
+
+	return 0;
+}
+
+#define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_48000)
+
+/*
+ * There is only 1 physical AC97 interface for pxa2xx, but it
+ * has extra fifo's that can be used for aux DACs and ADCs.
+ */
+struct snd_soc_cpu_dai pxa_ac97_dai[] = {
+{
+	.name = "pxa2xx-ac97",
+	.id = 0,
+	.type = SND_SOC_DAI_AC97,
+	.probe = pxa2xx_ac97_probe,
+	.remove = pxa2xx_ac97_remove,
+	.suspend = pxa2xx_ac97_suspend,
+	.resume = pxa2xx_ac97_resume,
+	.playback = {
+		.stream_name = "AC97 Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "AC97 Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = {
+		.hw_params = pxa2xx_ac97_hw_params,},
+},
+{
+	.name = "pxa2xx-ac97-aux",
+	.id = 1,
+	.type = SND_SOC_DAI_AC97,
+	.playback = {
+		.stream_name = "AC97 Aux Playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "AC97 Aux Capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = {
+		.hw_params = pxa2xx_ac97_hw_aux_params,},
+},
+{
+	.name = "pxa2xx-ac97-mic",
+	.id = 2,
+	.type = SND_SOC_DAI_AC97,
+	.capture = {
+		.stream_name = "AC97 Mic Capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = PXA2XX_AC97_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = {
+		.hw_params = pxa2xx_ac97_hw_mic_params,},
+},
+};
+
+EXPORT_SYMBOL_GPL(pxa_ac97_dai);
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
new file mode 100644
index 00000000000..4c4b882316a
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -0,0 +1,22 @@
+/*
+ * linux/sound/arm/pxa2xx-ac97.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_AC97_H
+#define _PXA2XX_AC97_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_AC97_HIFI	0
+#define PXA2XX_DAI_AC97_AUX		1
+#define PXA2XX_DAI_AC97_MIC		2
+
+extern struct snd_soc_cpu_dai pxa_ac97_dai[3];
+
+/* platform data */
+extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
new file mode 100644
index 00000000000..575a6137c04
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -0,0 +1,318 @@
+/*
+ * pxa2xx-i2s.c  --  ALSA Soc Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    12th Aug 2005   Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <asm/hardware.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/audio.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+struct pxa_i2s_port {
+	u32 sadiv;
+	u32 sacr0;
+	u32 sacr1;
+	u32 saimr;
+	int master;
+	u32 fmt;
+};
+static struct pxa_i2s_port pxa_i2s;
+
+static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
+	.name			= "I2S PCM Stereo out",
+	.dev_addr		= __PREG(SADR),
+	.drcmr			= &DRCMRTXSADR,
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
+	.name			= "I2S PCM Stereo in",
+	.dev_addr		= __PREG(SADR),
+	.drcmr			= &DRCMRRXSADR,
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST32 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_gpio gpio_bus[] = {
+	{ /* I2S SoC Slave */
+		.rx = GPIO29_SDATA_IN_I2S_MD,
+		.tx = GPIO30_SDATA_OUT_I2S_MD,
+		.clk = GPIO28_BITCLK_IN_I2S_MD,
+		.frm = GPIO31_SYNC_I2S_MD,
+	},
+	{ /* I2S SoC Master */
+#ifdef CONFIG_PXA27x
+		.sys = GPIO113_I2S_SYSCLK_MD,
+#else
+		.sys = GPIO32_SYSCLK_I2S_MD,
+#endif
+		.rx = GPIO29_SDATA_IN_I2S_MD,
+		.tx = GPIO30_SDATA_OUT_I2S_MD,
+		.clk = GPIO28_BITCLK_OUT_I2S_MD,
+		.frm = GPIO31_SYNC_I2S_MD,
+	},
+};
+
+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	if (!cpu_dai->active) {
+		SACR0 |= SACR0_RST;
+		SACR0 = 0;
+	}
+
+	return 0;
+}
+
+/* wait for I2S controller to be ready */
+static int pxa_i2s_wait(void)
+{
+	int i;
+
+	/* flush the Rx FIFO */
+	for(i = 0; i < 16; i++)
+		SADR;
+	return 0;
+}
+
+static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+		unsigned int fmt)
+{
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		pxa_i2s.fmt = 0;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		pxa_i2s.fmt = SACR1_AMSL;
+		break;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		pxa_i2s.master = 1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		pxa_i2s.master = 0;
+		break;
+	default:
+		break;
+	}
+	return 0;
+}
+
+static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	if (clk_id != PXA2XX_I2S_SYSCLK)
+		return -ENODEV;
+
+	if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT)
+		pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys);
+
+	return 0;
+}
+
+static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
+	pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
+	pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
+	pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
+	pxa_set_cken(CKEN8_I2S, 1);
+	pxa_i2s_wait();
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out;
+	else
+		cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in;
+
+	/* is port used by another stream */
+	if (!(SACR0 & SACR0_ENB)) {
+
+		SACR0 = 0;
+		SACR1 = 0;
+		if (pxa_i2s.master)
+			SACR0 |= SACR0_BCKD;
+
+		SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1);
+		SACR1 |= pxa_i2s.fmt;
+	}
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		SAIMR |= SAIMR_TFS;
+	else
+		SAIMR |= SAIMR_RFS;
+
+	switch (params_rate(params)) {
+	case 8000:
+		SADIV = 0x48;
+		break;
+	case 11025:
+		SADIV = 0x34;
+		break;
+	case 16000:
+		SADIV = 0x24;
+		break;
+	case 22050:
+		SADIV = 0x1a;
+		break;
+	case 44100:
+		SADIV = 0xd;
+		break;
+	case 48000:
+		SADIV = 0xc;
+		break;
+	case 96000: /* not in manual and possibly slightly inaccurate */
+		SADIV = 0x6;
+		break;
+	}
+
+	return 0;
+}
+
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		SACR0 |= SACR0_ENB;
+		break;
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		SACR1 |= SACR1_DRPL;
+		SAIMR &= ~SAIMR_TFS;
+	} else {
+		SACR1 |= SACR1_DREC;
+		SAIMR &= ~SAIMR_RFS;
+	}
+
+	if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
+		SACR0 &= ~SACR0_ENB;
+		pxa_i2s_wait();
+		pxa_set_cken(CKEN8_I2S, 0);
+	}
+}
+
+#ifdef CONFIG_PM
+static int pxa2xx_i2s_suspend(struct platform_device *dev,
+	struct snd_soc_cpu_dai *dai)
+{
+	if (!dai->active)
+		return 0;
+
+	/* store registers */
+	pxa_i2s.sacr0 = SACR0;
+	pxa_i2s.sacr1 = SACR1;
+	pxa_i2s.saimr = SAIMR;
+	pxa_i2s.sadiv = SADIV;
+
+	/* deactivate link */
+	SACR0 &= ~SACR0_ENB;
+	pxa_i2s_wait();
+	return 0;
+}
+
+static int pxa2xx_i2s_resume(struct platform_device *pdev,
+	struct snd_soc_cpu_dai *dai)
+{
+	if (!dai->active)
+		return 0;
+
+	pxa_i2s_wait();
+
+	SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB;
+	SACR1 = pxa_i2s.sacr1;
+	SAIMR = pxa_i2s.saimr;
+	SADIV = pxa_i2s.sadiv;
+	SACR0 |= SACR0_ENB;
+
+	return 0;
+}
+
+#else
+#define pxa2xx_i2s_suspend	NULL
+#define pxa2xx_i2s_resume	NULL
+#endif
+
+#define PXA2XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+struct snd_soc_cpu_dai pxa_i2s_dai = {
+	.name = "pxa2xx-i2s",
+	.id = 0,
+	.type = SND_SOC_DAI_I2S,
+	.suspend = pxa2xx_i2s_suspend,
+	.resume = pxa2xx_i2s_resume,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PXA2XX_I2S_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = PXA2XX_I2S_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = {
+		.startup = pxa2xx_i2s_startup,
+		.shutdown = pxa2xx_i2s_shutdown,
+		.trigger = pxa2xx_i2s_trigger,
+		.hw_params = pxa2xx_i2s_hw_params,},
+	.dai_ops = {
+		.set_fmt = pxa2xx_i2s_set_dai_fmt,
+		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
+	},
+};
+
+EXPORT_SYMBOL_GPL(pxa_i2s_dai);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h
new file mode 100644
index 00000000000..a2484f0881f
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-i2s.h
@@ -0,0 +1,20 @@
+/*
+ * linux/sound/arm/pxa2xx-i2s.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_I2S_H
+#define _PXA2XX_I2S_H
+
+/* pxa2xx DAI ID's */
+#define PXA2XX_DAI_I2S			0
+
+/* I2S clock */
+#define PXA2XX_I2S_SYSCLK		0
+
+extern struct snd_soc_cpu_dai pxa_i2s_dai;
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
new file mode 100644
index 00000000000..35e8fa3a469
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -0,0 +1,372 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Nov 30, 2004
+ * Copyright:	(C) 2004 MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+#include <asm/hardware.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/audio.h>
+
+#include "pxa2xx-pcm.h"
+
+static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_INTERLEAVED |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
+					SNDRV_PCM_FMTBIT_S24_LE |
+					SNDRV_PCM_FMTBIT_S32_LE,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192 - 32,
+	.periods_min		= 1,
+	.periods_max		= PAGE_SIZE/sizeof(pxa_dma_desc),
+	.buffer_bytes_max	= 128 * 1024,
+	.fifo_size		= 32,
+};
+
+struct pxa2xx_runtime_data {
+	int dma_ch;
+	struct pxa2xx_pcm_dma_params *params;
+	pxa_dma_desc *dma_desc_array;
+	dma_addr_t dma_desc_array_phys;
+};
+
+static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
+{
+	struct snd_pcm_substream *substream = dev_id;
+	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+	int dcsr;
+
+	dcsr = DCSR(dma_ch);
+	DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN;
+
+	if (dcsr & DCSR_ENDINTR) {
+		snd_pcm_period_elapsed(substream);
+	} else {
+		printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n",
+			prtd->params->name, dma_ch, dcsr );
+	}
+}
+
+static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct pxa2xx_runtime_data *prtd = runtime->private_data;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+	size_t totsize = params_buffer_bytes(params);
+	size_t period = params_period_bytes(params);
+	pxa_dma_desc *dma_desc;
+	dma_addr_t dma_buff_phys, next_desc_phys;
+	int ret;
+
+	/* return if this is a bufferless transfer e.g.
+	 * codec <--> BT codec or GSM modem -- lg FIXME */
+	 if (!dma)
+	 	return 0;
+
+	/* this may get called several times by oss emulation
+	 * with different params */
+	if (prtd->params == NULL) {
+		prtd->params = dma;
+		ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+			      pxa2xx_pcm_dma_irq, substream);
+		if (ret < 0)
+			return ret;
+		prtd->dma_ch = ret;
+	} else if (prtd->params != dma) {
+		pxa_free_dma(prtd->dma_ch);
+		prtd->params = dma;
+		ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+			      pxa2xx_pcm_dma_irq, substream);
+		if (ret < 0)
+			return ret;
+		prtd->dma_ch = ret;
+	}
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	runtime->dma_bytes = totsize;
+
+	dma_desc = prtd->dma_desc_array;
+	next_desc_phys = prtd->dma_desc_array_phys;
+	dma_buff_phys = runtime->dma_addr;
+	do {
+		next_desc_phys += sizeof(pxa_dma_desc);
+		dma_desc->ddadr = next_desc_phys;
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			dma_desc->dsadr = dma_buff_phys;
+			dma_desc->dtadr = prtd->params->dev_addr;
+		} else {
+			dma_desc->dsadr = prtd->params->dev_addr;
+			dma_desc->dtadr = dma_buff_phys;
+		}
+		if (period > totsize)
+			period = totsize;
+		dma_desc->dcmd = prtd->params->dcmd | period | DCMD_ENDIRQEN;
+		dma_desc++;
+		dma_buff_phys += period;
+	} while (totsize -= period);
+	dma_desc[-1].ddadr = prtd->dma_desc_array_phys;
+
+	return 0;
+}
+
+static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+
+	if (prtd && prtd->params)
+		*prtd->params->drcmr = 0;
+
+	if (prtd->dma_ch) {
+		snd_pcm_set_runtime_buffer(substream, NULL);
+		pxa_free_dma(prtd->dma_ch);
+		prtd->dma_ch = 0;
+	}
+
+	return 0;
+}
+
+static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+
+	DCSR(prtd->dma_ch) &= ~DCSR_RUN;
+	DCSR(prtd->dma_ch) = 0;
+	DCMD(prtd->dma_ch) = 0;
+	*prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD;
+
+	return 0;
+}
+
+static int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
+		DCSR(prtd->dma_ch) = DCSR_RUN;
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		DCSR(prtd->dma_ch) &= ~DCSR_RUN;
+		break;
+
+	case SNDRV_PCM_TRIGGER_RESUME:
+		DCSR(prtd->dma_ch) |= DCSR_RUN;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
+		DCSR(prtd->dma_ch) |= DCSR_RUN;
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static snd_pcm_uframes_t
+pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct pxa2xx_runtime_data *prtd = runtime->private_data;
+
+	dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+			 DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch);
+	snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr);
+
+	if (x == runtime->buffer_size)
+		x = 0;
+	return x;
+}
+
+static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct pxa2xx_runtime_data *prtd;
+	int ret;
+
+	snd_soc_set_runtime_hwparams(substream, &pxa2xx_pcm_hardware);
+
+	/*
+	 * For mysterious reasons (and despite what the manual says)
+	 * playback samples are lost if the DMA count is not a multiple
+	 * of the DMA burst size.  Let's add a rule to enforce that.
+	 */
+	ret = snd_pcm_hw_constraint_step(runtime, 0,
+		SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+	if (ret)
+		goto out;
+
+	ret = snd_pcm_hw_constraint_step(runtime, 0,
+		SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+	if (ret)
+		goto out;
+
+	ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0)
+		goto out;
+
+	prtd = kzalloc(sizeof(struct pxa2xx_runtime_data), GFP_KERNEL);
+	if (prtd == NULL) {
+		ret = -ENOMEM;
+		goto out;
+	}
+
+	prtd->dma_desc_array =
+		dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE,
+				       &prtd->dma_desc_array_phys, GFP_KERNEL);
+	if (!prtd->dma_desc_array) {
+		ret = -ENOMEM;
+		goto err1;
+	}
+
+	runtime->private_data = prtd;
+	return 0;
+
+ err1:
+	kfree(prtd);
+ out:
+	return ret;
+}
+
+static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct pxa2xx_runtime_data *prtd = runtime->private_data;
+
+	dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE,
+			      prtd->dma_desc_array, prtd->dma_desc_array_phys);
+	kfree(prtd);
+	return 0;
+}
+
+static int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream,
+	struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+				     runtime->dma_area,
+				     runtime->dma_addr,
+				     runtime->dma_bytes);
+}
+
+struct snd_pcm_ops pxa2xx_pcm_ops = {
+	.open		= pxa2xx_pcm_open,
+	.close		= pxa2xx_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= pxa2xx_pcm_hw_params,
+	.hw_free	= pxa2xx_pcm_hw_free,
+	.prepare	= pxa2xx_pcm_prepare,
+	.trigger	= pxa2xx_pcm_trigger,
+	.pointer	= pxa2xx_pcm_pointer,
+	.mmap		= pxa2xx_pcm_mmap,
+};
+
+static int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = pxa2xx_pcm_hardware.buffer_bytes_max;
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+					   &buf->addr, GFP_KERNEL);
+	if (!buf->area)
+		return -ENOMEM;
+	buf->bytes = size;
+	return 0;
+}
+
+static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+
+		dma_free_writecombine(pcm->card->dev, buf->bytes,
+				      buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK;
+
+int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+	struct snd_pcm *pcm)
+{
+	int ret = 0;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &pxa2xx_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+	if (dai->playback.channels_min) {
+		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (dai->capture.channels_min) {
+		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+ out:
+	return ret;
+}
+
+struct snd_soc_platform pxa2xx_soc_platform = {
+	.name		= "pxa2xx-audio",
+	.pcm_ops 	= &pxa2xx_pcm_ops,
+	.pcm_new	= pxa2xx_pcm_new,
+	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
+};
+
+EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
+
+MODULE_AUTHOR("Nicolas Pitre");
+MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-pcm.h b/sound/soc/pxa/pxa2xx-pcm.h
new file mode 100644
index 00000000000..54c9c755e50
--- /dev/null
+++ b/sound/soc/pxa/pxa2xx-pcm.h
@@ -0,0 +1,34 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
+ *
+ * Author:	Nicolas Pitre
+ * Created:	Nov 30, 2004
+ * Copyright:	MontaVista Software, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _PXA2XX_PCM_H
+#define _PXA2XX_PCM_H
+
+struct pxa2xx_pcm_dma_params {
+	char *name;			/* stream identifier */
+	u32 dcmd;			/* DMA descriptor dcmd field */
+	volatile u32 *drcmr;		/* the DMA request channel to use */
+	u32 dev_addr;			/* device physical address for DMA */
+};
+
+struct pxa2xx_gpio {
+	u32 sys;
+	u32	rx;
+	u32 tx;
+	u32 clk;
+	u32 frm;
+};
+
+/* platform data */
+extern struct snd_soc_platform pxa2xx_soc_platform;
+
+#endif
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
new file mode 100644
index 00000000000..80e82109fef
--- /dev/null
+++ b/sound/soc/pxa/spitz.c
@@ -0,0 +1,394 @@
+/*
+ * spitz.c  --  SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    30th Nov 2005   Initial version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#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 <asm/mach-types.h>
+#include "../codecs/wm8750.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+
+#define SPITZ_HP        0
+#define SPITZ_MIC       1
+#define SPITZ_LINE      2
+#define SPITZ_HEADSET   3
+#define SPITZ_HP_OFF    4
+#define SPITZ_SPK_ON    0
+#define SPITZ_SPK_OFF   1
+
+ /* audio clock in Hz - rounded from 12.235MHz */
+#define SPITZ_AUDIO_CLOCK 12288000
+
+static int spitz_jack_func;
+static int spitz_spk_func;
+
+static void spitz_ext_control(struct snd_soc_codec *codec)
+{
+	if (spitz_spk_func == SPITZ_SPK_ON)
+		snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+	else
+		snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0);
+
+	/* set up jack connection */
+	switch (spitz_jack_func) {
+	case SPITZ_HP:
+		/* enable and unmute hp jack, disable mic bias */
+		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		break;
+	case SPITZ_MIC:
+		/* enable mic jack and bias, mute hp */
+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
+		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		break;
+	case SPITZ_LINE:
+		/* enable line jack, disable mic bias and mute hp */
+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Line Jack", 1);
+		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		break;
+	case SPITZ_HEADSET:
+		/* enable and unmute headset jack enable mic bias, mute L hp */
+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
+		snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1);
+		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		break;
+	case SPITZ_HP_OFF:
+
+		/* jack removed, everything off */
+		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
+		snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
+		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
+		break;
+	}
+	snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int spitz_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	spitz_ext_control(codec);
+	return 0;
+}
+
+static int spitz_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as input (unused) */
+	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+		SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops spitz_ops = {
+	.startup = spitz_startup,
+	.hw_params = spitz_hw_params,
+};
+
+static int spitz_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = spitz_jack_func;
+	return 0;
+}
+
+static int spitz_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (spitz_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	spitz_jack_func = ucontrol->value.integer.value[0];
+	spitz_ext_control(codec);
+	return 1;
+}
+
+static int spitz_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = spitz_spk_func;
+	return 0;
+}
+
+static int spitz_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (spitz_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	spitz_spk_func = ucontrol->value.integer.value[0];
+	spitz_ext_control(codec);
+	return 1;
+}
+
+static int spitz_mic_bias(struct snd_soc_dapm_widget *w, int event)
+{
+	if (machine_is_borzoi() || machine_is_spitz()) {
+		if (SND_SOC_DAPM_EVENT_ON(event))
+			set_scoop_gpio(&spitzscoop2_device.dev,
+				SPITZ_SCP2_MIC_BIAS);
+		else
+			reset_scoop_gpio(&spitzscoop2_device.dev,
+				SPITZ_SCP2_MIC_BIAS);
+	}
+
+	if (machine_is_akita()) {
+		if (SND_SOC_DAPM_EVENT_ON(event))
+			akita_set_ioexp(&akitaioexp_device.dev,
+				AKITA_IOEXP_MIC_BIAS);
+		else
+			akita_reset_ioexp(&akitaioexp_device.dev,
+				AKITA_IOEXP_MIC_BIAS);
+	}
+	return 0;
+}
+
+/* spitz machine dapm widgets */
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_LINE("Line Jack", NULL),
+
+	/* headset is a mic and mono headphone */
+	SND_SOC_DAPM_HP("Headset Jack", NULL),
+};
+
+/* Spitz machine audio_map */
+static const char *audio_map[][3] = {
+
+	/* headphone connected to LOUT1, ROUT1 */
+	{"Headphone Jack", NULL, "LOUT1"},
+	{"Headphone Jack", NULL, "ROUT1"},
+
+	/* headset connected to ROUT1 and LINPUT1 with bias (def below) */
+	{"Headset Jack", NULL, "ROUT1"},
+
+	/* ext speaker connected to LOUT2, ROUT2  */
+	{"Ext Spk", NULL , "ROUT2"},
+	{"Ext Spk", NULL , "LOUT2"},
+
+	/* mic is connected to input 1 - with bias */
+	{"LINPUT1", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Mic Jack"},
+
+	/* line is connected to input 1 - no bias */
+	{"LINPUT1", NULL, "Line Jack"},
+
+	{NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+	"Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum spitz_enum[] = {
+	SOC_ENUM_SINGLE_EXT(5, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
+	SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack,
+		spitz_set_jack),
+	SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk,
+		spitz_set_spk),
+};
+
+/*
+ * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device
+ */
+static int spitz_wm8750_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	/* NC codec pins */
+	snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0);
+	snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0);
+	snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0);
+	snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0);
+	snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0);
+	snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
+	snd_soc_dapm_set_endpoint(codec, "MONO", 0);
+
+	/* Add spitz specific controls */
+	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
+		err = snd_ctl_add(codec->card,
+			snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* Add spitz specific widgets */
+	for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+	}
+
+	/* Set up spitz specific audio path audio_map */
+	for (i = 0; audio_map[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+			audio_map[i][1], audio_map[i][2]);
+	}
+
+	snd_soc_dapm_sync_endpoints(codec);
+	return 0;
+}
+
+/* spitz digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link spitz_dai = {
+	.name = "wm8750",
+	.stream_name = "WM8750",
+	.cpu_dai = &pxa_i2s_dai,
+	.codec_dai = &wm8750_dai,
+	.init = spitz_wm8750_init,
+	.ops = &spitz_ops,
+};
+
+/* spitz audio machine driver */
+static struct snd_soc_machine snd_soc_machine_spitz = {
+	.name = "Spitz",
+	.dai_link = &spitz_dai,
+	.num_links = 1,
+};
+
+/* spitz audio private data */
+static struct wm8750_setup_data spitz_wm8750_setup = {
+	.i2c_address = 0x1b,
+};
+
+/* spitz audio subsystem */
+static struct snd_soc_device spitz_snd_devdata = {
+	.machine = &snd_soc_machine_spitz,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8750,
+	.codec_data = &spitz_wm8750_setup,
+};
+
+static struct platform_device *spitz_snd_device;
+
+static int __init spitz_init(void)
+{
+	int ret;
+
+	if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita()))
+		return -ENODEV;
+
+	spitz_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!spitz_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(spitz_snd_device, &spitz_snd_devdata);
+	spitz_snd_devdata.dev = &spitz_snd_device->dev;
+	ret = platform_device_add(spitz_snd_device);
+
+	if (ret)
+		platform_device_put(spitz_snd_device);
+
+	return ret;
+}
+
+static void __exit spitz_exit(void)
+{
+	platform_device_unregister(spitz_snd_device);
+}
+
+module_init(spitz_init);
+module_exit(spitz_exit);
+
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Spitz");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
new file mode 100644
index 00000000000..5504e30acf1
--- /dev/null
+++ b/sound/soc/pxa/tosa.c
@@ -0,0 +1,289 @@
+/*
+ * tosa.c  --  SoC audio for Tosa
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    30th Nov 2005   Initial version.
+ *
+ * GPIO's
+ *  1 - Jack Insertion
+ *  5 - Hookswitch (headset answer/hang up switch)
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#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 "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_machine tosa;
+
+#define TOSA_HP        0
+#define TOSA_MIC_INT   1
+#define TOSA_HEADSET   2
+#define TOSA_HP_OFF    3
+#define TOSA_SPK_ON    0
+#define TOSA_SPK_OFF   1
+
+static int tosa_jack_func;
+static int tosa_spk_func;
+
+static void tosa_ext_control(struct snd_soc_codec *codec)
+{
+	int spk = 0, mic_int = 0, hp = 0, hs = 0;
+
+	/* set up jack connection */
+	switch (tosa_jack_func) {
+	case TOSA_HP:
+		hp = 1;
+		break;
+	case TOSA_MIC_INT:
+		mic_int = 1;
+		break;
+	case TOSA_HEADSET:
+		hs = 1;
+		break;
+	}
+
+	if (tosa_spk_func == TOSA_SPK_ON)
+		spk = 1;
+
+	snd_soc_dapm_set_endpoint(codec, "Speaker", spk);
+	snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int);
+	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
+	snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
+	snd_soc_dapm_sync_endpoints(codec);
+}
+
+static int tosa_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->socdev->codec;
+
+	/* check the jack status at stream startup */
+	tosa_ext_control(codec);
+	return 0;
+}
+
+static struct snd_soc_ops tosa_ops = {
+	.startup = tosa_startup,
+};
+
+static int tosa_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = tosa_jack_func;
+	return 0;
+}
+
+static int tosa_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (tosa_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	tosa_jack_func = ucontrol->value.integer.value[0];
+	tosa_ext_control(codec);
+	return 1;
+}
+
+static int tosa_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = tosa_spk_func;
+	return 0;
+}
+
+static int tosa_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (tosa_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	tosa_spk_func = ucontrol->value.integer.value[0];
+	tosa_ext_control(codec);
+	return 1;
+}
+
+/* tosa dapm event handlers */
+static int tosa_hp_event(struct snd_soc_dapm_widget *w, 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);
+	return 0;
+}
+
+/* tosa machine dapm widgets */
+static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
+SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
+SND_SOC_DAPM_HP("Headset Jack", NULL),
+SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* tosa audio map */
+static const char *audio_map[][3] = {
+
+	/* headphone connected to HPOUTL, HPOUTR */
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+
+	/* ext speaker connected to LOUT2, ROUT2 */
+	{"Speaker", NULL, "LOUT2"},
+	{"Speaker", NULL, "ROUT2"},
+
+	/* internal mic is connected to mic1, mic2 differential - with bias */
+	{"MIC1", NULL, "Mic Bias"},
+	{"MIC2", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Mic (Internal)"},
+
+	/* headset is connected to HPOUTR, and LINEINR with bias */
+	{"Headset Jack", NULL, "HPOUTR"},
+	{"LINEINR", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Headset Jack"},
+
+	{NULL, NULL, NULL},
+};
+
+static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
+	"Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum tosa_enum[] = {
+	SOC_ENUM_SINGLE_EXT(5, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new tosa_controls[] = {
+	SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
+		tosa_set_jack),
+	SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
+		tosa_set_spk),
+};
+
+static int tosa_ac97_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
+	snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0);
+
+	/* add tosa specific controls */
+	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&tosa_controls[i],codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* add tosa specific widgets */
+	for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) {
+		snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]);
+	}
+
+	/* set up tosa specific audio path audio_map */
+	for (i = 0; audio_map[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec, audio_map[i][0],
+			audio_map[i][1], audio_map[i][2]);
+	}
+
+	snd_soc_dapm_sync_endpoints(codec);
+	return 0;
+}
+
+static struct snd_soc_dai_link tosa_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.init = tosa_ac97_init,
+	.ops = &tosa_ops,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	.ops = &tosa_ops,
+},
+};
+
+static struct snd_soc_machine tosa = {
+	.name = "Tosa",
+	.dai_link = tosa_dai,
+	.num_links = ARRAY_SIZE(tosa_dai),
+};
+
+static struct snd_soc_device tosa_snd_devdata = {
+	.machine = &tosa,
+	.platform = &pxa2xx_soc_platform,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *tosa_snd_device;
+
+static int __init tosa_init(void)
+{
+	int ret;
+
+	if (!machine_is_tosa())
+		return -ENODEV;
+
+	tosa_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!tosa_snd_device)
+		return -ENOMEM;
+
+	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);
+
+	return ret;
+}
+
+static void __exit tosa_exit(void)
+{
+	platform_device_unregister(tosa_snd_device);
+}
+
+module_init(tosa_init);
+module_exit(tosa_exit);
+
+/* Module information */
+MODULE_AUTHOR("Richard Purdie");
+MODULE_DESCRIPTION("ALSA SoC Tosa");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
new file mode 100644
index 00000000000..36519aef55d
--- /dev/null
+++ b/sound/soc/soc-core.c
@@ -0,0 +1,1587 @@
+/*
+ * soc-core.c  --  ALSA SoC Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *         with code, comments and ideas from :-
+ *         Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    12th Aug 2005   Initial version.
+ *    25th Oct 2005   Working Codec, Interface and Platform registration.
+ *
+ *  TODO:
+ *   o Add hw rules to enforce rates, etc.
+ *   o More testing with other codecs/machines.
+ *   o Add more codecs and platforms to ensure good API coverage.
+ *   o Support TDM on PCM and I2S
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+/* debug */
+#define SOC_DEBUG 0
+#if SOC_DEBUG
+#define dbg(format, arg...) printk(format, ## arg)
+#else
+#define dbg(format, arg...)
+#endif
+
+static DEFINE_MUTEX(pcm_mutex);
+static DEFINE_MUTEX(io_mutex);
+static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
+
+/*
+ * This is a timeout to do a DAPM powerdown after a stream is closed().
+ * It can be used to eliminate pops between different playback streams, e.g.
+ * between two audio tracks.
+ */
+static int pmdown_time = 5000;
+module_param(pmdown_time, int, 0);
+MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");
+
+/*
+ * This function forces any delayed work to be queued and run.
+ */
+static int run_delayed_work(struct delayed_work *dwork)
+{
+	int ret;
+
+	/* cancel any work waiting to be queued. */
+	ret = cancel_delayed_work(dwork);
+
+	/* if there was any work waiting then we run it now and
+	 * wait for it's completion */
+	if (ret) {
+		schedule_delayed_work(dwork, 0);
+		flush_scheduled_work();
+	}
+	return ret;
+}
+
+#ifdef CONFIG_SND_SOC_AC97_BUS
+/* unregister ac97 codec */
+static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
+{
+	if (codec->ac97->dev.bus)
+		device_unregister(&codec->ac97->dev);
+	return 0;
+}
+
+/* stop no dev release warning */
+static void soc_ac97_device_release(struct device *dev){}
+
+/* register ac97 codec to bus */
+static int soc_ac97_dev_register(struct snd_soc_codec *codec)
+{
+	int err;
+
+	codec->ac97->dev.bus = &ac97_bus_type;
+	codec->ac97->dev.parent = NULL;
+	codec->ac97->dev.release = soc_ac97_device_release;
+
+	snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
+		 codec->card->number, 0, codec->name);
+	err = device_register(&codec->ac97->dev);
+	if (err < 0) {
+		snd_printk(KERN_ERR "Can't register ac97 bus\n");
+		codec->ac97->dev.bus = NULL;
+		return err;
+	}
+	return 0;
+}
+#endif
+
+static inline const char* get_dai_name(int type)
+{
+	switch(type) {
+	case SND_SOC_DAI_AC97:
+		return "AC97";
+	case SND_SOC_DAI_I2S:
+		return "I2S";
+	case SND_SOC_DAI_PCM:
+		return "PCM";
+	}
+	return NULL;
+}
+
+/*
+ * Called by ALSA when a PCM substream is opened, the runtime->hw record is
+ * then initialized and any private data can be allocated. This also calls
+ * startup for the cpu DAI, platform, machine and codec DAI.
+ */
+static int soc_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	int ret = 0;
+
+	mutex_lock(&pcm_mutex);
+
+	/* startup the audio subsystem */
+	if (cpu_dai->ops.startup) {
+		ret = cpu_dai->ops.startup(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't open interface %s\n",
+				cpu_dai->name);
+			goto out;
+		}
+	}
+
+	if (platform->pcm_ops->open) {
+		ret = platform->pcm_ops->open(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
+			goto platform_err;
+		}
+	}
+
+	if (codec_dai->ops.startup) {
+		ret = codec_dai->ops.startup(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't open codec %s\n",
+				codec_dai->name);
+			goto codec_dai_err;
+		}
+	}
+
+	if (machine->ops && machine->ops->startup) {
+		ret = machine->ops->startup(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
+			goto machine_err;
+		}
+	}
+
+	/* Check that the codec and cpu DAI's are compatible */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		runtime->hw.rate_min =
+			max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min);
+		runtime->hw.rate_max =
+			min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max);
+		runtime->hw.channels_min =
+			max(codec_dai->playback.channels_min,
+				cpu_dai->playback.channels_min);
+		runtime->hw.channels_max =
+			min(codec_dai->playback.channels_max,
+				cpu_dai->playback.channels_max);
+		runtime->hw.formats =
+			codec_dai->playback.formats & cpu_dai->playback.formats;
+		runtime->hw.rates =
+			codec_dai->playback.rates & cpu_dai->playback.rates;
+	} else {
+		runtime->hw.rate_min =
+			max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min);
+		runtime->hw.rate_max =
+			min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max);
+		runtime->hw.channels_min =
+			max(codec_dai->capture.channels_min,
+				cpu_dai->capture.channels_min);
+		runtime->hw.channels_max =
+			min(codec_dai->capture.channels_max,
+				cpu_dai->capture.channels_max);
+		runtime->hw.formats =
+			codec_dai->capture.formats & cpu_dai->capture.formats;
+		runtime->hw.rates =
+			codec_dai->capture.rates & cpu_dai->capture.rates;
+	}
+
+	snd_pcm_limit_hw_rates(runtime);
+	if (!runtime->hw.rates) {
+		printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
+			codec_dai->name, cpu_dai->name);
+		goto machine_err;
+	}
+	if (!runtime->hw.formats) {
+		printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
+			codec_dai->name, cpu_dai->name);
+		goto machine_err;
+	}
+	if (!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 machine_err;
+	}
+
+	dbg("asoc: %s <-> %s info:\n",codec_dai->name, cpu_dai->name);
+	dbg("asoc: rate mask 0x%x\n", runtime->hw.rates);
+	dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+		runtime->hw.channels_max);
+	dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
+		runtime->hw.rate_max);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		cpu_dai->playback.active = codec_dai->playback.active = 1;
+	else
+		cpu_dai->capture.active = codec_dai->capture.active = 1;
+	cpu_dai->active = codec_dai->active = 1;
+	cpu_dai->runtime = runtime;
+	socdev->codec->active++;
+	mutex_unlock(&pcm_mutex);
+	return 0;
+
+machine_err:
+	if (machine->ops && machine->ops->shutdown)
+		machine->ops->shutdown(substream);
+
+codec_dai_err:
+	if (platform->pcm_ops->close)
+		platform->pcm_ops->close(substream);
+
+platform_err:
+	if (cpu_dai->ops.shutdown)
+		cpu_dai->ops.shutdown(substream);
+out:
+	mutex_unlock(&pcm_mutex);
+	return ret;
+}
+
+/*
+ * Power down the audio subsytem pmdown_time msecs after close is called.
+ * This is to ensure there are no pops or clicks in between any music tracks
+ * due to DAPM power cycling.
+ */
+static void close_delayed_work(struct work_struct *work)
+{
+	struct snd_soc_device *socdev =
+		container_of(work, struct snd_soc_device, delayed_work.work);
+	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec_dai *codec_dai;
+	int i;
+
+	mutex_lock(&pcm_mutex);
+	for(i = 0; i < codec->num_dai; i++) {
+		codec_dai = &codec->dai[i];
+
+		dbg("pop wq checking: %s status: %s waiting: %s\n",
+			codec_dai->playback.stream_name,
+			codec_dai->playback.active ? "active" : "inactive",
+			codec_dai->pop_wait ? "yes" : "no");
+
+		/* are we waiting on this codec DAI stream */
+		if (codec_dai->pop_wait == 1) {
+
+			codec_dai->pop_wait = 0;
+			snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name,
+				SND_SOC_DAPM_STREAM_STOP);
+
+			/* power down the codec power domain if no longer active */
+			if (codec->active == 0) {
+				dbg("pop wq D3 %s %s\n", codec->name,
+					codec_dai->playback.stream_name);
+		 		if (codec->dapm_event)
+					codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+			}
+		}
+	}
+	mutex_unlock(&pcm_mutex);
+}
+
+/*
+ * Called by ALSA when a PCM substream is closed. Private data can be
+ * freed here. The cpu DAI, codec DAI, machine and platform are also
+ * shutdown.
+ */
+static int soc_codec_close(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	mutex_lock(&pcm_mutex);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		cpu_dai->playback.active = codec_dai->playback.active = 0;
+	else
+		cpu_dai->capture.active = codec_dai->capture.active = 0;
+
+	if (codec_dai->playback.active == 0 &&
+		codec_dai->capture.active == 0) {
+		cpu_dai->active = codec_dai->active = 0;
+	}
+	codec->active--;
+
+	if (cpu_dai->ops.shutdown)
+		cpu_dai->ops.shutdown(substream);
+
+	if (codec_dai->ops.shutdown)
+		codec_dai->ops.shutdown(substream);
+
+	if (machine->ops && machine->ops->shutdown)
+		machine->ops->shutdown(substream);
+
+	if (platform->pcm_ops->close)
+		platform->pcm_ops->close(substream);
+	cpu_dai->runtime = NULL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* start delayed pop wq here for playback streams */
+		codec_dai->pop_wait = 1;
+		schedule_delayed_work(&socdev->delayed_work,
+			msecs_to_jiffies(pmdown_time));
+	} else {
+		/* capture streams can be powered down now */
+		snd_soc_dapm_stream_event(codec,
+			codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_STOP);
+
+		if (codec->active == 0 && codec_dai->pop_wait == 0){
+			if (codec->dapm_event)
+				codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+		}
+	}
+
+	mutex_unlock(&pcm_mutex);
+	return 0;
+}
+
+/*
+ * Called by ALSA when the PCM substream is prepared, can set format, sample
+ * rate, etc.  This function is non atomic and can be called multiple times,
+ * it can refer to the runtime info.
+ */
+static int soc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	mutex_lock(&pcm_mutex);
+
+	if (machine->ops && machine->ops->prepare) {
+		ret = machine->ops->prepare(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: machine prepare error\n");
+			goto out;
+		}
+	}
+
+	if (platform->pcm_ops->prepare) {
+		ret = platform->pcm_ops->prepare(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: platform prepare error\n");
+			goto out;
+		}
+	}
+
+	if (codec_dai->ops.prepare) {
+		ret = codec_dai->ops.prepare(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: codec DAI prepare error\n");
+			goto out;
+		}
+	}
+
+	if (cpu_dai->ops.prepare) {
+		ret = cpu_dai->ops.prepare(substream);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
+			goto out;
+		}
+	}
+
+	/* we only want to start a DAPM playback stream if we are not waiting
+	 * on an existing one stopping */
+	if (codec_dai->pop_wait) {
+		/* we are waiting for the delayed work to start */
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+				snd_soc_dapm_stream_event(socdev->codec,
+					codec_dai->capture.stream_name,
+					SND_SOC_DAPM_STREAM_START);
+		else {
+			codec_dai->pop_wait = 0;
+			cancel_delayed_work(&socdev->delayed_work);
+			if (codec_dai->dai_ops.digital_mute)
+				codec_dai->dai_ops.digital_mute(codec_dai, 0);
+		}
+	} else {
+		/* no delayed work - do we need to power up codec */
+		if (codec->dapm_state != SNDRV_CTL_POWER_D0) {
+
+			if (codec->dapm_event)
+				codec->dapm_event(codec, SNDRV_CTL_POWER_D1);
+
+			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+				snd_soc_dapm_stream_event(codec,
+					codec_dai->playback.stream_name,
+					SND_SOC_DAPM_STREAM_START);
+			else
+				snd_soc_dapm_stream_event(codec,
+					codec_dai->capture.stream_name,
+					SND_SOC_DAPM_STREAM_START);
+
+			if (codec->dapm_event)
+				codec->dapm_event(codec, SNDRV_CTL_POWER_D0);
+			if (codec_dai->dai_ops.digital_mute)
+				codec_dai->dai_ops.digital_mute(codec_dai, 0);
+
+		} else {
+			/* codec already powered - power on widgets */
+			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+				snd_soc_dapm_stream_event(codec,
+					codec_dai->playback.stream_name,
+					SND_SOC_DAPM_STREAM_START);
+			else
+				snd_soc_dapm_stream_event(codec,
+					codec_dai->capture.stream_name,
+					SND_SOC_DAPM_STREAM_START);
+			if (codec_dai->dai_ops.digital_mute)
+				codec_dai->dai_ops.digital_mute(codec_dai, 0);
+		}
+	}
+
+out:
+	mutex_unlock(&pcm_mutex);
+	return ret;
+}
+
+/*
+ * Called by ALSA when the hardware params are set by application. This
+ * function can also be called multiple times and can allocate buffers
+ * (using snd_pcm_lib_* ). It's non-atomic.
+ */
+static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	int ret = 0;
+
+	mutex_lock(&pcm_mutex);
+
+	if (machine->ops && machine->ops->hw_params) {
+		ret = machine->ops->hw_params(substream, params);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: machine hw_params failed\n");
+			goto out;
+		}
+	}
+
+	if (codec_dai->ops.hw_params) {
+		ret = codec_dai->ops.hw_params(substream, params);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
+				codec_dai->name);
+			goto codec_err;
+		}
+	}
+
+	if (cpu_dai->ops.hw_params) {
+		ret = cpu_dai->ops.hw_params(substream, params);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't set interface %s hw params\n",
+				cpu_dai->name);
+			goto interface_err;
+		}
+	}
+
+	if (platform->pcm_ops->hw_params) {
+		ret = platform->pcm_ops->hw_params(substream, params);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't set platform %s hw params\n",
+				platform->name);
+			goto platform_err;
+		}
+	}
+
+out:
+	mutex_unlock(&pcm_mutex);
+	return ret;
+
+platform_err:
+	if (cpu_dai->ops.hw_free)
+		cpu_dai->ops.hw_free(substream);
+
+interface_err:
+	if (codec_dai->ops.hw_free)
+		codec_dai->ops.hw_free(substream);
+
+codec_err:
+	if(machine->ops && machine->ops->hw_free)
+		machine->ops->hw_free(substream);
+
+	mutex_unlock(&pcm_mutex);
+	return ret;
+}
+
+/*
+ * Free's resources allocated by hw_params, can be called multiple times
+ */
+static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	mutex_lock(&pcm_mutex);
+
+	/* apply codec digital mute */
+	if (!codec->active && codec_dai->dai_ops.digital_mute)
+		codec_dai->dai_ops.digital_mute(codec_dai, 1);
+
+	/* free any machine hw params */
+	if (machine->ops && machine->ops->hw_free)
+		machine->ops->hw_free(substream);
+
+	/* free any DMA resources */
+	if (platform->pcm_ops->hw_free)
+		platform->pcm_ops->hw_free(substream);
+
+	/* now free hw params for the DAI's  */
+	if (codec_dai->ops.hw_free)
+		codec_dai->ops.hw_free(substream);
+
+	if (cpu_dai->ops.hw_free)
+		cpu_dai->ops.hw_free(substream);
+
+	mutex_unlock(&pcm_mutex);
+	return 0;
+}
+
+static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_dai_link *machine = rtd->dai;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	int ret;
+
+	if (codec_dai->ops.trigger) {
+		ret = codec_dai->ops.trigger(substream, cmd);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (platform->pcm_ops->trigger) {
+		ret = platform->pcm_ops->trigger(substream, cmd);
+		if (ret < 0)
+			return ret;
+	}
+
+	if (cpu_dai->ops.trigger) {
+		ret = cpu_dai->ops.trigger(substream, cmd);
+		if (ret < 0)
+			return ret;
+	}
+	return 0;
+}
+
+/* ASoC PCM operations */
+static struct snd_pcm_ops soc_pcm_ops = {
+	.open		= soc_pcm_open,
+	.close		= soc_codec_close,
+	.hw_params	= soc_pcm_hw_params,
+	.hw_free	= soc_pcm_hw_free,
+	.prepare	= soc_pcm_prepare,
+	.trigger	= soc_pcm_trigger,
+};
+
+#ifdef CONFIG_PM
+/* powers down audio subsystem for suspend */
+static int soc_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ 	struct snd_soc_machine *machine = socdev->machine;
+ 	struct snd_soc_platform *platform = socdev->platform;
+ 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+
+	/* mute any active DAC's */
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
+		if (dai->dai_ops.digital_mute && dai->playback.active)
+			dai->dai_ops.digital_mute(dai, 1);
+	}
+
+	if (machine->suspend_pre)
+		machine->suspend_pre(pdev, state);
+
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_cpu_dai  *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
+			cpu_dai->suspend(pdev, cpu_dai);
+		if (platform->suspend)
+			platform->suspend(pdev, cpu_dai);
+	}
+
+	/* close any waiting streams and save state */
+	run_delayed_work(&socdev->delayed_work);
+	codec->suspend_dapm_state = codec->dapm_state;
+
+	for(i = 0; i < codec->num_dai; i++) {
+		char *stream = codec->dai[i].playback.stream_name;
+		if (stream != NULL)
+			snd_soc_dapm_stream_event(codec, stream,
+				SND_SOC_DAPM_STREAM_SUSPEND);
+		stream = codec->dai[i].capture.stream_name;
+		if (stream != NULL)
+			snd_soc_dapm_stream_event(codec, stream,
+				SND_SOC_DAPM_STREAM_SUSPEND);
+	}
+
+	if (codec_dev->suspend)
+		codec_dev->suspend(pdev, state);
+
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
+			cpu_dai->suspend(pdev, cpu_dai);
+	}
+
+	if (machine->suspend_post)
+		machine->suspend_post(pdev, state);
+
+	return 0;
+}
+
+/* powers up audio subsystem after a suspend */
+static int soc_resume(struct platform_device *pdev)
+{
+ 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ 	struct snd_soc_machine *machine = socdev->machine;
+ 	struct snd_soc_platform *platform = socdev->platform;
+ 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+
+	if (machine->resume_pre)
+		machine->resume_pre(pdev);
+
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
+			cpu_dai->resume(pdev, cpu_dai);
+	}
+
+	if (codec_dev->resume)
+		codec_dev->resume(pdev);
+
+	for(i = 0; i < codec->num_dai; i++) {
+		char* stream = codec->dai[i].playback.stream_name;
+		if (stream != NULL)
+			snd_soc_dapm_stream_event(codec, stream,
+				SND_SOC_DAPM_STREAM_RESUME);
+		stream = codec->dai[i].capture.stream_name;
+		if (stream != NULL)
+			snd_soc_dapm_stream_event(codec, stream,
+				SND_SOC_DAPM_STREAM_RESUME);
+	}
+
+	/* unmute any active DAC's */
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
+		if (dai->dai_ops.digital_mute && dai->playback.active)
+			dai->dai_ops.digital_mute(dai, 0);
+	}
+
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
+			cpu_dai->resume(pdev, cpu_dai);
+		if (platform->resume)
+			platform->resume(pdev, cpu_dai);
+	}
+
+	if (machine->resume_post)
+		machine->resume_post(pdev);
+
+	return 0;
+}
+
+#else
+#define soc_suspend	NULL
+#define soc_resume	NULL
+#endif
+
+/* probes a new socdev */
+static int soc_probe(struct platform_device *pdev)
+{
+	int ret = 0, i;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+
+	if (machine->probe) {
+		ret = machine->probe(pdev);
+		if(ret < 0)
+			return ret;
+	}
+
+	for (i = 0; i < machine->num_links; i++) {
+		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->probe) {
+			ret = cpu_dai->probe(pdev);
+			if(ret < 0)
+				goto cpu_dai_err;
+		}
+	}
+
+	if (codec_dev->probe) {
+		ret = codec_dev->probe(pdev);
+		if(ret < 0)
+			goto cpu_dai_err;
+	}
+
+	if (platform->probe) {
+		ret = platform->probe(pdev);
+		if(ret < 0)
+			goto platform_err;
+	}
+
+	/* DAPM stream work */
+	INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
+	return 0;
+
+platform_err:
+	if (codec_dev->remove)
+		codec_dev->remove(pdev);
+
+cpu_dai_err:
+	for (i--; i >= 0; i--) {
+		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->remove)
+			cpu_dai->remove(pdev);
+	}
+
+	if (machine->remove)
+		machine->remove(pdev);
+
+	return ret;
+}
+
+/* removes a socdev */
+static int soc_remove(struct platform_device *pdev)
+{
+	int i;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+
+	run_delayed_work(&socdev->delayed_work);
+
+	if (platform->remove)
+		platform->remove(pdev);
+
+	if (codec_dev->remove)
+		codec_dev->remove(pdev);
+
+	for (i = 0; i < machine->num_links; i++) {
+		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->remove)
+			cpu_dai->remove(pdev);
+	}
+
+	if (machine->remove)
+		machine->remove(pdev);
+
+	return 0;
+}
+
+/* ASoC platform driver */
+static struct platform_driver soc_driver = {
+	.driver		= {
+		.name		= "soc-audio",
+	},
+	.probe		= soc_probe,
+	.remove		= soc_remove,
+	.suspend	= soc_suspend,
+	.resume		= soc_resume,
+};
+
+/* create a new pcm */
+static int soc_new_pcm(struct snd_soc_device *socdev,
+	struct snd_soc_dai_link *dai_link, int num)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai;
+	struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai;
+	struct snd_soc_pcm_runtime *rtd;
+	struct snd_pcm *pcm;
+	char new_name[64];
+	int ret = 0, playback = 0, capture = 0;
+
+	rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL);
+	if (rtd == NULL)
+		return -ENOMEM;
+
+	rtd->dai = dai_link;
+	rtd->socdev = socdev;
+	codec_dai->codec = socdev->codec;
+
+	/* check client and interface hw capabilities */
+	sprintf(new_name, "%s %s-%s-%d",dai_link->stream_name, codec_dai->name,
+		get_dai_name(cpu_dai->type), num);
+
+	if (codec_dai->playback.channels_min)
+		playback = 1;
+	if (codec_dai->capture.channels_min)
+		capture = 1;
+
+	ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
+		capture, &pcm);
+	if (ret < 0) {
+		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+		kfree(rtd);
+		return ret;
+	}
+
+	pcm->private_data = rtd;
+	soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
+	soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
+	soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
+	soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
+	soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
+	soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
+	soc_pcm_ops.page = socdev->platform->pcm_ops->page;
+
+	if (playback)
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
+
+	if (capture)
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
+
+	ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
+	if (ret < 0) {
+		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
+		kfree(rtd);
+		return ret;
+	}
+
+	pcm->private_free = socdev->platform->pcm_free;
+	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+		cpu_dai->name);
+	return ret;
+}
+
+/* codec register dump */
+static ssize_t codec_reg_show(struct device *dev,
+	struct device_attribute *attr, char *buf)
+{
+	struct snd_soc_device *devdata = dev_get_drvdata(dev);
+	struct snd_soc_codec *codec = devdata->codec;
+	int i, step = 1, count = 0;
+
+	if (!codec->reg_cache_size)
+		return 0;
+
+	if (codec->reg_cache_step)
+		step = codec->reg_cache_step;
+
+	count += sprintf(buf, "%s registers\n", codec->name);
+	for(i = 0; i < codec->reg_cache_size; i += step)
+		count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i));
+
+	return count;
+}
+static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
+
+/**
+ * snd_soc_new_ac97_codec - initailise AC97 device
+ * @codec: audio codec
+ * @ops: AC97 bus operations
+ * @num: AC97 codec number
+ *
+ * Initialises AC97 codec resources for use by ad-hoc devices only.
+ */
+int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
+	struct snd_ac97_bus_ops *ops, int num)
+{
+	mutex_lock(&codec->mutex);
+
+	codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
+	if (codec->ac97 == NULL) {
+		mutex_unlock(&codec->mutex);
+		return -ENOMEM;
+	}
+
+	codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
+	if (codec->ac97->bus == NULL) {
+		kfree(codec->ac97);
+		codec->ac97 = NULL;
+		mutex_unlock(&codec->mutex);
+		return -ENOMEM;
+	}
+
+	codec->ac97->bus->ops = ops;
+	codec->ac97->num = num;
+	mutex_unlock(&codec->mutex);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);
+
+/**
+ * snd_soc_free_ac97_codec - free AC97 codec device
+ * @codec: audio codec
+ *
+ * Frees AC97 codec device resources.
+ */
+void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
+{
+	mutex_lock(&codec->mutex);
+	kfree(codec->ac97->bus);
+	kfree(codec->ac97);
+	codec->ac97 = NULL;
+	mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);
+
+/**
+ * snd_soc_update_bits - update codec register bits
+ * @codec: audio codec
+ * @reg: codec register
+ * @mask: register mask
+ * @value: new value
+ *
+ * Writes new register value.
+ *
+ * Returns 1 for change else 0.
+ */
+int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
+				unsigned short mask, unsigned short value)
+{
+	int change;
+	unsigned short old, new;
+
+	mutex_lock(&io_mutex);
+	old = snd_soc_read(codec, reg);
+	new = (old & ~mask) | value;
+	change = old != new;
+	if (change)
+		snd_soc_write(codec, reg, new);
+
+	mutex_unlock(&io_mutex);
+	return change;
+}
+EXPORT_SYMBOL_GPL(snd_soc_update_bits);
+
+/**
+ * snd_soc_test_bits - test register for change
+ * @codec: audio codec
+ * @reg: codec register
+ * @mask: register mask
+ * @value: new value
+ *
+ * Tests a register with a new value and checks if the new value is
+ * different from the old value.
+ *
+ * Returns 1 for change else 0.
+ */
+int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
+				unsigned short mask, unsigned short value)
+{
+	int change;
+	unsigned short old, new;
+
+	mutex_lock(&io_mutex);
+	old = snd_soc_read(codec, reg);
+	new = (old & ~mask) | value;
+	change = old != new;
+	mutex_unlock(&io_mutex);
+
+	return change;
+}
+EXPORT_SYMBOL_GPL(snd_soc_test_bits);
+
+/**
+ * snd_soc_new_pcms - create new sound card and pcms
+ * @socdev: the SoC audio device
+ *
+ * Create a new sound card based upon the codec and interface pcms.
+ *
+ * Returns 0 for success, else error.
+ */
+int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_machine *machine = socdev->machine;
+	int ret = 0, i;
+
+	mutex_lock(&codec->mutex);
+
+	/* register a sound card */
+	codec->card = snd_card_new(idx, xid, codec->owner, 0);
+	if (!codec->card) {
+		printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
+			codec->name);
+		mutex_unlock(&codec->mutex);
+		return -ENODEV;
+	}
+
+	codec->card->dev = socdev->dev;
+	codec->card->private_data = codec;
+	strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
+
+	/* create the pcms */
+	for(i = 0; i < machine->num_links; i++) {
+		ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: can't create pcm %s\n",
+				machine->dai_link[i].stream_name);
+			mutex_unlock(&codec->mutex);
+			return ret;
+		}
+	}
+
+	mutex_unlock(&codec->mutex);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
+
+/**
+ * snd_soc_register_card - register sound card
+ * @socdev: the SoC audio device
+ *
+ * Register a SoC sound card. Also registers an AC97 device if the
+ * codec is AC97 for ad hoc devices.
+ *
+ * Returns 0 for success, else error.
+ */
+int snd_soc_register_card(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_machine *machine = socdev->machine;
+	int ret = 0, i, ac97 = 0, err = 0;
+
+	mutex_lock(&codec->mutex);
+	for(i = 0; i < machine->num_links; i++) {
+		if (socdev->machine->dai_link[i].init) {
+			err = socdev->machine->dai_link[i].init(codec);
+			if (err < 0) {
+				printk(KERN_ERR "asoc: failed to init %s\n",
+					socdev->machine->dai_link[i].stream_name);
+				continue;
+			}
+		}
+		if (socdev->machine->dai_link[i].cpu_dai->type == SND_SOC_DAI_AC97)
+			ac97 = 1;
+	}
+	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
+		 "%s", machine->name);
+	snprintf(codec->card->longname, sizeof(codec->card->longname),
+		 "%s (%s)", machine->name, codec->name);
+
+	ret = snd_card_register(codec->card);
+	if (ret < 0) {
+		printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n",
+				codec->name);
+		goto out;
+	}
+
+#ifdef CONFIG_SND_SOC_AC97_BUS
+	if (ac97) {
+		ret = soc_ac97_dev_register(codec);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: AC97 device register failed\n");
+			snd_card_free(codec->card);
+			goto out;
+		}
+	}
+#endif
+
+	err = snd_soc_dapm_sys_add(socdev->dev);
+	if (err < 0)
+		printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
+
+	err = device_create_file(socdev->dev, &dev_attr_codec_reg);
+	if (err < 0)
+		printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n");
+out:
+	mutex_unlock(&codec->mutex);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_card);
+
+/**
+ * snd_soc_free_pcms - free sound card and pcms
+ * @socdev: the SoC audio device
+ *
+ * Frees sound card and pcms associated with the socdev.
+ * Also unregister the codec if it is an AC97 device.
+ */
+void snd_soc_free_pcms(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+
+	mutex_lock(&codec->mutex);
+#ifdef CONFIG_SND_SOC_AC97_BUS
+	if (codec->ac97)
+		soc_ac97_dev_unregister(codec);
+#endif
+
+	if (codec->card)
+		snd_card_free(codec->card);
+	device_remove_file(socdev->dev, &dev_attr_codec_reg);
+	mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_free_pcms);
+
+/**
+ * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
+ * @substream: the pcm substream
+ * @hw: the hardware parameters
+ *
+ * Sets the substream runtime hardware parameters.
+ */
+int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
+	const struct snd_pcm_hardware *hw)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	runtime->hw.info = hw->info;
+	runtime->hw.formats = hw->formats;
+	runtime->hw.period_bytes_min = hw->period_bytes_min;
+	runtime->hw.period_bytes_max = hw->period_bytes_max;
+	runtime->hw.periods_min = hw->periods_min;
+	runtime->hw.periods_max = hw->periods_max;
+	runtime->hw.buffer_bytes_max = hw->buffer_bytes_max;
+	runtime->hw.fifo_size = hw->fifo_size;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);
+
+/**
+ * snd_soc_cnew - create new control
+ * @_template: control template
+ * @data: control private data
+ * @lnng_name: control long name
+ *
+ * Create a new mixer control from a template control.
+ *
+ * Returns 0 for success, else error.
+ */
+struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
+	void *data, char *long_name)
+{
+	struct snd_kcontrol_new template;
+
+	memcpy(&template, _template, sizeof(template));
+	if (long_name)
+		template.name = long_name;
+	template.access = SNDRV_CTL_ELEM_ACCESS_READWRITE;
+	template.index = 0;
+
+	return snd_ctl_new1(&template, data);
+}
+EXPORT_SYMBOL_GPL(snd_soc_cnew);
+
+/**
+ * snd_soc_info_enum_double - enumerated double mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a double enumerated
+ * mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
+	uinfo->value.enumerated.items = e->mask;
+
+	if (uinfo->value.enumerated.item > e->mask - 1)
+		uinfo->value.enumerated.item = e->mask - 1;
+	strcpy(uinfo->value.enumerated.name,
+		e->texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);
+
+/**
+ * snd_soc_get_enum_double - enumerated double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a double enumerated mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned short val, bitmask;
+
+	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+		;
+	val = snd_soc_read(codec, e->reg);
+	ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
+	if (e->shift_l != e->shift_r)
+		ucontrol->value.enumerated.item[1] =
+			(val >> e->shift_r) & (bitmask - 1);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);
+
+/**
+ * snd_soc_put_enum_double - enumerated double mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a double enumerated mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned short val;
+	unsigned short mask, bitmask;
+
+	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+		;
+	if (ucontrol->value.enumerated.item[0] > e->mask - 1)
+		return -EINVAL;
+	val = ucontrol->value.enumerated.item[0] << e->shift_l;
+	mask = (bitmask - 1) << e->shift_l;
+	if (e->shift_l != e->shift_r) {
+		if (ucontrol->value.enumerated.item[1] > e->mask - 1)
+			return -EINVAL;
+		val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+		mask |= (bitmask - 1) << e->shift_r;
+	}
+
+	return snd_soc_update_bits(codec, e->reg, mask, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
+
+/**
+ * snd_soc_info_enum_ext - external enumerated single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about an external enumerated
+ * single mixer.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = e->mask;
+
+	if (uinfo->value.enumerated.item > e->mask - 1)
+		uinfo->value.enumerated.item = e->mask - 1;
+	strcpy(uinfo->value.enumerated.name,
+		e->texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);
+
+/**
+ * snd_soc_info_volsw_ext - external single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single external mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	int mask = kcontrol->private_value;
+
+	uinfo->type =
+		mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = mask;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);
+
+/**
+ * snd_soc_info_bool_ext - external single boolean mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single boolean external mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_bool_ext);
+
+/**
+ * snd_soc_info_volsw - single mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	int mask = (kcontrol->private_value >> 16) & 0xff;
+	int shift = (kcontrol->private_value >> 8) & 0x0f;
+	int rshift = (kcontrol->private_value >> 12) & 0x0f;
+
+	uinfo->type =
+		mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = shift == rshift ? 1 : 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = mask;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw);
+
+/**
+ * snd_soc_get_volsw - single mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int shift = (kcontrol->private_value >> 8) & 0x0f;
+	int rshift = (kcontrol->private_value >> 12) & 0x0f;
+	int mask = (kcontrol->private_value >> 16) & 0xff;
+	int invert = (kcontrol->private_value >> 24) & 0x01;
+
+	ucontrol->value.integer.value[0] =
+		(snd_soc_read(codec, reg) >> shift) & mask;
+	if (shift != rshift)
+		ucontrol->value.integer.value[1] =
+			(snd_soc_read(codec, reg) >> rshift) & mask;
+	if (invert) {
+		ucontrol->value.integer.value[0] =
+			mask - ucontrol->value.integer.value[0];
+		if (shift != rshift)
+			ucontrol->value.integer.value[1] =
+				mask - ucontrol->value.integer.value[1];
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw);
+
+/**
+ * snd_soc_put_volsw - single mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int shift = (kcontrol->private_value >> 8) & 0x0f;
+	int rshift = (kcontrol->private_value >> 12) & 0x0f;
+	int mask = (kcontrol->private_value >> 16) & 0xff;
+	int invert = (kcontrol->private_value >> 24) & 0x01;
+	int err;
+	unsigned short val, val2, val_mask;
+
+	val = (ucontrol->value.integer.value[0] & mask);
+	if (invert)
+		val = mask - val;
+	val_mask = mask << shift;
+	val = val << shift;
+	if (shift != rshift) {
+		val2 = (ucontrol->value.integer.value[1] & mask);
+		if (invert)
+			val2 = mask - val2;
+		val_mask |= mask << rshift;
+		val |= val2 << rshift;
+	}
+	err = snd_soc_update_bits(codec, reg, val_mask, val);
+	return err;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
+
+/**
+ * snd_soc_info_volsw_2r - double mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a double mixer control that
+ * spans 2 codec registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	int mask = (kcontrol->private_value >> 12) & 0xff;
+
+	uinfo->type =
+		mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = mask;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);
+
+/**
+ * snd_soc_get_volsw_2r - double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a double mixer control that spans 2 registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int reg2 = (kcontrol->private_value >> 24) & 0xff;
+	int shift = (kcontrol->private_value >> 8) & 0x0f;
+	int mask = (kcontrol->private_value >> 12) & 0xff;
+	int invert = (kcontrol->private_value >> 20) & 0x01;
+
+	ucontrol->value.integer.value[0] =
+		(snd_soc_read(codec, reg) >> shift) & mask;
+	ucontrol->value.integer.value[1] =
+		(snd_soc_read(codec, reg2) >> shift) & mask;
+	if (invert) {
+		ucontrol->value.integer.value[0] =
+			mask - ucontrol->value.integer.value[0];
+		ucontrol->value.integer.value[1] =
+			mask - ucontrol->value.integer.value[1];
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);
+
+/**
+ * snd_soc_put_volsw_2r - double mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a double mixer control that spans 2 registers.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int reg2 = (kcontrol->private_value >> 24) & 0xff;
+	int shift = (kcontrol->private_value >> 8) & 0x0f;
+	int mask = (kcontrol->private_value >> 12) & 0xff;
+	int invert = (kcontrol->private_value >> 20) & 0x01;
+	int err;
+	unsigned short val, val2, val_mask;
+
+	val_mask = mask << shift;
+	val = (ucontrol->value.integer.value[0] & mask);
+	val2 = (ucontrol->value.integer.value[1] & mask);
+
+	if (invert) {
+		val = mask - val;
+		val2 = mask - val2;
+	}
+
+	val = val << shift;
+	val2 = val2 << shift;
+
+	if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0)
+		return err;
+
+	err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+	return err;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
+
+static int __devinit snd_soc_init(void)
+{
+	printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
+	return platform_driver_register(&soc_driver);
+}
+
+static void snd_soc_exit(void)
+{
+ 	platform_driver_unregister(&soc_driver);
+}
+
+module_init(snd_soc_init);
+module_exit(snd_soc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("ALSA SoC Core");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
new file mode 100644
index 00000000000..7caf8c7b0ac
--- /dev/null
+++ b/sound/soc/soc-dapm.c
@@ -0,0 +1,1323 @@
+/*
+ * soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Revision history
+ *    12th Aug 2005   Initial version.
+ *    25th Oct 2005   Implemented path power domain.
+ *    18th Dec 2005   Implemented machine and stream level power domain.
+ *
+ *  Features:
+ *    o Changes power status of internal codec blocks depending on the
+ *      dynamic configuration of codec internal audio paths and active
+ *      DAC's/ADC's.
+ *    o Platform power domain - can support external components i.e. amps and
+ *      mic/meadphone insertion events.
+ *    o Automatic Mic Bias support
+ *    o Jack insertion power event initiation - e.g. hp insertion will enable
+ *      sinks, dacs, etc
+ *    o Delayed powerdown of audio susbsytem to reduce pops between a quick
+ *      device reopen.
+ *
+ *  Todo:
+ *    o DAPM power change sequencing - allow for configurable per
+ *      codec sequences.
+ *    o Support for analogue bias optimisation.
+ *    o Support for reduced codec oversampling rates.
+ *    o Support for reduced codec bias currents.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/bitops.h>
+#include <linux/platform_device.h>
+#include <linux/jiffies.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+/* debug */
+#define DAPM_DEBUG 0
+#if DAPM_DEBUG
+#define dump_dapm(codec, action) dbg_dump_dapm(codec, action)
+#define dbg(format, arg...) printk(format, ## arg)
+#else
+#define dump_dapm(codec, action)
+#define dbg(format, arg...)
+#endif
+
+#define POP_DEBUG 0
+#if POP_DEBUG
+#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */
+#define pop_wait(time) schedule_timeout_interruptible(msecs_to_jiffies(time))
+#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME)
+#else
+#define pop_dbg(format, arg...)
+#define pop_wait(time)
+#endif
+
+/* dapm power sequences - make this per codec in the future */
+static int dapm_up_seq[] = {
+	snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
+	snd_soc_dapm_mux, snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_pga,
+	snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
+};
+static int dapm_down_seq[] = {
+	snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
+	snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic,
+	snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_post
+};
+
+static int dapm_status = 1;
+module_param(dapm_status, int, 0);
+MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
+
+/* create a new dapm widget */
+static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
+	const struct snd_soc_dapm_widget *_widget)
+{
+	return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
+}
+
+/* set up initial codec paths */
+static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
+	struct snd_soc_dapm_path *p, int i)
+{
+	switch (w->id) {
+	case snd_soc_dapm_switch:
+	case snd_soc_dapm_mixer: {
+		int val;
+		int reg = w->kcontrols[i].private_value & 0xff;
+		int shift = (w->kcontrols[i].private_value >> 8) & 0x0f;
+		int mask = (w->kcontrols[i].private_value >> 16) & 0xff;
+		int invert = (w->kcontrols[i].private_value >> 24) & 0x01;
+
+		val = snd_soc_read(w->codec, reg);
+		val = (val >> shift) & mask;
+
+		if ((invert && !val) || (!invert && val))
+			p->connect = 1;
+		else
+			p->connect = 0;
+	}
+	break;
+	case snd_soc_dapm_mux: {
+		struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value;
+		int val, item, bitmask;
+
+		for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+		;
+		val = snd_soc_read(w->codec, e->reg);
+		item = (val >> e->shift_l) & (bitmask - 1);
+
+		p->connect = 0;
+		for (i = 0; i < e->mask; i++) {
+			if (!(strcmp(p->name, e->texts[i])) && item == i)
+				p->connect = 1;
+		}
+	}
+	break;
+	/* does not effect routing - always connected */
+	case snd_soc_dapm_pga:
+	case snd_soc_dapm_output:
+	case snd_soc_dapm_adc:
+	case snd_soc_dapm_input:
+	case snd_soc_dapm_dac:
+	case snd_soc_dapm_micbias:
+	case snd_soc_dapm_vmid:
+		p->connect = 1;
+	break;
+	/* does effect routing - dynamically connected */
+	case snd_soc_dapm_hp:
+	case snd_soc_dapm_mic:
+	case snd_soc_dapm_spk:
+	case snd_soc_dapm_line:
+	case snd_soc_dapm_pre:
+	case snd_soc_dapm_post:
+		p->connect = 0;
+	break;
+	}
+}
+
+/* connect mux widget to it's interconnecting audio paths */
+static int dapm_connect_mux(struct snd_soc_codec *codec,
+	struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
+	struct snd_soc_dapm_path *path, const char *control_name,
+	const struct snd_kcontrol_new *kcontrol)
+{
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	int i;
+
+	for (i = 0; i < e->mask; i++) {
+		if (!(strcmp(control_name, e->texts[i]))) {
+			list_add(&path->list, &codec->dapm_paths);
+			list_add(&path->list_sink, &dest->sources);
+			list_add(&path->list_source, &src->sinks);
+			path->name = (char*)e->texts[i];
+			dapm_set_path_status(dest, path, 0);
+			return 0;
+		}
+	}
+
+	return -ENODEV;
+}
+
+/* connect mixer widget to it's interconnecting audio paths */
+static int dapm_connect_mixer(struct snd_soc_codec *codec,
+	struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
+	struct snd_soc_dapm_path *path, const char *control_name)
+{
+	int i;
+
+	/* search for mixer kcontrol */
+	for (i = 0; i < dest->num_kcontrols; i++) {
+		if (!strcmp(control_name, dest->kcontrols[i].name)) {
+			list_add(&path->list, &codec->dapm_paths);
+			list_add(&path->list_sink, &dest->sources);
+			list_add(&path->list_source, &src->sinks);
+			path->name = dest->kcontrols[i].name;
+			dapm_set_path_status(dest, path, i);
+			return 0;
+		}
+	}
+	return -ENODEV;
+}
+
+/* update dapm codec register bits */
+static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
+{
+	int change, power;
+	unsigned short old, new;
+	struct snd_soc_codec *codec = widget->codec;
+
+	/* check for valid widgets */
+	if (widget->reg < 0 || widget->id == snd_soc_dapm_input ||
+		widget->id == snd_soc_dapm_output ||
+		widget->id == snd_soc_dapm_hp ||
+		widget->id == snd_soc_dapm_mic ||
+		widget->id == snd_soc_dapm_line ||
+		widget->id == snd_soc_dapm_spk)
+		return 0;
+
+	power = widget->power;
+	if (widget->invert)
+		power = (power ? 0:1);
+
+	old = snd_soc_read(codec, widget->reg);
+	new = (old & ~(0x1 << widget->shift)) | (power << widget->shift);
+
+	change = old != new;
+	if (change) {
+		pop_dbg("pop test %s : %s in %d ms\n", widget->name,
+			widget->power ? "on" : "off", POP_TIME);
+		snd_soc_write(codec, widget->reg, new);
+		pop_wait(POP_TIME);
+	}
+	dbg("reg old %x new %x change %d\n", old, new, change);
+	return change;
+}
+
+/* ramps the volume up or down to minimise pops before or after a
+ * DAPM power event */
+static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power)
+{
+	const struct snd_kcontrol_new *k = widget->kcontrols;
+
+	if (widget->muted && !power)
+		return 0;
+	if (!widget->muted && power)
+		return 0;
+
+	if (widget->num_kcontrols && k) {
+		int reg = k->private_value & 0xff;
+		int shift = (k->private_value >> 8) & 0x0f;
+		int mask = (k->private_value >> 16) & 0xff;
+		int invert = (k->private_value >> 24) & 0x01;
+
+		if (power) {
+			int i;
+			/* power up has happended, increase volume to last level */
+			if (invert) {
+				for (i = mask; i > widget->saved_value; i--)
+					snd_soc_update_bits(widget->codec, reg, mask, i);
+			} else {
+				for (i = 0; i < widget->saved_value; i++)
+					snd_soc_update_bits(widget->codec, reg, mask, i);
+			}
+			widget->muted = 0;
+		} else {
+			/* power down is about to occur, decrease volume to mute */
+			int val = snd_soc_read(widget->codec, reg);
+			int i = widget->saved_value = (val >> shift) & mask;
+			if (invert) {
+				for (; i < mask; i++)
+					snd_soc_update_bits(widget->codec, reg, mask, i);
+			} else {
+				for (; i > 0; i--)
+					snd_soc_update_bits(widget->codec, reg, mask, i);
+			}
+			widget->muted = 1;
+		}
+	}
+	return 0;
+}
+
+/* create new dapm mixer control */
+static int dapm_new_mixer(struct snd_soc_codec *codec,
+	struct snd_soc_dapm_widget *w)
+{
+	int i, ret = 0;
+	char name[32];
+	struct snd_soc_dapm_path *path;
+
+	/* add kcontrol */
+	for (i = 0; i < w->num_kcontrols; i++) {
+
+		/* match name */
+		list_for_each_entry(path, &w->sources, list_sink) {
+
+			/* mixer/mux paths name must match control name */
+			if (path->name != (char*)w->kcontrols[i].name)
+				continue;
+
+			/* add dapm control with long name */
+			snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name);
+			path->long_name = kstrdup (name, GFP_KERNEL);
+			if (path->long_name == NULL)
+				return -ENOMEM;
+
+			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
+				path->long_name);
+			ret = snd_ctl_add(codec->card, path->kcontrol);
+			if (ret < 0) {
+				printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n",
+						path->long_name);
+				kfree(path->long_name);
+				path->long_name = NULL;
+				return ret;
+			}
+		}
+	}
+	return ret;
+}
+
+/* create new dapm mux control */
+static int dapm_new_mux(struct snd_soc_codec *codec,
+	struct snd_soc_dapm_widget *w)
+{
+	struct snd_soc_dapm_path *path = NULL;
+	struct snd_kcontrol *kcontrol;
+	int ret = 0;
+
+	if (!w->num_kcontrols) {
+		printk(KERN_ERR "asoc: mux %s has no controls\n", w->name);
+		return -EINVAL;
+	}
+
+	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+	ret = snd_ctl_add(codec->card, kcontrol);
+	if (ret < 0)
+		goto err;
+
+	list_for_each_entry(path, &w->sources, list_sink)
+		path->kcontrol = kcontrol;
+
+	return ret;
+
+err:
+	printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
+	return ret;
+}
+
+/* create new dapm volume control */
+static int dapm_new_pga(struct snd_soc_codec *codec,
+	struct snd_soc_dapm_widget *w)
+{
+	struct snd_kcontrol *kcontrol;
+	int ret = 0;
+
+	if (!w->num_kcontrols)
+		return -EINVAL;
+
+	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+	ret = snd_ctl_add(codec->card, kcontrol);
+	if (ret < 0) {
+		printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
+		return ret;
+	}
+
+	return ret;
+}
+
+/* reset 'walked' bit for each dapm path */
+static inline void dapm_clear_walk(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_path *p;
+
+	list_for_each_entry(p, &codec->dapm_paths, list)
+		p->walked = 0;
+}
+
+/*
+ * Recursively check for a completed path to an active or physically connected
+ * output widget. Returns number of complete paths.
+ */
+static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
+{
+	struct snd_soc_dapm_path *path;
+	int con = 0;
+
+	if (widget->id == snd_soc_dapm_adc && widget->active)
+		return 1;
+
+	if (widget->connected) {
+		/* connected pin ? */
+		if (widget->id == snd_soc_dapm_output && !widget->ext)
+			return 1;
+
+		/* connected jack or spk ? */
+		if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
+			widget->id == snd_soc_dapm_line)
+			return 1;
+	}
+
+	list_for_each_entry(path, &widget->sinks, list_source) {
+		if (path->walked)
+			continue;
+
+		if (path->sink && path->connect) {
+			path->walked = 1;
+			con += is_connected_output_ep(path->sink);
+		}
+	}
+
+	return con;
+}
+
+/*
+ * Recursively check for a completed path to an active or physically connected
+ * input widget. Returns number of complete paths.
+ */
+static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
+{
+	struct snd_soc_dapm_path *path;
+	int con = 0;
+
+	/* active stream ? */
+	if (widget->id == snd_soc_dapm_dac && widget->active)
+		return 1;
+
+	if (widget->connected) {
+		/* connected pin ? */
+		if (widget->id == snd_soc_dapm_input && !widget->ext)
+			return 1;
+
+		/* connected VMID/Bias for lower pops */
+		if (widget->id == snd_soc_dapm_vmid)
+			return 1;
+
+		/* connected jack ? */
+		if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+			return 1;
+	}
+
+	list_for_each_entry(path, &widget->sources, list_sink) {
+		if (path->walked)
+			continue;
+
+		if (path->source && path->connect) {
+			path->walked = 1;
+			con += is_connected_input_ep(path->source);
+		}
+	}
+
+	return con;
+}
+
+/*
+ * Scan each dapm widget for complete audio path.
+ * A complete path is a route that has valid endpoints i.e.:-
+ *
+ *  o DAC to output pin.
+ *  o Input Pin to ADC.
+ *  o Input pin to Output pin (bypass, sidetone)
+ *  o DAC to ADC (loopback).
+ */
+static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
+{
+	struct snd_soc_dapm_widget *w;
+	int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power;
+
+	/* do we have a sequenced stream event */
+	if (event == SND_SOC_DAPM_STREAM_START) {
+		c = ARRAY_SIZE(dapm_up_seq);
+		seq = dapm_up_seq;
+	} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+		c = ARRAY_SIZE(dapm_down_seq);
+		seq = dapm_down_seq;
+	}
+
+	for(i = 0; i < c; i++) {
+		list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+			/* is widget in stream order */
+			if (seq && seq[i] && w->id != seq[i])
+				continue;
+
+			/* vmid - no action */
+			if (w->id == snd_soc_dapm_vmid)
+				continue;
+
+			/* active ADC */
+			if (w->id == snd_soc_dapm_adc && w->active) {
+				in = is_connected_input_ep(w);
+				dapm_clear_walk(w->codec);
+				w->power = (in != 0) ? 1 : 0;
+				dapm_update_bits(w);
+				continue;
+			}
+
+			/* active DAC */
+			if (w->id == snd_soc_dapm_dac && w->active) {
+				out = is_connected_output_ep(w);
+				dapm_clear_walk(w->codec);
+				w->power = (out != 0) ? 1 : 0;
+				dapm_update_bits(w);
+				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)
+					continue;
+
+				if (event == SND_SOC_DAPM_STREAM_START) {
+					ret = w->event(w, SND_SOC_DAPM_PRE_PMU);
+					if (ret < 0)
+						return ret;
+				} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+					ret = w->event(w, SND_SOC_DAPM_PRE_PMD);
+					if (ret < 0)
+						return ret;
+				}
+				continue;
+			}
+			if (w->id == snd_soc_dapm_post) {
+				if (!w->event)
+					continue;
+
+				if (event == SND_SOC_DAPM_STREAM_START) {
+					ret = w->event(w, SND_SOC_DAPM_POST_PMU);
+					if (ret < 0)
+						return ret;
+				} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+					ret = w->event(w, SND_SOC_DAPM_POST_PMD);
+					if (ret < 0)
+						return ret;
+				}
+				continue;
+			}
+
+			/* all other widgets */
+			in = is_connected_input_ep(w);
+			dapm_clear_walk(w->codec);
+			out = is_connected_output_ep(w);
+			dapm_clear_walk(w->codec);
+			power = (out != 0 && in != 0) ? 1 : 0;
+			power_change = (w->power == power) ? 0: 1;
+			w->power = power;
+
+			/* call any power change event handlers */
+			if (power_change) {
+				if (w->event) {
+					dbg("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, 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, 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, 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, SND_SOC_DAPM_POST_PMD);
+							if (ret < 0)
+								return ret;
+						}
+					}
+				} else
+					/* no event handler */
+					dapm_update_bits(w);
+			}
+		}
+	}
+
+	return ret;
+}
+
+#if DAPM_DEBUG
+static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
+{
+	struct snd_soc_dapm_widget *w;
+	struct snd_soc_dapm_path *p = NULL;
+	int in, out;
+
+	printk("DAPM %s %s\n", codec->name, action);
+
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+		/* only display widgets that effect routing */
+		switch (w->id) {
+		case snd_soc_dapm_pre:
+		case snd_soc_dapm_post:
+		case snd_soc_dapm_vmid:
+			continue;
+		case snd_soc_dapm_mux:
+		case snd_soc_dapm_output:
+		case snd_soc_dapm_input:
+		case snd_soc_dapm_switch:
+		case snd_soc_dapm_hp:
+		case snd_soc_dapm_mic:
+		case snd_soc_dapm_spk:
+		case snd_soc_dapm_line:
+		case snd_soc_dapm_micbias:
+		case snd_soc_dapm_dac:
+		case snd_soc_dapm_adc:
+		case snd_soc_dapm_pga:
+		case snd_soc_dapm_mixer:
+			if (w->name) {
+				in = is_connected_input_ep(w);
+				dapm_clear_walk(w->codec);
+				out = is_connected_output_ep(w);
+				dapm_clear_walk(w->codec);
+				printk("%s: %s  in %d out %d\n", w->name,
+					w->power ? "On":"Off",in, out);
+
+				list_for_each_entry(p, &w->sources, list_sink) {
+					if (p->connect)
+						printk(" in  %s %s\n", p->name ? p->name : "static",
+							p->source->name);
+				}
+				list_for_each_entry(p, &w->sinks, list_source) {
+					if (p->connect)
+						printk(" out %s %s\n", p->name ? p->name : "static",
+							p->sink->name);
+				}
+			}
+		break;
+		}
+	}
+}
+#endif
+
+/* test and update the power status of a mux widget */
+static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
+				 struct snd_kcontrol *kcontrol, int mask,
+				 int val, struct soc_enum* e)
+{
+	struct snd_soc_dapm_path *path;
+	int found = 0;
+
+	if (widget->id != snd_soc_dapm_mux)
+		return -ENODEV;
+
+	if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
+		return 0;
+
+	/* find dapm widget path assoc with kcontrol */
+	list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+		if (path->kcontrol != kcontrol)
+			continue;
+
+		if (!path->name || ! e->texts[val])
+			continue;
+
+		found = 1;
+		/* we now need to match the string in the enum to the path */
+		if (!(strcmp(path->name, e->texts[val])))
+			path->connect = 1; /* new connection */
+		else
+			path->connect = 0; /* old connection must be powered down */
+	}
+
+	if (found)
+		dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+
+	return 0;
+}
+
+/* test and update the power status of a mixer widget */
+static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
+				   struct snd_kcontrol *kcontrol, int reg,
+				   int val_mask, int val, int invert)
+{
+	struct snd_soc_dapm_path *path;
+	int found = 0;
+
+	if (widget->id != snd_soc_dapm_mixer)
+		return -ENODEV;
+
+	if (!snd_soc_test_bits(widget->codec, reg, val_mask, val))
+		return 0;
+
+	/* find dapm widget path assoc with kcontrol */
+	list_for_each_entry(path, &widget->codec->dapm_paths, list) {
+		if (path->kcontrol != kcontrol)
+			continue;
+
+		/* found, now check type */
+		found = 1;
+		if (val)
+			/* new connection */
+			path->connect = invert ? 0:1;
+		else
+			/* old connection must be powered down */
+			path->connect = invert ? 1:0;
+		break;
+	}
+
+	if (found)
+		dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+
+	return 0;
+}
+
+/* show dapm widget status in sys fs */
+static ssize_t dapm_widget_show(struct device *dev,
+	struct device_attribute *attr, char *buf)
+{
+	struct snd_soc_device *devdata = dev_get_drvdata(dev);
+	struct snd_soc_codec *codec = devdata->codec;
+	struct snd_soc_dapm_widget *w;
+	int count = 0;
+	char *state = "not set";
+
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+		/* only display widgets that burnm power */
+		switch (w->id) {
+		case snd_soc_dapm_hp:
+		case snd_soc_dapm_mic:
+		case snd_soc_dapm_spk:
+		case snd_soc_dapm_line:
+		case snd_soc_dapm_micbias:
+		case snd_soc_dapm_dac:
+		case snd_soc_dapm_adc:
+		case snd_soc_dapm_pga:
+		case snd_soc_dapm_mixer:
+			if (w->name)
+				count += sprintf(buf + count, "%s: %s\n",
+					w->name, w->power ? "On":"Off");
+		break;
+		default:
+		break;
+		}
+	}
+
+	switch(codec->dapm_state){
+	case SNDRV_CTL_POWER_D0:
+		state = "D0";
+		break;
+	case SNDRV_CTL_POWER_D1:
+		state = "D1";
+		break;
+	case SNDRV_CTL_POWER_D2:
+		state = "D2";
+		break;
+	case SNDRV_CTL_POWER_D3hot:
+		state = "D3hot";
+		break;
+	case SNDRV_CTL_POWER_D3cold:
+		state = "D3cold";
+		break;
+	}
+	count += sprintf(buf + count, "PM State: %s\n", state);
+
+	return count;
+}
+
+static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
+
+int snd_soc_dapm_sys_add(struct device *dev)
+{
+	int ret = 0;
+
+	if (dapm_status)
+		ret = device_create_file(dev, &dev_attr_dapm_widget);
+
+	return ret;
+}
+
+static void snd_soc_dapm_sys_remove(struct device *dev)
+{
+	if (dapm_status)
+		device_remove_file(dev, &dev_attr_dapm_widget);
+}
+
+/* free all dapm widgets and resources */
+static void dapm_free_widgets(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_widget *w, *next_w;
+	struct snd_soc_dapm_path *p, *next_p;
+
+	list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) {
+		list_del(&w->list);
+		kfree(w);
+	}
+
+	list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) {
+		list_del(&p->list);
+		kfree(p->long_name);
+		kfree(p);
+	}
+}
+
+/**
+ * snd_soc_dapm_sync_endpoints - scan and power dapm paths
+ * @codec: audio codec
+ *
+ * Walks all dapm audio paths and powers widgets according to their
+ * stream or path usage.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec)
+{
+	return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints);
+
+/**
+ * snd_soc_dapm_connect_input - connect dapm widgets
+ * @codec: audio codec
+ * @sink: name of target widget
+ * @control: mixer control name
+ * @source: name of source name
+ *
+ * Connects 2 dapm widgets together via a named audio path. The sink is
+ * the widget receiving the audio signal, whilst the source is the sender
+ * of the audio signal.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink,
+	const char * control, const char *source)
+{
+	struct snd_soc_dapm_path *path;
+	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
+	int ret = 0;
+
+	/* find src and dest widgets */
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+
+		if (!wsink && !(strcmp(w->name, sink))) {
+			wsink = w;
+			continue;
+		}
+		if (!wsource && !(strcmp(w->name, source))) {
+			wsource = w;
+		}
+	}
+
+	if (wsource == NULL || wsink == NULL)
+		return -ENODEV;
+
+	path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL);
+	if (!path)
+		return -ENOMEM;
+
+	path->source = wsource;
+	path->sink = wsink;
+	INIT_LIST_HEAD(&path->list);
+	INIT_LIST_HEAD(&path->list_source);
+	INIT_LIST_HEAD(&path->list_sink);
+
+	/* check for external widgets */
+	if (wsink->id == snd_soc_dapm_input) {
+		if (wsource->id == snd_soc_dapm_micbias ||
+			wsource->id == snd_soc_dapm_mic ||
+			wsink->id == snd_soc_dapm_line)
+			wsink->ext = 1;
+	}
+	if (wsource->id == snd_soc_dapm_output) {
+		if (wsink->id == snd_soc_dapm_spk ||
+			wsink->id == snd_soc_dapm_hp ||
+			wsink->id == snd_soc_dapm_line)
+			wsource->ext = 1;
+	}
+
+	/* connect static paths */
+	if (control == NULL) {
+		list_add(&path->list, &codec->dapm_paths);
+		list_add(&path->list_sink, &wsink->sources);
+		list_add(&path->list_source, &wsource->sinks);
+		path->connect = 1;
+		return 0;
+	}
+
+	/* connect dynamic paths */
+	switch(wsink->id) {
+	case snd_soc_dapm_adc:
+	case snd_soc_dapm_dac:
+	case snd_soc_dapm_pga:
+	case snd_soc_dapm_input:
+	case snd_soc_dapm_output:
+	case snd_soc_dapm_micbias:
+	case snd_soc_dapm_vmid:
+	case snd_soc_dapm_pre:
+	case snd_soc_dapm_post:
+		list_add(&path->list, &codec->dapm_paths);
+		list_add(&path->list_sink, &wsink->sources);
+		list_add(&path->list_source, &wsource->sinks);
+		path->connect = 1;
+		return 0;
+	case snd_soc_dapm_mux:
+		ret = dapm_connect_mux(codec, wsource, wsink, path, control,
+			&wsink->kcontrols[0]);
+		if (ret != 0)
+			goto err;
+		break;
+	case snd_soc_dapm_switch:
+	case snd_soc_dapm_mixer:
+		ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
+		if (ret != 0)
+			goto err;
+		break;
+	case snd_soc_dapm_hp:
+	case snd_soc_dapm_mic:
+	case snd_soc_dapm_line:
+	case snd_soc_dapm_spk:
+		list_add(&path->list, &codec->dapm_paths);
+		list_add(&path->list_sink, &wsink->sources);
+		list_add(&path->list_source, &wsource->sinks);
+		path->connect = 0;
+		return 0;
+	}
+	return 0;
+
+err:
+	printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source,
+		control, sink);
+	kfree(path);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input);
+
+/**
+ * snd_soc_dapm_new_widgets - add new dapm widgets
+ * @codec: audio codec
+ *
+ * Checks the codec for any new dapm widgets and creates them if found.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_widget *w;
+
+	mutex_lock(&codec->mutex);
+	list_for_each_entry(w, &codec->dapm_widgets, list)
+	{
+		if (w->new)
+			continue;
+
+		switch(w->id) {
+		case snd_soc_dapm_switch:
+		case snd_soc_dapm_mixer:
+			dapm_new_mixer(codec, w);
+			break;
+		case snd_soc_dapm_mux:
+			dapm_new_mux(codec, w);
+			break;
+		case snd_soc_dapm_adc:
+		case snd_soc_dapm_dac:
+		case snd_soc_dapm_pga:
+			dapm_new_pga(codec, w);
+			break;
+		case snd_soc_dapm_input:
+		case snd_soc_dapm_output:
+		case snd_soc_dapm_micbias:
+		case snd_soc_dapm_spk:
+		case snd_soc_dapm_hp:
+		case snd_soc_dapm_mic:
+		case snd_soc_dapm_line:
+		case snd_soc_dapm_vmid:
+		case snd_soc_dapm_pre:
+		case snd_soc_dapm_post:
+			break;
+		}
+		w->new = 1;
+	}
+
+	dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+	mutex_unlock(&codec->mutex);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
+
+/**
+ * snd_soc_dapm_get_volsw - dapm mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a dapm mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int shift = (kcontrol->private_value >> 8) & 0x0f;
+	int rshift = (kcontrol->private_value >> 12) & 0x0f;
+	int mask = (kcontrol->private_value >> 16) & 0xff;
+	int invert = (kcontrol->private_value >> 24) & 0x01;
+
+	/* return the saved value if we are powered down */
+	if (widget->id == snd_soc_dapm_pga && !widget->power) {
+		ucontrol->value.integer.value[0] = widget->saved_value;
+		return 0;
+	}
+
+	ucontrol->value.integer.value[0] =
+		(snd_soc_read(widget->codec, reg) >> shift) & mask;
+	if (shift != rshift)
+		ucontrol->value.integer.value[1] =
+			(snd_soc_read(widget->codec, reg) >> rshift) & mask;
+	if (invert) {
+		ucontrol->value.integer.value[0] =
+			mask - ucontrol->value.integer.value[0];
+		if (shift != rshift)
+			ucontrol->value.integer.value[1] =
+				mask - ucontrol->value.integer.value[1];
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
+
+/**
+ * snd_soc_dapm_put_volsw - dapm mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a dapm mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int shift = (kcontrol->private_value >> 8) & 0x0f;
+	int rshift = (kcontrol->private_value >> 12) & 0x0f;
+	int mask = (kcontrol->private_value >> 16) & 0xff;
+	int invert = (kcontrol->private_value >> 24) & 0x01;
+	unsigned short val, val2, val_mask;
+	int ret;
+
+	val = (ucontrol->value.integer.value[0] & mask);
+
+	if (invert)
+		val = mask - val;
+	val_mask = mask << shift;
+	val = val << shift;
+	if (shift != rshift) {
+		val2 = (ucontrol->value.integer.value[1] & mask);
+		if (invert)
+			val2 = mask - val2;
+		val_mask |= mask << rshift;
+		val |= val2 << rshift;
+	}
+
+	mutex_lock(&widget->codec->mutex);
+	widget->value = val;
+
+	/* save volume value if the widget is powered down */
+	if (widget->id == snd_soc_dapm_pga && !widget->power) {
+		widget->saved_value = val;
+		mutex_unlock(&widget->codec->mutex);
+		return 1;
+	}
+
+	dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert);
+	if (widget->event) {
+		if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+			ret = widget->event(widget, SND_SOC_DAPM_PRE_REG);
+			if (ret < 0)
+				goto out;
+		}
+		ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
+		if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+			ret = widget->event(widget, SND_SOC_DAPM_POST_REG);
+	} else
+		ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
+
+out:
+	mutex_unlock(&widget->codec->mutex);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
+
+/**
+ * snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a dapm enumerated double mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned short val, bitmask;
+
+	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+		;
+	val = snd_soc_read(widget->codec, e->reg);
+	ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
+	if (e->shift_l != e->shift_r)
+		ucontrol->value.enumerated.item[1] =
+			(val >> e->shift_r) & (bitmask - 1);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
+
+/**
+ * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a dapm enumerated double mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	unsigned short val, mux;
+	unsigned short mask, bitmask;
+	int ret = 0;
+
+	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
+		;
+	if (ucontrol->value.enumerated.item[0] > e->mask - 1)
+		return -EINVAL;
+	mux = ucontrol->value.enumerated.item[0];
+	val = mux << e->shift_l;
+	mask = (bitmask - 1) << e->shift_l;
+	if (e->shift_l != e->shift_r) {
+		if (ucontrol->value.enumerated.item[1] > e->mask - 1)
+			return -EINVAL;
+		val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+		mask |= (bitmask - 1) << e->shift_r;
+	}
+
+	mutex_lock(&widget->codec->mutex);
+	widget->value = val;
+	dapm_mux_update_power(widget, kcontrol, mask, mux, e);
+	if (widget->event) {
+		if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+			ret = widget->event(widget, SND_SOC_DAPM_PRE_REG);
+			if (ret < 0)
+				goto out;
+		}
+		ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+		if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+			ret = widget->event(widget, SND_SOC_DAPM_POST_REG);
+	} else
+		ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+
+out:
+	mutex_unlock(&widget->codec->mutex);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
+
+/**
+ * snd_soc_dapm_new_control - create new dapm control
+ * @codec: audio codec
+ * @widget: widget template
+ *
+ * Creates a new dapm control based upon the template.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
+	const struct snd_soc_dapm_widget *widget)
+{
+	struct snd_soc_dapm_widget *w;
+
+	if ((w = dapm_cnew_widget(widget)) == NULL)
+		return -ENOMEM;
+
+	w->codec = codec;
+	INIT_LIST_HEAD(&w->sources);
+	INIT_LIST_HEAD(&w->sinks);
+	INIT_LIST_HEAD(&w->list);
+	list_add(&w->list, &codec->dapm_widgets);
+
+	/* machine layer set ups unconnected pins and insertions */
+	w->connected = 1;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
+
+/**
+ * snd_soc_dapm_stream_event - send a stream event to the dapm core
+ * @codec: audio codec
+ * @stream: stream name
+ * @event: stream event
+ *
+ * Sends a stream event to the dapm core. The core then makes any
+ * necessary widget power changes.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
+	char *stream, int event)
+{
+	struct snd_soc_dapm_widget *w;
+
+	if (stream == NULL)
+		return 0;
+
+	mutex_lock(&codec->mutex);
+	list_for_each_entry(w, &codec->dapm_widgets, list)
+	{
+		if (!w->sname)
+			continue;
+		dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname,
+			stream, event);
+		if (strstr(w->sname, stream)) {
+			switch(event) {
+			case SND_SOC_DAPM_STREAM_START:
+				w->active = 1;
+				break;
+			case SND_SOC_DAPM_STREAM_STOP:
+				w->active = 0;
+				break;
+			case SND_SOC_DAPM_STREAM_SUSPEND:
+				if (w->active)
+					w->suspend = 1;
+				w->active = 0;
+				break;
+			case SND_SOC_DAPM_STREAM_RESUME:
+				if (w->suspend) {
+					w->active = 1;
+					w->suspend = 0;
+				}
+				break;
+			case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
+				break;
+			case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
+				break;
+			}
+		}
+	}
+	mutex_unlock(&codec->mutex);
+
+	dapm_power_widgets(codec, event);
+	dump_dapm(codec, __FUNCTION__);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
+
+/**
+ * snd_soc_dapm_set_endpoint - set audio endpoint status
+ * @codec: audio codec
+ * @endpoint: audio signal endpoint (or start point)
+ * @status: point status
+ *
+ * Set audio endpoint status - connected or disconnected.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
+	char *endpoint, int status)
+{
+	struct snd_soc_dapm_widget *w;
+
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+		if (!strcmp(w->name, endpoint)) {
+			w->connected = status;
+		}
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
+
+/**
+ * snd_soc_dapm_free - free dapm resources
+ * @socdev: SoC device
+ *
+ * Free all dapm widgets and resources.
+ */
+void snd_soc_dapm_free(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+
+	snd_soc_dapm_sys_remove(socdev->dev);
+	dapm_free_widgets(codec);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
+MODULE_LICENSE("GPL");