Merge branch 'topic/asoc' into to-push

129 files changed, 12273 insertions, 4204 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 4dfda6674be..ef025c66cc6 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -22,17 +22,16 @@ if SND_SOC
 config SND_SOC_AC97_BUS
 	bool
 
-# All the supported Soc's
-source "sound/soc/at32/Kconfig"
-source "sound/soc/at91/Kconfig"
+# All the supported SoCs
+source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
+source "sound/soc/blackfin/Kconfig"
+source "sound/soc/davinci/Kconfig"
+source "sound/soc/fsl/Kconfig"
+source "sound/soc/omap/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
 source "sound/soc/sh/Kconfig"
-source "sound/soc/fsl/Kconfig"
-source "sound/soc/davinci/Kconfig"
-source "sound/soc/omap/Kconfig"
-source "sound/soc/blackfin/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index d849349f2c6..86a9b1f5b0f 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,13 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
-obj-$(CONFIG_SND_SOC)	+= codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
-obj-$(CONFIG_SND_SOC)	+= omap/ au1x/ blackfin/
+obj-$(CONFIG_SND_SOC)	+= codecs/
+obj-$(CONFIG_SND_SOC)	+= atmel/
+obj-$(CONFIG_SND_SOC)	+= au1x/
+obj-$(CONFIG_SND_SOC)	+= blackfin/
+obj-$(CONFIG_SND_SOC)	+= davinci/
+obj-$(CONFIG_SND_SOC)	+= fsl/
+obj-$(CONFIG_SND_SOC)	+= omap/
+obj-$(CONFIG_SND_SOC)	+= pxa/
+obj-$(CONFIG_SND_SOC)	+= s3c24xx/
+obj-$(CONFIG_SND_SOC)	+= sh/
diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig
deleted file mode 100644
index b0765e86c08..00000000000
--- a/sound/soc/at32/Kconfig
+++ /dev/null
@@ -1,34 +0,0 @@
-config SND_AT32_SOC
-        tristate "SoC Audio for the Atmel AT32 System-on-a-Chip"
-        depends on AVR32 && SND_SOC
-        help
-          Say Y or M if you want to add support for codecs attached to 
-          the AT32 SSC interface.  You will also need to
-          to select the audio interfaces to support below.
-
-
-config SND_AT32_SOC_SSC
-        tristate
-
-
-
-config SND_AT32_SOC_PLAYPAQ
-        tristate "SoC Audio support for PlayPaq with WM8510"
-        depends on SND_AT32_SOC && BOARD_PLAYPAQ
-        select SND_AT32_SOC_SSC
-        select SND_SOC_WM8510
-        help
-          Say Y or M here if you want to add support for SoC audio
-          on the LRS PlayPaq.
-
-
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
-        bool "Run CODEC on PlayPaq in slave mode"
-        depends on SND_AT32_SOC_PLAYPAQ
-        default n
-        help
-          Say Y if you want to run with the AT32 SSC generating the BCLK
-          and FRAME signals on the PlayPaq.  Unless you want to play
-          with the AT32 as the SSC master, you probably want to say N here,
-          as this will give you better sound quality.
diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile
deleted file mode 100644
index c03e55ecece..00000000000
--- a/sound/soc/at32/Makefile
+++ /dev/null
@@ -1,11 +0,0 @@
-# AT32 Platform Support
-snd-soc-at32-objs := at32-pcm.o
-snd-soc-at32-ssc-objs := at32-ssc.o
-
-obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o
-obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o
-
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c
deleted file mode 100644
index c83584f989a..00000000000
--- a/sound/soc/at32/at32-pcm.c
+++ /dev/null
@@ -1,492 +0,0 @@
-/* sound/soc/at32/at32-pcm.c
- * ASoC PCM interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * 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.
- *
- * Note that this is basically a port of the sound/soc/at91-pcm.c to
- * the AVR32 kernel.  Thanks to Frank Mandarino for that code.
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dma-mapping.h>
-#include <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include "at32-pcm.h"
-
-
-
-/*--------------------------------------------------------------------------*\
- * Hardware definition
-\*--------------------------------------------------------------------------*/
-/* TODO: These values were taken from the AT91 platform driver, check
- *	 them against real values for AT32
- */
-static const struct snd_pcm_hardware at32_pcm_hardware = {
-	.info = (SNDRV_PCM_INFO_MMAP |
-		 SNDRV_PCM_INFO_MMAP_VALID |
-		 SNDRV_PCM_INFO_INTERLEAVED |
-		 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		 SNDRV_PCM_INFO_PAUSE),
-
-	.formats = SNDRV_PCM_FMTBIT_S16,
-	.period_bytes_min = 32,
-	.period_bytes_max = 8192,	/* 512 frames * 16 bytes / frame */
-	.periods_min = 2,
-	.periods_max = 1024,
-	.buffer_bytes_max = 32 * 1024,
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * Data types
-\*--------------------------------------------------------------------------*/
-struct at32_runtime_data {
-	struct at32_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 */
-	int periods;		/* period index of period_ptr */
-
-	/* Save PDC registers (for power management) */
-	u32 pdc_xpr_save;
-	u32 pdc_xcr_save;
-	u32 pdc_xnpr_save;
-	u32 pdc_xncr_save;
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * Helper functions
-\*--------------------------------------------------------------------------*/
-static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
-{
-	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
-	struct snd_dma_buffer *dmabuf = &substream->dma_buffer;
-	size_t size = at32_pcm_hardware.buffer_bytes_max;
-
-	dmabuf->dev.type = SNDRV_DMA_TYPE_DEV;
-	dmabuf->dev.dev = pcm->card->dev;
-	dmabuf->private_data = NULL;
-	dmabuf->area = dma_alloc_coherent(pcm->card->dev, size,
-					  &dmabuf->addr, GFP_KERNEL);
-	pr_debug("at32_pcm: preallocate_dma_buffer: "
-		 "area=%p, addr=%p, size=%ld\n",
-		 (void *)dmabuf->area, (void *)dmabuf->addr, size);
-
-	if (!dmabuf->area)
-		return -ENOMEM;
-
-	dmabuf->bytes = size;
-	return 0;
-}
-
-
-
-/*--------------------------------------------------------------------------*\
- * ISR
-\*--------------------------------------------------------------------------*/
-static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *rtd = substream->runtime;
-	struct at32_runtime_data *prtd = rtd->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-	static int count;
-
-	count++;
-	if (ssc_sr & params->mask->ssc_endbuf) {
-		pr_warning("at32-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 */
-		ssc_writex(params->ssc->regs, ATMEL_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;
-
-
-		ssc_writex(params->ssc->regs, params->pdc->xpr,
-			   prtd->period_ptr);
-		ssc_writex(params->ssc->regs, params->pdc->xcr,
-			   prtd->period_size / params->pdc_xfer_size);
-		ssc_writex(params->ssc->regs, ATMEL_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;
-		ssc_writex(params->ssc->regs, params->pdc->xnpr,
-			   prtd->period_ptr);
-		ssc_writex(params->ssc->regs, params->pdc->xncr,
-			   prtd->period_size / params->pdc_xfer_size);
-	}
-
-
-	snd_pcm_period_elapsed(substream);
-}
-
-
-
-/*--------------------------------------------------------------------------*\
- * PCM operations
-\*--------------------------------------------------------------------------*/
-static int at32_pcm_hw_params(struct snd_pcm_substream *substream,
-			      struct snd_pcm_hw_params *params)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at32_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 = at32_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);
-
-	pr_debug("hw_params: DMA for %s initialized "
-		 "(dma_bytes=%ld, period_size=%ld)\n",
-		 prtd->params->name, runtime->dma_bytes, prtd->period_size);
-
-	return 0;
-}
-
-
-
-static int at32_pcm_hw_free(struct snd_pcm_substream *substream)
-{
-	struct at32_runtime_data *prtd = substream->runtime->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-
-	if (params != NULL) {
-		ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
-			   params->mask->pdc_disable);
-		prtd->params->dma_intr_handler = NULL;
-	}
-
-	return 0;
-}
-
-
-
-static int at32_pcm_prepare(struct snd_pcm_substream *substream)
-{
-	struct at32_runtime_data *prtd = substream->runtime->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-
-	ssc_writex(params->ssc->regs, SSC_IDR,
-		   params->mask->ssc_endx | params->mask->ssc_endbuf);
-	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-		   params->mask->pdc_disable);
-
-	return 0;
-}
-
-
-static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct snd_pcm_runtime *rtd = substream->runtime;
-	struct at32_runtime_data *prtd = rtd->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-	int ret = 0;
-
-	pr_debug("at32_pcm_trigger: buffer_size = %ld, "
-		 "dma_area = %p, dma_bytes = %ld\n",
-		 rtd->buffer_size, rtd->dma_area, rtd->dma_bytes);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		prtd->period_ptr = prtd->dma_buffer;
-
-		ssc_writex(params->ssc->regs, params->pdc->xpr,
-			   prtd->period_ptr);
-		ssc_writex(params->ssc->regs, params->pdc->xcr,
-			   prtd->period_size / params->pdc_xfer_size);
-
-		prtd->period_ptr += prtd->period_size;
-		ssc_writex(params->ssc->regs, params->pdc->xnpr,
-			   prtd->period_ptr);
-		ssc_writex(params->ssc->regs, params->pdc->xncr,
-			   prtd->period_size / params->pdc_xfer_size);
-
-		pr_debug("trigger: period_ptr=%lx, xpr=%x, "
-			 "xcr=%d, xnpr=%x, xncr=%d\n",
-			 (unsigned long)prtd->period_ptr,
-			 ssc_readx(params->ssc->regs, params->pdc->xpr),
-			 ssc_readx(params->ssc->regs, params->pdc->xcr),
-			 ssc_readx(params->ssc->regs, params->pdc->xnpr),
-			 ssc_readx(params->ssc->regs, params->pdc->xncr));
-
-		ssc_writex(params->ssc->regs, SSC_IER,
-			   params->mask->ssc_endx | params->mask->ssc_endbuf);
-		ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
-			   params->mask->pdc_enable);
-
-		pr_debug("sr=%x, imr=%x\n",
-			 ssc_readx(params->ssc->regs, SSC_SR),
-			 ssc_readx(params->ssc->regs, SSC_IER));
-		break;		/* SNDRV_PCM_TRIGGER_START */
-
-
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-			   params->mask->pdc_disable);
-		break;
-
-
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-			   params->mask->pdc_enable);
-		break;
-
-	default:
-		ret = -EINVAL;
-	}
-
-	return ret;
-}
-
-
-
-static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at32_runtime_data *prtd = runtime->private_data;
-	struct at32_pcm_dma_params *params = prtd->params;
-	dma_addr_t ptr;
-	snd_pcm_uframes_t x;
-
-	ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr);
-	x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
-
-	if (x == runtime->buffer_size)
-		x = 0;
-
-	return x;
-}
-
-
-
-static int at32_pcm_open(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct at32_runtime_data *prtd;
-	int ret = 0;
-
-	snd_soc_set_runtime_hwparams(substream, &at32_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(*prtd), GFP_KERNEL);
-	if (prtd == NULL) {
-		ret = -ENOMEM;
-		goto out;
-	}
-	runtime->private_data = prtd;
-
-
-out:
-	return ret;
-}
-
-
-
-static int at32_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct at32_runtime_data *prtd = substream->runtime->private_data;
-
-	kfree(prtd);
-	return 0;
-}
-
-
-static int at32_pcm_mmap(struct snd_pcm_substream *substream,
-			 struct vm_area_struct *vma)
-{
-	return remap_pfn_range(vma, vma->vm_start,
-			       substream->dma_buffer.addr >> PAGE_SHIFT,
-			       vma->vm_end - vma->vm_start, vma->vm_page_prot);
-}
-
-
-
-static struct snd_pcm_ops at32_pcm_ops = {
-	.open = at32_pcm_open,
-	.close = at32_pcm_close,
-	.ioctl = snd_pcm_lib_ioctl,
-	.hw_params = at32_pcm_hw_params,
-	.hw_free = at32_pcm_hw_free,
-	.prepare = at32_pcm_prepare,
-	.trigger = at32_pcm_trigger,
-	.pointer = at32_pcm_pointer,
-	.mmap = at32_pcm_mmap,
-};
-
-
-
-/*--------------------------------------------------------------------------*\
- * ASoC platform driver
-\*--------------------------------------------------------------------------*/
-static u64 at32_pcm_dmamask = 0xffffffff;
-
-static int at32_pcm_new(struct snd_card *card,
-			struct snd_soc_dai *dai,
-			struct snd_pcm *pcm)
-{
-	int ret = 0;
-
-	if (!card->dev->dma_mask)
-		card->dev->dma_mask = &at32_pcm_dmamask;
-	if (!card->dev->coherent_dma_mask)
-		card->dev->coherent_dma_mask = 0xffffffff;
-
-	if (dai->playback.channels_min) {
-		ret = at32_pcm_preallocate_dma_buffer(
-			  pcm, SNDRV_PCM_STREAM_PLAYBACK);
-		if (ret)
-			goto out;
-	}
-
-	if (dai->capture.channels_min) {
-		pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n");
-		ret = at32_pcm_preallocate_dma_buffer(
-			  pcm, SNDRV_PCM_STREAM_CAPTURE);
-		if (ret)
-			goto out;
-	}
-
-
-out:
-	return ret;
-}
-
-
-
-static void at32_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 == NULL)
-			continue;
-
-		buf = &substream->dma_buffer;
-		if (!buf->area)
-			continue;
-		dma_free_coherent(pcm->card->dev, buf->bytes,
-				  buf->area, buf->addr);
-		buf->area = NULL;
-	}
-}
-
-
-
-#ifdef CONFIG_PM
-static int at32_pcm_suspend(struct platform_device *pdev,
-			    struct snd_soc_dai *dai)
-{
-	struct snd_pcm_runtime *runtime = dai->runtime;
-	struct at32_runtime_data *prtd;
-	struct at32_pcm_dma_params *params;
-
-	if (runtime == NULL)
-		return 0;
-	prtd = runtime->private_data;
-	params = prtd->params;
-
-	/* Disable the PDC and save the PDC registers */
-	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
-		   params->mask->pdc_disable);
-
-	prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
-	prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
-	prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr);
-	prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr);
-
-	return 0;
-}
-
-
-
-static int at32_pcm_resume(struct platform_device *pdev,
-			   struct snd_soc_dai *dai)
-{
-	struct snd_pcm_runtime *runtime = dai->runtime;
-	struct at32_runtime_data *prtd;
-	struct at32_pcm_dma_params *params;
-
-	if (runtime == NULL)
-		return 0;
-	prtd = runtime->private_data;
-	params = prtd->params;
-
-	/* Restore the PDC registers and enable the PDC */
-	ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save);
-	ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save);
-	ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
-	ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
-
-	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable);
-	return 0;
-}
-#else /* CONFIG_PM */
-#  define at32_pcm_suspend	NULL
-#  define at32_pcm_resume	NULL
-#endif /* CONFIG_PM */
-
-
-
-struct snd_soc_platform at32_soc_platform = {
-	.name = "at32-audio",
-	.pcm_ops = &at32_pcm_ops,
-	.pcm_new = at32_pcm_new,
-	.pcm_free = at32_pcm_free_dma_buffers,
-	.suspend = at32_pcm_suspend,
-	.resume = at32_pcm_resume,
-};
-EXPORT_SYMBOL_GPL(at32_soc_platform);
-
-
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("Atmel AT32 PCM module");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h
deleted file mode 100644
index 2a52430417d..00000000000
--- a/sound/soc/at32/at32-pcm.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/* sound/soc/at32/at32-pcm.h
- * ASoC PCM interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * 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 __SOUND_SOC_AT32_AT32_PCM_H
-#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__
-
-#include <linux/atmel-ssc.h>
-
-
-/*
- * Registers and status bits that are required by the PCM driver
- * TODO: Is ptcr really used?
- */
-struct at32_pdc_regs {
-	u32 xpr;		/* PDC RX/TX pointer */
-	u32 xcr;		/* PDC RX/TX counter */
-	u32 xnpr;		/* PDC next RX/TX pointer */
-	u32 xncr;		/* PDC next RX/TX counter */
-	u32 ptcr;		/* PDC transfer control */
-};
-
-
-
-/*
- * SSC mask info
- */
-struct at32_ssc_mask {
-	u32 ssc_enable;		/* SSC RX/TX enable */
-	u32 ssc_disable;	/* SSC RX/TX disable */
-	u32 ssc_endx;		/* SSC ENDTX or ENDRX */
-	u32 ssc_endbuf;		/* SSC TXBUFF or RXBUFF */
-	u32 pdc_enable;		/* PDC RX/TX enable */
-	u32 pdc_disable;	/* PDC RX/TX 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 at32_pcm_dma_params {
-	char *name;		/* stream identifier */
-	int pdc_xfer_size;	/* PDC counter increment in bytes */
-	struct ssc_device *ssc;	/* SSC device for stream */
-	struct at32_pdc_regs *pdc;	/* PDC register info */
-	struct at32_ssc_mask *mask;	/* SSC mask info */
-	struct snd_pcm_substream *substream;
-	void (*dma_intr_handler) (u32, struct snd_pcm_substream *);
-};
-
-
-
-/*
- * The AT32 ASoC platform driver
- */
-extern struct snd_soc_platform at32_soc_platform;
-
-
-
-/*
- * SSC register access (since ssc_writel() / ssc_readl() require literal name)
- */
-#define ssc_readx(base, reg)            (__raw_readl((base) + (reg)))
-#define ssc_writex(base, reg, value)    __raw_writel((value), (base) + (reg))
-
-#endif /* __SOUND_SOC_AT32_AT32_PCM_H */
diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c
deleted file mode 100644
index 4ef6492c902..00000000000
--- a/sound/soc/at32/at32-ssc.c
+++ /dev/null
@@ -1,849 +0,0 @@
-/* sound/soc/at32/at32-ssc.c
- * ASoC platform driver for AT32 using SSC as DAI
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * 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.
- *
- * Note that this is basically a port of the sound/soc/at91-ssc.c to
- * the AVR32 kernel.  Thanks to Frank Mandarino for that code.
- */
-
-/* #define DEBUG */
-
-#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 <linux/io.h>
-#include <linux/atmel_pdc.h>
-#include <linux/atmel-ssc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include "at32-pcm.h"
-#include "at32-ssc.h"
-
-
-
-/*-------------------------------------------------------------------------*\
- * Constants
-\*-------------------------------------------------------------------------*/
-#define NUM_SSC_DEVICES		3
-
-/*
- * SSC direction masks
- */
-#define SSC_DIR_MASK_UNUSED	0
-#define SSC_DIR_MASK_PLAYBACK	1
-#define SSC_DIR_MASK_CAPTURE	2
-
-/*
- * SSC register values that Atmel left out of <linux/atmel-ssc.h>.  These
- * are expected to be used with SSC_BF
- */
-/* START bit field values */
-#define SSC_START_CONTINUOUS	0
-#define SSC_START_TX_RX		1
-#define SSC_START_LOW_RF	2
-#define SSC_START_HIGH_RF	3
-#define SSC_START_FALLING_RF	4
-#define SSC_START_RISING_RF	5
-#define SSC_START_LEVEL_RF	6
-#define SSC_START_EDGE_RF	7
-#define SSS_START_COMPARE_0	8
-
-/* CKI bit field values */
-#define SSC_CKI_FALLING		0
-#define SSC_CKI_RISING		1
-
-/* CKO bit field values */
-#define SSC_CKO_NONE		0
-#define SSC_CKO_CONTINUOUS	1
-#define SSC_CKO_TRANSFER	2
-
-/* CKS bit field values */
-#define SSC_CKS_DIV		0
-#define SSC_CKS_CLOCK		1
-#define SSC_CKS_PIN		2
-
-/* FSEDGE bit field values */
-#define SSC_FSEDGE_POSITIVE	0
-#define SSC_FSEDGE_NEGATIVE	1
-
-/* FSOS bit field values */
-#define SSC_FSOS_NONE		0
-#define SSC_FSOS_NEGATIVE	1
-#define SSC_FSOS_POSITIVE	2
-#define SSC_FSOS_LOW		3
-#define SSC_FSOS_HIGH		4
-#define SSC_FSOS_TOGGLE		5
-
-#define START_DELAY		1
-
-
-
-/*-------------------------------------------------------------------------*\
- * Module data
-\*-------------------------------------------------------------------------*/
-/*
- * SSC PDC registered required by the PCM DMA engine
- */
-static struct at32_pdc_regs pdc_tx_reg = {
-	.xpr = SSC_PDC_TPR,
-	.xcr = SSC_PDC_TCR,
-	.xnpr = SSC_PDC_TNPR,
-	.xncr = SSC_PDC_TNCR,
-};
-
-
-
-static struct at32_pdc_regs pdc_rx_reg = {
-	.xpr = SSC_PDC_RPR,
-	.xcr = SSC_PDC_RCR,
-	.xnpr = SSC_PDC_RNPR,
-	.xncr = SSC_PDC_RNCR,
-};
-
-
-
-/*
- * SSC and PDC status bits for transmit and receive
- */
-static struct at32_ssc_mask ssc_tx_mask = {
-	.ssc_enable = SSC_BIT(CR_TXEN),
-	.ssc_disable = SSC_BIT(CR_TXDIS),
-	.ssc_endx = SSC_BIT(SR_ENDTX),
-	.ssc_endbuf = SSC_BIT(SR_TXBUFE),
-	.pdc_enable = SSC_BIT(PDC_PTCR_TXTEN),
-	.pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS),
-};
-
-
-
-static struct at32_ssc_mask ssc_rx_mask = {
-	.ssc_enable = SSC_BIT(CR_RXEN),
-	.ssc_disable = SSC_BIT(CR_RXDIS),
-	.ssc_endx = SSC_BIT(SR_ENDRX),
-	.ssc_endbuf = SSC_BIT(SR_RXBUFF),
-	.pdc_enable = SSC_BIT(PDC_PTCR_RXTEN),
-	.pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS),
-};
-
-
-
-/*
- * DMA parameters for each SSC
- */
-static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
-	{
-	 {
-	  .name = "SSC0 PCM out",
-	  .pdc = &pdc_tx_reg,
-	  .mask = &ssc_tx_mask,
-	  },
-	 {
-	  .name = "SSC0 PCM in",
-	  .pdc = &pdc_rx_reg,
-	  .mask = &ssc_rx_mask,
-	  },
-	 },
-	{
-	 {
-	  .name = "SSC1 PCM out",
-	  .pdc = &pdc_tx_reg,
-	  .mask = &ssc_tx_mask,
-	  },
-	 {
-	  .name = "SSC1 PCM in",
-	  .pdc = &pdc_rx_reg,
-	  .mask = &ssc_rx_mask,
-	  },
-	 },
-	{
-	 {
-	  .name = "SSC2 PCM out",
-	  .pdc = &pdc_tx_reg,
-	  .mask = &ssc_tx_mask,
-	  },
-	 {
-	  .name = "SSC2 PCM in",
-	  .pdc = &pdc_rx_reg,
-	  .mask = &ssc_rx_mask,
-	  },
-	 },
-};
-
-
-
-static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = {
-	{
-	 .name = "ssc0",
-	 .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
-	 .dir_mask = SSC_DIR_MASK_UNUSED,
-	 .initialized = 0,
-	 },
-	{
-	 .name = "ssc1",
-	 .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
-	 .dir_mask = SSC_DIR_MASK_UNUSED,
-	 .initialized = 0,
-	 },
-	{
-	 .name = "ssc2",
-	 .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
-	 .dir_mask = SSC_DIR_MASK_UNUSED,
-	 .initialized = 0,
-	 },
-};
-
-
-
-
-/*-------------------------------------------------------------------------*\
- * ISR
-\*-------------------------------------------------------------------------*/
-/*
- * SSC interrupt handler.  Passes PDC interrupts to the DMA interrupt
- * handler in the PCM driver.
- */
-static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id)
-{
-	struct at32_ssc_info *ssc_p = dev_id;
-	struct at32_pcm_dma_params *dma_params;
-	u32 ssc_sr;
-	u32 ssc_substream_mask;
-	int i;
-
-	ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) &
-		  ssc_readl(ssc_p->ssc->regs, IMR));
-
-	/*
-	 * Loop through substreams attached to this SSC.  If a DMA-related
-	 * interrupt occured on that substream, call the DMA interrupt
-	 * handler function, if one has been registered in the dma_param
-	 * 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_substream_mask = (dma_params->mask->ssc_endx |
-					      dma_params->mask->ssc_endbuf);
-			if (ssc_sr & ssc_substream_mask) {
-				dma_params->dma_intr_handler(ssc_sr,
-							     dma_params->
-							     substream);
-			}
-		}
-	}
-
-
-	return IRQ_HANDLED;
-}
-
-/*-------------------------------------------------------------------------*\
- * DAI functions
-\*-------------------------------------------------------------------------*/
-/*
- * Startup.  Only that one substream allowed in each direction.
- */
-static int at32_ssc_startup(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	int dir_mask;
-
-	dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-		    SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE);
-
-	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 at32_ssc_shutdown(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	struct at32_pcm_dma_params *dma_params;
-	int dir_mask;
-
-	dma_params = ssc_p->dma_params[substream->stream];
-
-	if (dma_params != NULL) {
-		ssc_writel(dma_params->ssc->regs, CR,
-			   dma_params->mask->ssc_disable);
-		pr_debug("%s disabled SSC_SR=0x%08x\n",
-			 (substream->stream ? "receiver" : "transmit"),
-			 ssc_readl(ssc_p->ssc->regs, SR));
-
-		dma_params->ssc = NULL;
-		dma_params->substream = NULL;
-		ssc_p->dma_params[substream->stream] = NULL;
-	}
-
-
-	dir_mask = 1 << substream->stream;
-	spin_lock_irq(&ssc_p->lock);
-	ssc_p->dir_mask &= ~dir_mask;
-	if (!ssc_p->dir_mask) {
-		/* Shutdown the SSC clock */
-		pr_debug("at32-ssc: Stopping user %d clock\n",
-			 ssc_p->ssc->user);
-		clk_disable(ssc_p->ssc->clk);
-
-		if (ssc_p->initialized) {
-			free_irq(ssc_p->ssc->irq, ssc_p);
-			ssc_p->initialized = 0;
-		}
-
-		/* Reset the SSC */
-		ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
-
-		/* clear the SSC dividers */
-		ssc_p->cmr_div = 0;
-		ssc_p->tcmr_period = 0;
-		ssc_p->rcmr_period = 0;
-	}
-	spin_unlock_irq(&ssc_p->lock);
-}
-
-
-
-/*
- * Set the SSC system clock rate
- */
-static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
-				   int clk_id, unsigned int freq, int dir)
-{
-	/* TODO: What the heck do I do here? */
-	return 0;
-}
-
-
-
-/*
- * Record DAI format for use by hw_params()
- */
-static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
-				unsigned int fmt)
-{
-	struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
-	ssc_p->daifmt = fmt;
-	return 0;
-}
-
-
-
-/*
- * Record SSC clock dividers for use in hw_params()
- */
-static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
-				   int div_id, int div)
-{
-	struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
-	switch (div_id) {
-	case AT32_SSC_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 AT32_SSC_TCMR_PERIOD:
-		ssc_p->tcmr_period = div;
-		break;
-
-	case AT32_SSC_RCMR_PERIOD:
-		ssc_p->rcmr_period = div;
-		break;
-
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
-
-
-/*
- * Configure the SSC
- */
-static int at32_ssc_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 at32_ssc_info *ssc_p = &ssc_info[id];
-	struct at32_pcm_dma_params *dma_params;
-	int 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
-	 */
-	dma_params = &ssc_dma_params[id][substream->stream];
-	dma_params->ssc = ssc_p->ssc;
-	dma_params->substream = substream;
-
-	ssc_p->dma_params[substream->stream] = dma_params;
-
-
-	/*
-	 * The cpu_dai->dma_data field is only used to communicate the
-	 * appropriate DMA parameters to the PCM driver's 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);
-
-
-	/*
-	 * Determine sample size in bits and the PDC increment
-	 */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S8:
-		bits = 8;
-		dma_params->pdc_xfer_size = 1;
-		break;
-
-	case SNDRV_PCM_FORMAT_S16:
-		bits = 16;
-		dma_params->pdc_xfer_size = 2;
-		break;
-
-	case SNDRV_PCM_FORMAT_S24:
-		bits = 24;
-		dma_params->pdc_xfer_size = 4;
-		break;
-
-	case SNDRV_PCM_FORMAT_S32:
-		bits = 32;
-		dma_params->pdc_xfer_size = 4;
-		break;
-
-	default:
-		pr_warning("at32-ssc: Unsupported PCM format %d",
-			   params_format(params));
-		return -EINVAL;
-	}
-	pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n",
-		 bits, dma_params->pdc_xfer_size, channels);
-
-
-	/*
-	 * The SSC only supports up to 16-bit samples in I2S format, due
-	 * to the size of the Frame Mode Register FSLEN field.
-	 */
-	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S)
-		if (bits > 16) {
-			pr_warning("at32-ssc: "
-				   "sample size %d is too large for I2S\n",
-				   bits);
-			return -EINVAL;
-		}
-
-
-	/*
-	 * Compute the SSC register settings
-	 */
-	switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK |
-				 SND_SOC_DAIFMT_MASTER_MASK)) {
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
-		/*
-		 * I2S format, SSC provides BCLK and LRS clocks.
-		 *
-		 * The SSC transmit and receive clocks are generated from the
-		 * MCK divider, and the BCLK signal is output on the SSC TK line
-		 */
-		pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n");
-		rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) |
-			SSC_BF(RCMR_STTDLY, START_DELAY) |
-			SSC_BF(RCMR_START, SSC_START_FALLING_RF) |
-			SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
-			SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
-			SSC_BF(RCMR_CKS, SSC_CKS_DIV));
-
-		rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) |
-			SSC_BF(RFMR_FSLEN, bits - 1) |
-			SSC_BF(RFMR_DATNB, channels - 1) |
-			SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
-		tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) |
-			SSC_BF(TCMR_STTDLY, START_DELAY) |
-			SSC_BF(TCMR_START, SSC_START_FALLING_RF) |
-			SSC_BF(TCMR_CKI, SSC_CKI_FALLING) |
-			SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) |
-			SSC_BF(TCMR_CKS, SSC_CKS_DIV));
-
-		tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) |
-			SSC_BF(TFMR_FSLEN, bits - 1) |
-			SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) |
-			SSC_BF(TFMR_DATLEN, bits - 1));
-		break;
-
-
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
-		/*
-		 * I2S format, CODEC supplies BCLK and LRC clock.
-		 *
-		 * The SSC transmit clock is obtained from the BCLK signal
-		 * 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.
-		 */
-		pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n");
-		start_event = ((channels == 1) ?
-			       SSC_START_FALLING_RF : SSC_START_EDGE_RF);
-
-		rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) |
-			SSC_BF(RCMR_START, start_event) |
-			SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
-			SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
-			SSC_BF(RCMR_CKS, SSC_CKS_CLOCK));
-
-		rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) |
-			SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
-		tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) |
-			SSC_BF(TCMR_START, start_event) |
-			SSC_BF(TCMR_CKI, SSC_CKI_FALLING) |
-			SSC_BF(TCMR_CKO, SSC_CKO_NONE) |
-			SSC_BF(TCMR_CKS, SSC_CKS_PIN));
-
-		tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) |
-			SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1));
-		break;
-
-
-	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
-		/*
-		 * DSP/PCM Mode A format, 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
-		 */
-		pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n");
-		rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) |
-			SSC_BF(RCMR_STTDLY, 1) |
-			SSC_BF(RCMR_START, SSC_START_RISING_RF) |
-			SSC_BF(RCMR_CKI, SSC_CKI_RISING) |
-			SSC_BF(RCMR_CKO, SSC_CKO_NONE) |
-			SSC_BF(RCMR_CKS, SSC_CKS_DIV));
-
-		rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) |
-			SSC_BF(RFMR_DATNB, channels - 1) |
-			SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1));
-
-		tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) |
-			SSC_BF(TCMR_STTDLY, 1) |
-			SSC_BF(TCMR_START, SSC_START_RISING_RF) |
-			SSC_BF(TCMR_CKI, SSC_CKI_RISING) |
-			SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) |
-			SSC_BF(TCMR_CKS, SSC_CKS_DIV));
-
-		tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) |
-			SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) |
-			SSC_BF(TFMR_DATNB, channels - 1) |
-			SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1));
-		break;
-
-
-	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
-	default:
-		pr_warning("at32-ssc: unsupported DAI format 0x%x\n",
-			   ssc_p->daifmt);
-		return -EINVAL;
-		break;
-	}
-	pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
-		 rcmr, rfmr, tcmr, tfmr);
-
-
-	if (!ssc_p->initialized) {
-		/* enable peripheral clock */
-		pr_debug("at32-ssc: Starting clock\n");
-		clk_enable(ssc_p->ssc->clk);
-
-		/* Reset the SSC and its PDC registers */
-		ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
-
-		ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
-
-		ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
-		ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
-
-		ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0,
-				  ssc_p->name, ssc_p);
-		if (ret < 0) {
-			pr_warning("at32-ssc: request irq failed (%d)\n", ret);
-			pr_debug("at32-ssc: Stopping clock\n");
-			clk_disable(ssc_p->ssc->clk);
-			return ret;
-		}
-
-		ssc_p->initialized = 1;
-	}
-
-	/* Set SSC clock mode register */
-	ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div);
-
-	/* set receive clock mode and format */
-	ssc_writel(ssc_p->ssc->regs, RCMR, rcmr);
-	ssc_writel(ssc_p->ssc->regs, RFMR, rfmr);
-
-	/* set transmit clock mode and format */
-	ssc_writel(ssc_p->ssc->regs, TCMR, tcmr);
-	ssc_writel(ssc_p->ssc->regs, TFMR, tfmr);
-
-	pr_debug("at32-ssc: SSC initialized\n");
-	return 0;
-}
-
-
-
-static int at32_ssc_prepare(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
-	struct at32_pcm_dma_params *dma_params;
-
-	dma_params = ssc_p->dma_params[substream->stream];
-
-	ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable);
-
-	return 0;
-}
-
-
-
-#ifdef CONFIG_PM
-static int at32_ssc_suspend(struct platform_device *pdev,
-			    struct snd_soc_dai *cpu_dai)
-{
-	struct at32_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 = ssc_readl(ssc_p->ssc->regs, SR);
-	ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS));
-
-	/* Save the current interrupt mask, then disable unmasked interrupts */
-	ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR);
-	ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr);
-
-	ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR);
-	ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR);
-	ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR);
-	ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR);
-	ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR);
-
-	return 0;
-}
-
-
-
-static int at32_ssc_resume(struct platform_device *pdev,
-			   struct snd_soc_dai *cpu_dai)
-{
-	struct at32_ssc_info *ssc_p;
-	u32 cr;
-
-	if (!cpu_dai->active)
-		return 0;
-
-	ssc_p = &ssc_info[cpu_dai->id];
-
-	/* restore SSC register settings */
-	ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr);
-	ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr);
-	ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr);
-	ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr);
-	ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr);
-
-	/* re-enable interrupts */
-	ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
-
-	/* Re-enable recieve and transmit as appropriate */
-	cr = 0;
-	cr |=
-	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
-	cr |=
-	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0;
-	ssc_writel(ssc_p->ssc->regs, CR, cr);
-
-	return 0;
-}
-#else /* CONFIG_PM */
-#  define at32_ssc_suspend	NULL
-#  define at32_ssc_resume	NULL
-#endif /* CONFIG_PM */
-
-
-#define AT32_SSC_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 AT32_SSC_FORMATS \
-    (SNDRV_PCM_FMTBIT_S8  | SNDRV_PCM_FMTBIT_S16 | \
-     SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32)
-
-
-struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = {
-	{
-	 .name = "at32-ssc0",
-	 .id = 0,
-	 .type = SND_SOC_DAI_PCM,
-	 .suspend = at32_ssc_suspend,
-	 .resume = at32_ssc_resume,
-	 .playback = {
-		      .channels_min = 1,
-		      .channels_max = 2,
-		      .rates = AT32_SSC_RATES,
-		      .formats = AT32_SSC_FORMATS,
-		      },
-	 .capture = {
-		     .channels_min = 1,
-		     .channels_max = 2,
-		     .rates = AT32_SSC_RATES,
-		     .formats = AT32_SSC_FORMATS,
-		     },
-	 .ops = {
-		 .startup = at32_ssc_startup,
-		 .shutdown = at32_ssc_shutdown,
-		 .prepare = at32_ssc_prepare,
-		 .hw_params = at32_ssc_hw_params,
-		 },
-	 .dai_ops = {
-		     .set_sysclk = at32_ssc_set_dai_sysclk,
-		     .set_fmt = at32_ssc_set_dai_fmt,
-		     .set_clkdiv = at32_ssc_set_dai_clkdiv,
-		     },
-	 .private_data = &ssc_info[0],
-	 },
-	{
-	 .name = "at32-ssc1",
-	 .id = 1,
-	 .type = SND_SOC_DAI_PCM,
-	 .suspend = at32_ssc_suspend,
-	 .resume = at32_ssc_resume,
-	 .playback = {
-		      .channels_min = 1,
-		      .channels_max = 2,
-		      .rates = AT32_SSC_RATES,
-		      .formats = AT32_SSC_FORMATS,
-		      },
-	 .capture = {
-		     .channels_min = 1,
-		     .channels_max = 2,
-		     .rates = AT32_SSC_RATES,
-		     .formats = AT32_SSC_FORMATS,
-		     },
-	 .ops = {
-		 .startup = at32_ssc_startup,
-		 .shutdown = at32_ssc_shutdown,
-		 .prepare = at32_ssc_prepare,
-		 .hw_params = at32_ssc_hw_params,
-		 },
-	 .dai_ops = {
-		     .set_sysclk = at32_ssc_set_dai_sysclk,
-		     .set_fmt = at32_ssc_set_dai_fmt,
-		     .set_clkdiv = at32_ssc_set_dai_clkdiv,
-		     },
-	 .private_data = &ssc_info[1],
-	 },
-	{
-	 .name = "at32-ssc2",
-	 .id = 2,
-	 .type = SND_SOC_DAI_PCM,
-	 .suspend = at32_ssc_suspend,
-	 .resume = at32_ssc_resume,
-	 .playback = {
-		      .channels_min = 1,
-		      .channels_max = 2,
-		      .rates = AT32_SSC_RATES,
-		      .formats = AT32_SSC_FORMATS,
-		      },
-	 .capture = {
-		     .channels_min = 1,
-		     .channels_max = 2,
-		     .rates = AT32_SSC_RATES,
-		     .formats = AT32_SSC_FORMATS,
-		     },
-	 .ops = {
-		 .startup = at32_ssc_startup,
-		 .shutdown = at32_ssc_shutdown,
-		 .prepare = at32_ssc_prepare,
-		 .hw_params = at32_ssc_hw_params,
-		 },
-	 .dai_ops = {
-		     .set_sysclk = at32_ssc_set_dai_sysclk,
-		     .set_fmt = at32_ssc_set_dai_fmt,
-		     .set_clkdiv = at32_ssc_set_dai_clkdiv,
-		     },
-	 .private_data = &ssc_info[2],
-	 },
-};
-EXPORT_SYMBOL_GPL(at32_ssc_dai);
-
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("AT32 SSC ASoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h
deleted file mode 100644
index 3c052dbbe46..00000000000
--- a/sound/soc/at32/at32-ssc.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/* sound/soc/at32/at32-ssc.h
- * ASoC SSC interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * 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 __SOUND_SOC_AT32_AT32_SSC_H
-#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__
-
-#include <linux/types.h>
-#include <linux/atmel-ssc.h>
-
-#include "at32-pcm.h"
-
-
-
-struct at32_ssc_state {
-	u32 ssc_cmr;
-	u32 ssc_rcmr;
-	u32 ssc_rfmr;
-	u32 ssc_tcmr;
-	u32 ssc_tfmr;
-	u32 ssc_sr;
-	u32 ssc_imr;
-};
-
-
-
-struct at32_ssc_info {
-	char *name;
-	struct ssc_device *ssc;
-	spinlock_t lock;	/* lock for dir_mask */
-	unsigned short dir_mask;	/* 0=unused, 1=playback, 2=capture */
-	unsigned short initialized;	/* true if SSC has been initialized */
-	unsigned short daifmt;
-	unsigned short cmr_div;
-	unsigned short tcmr_period;
-	unsigned short rcmr_period;
-	struct at32_pcm_dma_params *dma_params[2];
-	struct at32_ssc_state ssc_state;
-};
-
-
-/* SSC divider ids */
-#define AT32_SSC_CMR_DIV        0	/* MCK divider for BCLK */
-#define AT32_SSC_TCMR_PERIOD    1	/* BCLK divider for transmit FS */
-#define AT32_SSC_RCMR_PERIOD    2	/* BCLK divider for receive FS */
-
-
-extern struct snd_soc_dai at32_ssc_dai[];
-
-
-
-#endif /* __SOUND_SOC_AT32_AT32_SSC_H */
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
deleted file mode 100644
index 85a883299c2..00000000000
--- a/sound/soc/at91/Kconfig
+++ /dev/null
@@ -1,10 +0,0 @@
-config SND_AT91_SOC
-	tristate "SoC Audio for the Atmel AT91 System-on-Chip"
-	depends on ARCH_AT91
-	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_SSC
-	tristate
diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile
deleted file mode 100644
index b817f11df28..00000000000
--- a/sound/soc/at91/Makefile
+++ /dev/null
@@ -1,6 +0,0 @@
-# AT91 Platform Support
-snd-soc-at91-objs := at91-pcm.o
-snd-soc-at91-ssc-objs := at91-ssc.o
-
-obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
-obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
deleted file mode 100644
index 7ab48bd25e4..00000000000
--- a/sound/soc/at91/at91-pcm.c
+++ /dev/null
@@ -1,434 +0,0 @@
-/*
- * 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 <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/hardware.h>
-#include <mach/at91_ssc.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 + ATMEL_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 + ATMEL_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 + ATMEL_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 + ATMEL_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 + ATMEL_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_IMR));
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable);
-		break;
-
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		at91_ssc_write(params->ssc_base + ATMEL_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_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_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 + ATMEL_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_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 + ATMEL_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
deleted file mode 100644
index e5aada2cb10..00000000000
--- a/sound/soc/at91/at91-pcm.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * 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 <mach/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/at91-ssc.c b/sound/soc/at91/at91-ssc.c
deleted file mode 100644
index 1b61cc46126..00000000000
--- a/sound/soc/at91/at91-ssc.c
+++ /dev/null
@@ -1,791 +0,0 @@
-/*
- * at91-ssc.c  --  ALSA SoC AT91 SSC Audio Layer Platform driver
- *
- * Author: Frank Mandarino <fmandarino@endrelia.com>
- *         Endrelia Technologies Inc.
- *
- * Based on pxa2xx Platform drivers by
- * Liam Girdwood <lrg@slimlogic.co.uk>
- *
- *  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 <linux/atmel_pdc.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/initval.h>
-#include <sound/soc.h>
-
-#include <mach/hardware.h>
-#include <mach/at91_pmc.h>
-#include <mach/at91_ssc.h>
-
-#include "at91-pcm.h"
-#include "at91-ssc.h"
-
-#if 0
-#define	DBG(x...)	printk(KERN_DEBUG "at91-ssc:" x)
-#else
-#define	DBG(x...)
-#endif
-
-#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
-#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		= ATMEL_PDC_TPR,
-	.xcr		= ATMEL_PDC_TCR,
-	.xnpr		= ATMEL_PDC_TNPR,
-	.xncr		= ATMEL_PDC_TNCR,
-};
-
-static struct at91_pdc_regs pdc_rx_reg = {
-	.xpr		= ATMEL_PDC_RPR,
-	.xcr		= ATMEL_PDC_RCR,
-	.xnpr		= ATMEL_PDC_RNPR,
-	.xncr		= ATMEL_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	= ATMEL_PDC_TXTEN,
-	.pdc_disable	= ATMEL_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	= ATMEL_PDC_RXTEN,
-	.pdc_disable	= ATMEL_PDC_RXTDIS,
-};
-
-
-/*
- * DMA parameters.
- */
-static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
-	{{
-	.name		= "SSC0 PCM out",
-	.pdc		= &pdc_tx_reg,
-	.mask		= &ssc_tx_mask,
-	},
-	{
-	.name		= "SSC0 PCM in",
-	.pdc		= &pdc_rx_reg,
-	.mask		= &ssc_rx_mask,
-	}},
-#if NUM_SSC_DEVICES == 3
-	{{
-	.name		= "SSC1 PCM out",
-	.pdc		= &pdc_tx_reg,
-	.mask		= &ssc_tx_mask,
-	},
-	{
-	.name		= "SSC1 PCM in",
-	.pdc		= &pdc_rx_reg,
-	.mask		= &ssc_rx_mask,
-	}},
-	{{
-	.name		= "SSC2 PCM out",
-	.pdc		= &pdc_tx_reg,
-	.mask		= &ssc_tx_mask,
-	},
-	{
-	.name		= "SSC2 PCM 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(ssc_info[0].lock),
-	.dir_mask	= 0,
-	.initialized	= 0,
-	},
-#if NUM_SSC_DEVICES == 3
-	{
-	.name		= "ssc1",
-	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
-	.dir_mask	= 0,
-	.initialized	= 0,
-	},
-	{
-	.name		= "ssc2",
-	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
-	.dir_mask	= 0,
-	.initialized	= 0,
-	},
-#endif
-};
-
-static unsigned int at91_ssc_sysclk;
-
-/*
- * SSC interrupt handler.  Passes PDC interrupts to the DMA
- * interrupt handler in the PCM driver.
- */
-static irqreturn_t at91_ssc_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_ssc_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("ssc_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_ssc_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_ssc_set_dai_sysclk(struct snd_soc_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_ssc_sysclk = freq;
-		break;
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
-
-/*
- * Record the DAI format for use in hw_params().
- */
-static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
-		unsigned int fmt)
-{
-	struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
-
-	ssc_p->daifmt = fmt;
-	return 0;
-}
-
-/*
- * Record SSC clock dividers for use in hw_params().
- */
-static int at91_ssc_set_dai_clkdiv(struct snd_soc_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_ssc_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);
-
-	/*
-	 * Determine sample size in bits and the PDC increment.
-	 */
-	switch(params_format(params)) {
-	case SNDRV_PCM_FORMAT_S8:
-		bits = 8;
-		dma_params->pdc_xfer_size = 1;
-		break;
-	case SNDRV_PCM_FORMAT_S16_LE:
-		bits = 16;
-		dma_params->pdc_xfer_size = 2;
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-		bits = 24;
-		dma_params->pdc_xfer_size = 4;
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-		bits = 32;
-		dma_params->pdc_xfer_size = 4;
-		break;
-	default:
-		printk(KERN_WARNING "at91-ssc: unsupported PCM format\n");
-		return -EINVAL;
-	}
-
-	/*
-	 * The SSC only supports up to 16-bit samples in I2S format, due
-	 * to the size of the Frame Mode Register FSLEN field.
-	 */
-	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
-		&& bits > 16) {
-		printk(KERN_WARNING
-			"at91-ssc: sample size %d is too large for I2S\n", bits);
-		return -EINVAL;
-	}
-
-	/*
-	 * Compute SSC register settings.
-	 */
-	switch (ssc_p->daifmt
-		& (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
-
-	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
-		/*
-		 * I2S format, 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_I2S | SND_SOC_DAIFMT_CBM_CFM:
-		/*
-		 * I2S format, CODEC supplies BCLK and LRC clocks.
-		 *
-		 * The SSC transmit clock is obtained from the BCLK signal on
-		 * on the TK line, and the SSC receive clock is generated from the
-		 * transmit clock.
-		 *
-		 * 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_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
-		/*
-		 * DSP/PCM Mode A format, 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_RISING_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_POSITIVE	     ) & AT91_SSC_FSOS)
-			| (( 0				<< 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_RISING_RF        ) & AT91_SSC_START)
-			| (( AT91_SSC_CK_RISING		     ) & 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_POSITIVE	     ) & AT91_SSC_FSOS)
-			| (( 0				<< 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_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
-	default:
-		printk(KERN_WARNING "at91-ssc: 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 + ATMEL_PDC_RPR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0);
-		at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0);
-
-		if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt,
-					0, ssc_p->name, ssc_p)) < 0) {
-			printk(KERN_WARNING "at91-ssc: request_irq failure\n");
-
-			DBG("Stopping pid %d clock\n", ssc_p->ssc.pid);
-			at91_sys_write(AT91_PMC_PCDR, 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_ssc_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_ssc_suspend(struct platform_device *pdev,
-	struct snd_soc_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_ssc_resume(struct platform_device *pdev,
-	struct snd_soc_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_ssc_suspend	NULL
-#define at91_ssc_resume		NULL
-#endif
-
-#define AT91_SSC_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 AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
-			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
-
-struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = {
-	{	.name = "at91-ssc0",
-		.id = 0,
-		.type = SND_SOC_DAI_PCM,
-		.suspend = at91_ssc_suspend,
-		.resume = at91_ssc_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.ops = {
-			.startup = at91_ssc_startup,
-			.shutdown = at91_ssc_shutdown,
-			.prepare = at91_ssc_prepare,
-			.hw_params = at91_ssc_hw_params,},
-		.dai_ops = {
-			.set_sysclk = at91_ssc_set_dai_sysclk,
-			.set_fmt = at91_ssc_set_dai_fmt,
-			.set_clkdiv = at91_ssc_set_dai_clkdiv,},
-		.private_data = &ssc_info[0].ssc,
-	},
-#if NUM_SSC_DEVICES == 3
-	{	.name = "at91-ssc1",
-		.id = 1,
-		.type = SND_SOC_DAI_PCM,
-		.suspend = at91_ssc_suspend,
-		.resume = at91_ssc_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.ops = {
-			.startup = at91_ssc_startup,
-			.shutdown = at91_ssc_shutdown,
-			.prepare = at91_ssc_prepare,
-			.hw_params = at91_ssc_hw_params,},
-		.dai_ops = {
-			.set_sysclk = at91_ssc_set_dai_sysclk,
-			.set_fmt = at91_ssc_set_dai_fmt,
-			.set_clkdiv = at91_ssc_set_dai_clkdiv,},
-		.private_data = &ssc_info[1].ssc,
-	},
-	{	.name = "at91-ssc2",
-		.id = 2,
-		.type = SND_SOC_DAI_PCM,
-		.suspend = at91_ssc_suspend,
-		.resume = at91_ssc_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 2,
-			.rates = AT91_SSC_RATES,
-			.formats = AT91_SSC_FORMATS,},
-		.ops = {
-			.startup = at91_ssc_startup,
-			.shutdown = at91_ssc_shutdown,
-			.prepare = at91_ssc_prepare,
-			.hw_params = at91_ssc_hw_params,},
-		.dai_ops = {
-			.set_sysclk = at91_ssc_set_dai_sysclk,
-			.set_fmt = at91_ssc_set_dai_fmt,
-			.set_clkdiv = at91_ssc_set_dai_clkdiv,},
-		.private_data = &ssc_info[2].ssc,
-	},
-#endif
-};
-
-EXPORT_SYMBOL_GPL(at91_ssc_dai);
-
-/* Module information */
-MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com");
-MODULE_DESCRIPTION("AT91 SSC ASoC Interface");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h
deleted file mode 100644
index 6b7bf382d06..00000000000
--- a/sound/soc/at91/at91-ssc.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * at91-ssc.h - ALSA SSC 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_SSC_H
-#define _AT91_SSC_H
-
-/* SSC system clock ids */
-#define AT91_SYSCLK_MCK		0 /* SSC uses AT91 MCK as system clock */
-
-/* SSC 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_dai at91_ssc_dai[];
-
-#endif /* _AT91_SSC_H */
-
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
new file mode 100644
index 00000000000..a608d7009db
--- /dev/null
+++ b/sound/soc/atmel/Kconfig
@@ -0,0 +1,43 @@
+config SND_ATMEL_SOC
+	tristate "SoC Audio for the Atmel System-on-Chip"
+	depends on ARCH_AT91 || AVR32
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the ATMEL SSC interface. You will also need
+	  to select the audio interfaces to support below.
+
+config SND_ATMEL_SOC_SSC
+	tristate
+	depends on SND_ATMEL_SOC
+	help
+	  Say Y or M if you want to add support for codecs the
+	  ATMEL SSC interface. You will also needs to select the individual
+	  machine drivers to support below.
+
+config SND_AT91_SOC_SAM9G20_WM8731
+	tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
+	depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC
+	select SND_ATMEL_SOC_SSC
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on WM8731-based
+	  AT91sam9g20 evaluation board.
+
+config SND_AT32_SOC_PLAYPAQ
+        tristate "SoC Audio support for PlayPaq with WM8510"
+        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ
+        select SND_ATMEL_SOC_SSC
+        select SND_SOC_WM8510
+        help
+          Say Y or M here if you want to add support for SoC audio
+          on the LRS PlayPaq.
+
+config SND_AT32_SOC_PLAYPAQ_SLAVE
+        bool "Run CODEC on PlayPaq in slave mode"
+        depends on SND_AT32_SOC_PLAYPAQ
+        default n
+        help
+          Say Y if you want to run with the AT32 SSC generating the BCLK
+          and FRAME signals on the PlayPaq.  Unless you want to play
+          with the AT32 as the SSC master, you probably want to say N here,
+          as this will give you better sound quality.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
new file mode 100644
index 00000000000..f54a7cc68e6
--- /dev/null
+++ b/sound/soc/atmel/Makefile
@@ -0,0 +1,15 @@
+# AT91 Platform Support
+snd-soc-atmel-pcm-objs := atmel-pcm.o
+snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o
+
+obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o
+obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
+
+# AT91 Machine Support
+snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
+
+# AT32 Machine Support
+snd-soc-playpaq-objs := playpaq_wm8510.o
+
+obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
+obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
new file mode 100644
index 00000000000..1fac5efd285
--- /dev/null
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -0,0 +1,494 @@
+/*
+ * atmel-pcm.c  --  ALSA PCM interface for the Atmel atmel SoC.
+ *
+ *  Copyright (C) 2005 SAN People
+ *  Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on at91-pcm. by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ *
+ * 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 as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/atmel_pdc.h>
+#include <linux/atmel-ssc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+
+#include "atmel-pcm.h"
+
+
+/*--------------------------------------------------------------------------*\
+ * Hardware definition
+\*--------------------------------------------------------------------------*/
+/* TODO: These values were taken from the AT91 platform driver, check
+ *	 them against real values for AT32
+ */
+static const struct snd_pcm_hardware atmel_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,
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * Data types
+\*--------------------------------------------------------------------------*/
+struct atmel_runtime_data {
+	struct atmel_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 */
+	int periods;			/* period index of period_ptr */
+
+	/* PDC register save */
+	u32 pdc_xpr_save;
+	u32 pdc_xcr_save;
+	u32 pdc_xnpr_save;
+	u32 pdc_xncr_save;
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * Helper functions
+\*--------------------------------------------------------------------------*/
+static int atmel_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 = atmel_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_coherent(pcm->card->dev, size,
+					  &buf->addr, GFP_KERNEL);
+	pr_debug("atmel-pcm:"
+		"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;
+}
+/*--------------------------------------------------------------------------*\
+ * ISR
+\*--------------------------------------------------------------------------*/
+static void atmel_pcm_dma_irq(u32 ssc_sr,
+	struct snd_pcm_substream *substream)
+{
+	struct atmel_runtime_data *prtd = substream->runtime->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+	static int count;
+
+	count++;
+
+	if (ssc_sr & params->mask->ssc_endbuf) {
+		pr_warning("atmel-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 */
+		ssc_writex(params->ssc->regs, ATMEL_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;
+
+		ssc_writex(params->ssc->regs, params->pdc->xpr,
+			   prtd->period_ptr);
+		ssc_writex(params->ssc->regs, params->pdc->xcr,
+			   prtd->period_size / params->pdc_xfer_size);
+		ssc_writex(params->ssc->regs, ATMEL_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;
+
+		ssc_writex(params->ssc->regs, params->pdc->xnpr,
+			   prtd->period_ptr);
+		ssc_writex(params->ssc->regs, params->pdc->xncr,
+			   prtd->period_size / params->pdc_xfer_size);
+	}
+
+	snd_pcm_period_elapsed(substream);
+}
+
+
+/*--------------------------------------------------------------------------*\
+ * PCM operations
+\*--------------------------------------------------------------------------*/
+static int atmel_pcm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct atmel_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 = atmel_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);
+
+	pr_debug("atmel-pcm: "
+		"hw_params: DMA for %s initialized "
+		"(dma_bytes=%u, period_size=%u)\n",
+		prtd->params->name,
+		runtime->dma_bytes,
+		prtd->period_size);
+	return 0;
+}
+
+static int atmel_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct atmel_runtime_data *prtd = substream->runtime->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+
+	if (params != NULL) {
+		ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
+			   params->mask->pdc_disable);
+		prtd->params->dma_intr_handler = NULL;
+	}
+
+	return 0;
+}
+
+static int atmel_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct atmel_runtime_data *prtd = substream->runtime->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+
+	ssc_writex(params->ssc->regs, SSC_IDR,
+		   params->mask->ssc_endx | params->mask->ssc_endbuf);
+	ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+		   params->mask->pdc_disable);
+	return 0;
+}
+
+static int atmel_pcm_trigger(struct snd_pcm_substream *substream,
+	int cmd)
+{
+	struct snd_pcm_runtime *rtd = substream->runtime;
+	struct atmel_runtime_data *prtd = rtd->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+	int ret = 0;
+
+	pr_debug("atmel-pcm:buffer_size = %ld,"
+		"dma_area = %p, dma_bytes = %u\n",
+		rtd->buffer_size, rtd->dma_area, rtd->dma_bytes);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		prtd->period_ptr = prtd->dma_buffer;
+
+		ssc_writex(params->ssc->regs, params->pdc->xpr,
+			   prtd->period_ptr);
+		ssc_writex(params->ssc->regs, params->pdc->xcr,
+			   prtd->period_size / params->pdc_xfer_size);
+
+		prtd->period_ptr += prtd->period_size;
+		ssc_writex(params->ssc->regs, params->pdc->xnpr,
+			   prtd->period_ptr);
+		ssc_writex(params->ssc->regs, params->pdc->xncr,
+			   prtd->period_size / params->pdc_xfer_size);
+
+		pr_debug("atmel-pcm: trigger: "
+			"period_ptr=%lx, xpr=%u, "
+			"xcr=%u, xnpr=%u, xncr=%u\n",
+			(unsigned long)prtd->period_ptr,
+			ssc_readx(params->ssc->regs, params->pdc->xpr),
+			ssc_readx(params->ssc->regs, params->pdc->xcr),
+			ssc_readx(params->ssc->regs, params->pdc->xnpr),
+			ssc_readx(params->ssc->regs, params->pdc->xncr));
+
+		ssc_writex(params->ssc->regs, SSC_IER,
+			   params->mask->ssc_endx | params->mask->ssc_endbuf);
+		ssc_writex(params->ssc->regs, SSC_PDC_PTCR,
+			   params->mask->pdc_enable);
+
+		pr_debug("sr=%u imr=%u\n",
+			ssc_readx(params->ssc->regs, SSC_SR),
+			ssc_readx(params->ssc->regs, SSC_IER));
+		break;		/* SNDRV_PCM_TRIGGER_START */
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+			   params->mask->pdc_disable);
+		break;
+
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR,
+			   params->mask->pdc_enable);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static snd_pcm_uframes_t atmel_pcm_pointer(
+	struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct atmel_runtime_data *prtd = runtime->private_data;
+	struct atmel_pcm_dma_params *params = prtd->params;
+	dma_addr_t ptr;
+	snd_pcm_uframes_t x;
+
+	ptr = (dma_addr_t) ssc_readx(params->ssc->regs, params->pdc->xpr);
+	x = bytes_to_frames(runtime, ptr - prtd->dma_buffer);
+
+	if (x == runtime->buffer_size)
+		x = 0;
+
+	return x;
+}
+
+static int atmel_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct atmel_runtime_data *prtd;
+	int ret = 0;
+
+	snd_soc_set_runtime_hwparams(substream, &atmel_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 atmel_runtime_data), GFP_KERNEL);
+	if (prtd == NULL) {
+		ret = -ENOMEM;
+		goto out;
+	}
+	runtime->private_data = prtd;
+
+ out:
+	return ret;
+}
+
+static int atmel_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct atmel_runtime_data *prtd = substream->runtime->private_data;
+
+	kfree(prtd);
+	return 0;
+}
+
+static int atmel_pcm_mmap(struct snd_pcm_substream *substream,
+	struct vm_area_struct *vma)
+{
+	return remap_pfn_range(vma, vma->vm_start,
+		       substream->dma_buffer.addr >> PAGE_SHIFT,
+		       vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+struct snd_pcm_ops atmel_pcm_ops = {
+	.open		= atmel_pcm_open,
+	.close		= atmel_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= atmel_pcm_hw_params,
+	.hw_free	= atmel_pcm_hw_free,
+	.prepare	= atmel_pcm_prepare,
+	.trigger	= atmel_pcm_trigger,
+	.pointer	= atmel_pcm_pointer,
+	.mmap		= atmel_pcm_mmap,
+};
+
+
+/*--------------------------------------------------------------------------*\
+ * ASoC platform driver
+\*--------------------------------------------------------------------------*/
+static u64 atmel_pcm_dmamask = 0xffffffff;
+
+static int atmel_pcm_new(struct snd_card *card,
+	struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+	int ret = 0;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &atmel_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = 0xffffffff;
+
+	if (dai->playback.channels_min) {
+		ret = atmel_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (dai->capture.channels_min) {
+		pr_debug("at32-pcm:"
+				"Allocating PCM capture DMA buffer\n");
+		ret = atmel_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+ out:
+	return ret;
+}
+
+static void atmel_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_coherent(pcm->card->dev, buf->bytes,
+				  buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+#ifdef CONFIG_PM
+static int atmel_pcm_suspend(struct snd_soc_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = dai->runtime;
+	struct atmel_runtime_data *prtd;
+	struct atmel_pcm_dma_params *params;
+
+	if (!runtime)
+		return 0;
+
+	prtd = runtime->private_data;
+	params = prtd->params;
+
+	/* disable the PDC and save the PDC registers */
+
+	ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable);
+
+	prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr);
+	prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr);
+	prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr);
+	prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr);
+
+	return 0;
+}
+
+static int atmel_pcm_resume(struct snd_soc_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = dai->runtime;
+	struct atmel_runtime_data *prtd;
+	struct atmel_pcm_dma_params *params;
+
+	if (!runtime)
+		return 0;
+
+	prtd = runtime->private_data;
+	params = prtd->params;
+
+	/* restore the PDC registers and enable the PDC */
+	ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save);
+	ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save);
+	ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save);
+	ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save);
+
+	ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable);
+	return 0;
+}
+#else
+#define atmel_pcm_suspend	NULL
+#define atmel_pcm_resume	NULL
+#endif
+
+struct snd_soc_platform atmel_soc_platform = {
+	.name		= "atmel-audio",
+	.pcm_ops 	= &atmel_pcm_ops,
+	.pcm_new	= atmel_pcm_new,
+	.pcm_free	= atmel_pcm_free_dma_buffers,
+	.suspend	= atmel_pcm_suspend,
+	.resume		= atmel_pcm_resume,
+};
+EXPORT_SYMBOL_GPL(atmel_soc_platform);
+
+static int __init atmel_pcm_modinit(void)
+{
+	return snd_soc_register_platform(&atmel_soc_platform);
+}
+module_init(atmel_pcm_modinit);
+
+static void __exit atmel_pcm_modexit(void)
+{
+	snd_soc_unregister_platform(&atmel_soc_platform);
+}
+module_exit(atmel_pcm_modexit);
+
+MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
+MODULE_DESCRIPTION("Atmel PCM module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h
new file mode 100644
index 00000000000..ec9b2824b66
--- /dev/null
+++ b/sound/soc/atmel/atmel-pcm.h
@@ -0,0 +1,86 @@
+/*
+ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC.
+ *
+ *  Copyright (C) 2005 SAN People
+ *  Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on at91-pcm. by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ *
+ * 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 as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#ifndef _ATMEL_PCM_H
+#define _ATMEL_PCM_H
+
+#include <linux/atmel-ssc.h>
+
+/*
+ * Registers and status bits that are required by the PCM driver.
+ */
+struct atmel_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 atmel_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 atmel_pcm_dma_params {
+	char *name;			/* stream identifier */
+	int pdc_xfer_size;		/* PDC counter increment in bytes */
+	struct ssc_device *ssc;		/* SSC device for stream */
+	struct atmel_pdc_regs *pdc;	/* PDC receive or transmit registers */
+	struct atmel_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 atmel_soc_platform;
+
+
+/*
+ * SSC register access (since ssc_writel() / ssc_readl() require literal name)
+ */
+#define ssc_readx(base, reg)            (__raw_readl((base) + (reg)))
+#define ssc_writex(base, reg, value)    __raw_writel((value), (base) + (reg))
+
+#endif /* _ATMEL_PCM_H */
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
new file mode 100644
index 00000000000..c5d67900d66
--- /dev/null
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -0,0 +1,790 @@
+/*
+ * atmel_ssc_dai.c  --  ALSA SoC ATMEL SSC Audio Layer Platform driver
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *         ATMEL CORP.
+ *
+ * Based on at91-ssc.c by
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#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 <linux/atmel_pdc.h>
+
+#include <linux/atmel-ssc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+
+#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
+#define NUM_SSC_DEVICES		1
+#else
+#define NUM_SSC_DEVICES		3
+#endif
+
+/*
+ * SSC PDC registers required by the PCM DMA engine.
+ */
+static struct atmel_pdc_regs pdc_tx_reg = {
+	.xpr		= ATMEL_PDC_TPR,
+	.xcr		= ATMEL_PDC_TCR,
+	.xnpr		= ATMEL_PDC_TNPR,
+	.xncr		= ATMEL_PDC_TNCR,
+};
+
+static struct atmel_pdc_regs pdc_rx_reg = {
+	.xpr		= ATMEL_PDC_RPR,
+	.xcr		= ATMEL_PDC_RCR,
+	.xnpr		= ATMEL_PDC_RNPR,
+	.xncr		= ATMEL_PDC_RNCR,
+};
+
+/*
+ * SSC & PDC status bits for transmit and receive.
+ */
+static struct atmel_ssc_mask ssc_tx_mask = {
+	.ssc_enable	= SSC_BIT(CR_TXEN),
+	.ssc_disable	= SSC_BIT(CR_TXDIS),
+	.ssc_endx	= SSC_BIT(SR_ENDTX),
+	.ssc_endbuf	= SSC_BIT(SR_TXBUFE),
+	.pdc_enable	= ATMEL_PDC_TXTEN,
+	.pdc_disable	= ATMEL_PDC_TXTDIS,
+};
+
+static struct atmel_ssc_mask ssc_rx_mask = {
+	.ssc_enable	= SSC_BIT(CR_RXEN),
+	.ssc_disable	= SSC_BIT(CR_RXDIS),
+	.ssc_endx	= SSC_BIT(SR_ENDRX),
+	.ssc_endbuf	= SSC_BIT(SR_RXBUFF),
+	.pdc_enable	= ATMEL_PDC_RXTEN,
+	.pdc_disable	= ATMEL_PDC_RXTDIS,
+};
+
+
+/*
+ * DMA parameters.
+ */
+static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = {
+	{{
+	.name		= "SSC0 PCM out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC0 PCM in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	} },
+#if NUM_SSC_DEVICES == 3
+	{{
+	.name		= "SSC1 PCM out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC1 PCM in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	} },
+	{{
+	.name		= "SSC2 PCM out",
+	.pdc		= &pdc_tx_reg,
+	.mask		= &ssc_tx_mask,
+	},
+	{
+	.name		= "SSC2 PCM in",
+	.pdc		= &pdc_rx_reg,
+	.mask		= &ssc_rx_mask,
+	} },
+#endif
+};
+
+
+static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = {
+	{
+	.name		= "ssc0",
+	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[0].lock),
+	.dir_mask	= SSC_DIR_MASK_UNUSED,
+	.initialized	= 0,
+	},
+#if NUM_SSC_DEVICES == 3
+	{
+	.name		= "ssc1",
+	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[1].lock),
+	.dir_mask	= SSC_DIR_MASK_UNUSED,
+	.initialized	= 0,
+	},
+	{
+	.name		= "ssc2",
+	.lock		= __SPIN_LOCK_UNLOCKED(ssc_info[2].lock),
+	.dir_mask	= SSC_DIR_MASK_UNUSED,
+	.initialized	= 0,
+	},
+#endif
+};
+
+
+/*
+ * SSC interrupt handler.  Passes PDC interrupts to the DMA
+ * interrupt handler in the PCM driver.
+ */
+static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id)
+{
+	struct atmel_ssc_info *ssc_p = dev_id;
+	struct atmel_pcm_dma_params *dma_params;
+	u32 ssc_sr;
+	u32 ssc_substream_mask;
+	int i;
+
+	ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR)
+			& (unsigned long)ssc_readl(ssc_p->ssc->regs, 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_substream_mask = (dma_params->mask->ssc_endx |
+					dma_params->mask->ssc_endbuf);
+			if (ssc_sr & ssc_substream_mask) {
+				dma_params->dma_intr_handler(ssc_sr,
+						dma_params->
+						substream);
+			}
+		}
+	}
+
+	return IRQ_HANDLED;
+}
+
+
+/*-------------------------------------------------------------------------*\
+ * DAI functions
+\*-------------------------------------------------------------------------*/
+/*
+ * Startup.  Only that one substream allowed in each direction.
+ */
+static int atmel_ssc_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	int dir_mask;
+
+	pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n",
+		ssc_readl(ssc_p->ssc->regs, SR));
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir_mask = SSC_DIR_MASK_PLAYBACK;
+	else
+		dir_mask = SSC_DIR_MASK_CAPTURE;
+
+	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 atmel_ssc_shutdown(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	struct atmel_pcm_dma_params *dma_params;
+	int dir, dir_mask;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = ssc_p->dma_params[dir];
+
+	if (dma_params != NULL) {
+		ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable);
+		pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n",
+			(dir ? "receive" : "transmit"),
+			ssc_readl(ssc_p->ssc->regs, SR));
+
+		dma_params->ssc = 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) {
+		if (ssc_p->initialized) {
+			/* Shutdown the SSC clock. */
+			pr_debug("atmel_ssc_dau: Stopping clock\n");
+			clk_disable(ssc_p->ssc->clk);
+
+			free_irq(ssc_p->ssc->irq, ssc_p);
+			ssc_p->initialized = 0;
+		}
+
+		/* Reset the SSC */
+		ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_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 DAI format for use in hw_params().
+ */
+static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+	ssc_p->daifmt = fmt;
+	return 0;
+}
+
+/*
+ * Record SSC clock dividers for use in hw_params().
+ */
+static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+	int div_id, int div)
+{
+	struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id];
+
+	switch (div_id) {
+	case ATMEL_SSC_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 ATMEL_SSC_TCMR_PERIOD:
+		ssc_p->tcmr_period = div;
+		break;
+
+	case ATMEL_SSC_RCMR_PERIOD:
+		ssc_p->rcmr_period = div;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Configure the SSC.
+ */
+static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	int id = rtd->dai->cpu_dai->id;
+	struct atmel_ssc_info *ssc_p = &ssc_info[id];
+	struct atmel_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.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = &ssc_dma_params[id][dir];
+	dma_params->ssc = ssc_p->ssc;
+	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);
+
+	/*
+	 * Determine sample size in bits and the PDC increment.
+	 */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		bits = 8;
+		dma_params->pdc_xfer_size = 1;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		bits = 16;
+		dma_params->pdc_xfer_size = 2;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		bits = 24;
+		dma_params->pdc_xfer_size = 4;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		bits = 32;
+		dma_params->pdc_xfer_size = 4;
+		break;
+	default:
+		printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format");
+		return -EINVAL;
+	}
+
+	/*
+	 * The SSC only supports up to 16-bit samples in I2S format, due
+	 * to the size of the Frame Mode Register FSLEN field.
+	 */
+	if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S
+		&& bits > 16) {
+		printk(KERN_WARNING
+				"atmel_ssc_dai: sample size %d"
+				"is too large for I2S\n", bits);
+		return -EINVAL;
+	}
+
+	/*
+	 * Compute SSC register settings.
+	 */
+	switch (ssc_p->daifmt
+		& (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) {
+
+	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS:
+		/*
+		 * I2S format, 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_BF(RCMR_PERIOD, ssc_p->rcmr_period)
+			| SSC_BF(RCMR_STTDLY, START_DELAY)
+			| SSC_BF(RCMR_START, SSC_START_FALLING_RF)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
+
+		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE)
+			| SSC_BF(RFMR_FSLEN, (bits - 1))
+			| SSC_BF(RFMR_DATNB, (channels - 1))
+			| SSC_BIT(RFMR_MSBF)
+			| SSC_BF(RFMR_LOOP, 0)
+			| SSC_BF(RFMR_DATLEN, (bits - 1));
+
+		tcmr =	  SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period)
+			| SSC_BF(TCMR_STTDLY, START_DELAY)
+			| SSC_BF(TCMR_START, SSC_START_FALLING_RF)
+			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
+			| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
+
+		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(TFMR_FSDEN, 0)
+			| SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE)
+			| SSC_BF(TFMR_FSLEN, (bits - 1))
+			| SSC_BF(TFMR_DATNB, (channels - 1))
+			| SSC_BIT(TFMR_MSBF)
+			| SSC_BF(TFMR_DATDEF, 0)
+			| SSC_BF(TFMR_DATLEN, (bits - 1));
+		break;
+
+	case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM:
+		/*
+		 * I2S format, CODEC supplies BCLK and LRC clocks.
+		 *
+		 * The SSC transmit clock is obtained from the BCLK signal on
+		 * on the TK line, and the SSC receive clock is
+		 * generated from the transmit clock.
+		 *
+		 *  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)
+				? SSC_START_FALLING_RF
+				: SSC_START_EDGE_RF);
+
+		rcmr =	  SSC_BF(RCMR_PERIOD, 0)
+			| SSC_BF(RCMR_STTDLY, START_DELAY)
+			| SSC_BF(RCMR_START, start_event)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(RCMR_CKS, SSC_CKS_CLOCK);
+
+		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(RFMR_FSOS, SSC_FSOS_NONE)
+			| SSC_BF(RFMR_FSLEN, 0)
+			| SSC_BF(RFMR_DATNB, 0)
+			| SSC_BIT(RFMR_MSBF)
+			| SSC_BF(RFMR_LOOP, 0)
+			| SSC_BF(RFMR_DATLEN, (bits - 1));
+
+		tcmr =	  SSC_BF(TCMR_PERIOD, 0)
+			| SSC_BF(TCMR_STTDLY, START_DELAY)
+			| SSC_BF(TCMR_START, start_event)
+			| SSC_BF(TCMR_CKI, SSC_CKI_FALLING)
+			| SSC_BF(TCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(TCMR_CKS, SSC_CKS_PIN);
+
+		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(TFMR_FSDEN, 0)
+			| SSC_BF(TFMR_FSOS, SSC_FSOS_NONE)
+			| SSC_BF(TFMR_FSLEN, 0)
+			| SSC_BF(TFMR_DATNB, 0)
+			| SSC_BIT(TFMR_MSBF)
+			| SSC_BF(TFMR_DATDEF, 0)
+			| SSC_BF(TFMR_DATLEN, (bits - 1));
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS:
+		/*
+		 * DSP/PCM Mode A format, 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_BF(RCMR_PERIOD, ssc_p->rcmr_period)
+			| SSC_BF(RCMR_STTDLY, 1)
+			| SSC_BF(RCMR_START, SSC_START_RISING_RF)
+			| SSC_BF(RCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(RCMR_CKO, SSC_CKO_NONE)
+			| SSC_BF(RCMR_CKS, SSC_CKS_DIV);
+
+		rfmr =	  SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE)
+			| SSC_BF(RFMR_FSLEN, 0)
+			| SSC_BF(RFMR_DATNB, (channels - 1))
+			| SSC_BIT(RFMR_MSBF)
+			| SSC_BF(RFMR_LOOP, 0)
+			| SSC_BF(RFMR_DATLEN, (bits - 1));
+
+		tcmr =	  SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period)
+			| SSC_BF(TCMR_STTDLY, 1)
+			| SSC_BF(TCMR_START, SSC_START_RISING_RF)
+			| SSC_BF(TCMR_CKI, SSC_CKI_RISING)
+			| SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS)
+			| SSC_BF(TCMR_CKS, SSC_CKS_DIV);
+
+		tfmr =	  SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE)
+			| SSC_BF(TFMR_FSDEN, 0)
+			| SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE)
+			| SSC_BF(TFMR_FSLEN, 0)
+			| SSC_BF(TFMR_DATNB, (channels - 1))
+			| SSC_BIT(TFMR_MSBF)
+			| SSC_BF(TFMR_DATDEF, 0)
+			| SSC_BF(TFMR_DATLEN, (bits - 1));
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM:
+	default:
+		printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
+			ssc_p->daifmt);
+		return -EINVAL;
+		break;
+	}
+	pr_debug("atmel_ssc_hw_params: "
+			"RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
+			rcmr, rfmr, tcmr, tfmr);
+
+	if (!ssc_p->initialized) {
+
+		/* Enable PMC peripheral clock for this SSC */
+		pr_debug("atmel_ssc_dai: Starting clock\n");
+		clk_enable(ssc_p->ssc->clk);
+
+		/* Reset the SSC and its PDC registers */
+		ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
+
+		ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0);
+
+		ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0);
+		ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0);
+
+		ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0,
+				ssc_p->name, ssc_p);
+		if (ret < 0) {
+			printk(KERN_WARNING
+					"atmel_ssc_dai: request_irq failure\n");
+			pr_debug("Atmel_ssc_dai: Stoping clock\n");
+			clk_disable(ssc_p->ssc->clk);
+			return ret;
+		}
+
+		ssc_p->initialized = 1;
+	}
+
+	/* set SSC clock mode register */
+	ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div);
+
+	/* set receive clock mode and format */
+	ssc_writel(ssc_p->ssc->regs, RCMR, rcmr);
+	ssc_writel(ssc_p->ssc->regs, RFMR, rfmr);
+
+	/* set transmit clock mode and format */
+	ssc_writel(ssc_p->ssc->regs, TCMR, tcmr);
+	ssc_writel(ssc_p->ssc->regs, TFMR, tfmr);
+
+	pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n");
+	return 0;
+}
+
+
+static int atmel_ssc_prepare(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id];
+	struct atmel_pcm_dma_params *dma_params;
+	int dir;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = 0;
+	else
+		dir = 1;
+
+	dma_params = ssc_p->dma_params[dir];
+
+	ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable);
+
+	pr_debug("%s enabled SSC_SR=0x%08x\n",
+			dir ? "receive" : "transmit",
+			ssc_readl(ssc_p->ssc->regs, SR));
+	return 0;
+}
+
+
+#ifdef CONFIG_PM
+static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai)
+{
+	struct atmel_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 = ssc_readl(ssc_p->ssc->regs, SR);
+	ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS));
+
+	/* Save the current interrupt mask, then disable unmasked interrupts */
+	ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR);
+	ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr);
+
+	ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR);
+	ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR);
+	ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR);
+	ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR);
+	ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR);
+
+	return 0;
+}
+
+
+
+static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
+{
+	struct atmel_ssc_info *ssc_p;
+	u32 cr;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	ssc_p = &ssc_info[cpu_dai->id];
+
+	/* restore SSC register settings */
+	ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr);
+	ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr);
+	ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr);
+	ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr);
+	ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr);
+
+	/* re-enable interrupts */
+	ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr);
+
+	/* Re-enable recieve and transmit as appropriate */
+	cr = 0;
+	cr |=
+	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0;
+	cr |=
+	    (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0;
+	ssc_writel(ssc_p->ssc->regs, CR, cr);
+
+	return 0;
+}
+#else /* CONFIG_PM */
+#  define atmel_ssc_suspend	NULL
+#  define atmel_ssc_resume	NULL
+#endif /* CONFIG_PM */
+
+
+#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
+			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
+	{	.name = "atmel-ssc0",
+		.id = 0,
+		.suspend = atmel_ssc_suspend,
+		.resume = atmel_ssc_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.ops = {
+			.startup = atmel_ssc_startup,
+			.shutdown = atmel_ssc_shutdown,
+			.prepare = atmel_ssc_prepare,
+			.hw_params = atmel_ssc_hw_params,
+			.set_fmt = atmel_ssc_set_dai_fmt,
+			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.private_data = &ssc_info[0],
+	},
+#if NUM_SSC_DEVICES == 3
+	{	.name = "atmel-ssc1",
+		.id = 1,
+		.suspend = atmel_ssc_suspend,
+		.resume = atmel_ssc_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.ops = {
+			.startup = atmel_ssc_startup,
+			.shutdown = atmel_ssc_shutdown,
+			.prepare = atmel_ssc_prepare,
+			.hw_params = atmel_ssc_hw_params,
+			.set_fmt = atmel_ssc_set_dai_fmt,
+			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.private_data = &ssc_info[1],
+	},
+	{	.name = "atmel-ssc2",
+		.id = 2,
+		.suspend = atmel_ssc_suspend,
+		.resume = atmel_ssc_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ATMEL_SSC_RATES,
+			.formats = ATMEL_SSC_FORMATS,},
+		.ops = {
+			.startup = atmel_ssc_startup,
+			.shutdown = atmel_ssc_shutdown,
+			.prepare = atmel_ssc_prepare,
+			.hw_params = atmel_ssc_hw_params,
+			.set_fmt = atmel_ssc_set_dai_fmt,
+			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.private_data = &ssc_info[2],
+	},
+#endif
+};
+EXPORT_SYMBOL_GPL(atmel_ssc_dai);
+
+static int __init atmel_ssc_modinit(void)
+{
+	return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+module_init(atmel_ssc_modinit);
+
+static void __exit atmel_ssc_modexit(void)
+{
+	snd_soc_unregister_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai));
+}
+module_exit(atmel_ssc_modexit);
+
+/* Module information */
+MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com");
+MODULE_DESCRIPTION("ATMEL SSC ASoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
new file mode 100644
index 00000000000..a828746e8a2
--- /dev/null
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -0,0 +1,121 @@
+/*
+ * atmel_ssc_dai.h - ALSA SSC interface for the Atmel  SoC
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *         ATMEL CORP.
+ *
+ * Based on at91-ssc.c by
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#ifndef _ATMEL_SSC_DAI_H
+#define _ATMEL_SSC_DAI_H
+
+#include <linux/types.h>
+#include <linux/atmel-ssc.h>
+
+#include "atmel-pcm.h"
+
+/* SSC system clock ids */
+#define ATMEL_SYSCLK_MCK	0 /* SSC uses AT91 MCK as system clock */
+
+/* SSC divider ids */
+#define ATMEL_SSC_CMR_DIV	0 /* MCK divider for BCLK */
+#define ATMEL_SSC_TCMR_PERIOD	1 /* BCLK divider for transmit FS */
+#define ATMEL_SSC_RCMR_PERIOD	2 /* BCLK divider for receive FS */
+/*
+ * SSC direction masks
+ */
+#define SSC_DIR_MASK_UNUSED	0
+#define SSC_DIR_MASK_PLAYBACK	1
+#define SSC_DIR_MASK_CAPTURE	2
+
+/*
+ * SSC register values that Atmel left out of <linux/atmel-ssc.h>.  These
+ * are expected to be used with SSC_BF
+ */
+/* START bit field values */
+#define SSC_START_CONTINUOUS	0
+#define SSC_START_TX_RX		1
+#define SSC_START_LOW_RF	2
+#define SSC_START_HIGH_RF	3
+#define SSC_START_FALLING_RF	4
+#define SSC_START_RISING_RF	5
+#define SSC_START_LEVEL_RF	6
+#define SSC_START_EDGE_RF	7
+#define SSS_START_COMPARE_0	8
+
+/* CKI bit field values */
+#define SSC_CKI_FALLING		0
+#define SSC_CKI_RISING		1
+
+/* CKO bit field values */
+#define SSC_CKO_NONE		0
+#define SSC_CKO_CONTINUOUS	1
+#define SSC_CKO_TRANSFER	2
+
+/* CKS bit field values */
+#define SSC_CKS_DIV		0
+#define SSC_CKS_CLOCK		1
+#define SSC_CKS_PIN		2
+
+/* FSEDGE bit field values */
+#define SSC_FSEDGE_POSITIVE	0
+#define SSC_FSEDGE_NEGATIVE	1
+
+/* FSOS bit field values */
+#define SSC_FSOS_NONE		0
+#define SSC_FSOS_NEGATIVE	1
+#define SSC_FSOS_POSITIVE	2
+#define SSC_FSOS_LOW		3
+#define SSC_FSOS_HIGH		4
+#define SSC_FSOS_TOGGLE		5
+
+#define START_DELAY		1
+
+struct atmel_ssc_state {
+	u32 ssc_cmr;
+	u32 ssc_rcmr;
+	u32 ssc_rfmr;
+	u32 ssc_tcmr;
+	u32 ssc_tfmr;
+	u32 ssc_sr;
+	u32 ssc_imr;
+};
+
+
+struct atmel_ssc_info {
+	char *name;
+	struct ssc_device *ssc;
+	spinlock_t lock;	/* lock for dir_mask */
+	unsigned short dir_mask;	/* 0=unused, 1=playback, 2=capture */
+	unsigned short initialized;	/* true if SSC has been initialized */
+	unsigned short daifmt;
+	unsigned short cmr_div;
+	unsigned short tcmr_period;
+	unsigned short rcmr_period;
+	struct atmel_pcm_dma_params *dma_params[2];
+	struct atmel_ssc_state ssc_state;
+};
+extern struct snd_soc_dai atmel_ssc_dai[];
+
+#endif /* _AT91_SSC_DAI_H */
diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index b1966e4dfcd..43dd8cee83c 100644
--- a/sound/soc/at32/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -22,7 +22,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/errno.h>
 #include <linux/clk.h>
@@ -40,8 +39,8 @@
 #include <mach/portmux.h>
 
 #include "../codecs/wm8510.h"
-#include "at32-pcm.h"
-#include "at32-ssc.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
 
 
 /*-------------------------------------------------------------------------*\
@@ -362,8 +361,9 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = {
 
 
 
-static struct snd_soc_machine snd_soc_machine_playpaq = {
+static struct snd_soc_card snd_soc_playpaq = {
 	.name = "LRS_PlayPaq_WM8510",
+	.platform = &at32_soc_platform,
 	.dai_link = &playpaq_wm8510_dai,
 	.num_links = 1,
 };
@@ -378,8 +378,7 @@ static struct wm8510_setup_data playpaq_wm8510_setup = {
 
 
 static struct snd_soc_device playpaq_wm8510_snd_devdata = {
-	.machine = &snd_soc_machine_playpaq,
-	.platform = &at32_soc_platform,
+	.card = &snd_soc_playpaq,
 	.codec_dev = &soc_codec_dev_wm8510,
 	.codec_data = &playpaq_wm8510_setup,
 };
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
new file mode 100644
index 00000000000..1fb59a9d371
--- /dev/null
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -0,0 +1,328 @@
+/*
+ * sam9g20_wm8731  --  SoC audio for AT91SAM9G20-based
+ * 			ATMEL AT91SAM9G20ek board.
+ *
+ *  Copyright (C) 2005 SAN People
+ *  Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on ati_b1_wm8731.c by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ * Based on corgi.c by:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+
+#include "../codecs/wm8731.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+
+static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	int ret;
+
+	/* codec system clock is supplied by PCK0, set to 12MHz */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+		12000000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+
+	dev_dbg(rtd->socdev->dev, "shutdown");
+}
+
+static int at91sam9g20ek_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_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct atmel_ssc_info *ssc_p = cpu_dai->private_data;
+	struct ssc_device *ssc = ssc_p->ssc;
+	int ret;
+
+	unsigned int rate;
+	int cmr_div, period;
+
+	if (ssc == NULL) {
+		printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n");
+		return -EINVAL;
+	}
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_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 = snd_soc_dai_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 = 55;	/* BCLK = 133MHz/(2*55) = 1.209MHz */
+		period = 74;	/* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */
+		break;
+	case 11025:
+		cmr_div = 67;	/* BCLK = 133MHz/(2*60) = 1.108MHz */
+		period = 45;	/* LRC = BCLK/(2*(49+1)) = 11083,3Hz */
+		break;
+	case 16000:
+		cmr_div = 63;	/* BCLK = 133MHz/(2*63) = 1.055MHz */
+		period = 32;	/* LRC = BCLK/(2*(32+1)) = 15993,2Hz */
+		break;
+	case 22050:
+		cmr_div = 52;	/* BCLK = 133MHz/(2*52) = 1.278MHz */
+		period = 28;	/* LRC = BCLK/(2*(28+1)) = 22049Hz */
+		break;
+	case 32000:
+		cmr_div = 66;	/* BCLK = 133MHz/(2*66) = 1.007MHz */
+		period = 15;	/* LRC = BCLK/(2*(15+1)) = 31486,742Hz */
+		break;
+	case 44100:
+		cmr_div = 29;	/* BCLK = 133MHz/(2*29) = 2.293MHz */
+		period = 25;	/* LRC = BCLK/(2*(25+1)) = 44098Hz */
+		break;
+	case 48000:
+		cmr_div = 33;	/* BCLK = 133MHz/(2*33) = 2.015MHz */
+		period = 20;	/* LRC = BCLK/(2*(20+1)) = 47979,79Hz */
+		break;
+	case 88200:
+		cmr_div = 29;	/* BCLK = 133MHz/(2*29) = 2.293MHz */
+		period = 12;	/* LRC = BCLK/(2*(12+1)) = 88196Hz */
+		break;
+	case 96000:
+		cmr_div = 23;	/* BCLK = 133MHz/(2*23) = 2.891MHz */
+		period = 14;	/* LRC = BCLK/(2*(14+1)) = 96376Hz */
+		break;
+	default:
+		printk(KERN_WARNING "unsupported rate %d"
+				" on at91sam9g20ek board\n", rate);
+		return -EINVAL;
+	}
+
+	/* set the MCK divider for BCLK */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div);
+	if (ret < 0)
+		return ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* set the BCLK divider for DACLRC */
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
+						ATMEL_SSC_TCMR_PERIOD, period);
+	} else {
+		/* set the BCLK divider for ADCLRC */
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
+						ATMEL_SSC_RCMR_PERIOD, period);
+	}
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops at91sam9g20ek_ops = {
+	.startup = at91sam9g20ek_startup,
+	.hw_params = at91sam9g20ek_hw_params,
+	.shutdown = at91sam9g20ek_shutdown,
+};
+
+
+static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+
+	/* 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"},
+};
+
+/*
+ * Logic for a wm8731 as connected on a at91sam9g20ek board.
+ */
+static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec)
+{
+	printk(KERN_DEBUG
+			"at91sam9g20ek_wm8731 "
+			": at91sam9g20ek_wm8731_init() called\n");
+
+	/* Add specific widgets */
+	snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
+				  ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
+	/* Set up specific audio path interconnects */
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	/* not connected */
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(codec, "Int Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link at91sam9g20ek_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731 PCM",
+	.cpu_dai = &atmel_ssc_dai[0],
+	.codec_dai = &wm8731_dai,
+	.init = at91sam9g20ek_wm8731_init,
+	.ops = &at91sam9g20ek_ops,
+};
+
+static struct snd_soc_card snd_soc_at91sam9g20ek = {
+	.name = "WM8731",
+	.platform = &atmel_soc_platform,
+	.dai_link = &at91sam9g20ek_dai,
+	.num_links = 1,
+};
+
+static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
+	.i2c_bus = 0,
+	.i2c_address = 0x1b,
+};
+
+static struct snd_soc_device at91sam9g20ek_snd_devdata = {
+	.card = &snd_soc_at91sam9g20ek,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &at91sam9g20ek_wm8731_setup,
+};
+
+static struct platform_device *at91sam9g20ek_snd_device;
+
+static int __init at91sam9g20ek_init(void)
+{
+	struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
+	struct ssc_device *ssc = NULL;
+	int ret;
+
+	/*
+	 * Request SSC device
+	 */
+	ssc = ssc_request(0);
+	if (IS_ERR(ssc)) {
+		ret = PTR_ERR(ssc);
+		ssc = NULL;
+		goto err_ssc;
+	}
+	ssc_p->ssc = ssc;
+
+	at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!at91sam9g20ek_snd_device) {
+		printk(KERN_DEBUG
+				"platform device allocation failed\n");
+		ret = -ENOMEM;
+	}
+
+	platform_set_drvdata(at91sam9g20ek_snd_device,
+			&at91sam9g20ek_snd_devdata);
+	at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev;
+
+	ret = platform_device_add(at91sam9g20ek_snd_device);
+	if (ret) {
+		printk(KERN_DEBUG
+				"platform device allocation failed\n");
+		platform_device_put(at91sam9g20ek_snd_device);
+	}
+
+	return ret;
+
+err_ssc:
+	return ret;
+}
+
+static void __exit at91sam9g20ek_exit(void)
+{
+	struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
+	struct ssc_device *ssc;
+
+	if (ssc_p != NULL) {
+		ssc = ssc_p->ssc;
+		if (ssc != NULL)
+			ssc_free(ssc);
+		ssc_p->ssc = NULL;
+	}
+
+	platform_device_unregister(at91sam9g20ek_snd_device);
+	at91sam9g20ek_snd_device = NULL;
+}
+
+module_init(at91sam9g20ek_init);
+module_exit(at91sam9g20ek_exit);
+
+/* Module information */
+MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
+MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 1466d932880..74c823d60f9 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -406,11 +406,12 @@ static int __init au1xpsc_audio_dbdma_init(void)
 {
 	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
 	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
-	return 0;
+	return snd_soc_register_platform(&au1xpsc_soc_platform);
 }
 
 static void __exit au1xpsc_audio_dbdma_exit(void)
 {
+	snd_soc_unregister_platform(&au1xpsc_soc_platform);
 }
 
 module_init(au1xpsc_audio_dbdma_init);
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 57facbad682..f0e30aec7f2 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -160,7 +160,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
-				  struct snd_pcm_hw_params *params)
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -210,7 +211,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
-				int cmd)
+				int cmd, struct snd_soc_dai *dai)
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -313,8 +314,7 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev,
 	au1xpsc_ac97_workdata = NULL;
 }
 
-static int au1xpsc_ac97_suspend(struct platform_device *pdev,
-				struct snd_soc_dai *dai)
+static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
 {
 	/* save interesting registers and disable PSC */
 	au1xpsc_ac97_workdata->pm[0] =
@@ -328,8 +328,7 @@ static int au1xpsc_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_ac97_resume(struct platform_device *pdev,
-			       struct snd_soc_dai *dai)
+static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 {
 	/* restore PSC clock config */
 	au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
@@ -345,7 +344,7 @@ static int au1xpsc_ac97_resume(struct platform_device *pdev,
 
 struct snd_soc_dai au1xpsc_ac97_dai = {
 	.name			= "au1xpsc_ac97",
-	.type			= SND_SOC_DAI_AC97,
+	.ac97_control		= 1,
 	.probe			= au1xpsc_ac97_probe,
 	.remove			= au1xpsc_ac97_remove,
 	.suspend		= au1xpsc_ac97_suspend,
@@ -372,11 +371,12 @@ EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 static int __init au1xpsc_ac97_init(void)
 {
 	au1xpsc_ac97_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_ac97_dai);
 }
 
 static void __exit au1xpsc_ac97_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
 }
 
 module_init(au1xpsc_ac97_init);
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 9384702c7eb..f916de4400e 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -116,7 +116,8 @@ out:
 }
 
 static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
 
@@ -240,7 +241,8 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
 	return 0;
 }
 
-static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
 	int ret, stype = SUBSTREAM_TYPE(substream);
@@ -337,8 +339,7 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev,
 	au1xpsc_i2s_workdata = NULL;
 }
 
-static int au1xpsc_i2s_suspend(struct platform_device *pdev,
-			       struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	/* save interesting register and disable PSC */
 	au1xpsc_i2s_workdata->pm[0] =
@@ -352,8 +353,7 @@ static int au1xpsc_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	/* select I2S mode and PSC clock */
 	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
@@ -369,7 +369,6 @@ static int au1xpsc_i2s_resume(struct platform_device *pdev,
 
 struct snd_soc_dai au1xpsc_i2s_dai = {
 	.name			= "au1xpsc_i2s",
-	.type			= SND_SOC_DAI_I2S,
 	.probe			= au1xpsc_i2s_probe,
 	.remove			= au1xpsc_i2s_remove,
 	.suspend		= au1xpsc_i2s_suspend,
@@ -389,8 +388,6 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 	.ops = {
 		.trigger	= au1xpsc_i2s_trigger,
 		.hw_params	= au1xpsc_i2s_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt	= au1xpsc_i2s_set_fmt,
 	},
 };
@@ -399,11 +396,12 @@ EXPORT_SYMBOL(au1xpsc_i2s_dai);
 static int __init au1xpsc_i2s_init(void)
 {
 	au1xpsc_i2s_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_i2s_dai);
 }
 
 static void __exit au1xpsc_i2s_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
 }
 
 module_init(au1xpsc_i2s_init);
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
index f75ae7f62c3..27683eb7905 100644
--- a/sound/soc/au1x/sample-ac97.c
+++ b/sound/soc/au1x/sample-ac97.c
@@ -42,14 +42,14 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
 	.ops		= NULL,
 };
 
-static struct snd_soc_machine au1xpsc_sample_ac97_machine = {
+static struct snd_soc_card au1xpsc_sample_ac97_machine = {
 	.name		= "Au1xxx PSC AC97 Audio",
 	.dai_link	= &au1xpsc_sample_ac97_dai,
 	.num_links	= 1,
 };
 
 static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
-	.machine	= &au1xpsc_sample_ac97_machine,
+	.card		= &au1xpsc_sample_ac97_machine,
 	.platform	= &au1xpsc_soc_platform, /* see dbdma2.c */
 	.codec_dev	= &soc_codec_dev_ac97,
 };
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index dc006206f62..0a2f8f9eff5 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -1,6 +1,6 @@
 config SND_BF5XX_I2S
 	tristate "SoC I2S Audio for the ADI BF5xx chip"
-	depends on BLACKFIN && SND_SOC
+	depends on BLACKFIN
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -13,7 +13,6 @@ config SND_BF5XX_SOC_SSM2602
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_SSM2602
 	select I2C
-	select I2C_BLACKFIN_TWI
 	help
 	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
 
@@ -35,7 +34,7 @@ config SND_BFIN_AD73311_SE
 
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
-	depends on BLACKFIN && SND_SOC
+	depends on BLACKFIN
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in slot 16
@@ -47,7 +46,7 @@ config SND_BF5XX_AC97
 	  properly with this driver. This driver is known to work with the
 	  Analog Devices line of AC97 codecs.
 
-config SND_MMAP_SUPPORT
+config SND_BF5XX_MMAP_SUPPORT
 	bool "Enable MMAP Support"
 	depends on SND_BF5XX_AC97
 	default y
@@ -55,9 +54,17 @@ config SND_MMAP_SUPPORT
 	  Say y if you want AC97 driver to support mmap mode.
 	  We introduce an intermediate buffer to simulate mmap.
 
+config SND_BF5XX_MULTICHAN_SUPPORT
+	bool "Enable Multichannel Support"
+	depends on SND_BF5XX_AC97
+	default n
+	help
+	  Say y if you want AC97 driver to support up to 5.1 channel audio.
+	  this mode will consume much more memory for DMA.
+
 config SND_BF5XX_SOC_SPORT
 	tristate
-	
+
 config SND_BF5XX_SOC_I2S
 	tristate
 	select SND_BF5XX_SOC_SPORT
@@ -80,7 +87,7 @@ config SND_BF5XX_SPORT_NUM
 	int "Set a SPORT for Sound chip"
 	depends on (SND_BF5XX_I2S || SND_BF5XX_AC97)
 	range 0 3 if BF54x
-	range 0 1 if (BF53x || BF561)
+	range 0 1 if !BF54x
 	default 0
 	help
 	  Set the correct SPORT for sound chip.
@@ -90,12 +97,13 @@ config SND_BF5XX_HAVE_COLD_RESET
 	depends on SND_BF5XX_AC97
 	default y if BFIN548_EZKIT
 	default n if !BFIN548_EZKIT
-	
+
 config SND_BF5XX_RESET_GPIO_NUM
 	int "Set a GPIO for cold reset"
 	depends on SND_BF5XX_HAVE_COLD_RESET
 	range 0 159
 	default 19 if BFIN548_EZKIT
 	default 5 if BFIN537_STAMP
+	default 0
 	help
 	  Set the correct GPIO for RESET the sound chip.
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 25e50d2ea1e..8067cfafa3a 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -43,24 +43,34 @@
 #include "bf5xx-ac97.h"
 #include "bf5xx-sport.h"
 
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+static unsigned int ac97_chan_mask[] = {
+	SP_FL, /* Mono */
+	SP_STEREO, /* Stereo */
+	SP_2DOT1, /* 2.1*/
+	SP_QUAD,/*Quadraquic*/
+	SP_FL | SP_FR | SP_FC | SP_SL | SP_SR,/*5 channels */
+	SP_5DOT1, /* 5.1 */
+};
+
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
 	 snd_pcm_uframes_t count)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct sport_device *sport = runtime->private_data;
+	unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1];
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		bf5xx_pcm_to_ac97(
-			(struct ac97_frame *)sport->tx_dma_buf + sport->tx_pos,
-			(__u32 *)runtime->dma_area + sport->tx_pos, count);
+		bf5xx_pcm_to_ac97((struct ac97_frame *)sport->tx_dma_buf +
+		sport->tx_pos, (__u16 *)runtime->dma_area + sport->tx_pos *
+		runtime->channels, count, chan_mask);
 		sport->tx_pos += runtime->period_size;
 		if (sport->tx_pos >= runtime->buffer_size)
 			sport->tx_pos %= runtime->buffer_size;
 		sport->tx_delay_pos = sport->tx_pos;
 	} else {
-		bf5xx_ac97_to_pcm(
-			(struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos,
-			(__u32 *)runtime->dma_area + sport->rx_pos, count);
+		bf5xx_ac97_to_pcm((struct ac97_frame *)sport->rx_dma_buf +
+		sport->rx_pos, (__u16 *)runtime->dma_area + sport->rx_pos *
+		runtime->channels, count);
 		sport->rx_pos += runtime->period_size;
 		if (sport->rx_pos >= runtime->buffer_size)
 			sport->rx_pos %= runtime->buffer_size;
@@ -71,7 +81,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
 static void bf5xx_dma_irq(void *data)
 {
 	struct snd_pcm_substream *pcm = data;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	struct snd_pcm_runtime *runtime = pcm->runtime;
 	struct sport_device *sport = runtime->private_data;
 	bf5xx_mmap_copy(pcm, runtime->period_size);
@@ -90,17 +100,14 @@ static void bf5xx_dma_irq(void *data)
  * The total rx/tx buffer is for ac97 frame to hold all pcm data
  * is  0x20000 * sizeof(struct ac97_frame) / 4.
  */
-#ifdef CONFIG_SND_MMAP_SUPPORT
 static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 				   SNDRV_PCM_INFO_MMAP |
 				   SNDRV_PCM_INFO_MMAP_VALID |
-				   SNDRV_PCM_INFO_BLOCK_TRANSFER,
-#else
-static const struct snd_pcm_hardware bf5xx_pcm_hardware = {
-	.info			= SNDRV_PCM_INFO_INTERLEAVED |
-				  SNDRV_PCM_INFO_BLOCK_TRANSFER,
 #endif
+				   SNDRV_PCM_INFO_BLOCK_TRANSFER,
+
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
 	.period_bytes_min	= 32,
 	.period_bytes_max	= 0x10000,
@@ -123,10 +130,20 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream,
 
 static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct sport_device *sport = runtime->private_data;
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-	memset(runtime->dma_area, 0, runtime->buffer_size);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		sport->once = 0;
+		if (runtime->dma_area)
+			memset(runtime->dma_area, 0, runtime->buffer_size);
+		memset(sport->tx_dma_buf, 0, runtime->buffer_size *
+			sizeof(struct ac97_frame));
+	} else
+		memset(sport->rx_dma_buf, 0, runtime->buffer_size *
+			sizeof(struct ac97_frame));
+#endif
 	snd_pcm_lib_free_pages(substream);
 	return 0;
 }
@@ -139,7 +156,7 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream)
 	/* An intermediate buffer is introduced for implementing mmap for
 	 * SPORT working in TMD mode(include AC97).
 	 */
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		sport_set_tx_callback(sport, bf5xx_dma_irq, substream);
 		sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods,
@@ -173,24 +190,24 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 			bf5xx_mmap_copy(substream, runtime->period_size);
-			snd_pcm_period_elapsed(substream);
 			sport->tx_delay_pos = 0;
+#endif
 			sport_tx_start(sport);
-		}
-		else
+		} else
 			sport_rx_start(sport);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 			sport->tx_pos = 0;
 #endif
 			sport_tx_stop(sport);
 		} else {
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 			sport->rx_pos = 0;
 #endif
 			sport_rx_stop(sport);
@@ -208,7 +225,7 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
 	struct sport_device *sport = runtime->private_data;
 	unsigned int curr;
 
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		curr = sport->tx_delay_pos;
 	else
@@ -249,22 +266,7 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream)
 	return ret;
 }
 
-static int bf5xx_pcm_close(struct snd_pcm_substream *substream)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct sport_device *sport = runtime->private_data;
-
-	pr_debug("%s enter\n", __func__);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		sport->once = 0;
-		memset(sport->tx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
-	} else
-		memset(sport->rx_dma_buf, 0, runtime->buffer_size * sizeof(struct ac97_frame));
-
-	return 0;
-}
-
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma)
 {
@@ -281,32 +283,29 @@ static	int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 		    void __user *buf, snd_pcm_uframes_t count)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-
+	unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1];
 	pr_debug("%s copy pos:0x%lx count:0x%lx\n",
 			substream->stream ? "Capture" : "Playback", pos, count);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		bf5xx_pcm_to_ac97(
-				(struct ac97_frame *)runtime->dma_area + pos,
-				buf, count);
+		bf5xx_pcm_to_ac97((struct ac97_frame *)runtime->dma_area + pos,
+			(__u16 *)buf, count, chan_mask);
 	else
-		bf5xx_ac97_to_pcm(
-				(struct ac97_frame *)runtime->dma_area + pos,
-				buf, count);
+		bf5xx_ac97_to_pcm((struct ac97_frame *)runtime->dma_area + pos,
+			(__u16 *)buf, count);
 	return 0;
 }
 #endif
 
 struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
 	.open		= bf5xx_pcm_open,
-	.close		= bf5xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
 	.hw_free	= bf5xx_pcm_hw_free,
 	.prepare	= bf5xx_pcm_prepare,
 	.trigger	= bf5xx_pcm_trigger,
 	.pointer	= bf5xx_pcm_pointer,
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	.mmap		= bf5xx_pcm_mmap,
 #else
 	.copy		= bf5xx_pcm_copy,
@@ -344,7 +343,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
  * Need to allocate local buffer when enable
  * MMAP for SPORT working in TMD mode (include AC97).
  */
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (!sport_handle->tx_dma_buf) {
 			sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \
@@ -381,7 +380,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 	struct snd_pcm_substream *substream;
 	struct snd_dma_buffer *buf;
 	int stream;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	size_t size = bf5xx_pcm_hardware.buffer_bytes_max *
 		sizeof(struct ac97_frame) / 4;
 #endif
@@ -395,7 +394,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 			continue;
 		dma_free_coherent(NULL, buf->bytes, buf->area, 0);
 		buf->area = NULL;
-#if defined(CONFIG_SND_MMAP_SUPPORT)
+#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		if (sport_handle->tx_dma_buf)
 			dma_free_coherent(NULL, size, \
@@ -452,6 +451,18 @@ struct snd_soc_platform bf5xx_ac97_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform);
 
+static int __init bfin_ac97_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_ac97_soc_platform);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_ac97_soc_platform);
+}
+module_exit(bfin_ac97_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 5e5aafb6485..3be2be60576 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -54,71 +54,103 @@
 static int *cmd_count;
 static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
 
-#if defined(CONFIG_BF54x)
+static u16 sport_req[][7] = {
+		PIN_REQ_SPORT_0,
+#ifdef PIN_REQ_SPORT_1
+		PIN_REQ_SPORT_1,
+#endif
+#ifdef PIN_REQ_SPORT_2
+		PIN_REQ_SPORT_2,
+#endif
+#ifdef PIN_REQ_SPORT_3
+		PIN_REQ_SPORT_3,
+#endif
+	};
+
 static struct sport_param sport_params[4] = {
 	{
 		.dma_rx_chan	= CH_SPORT0_RX,
 		.dma_tx_chan	= CH_SPORT0_TX,
-		.err_irq	= IRQ_SPORT0_ERR,
+		.err_irq	= IRQ_SPORT0_ERROR,
 		.regs		= (struct sport_register *)SPORT0_TCR1,
 	},
+#ifdef PIN_REQ_SPORT_1
 	{
 		.dma_rx_chan	= CH_SPORT1_RX,
 		.dma_tx_chan	= CH_SPORT1_TX,
-		.err_irq	= IRQ_SPORT1_ERR,
+		.err_irq	= IRQ_SPORT1_ERROR,
 		.regs		= (struct sport_register *)SPORT1_TCR1,
 	},
+#endif
+#ifdef PIN_REQ_SPORT_2
 	{
 		.dma_rx_chan	= CH_SPORT2_RX,
 		.dma_tx_chan	= CH_SPORT2_TX,
-		.err_irq	= IRQ_SPORT2_ERR,
+		.err_irq	= IRQ_SPORT2_ERROR,
 		.regs		= (struct sport_register *)SPORT2_TCR1,
 	},
+#endif
+#ifdef PIN_REQ_SPORT_3
 	{
 		.dma_rx_chan	= CH_SPORT3_RX,
 		.dma_tx_chan	= CH_SPORT3_TX,
-		.err_irq	= IRQ_SPORT3_ERR,
+		.err_irq	= IRQ_SPORT3_ERROR,
 		.regs		= (struct sport_register *)SPORT3_TCR1,
 	}
-};
-#else
-static struct sport_param sport_params[2] = {
-	{
-		.dma_rx_chan	= CH_SPORT0_RX,
-		.dma_tx_chan	= CH_SPORT0_TX,
-		.err_irq	= IRQ_SPORT0_ERROR,
-		.regs		= (struct sport_register *)SPORT0_TCR1,
-	},
-	{
-		.dma_rx_chan	= CH_SPORT1_RX,
-		.dma_tx_chan	= CH_SPORT1_TX,
-		.err_irq	= IRQ_SPORT1_ERROR,
-		.regs		= (struct sport_register *)SPORT1_TCR1,
-	}
-};
 #endif
+};
 
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
-		size_t count)
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src,
+		size_t count, unsigned int chan_mask)
 {
 	while (count--) {
-		dst->ac97_tag = TAG_VALID | TAG_PCM;
-		(dst++)->ac97_pcm = *src++;
+		dst->ac97_tag = TAG_VALID;
+		if (chan_mask & SP_FL) {
+			dst->ac97_pcm_r = *src++;
+			dst->ac97_tag |= TAG_PCM_RIGHT;
+		}
+		if (chan_mask & SP_FR) {
+			dst->ac97_pcm_l = *src++;
+			dst->ac97_tag |= TAG_PCM_LEFT;
+
+		}
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+		if (chan_mask & SP_SR) {
+			dst->ac97_sl = *src++;
+			dst->ac97_tag |= TAG_PCM_SL;
+		}
+		if (chan_mask & SP_SL) {
+			dst->ac97_sr = *src++;
+			dst->ac97_tag |= TAG_PCM_SR;
+		}
+		if (chan_mask & SP_LFE) {
+			dst->ac97_lfe = *src++;
+			dst->ac97_tag |= TAG_PCM_LFE;
+		}
+		if (chan_mask & SP_FC) {
+			dst->ac97_center = *src++;
+			dst->ac97_tag |= TAG_PCM_CENTER;
+		}
+#endif
+		dst++;
 	}
 }
 EXPORT_SYMBOL(bf5xx_pcm_to_ac97);
 
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst,
 		size_t count)
 {
-	while (count--)
-		*(dst++) = (src++)->ac97_pcm;
+	while (count--) {
+		*(dst++) = src->ac97_pcm_l;
+		*(dst++) = src->ac97_pcm_r;
+		src++;
+	}
 }
 EXPORT_SYMBOL(bf5xx_ac97_to_pcm);
 
 static unsigned int sport_tx_curr_frag(struct sport_device *sport)
 {
-	return sport->tx_curr_frag = sport_curr_offset_tx(sport) / \
+	return sport->tx_curr_frag = sport_curr_offset_tx(sport) /
 			sport->tx_fragsize;
 }
 
@@ -130,7 +162,7 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data)
 
 	sport_incfrag(sport, &nextfrag, 1);
 
-	nextwrite = (struct ac97_frame *)(sport->tx_buf + \
+	nextwrite = (struct ac97_frame *)(sport->tx_buf +
 			nextfrag * sport->tx_fragsize);
 	pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n",
 		sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]);
@@ -237,8 +269,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 #ifdef CONFIG_PM
-static int bf5xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	struct sport_device *sport =
 		(struct sport_device *)dai->private_data;
@@ -253,8 +284,7 @@ static int bf5xx_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int bf5xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	int ret;
 	struct sport_device *sport =
@@ -297,20 +327,15 @@ static int bf5xx_ac97_resume(struct platform_device *pdev,
 static int bf5xx_ac97_probe(struct platform_device *pdev,
 			    struct snd_soc_dai *dai)
 {
-	int ret;
-#if defined(CONFIG_BF54x)
-	u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1,
-				 PIN_REQ_SPORT_2, PIN_REQ_SPORT_3};
-#else
-	u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1};
-#endif
+	int ret = 0;
 	cmd_count = (int *)get_zeroed_page(GFP_KERNEL);
 	if (cmd_count == NULL)
 		return -ENOMEM;
 
 	if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
 		pr_err("Requesting Peripherals failed\n");
-		return -EFAULT;
+		ret =  -EFAULT;
+		goto peripheral_err;
 		}
 
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
@@ -318,54 +343,54 @@ static int bf5xx_ac97_probe(struct platform_device *pdev,
 	if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) {
 		pr_err("Failed to request GPIO_%d for reset\n",
 				CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-		peripheral_free_list(&sport_req[sport_num][0]);
-		return -1;
+		ret =  -1;
+		goto gpio_err;
 	}
 	gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1);
 #endif
 	sport_handle = sport_init(&sport_params[sport_num], 2, \
 			sizeof(struct ac97_frame), NULL);
 	if (!sport_handle) {
-		peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
-		gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
-		return -ENODEV;
+		ret = -ENODEV;
+		goto sport_err;
 	}
 	/*SPORT works in TDM mode to simulate AC97 transfers*/
 	ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1);
 	if (ret) {
 		pr_err("SPORT is busy!\n");
-		kfree(sport_handle);
-		peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
-		gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
-		return -EBUSY;
+		ret = -EBUSY;
+		goto sport_config_err;
 	}
 
 	ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1));
 	if (ret) {
 		pr_err("SPORT is busy!\n");
-		kfree(sport_handle);
-		peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
-		gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
-		return -EBUSY;
+		ret = -EBUSY;
+		goto sport_config_err;
 	}
 
 	ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1));
 	if (ret) {
 		pr_err("SPORT is busy!\n");
-		kfree(sport_handle);
-		peripheral_free_list(&sport_req[sport_num][0]);
-#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
-		gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
-		return -EBUSY;
+		ret = -EBUSY;
+		goto sport_config_err;
 	}
+
 	return 0;
+
+sport_config_err:
+	kfree(sport_handle);
+sport_err:
+#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
+	gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
+#endif
+gpio_err:
+	peripheral_free_list(&sport_req[sport_num][0]);
+peripheral_err:
+	free_page((unsigned long)cmd_count);
+	cmd_count = NULL;
+
+	return ret;
 }
 
 static void bf5xx_ac97_remove(struct platform_device *pdev,
@@ -373,6 +398,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev,
 {
 	free_page((unsigned long)cmd_count);
 	cmd_count = NULL;
+	peripheral_free_list(&sport_req[sport_num][0]);
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
 	gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
 #endif
@@ -381,7 +407,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev,
 struct snd_soc_dai bfin_ac97_dai = {
 	.name = "bf5xx-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = bf5xx_ac97_probe,
 	.remove = bf5xx_ac97_remove,
 	.suspend = bf5xx_ac97_suspend,
@@ -389,7 +415,11 @@ struct snd_soc_dai bfin_ac97_dai = {
 	.playback = {
 		.stream_name = "AC97 Playback",
 		.channels_min = 2,
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+		.channels_max = 6,
+#else
 		.channels_max = 2,
+#endif
 		.rates = SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
 	.capture = {
@@ -401,6 +431,18 @@ struct snd_soc_dai bfin_ac97_dai = {
 };
 EXPORT_SYMBOL_GPL(bfin_ac97_dai);
 
+static int __init bfin_ac97_init(void)
+{
+	return snd_soc_register_dai(&bfin_ac97_dai);
+}
+module_init(bfin_ac97_init);
+
+static void __exit bfin_ac97_exit(void)
+{
+	snd_soc_unregister_dai(&bfin_ac97_dai);
+}
+module_exit(bfin_ac97_exit);
+
 MODULE_AUTHOR("Roy Huang");
 MODULE_DESCRIPTION("AC97 driver for ADI Blackfin");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h
index 3f77cc558dc..3f2a911fe0c 100644
--- a/sound/soc/blackfin/bf5xx-ac97.h
+++ b/sound/soc/blackfin/bf5xx-ac97.h
@@ -16,21 +16,46 @@ struct ac97_frame {
 	u16 ac97_tag;		/* slot 0 */
 	u16 ac97_addr;		/* slot 1 */
 	u16 ac97_data;		/* slot 2 */
-	u32 ac97_pcm;		/* slot 3 and 4: left and right pcm data */
+	u16 ac97_pcm_l;		/*slot 3:front left*/
+	u16 ac97_pcm_r;		/*slot 4:front left*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+	u16 ac97_mdm_l1;
+	u16 ac97_center;	/*slot 6:center*/
+	u16 ac97_sl;		/*slot 7:surround left*/
+	u16 ac97_sr;		/*slot 8:surround right*/
+	u16 ac97_lfe;		/*slot 9:lfe*/
+#endif
 } __attribute__ ((packed));
 
+/* Speaker location */
+#define SP_FL		0x0001
+#define SP_FR		0x0010
+#define SP_FC		0x0002
+#define SP_LFE		0x0020
+#define SP_SL		0x0004
+#define SP_SR		0x0040
+
+#define SP_STEREO	(SP_FL | SP_FR)
+#define SP_2DOT1	(SP_FL | SP_FR | SP_LFE)
+#define SP_QUAD		(SP_FL | SP_FR | SP_SL | SP_SR)
+#define SP_5DOT1	(SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR)
+
 #define TAG_VALID		0x8000
 #define TAG_CMD			0x6000
 #define TAG_PCM_LEFT		0x1000
 #define TAG_PCM_RIGHT		0x0800
-#define TAG_PCM			(TAG_PCM_LEFT | TAG_PCM_RIGHT)
+#define TAG_PCM_MDM_L1		0x0400
+#define TAG_PCM_CENTER		0x0200
+#define TAG_PCM_SL		0x0100
+#define TAG_PCM_SR		0x0080
+#define TAG_PCM_LFE		0x0040
 
 extern struct snd_soc_dai bfin_ac97_dai;
 
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
-		size_t count);
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \
+		size_t count, unsigned int chan_mask);
 
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \
 		size_t count);
 
 #endif
diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c
index 124425d2232..d8f59127377 100644
--- a/sound/soc/blackfin/bf5xx-ad1980.c
+++ b/sound/soc/blackfin/bf5xx-ad1980.c
@@ -43,7 +43,7 @@
 #include "bf5xx-ac97-pcm.h"
 #include "bf5xx-ac97.h"
 
-static struct snd_soc_machine bf5xx_board;
+static struct snd_soc_card bf5xx_board;
 
 static int bf5xx_board_startup(struct snd_pcm_substream *substream)
 {
@@ -67,15 +67,15 @@ static struct snd_soc_dai_link bf5xx_board_dai = {
 	.ops = &bf5xx_board_ops,
 };
 
-static struct snd_soc_machine bf5xx_board = {
+static struct snd_soc_card bf5xx_board = {
 	.name = "bf5xx-board",
+	.platform = &bf5xx_ac97_soc_platform,
 	.dai_link = &bf5xx_board_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_board_snd_devdata = {
-	.machine = &bf5xx_board,
-	.platform = &bf5xx_ac97_soc_platform,
+	.card = &bf5xx_board,
 	.codec_dev = &soc_codec_dev_ad1980,
 };
 
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 622c9b90953..7f2a5e19907 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -65,7 +65,7 @@
 
 #define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
 
-static struct snd_soc_machine bf5xx_ad73311;
+static struct snd_soc_card bf5xx_ad73311;
 
 static int snd_ad73311_startup(void)
 {
@@ -168,7 +168,7 @@ static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream,
 		params_format(params));
 
 	/* set cpu DAI configuration */
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
@@ -190,16 +190,16 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = {
 	.ops = &bf5xx_ad73311_ops,
 };
 
-static struct snd_soc_machine bf5xx_ad73311 = {
+static struct snd_soc_card bf5xx_ad73311 = {
 	.name = "bf5xx_ad73311",
+	.platform = &bf5xx_i2s_soc_platform,
 	.probe = bf5xx_probe,
 	.dai_link = &bf5xx_ad73311_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
-	.machine = &bf5xx_ad73311,
-	.platform = &bf5xx_i2s_soc_platform,
+	.card = &bf5xx_ad73311,
 	.codec_dev = &soc_codec_dev_ad73311,
 };
 
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 61fccf92519..53d290b3ea4 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -283,6 +283,18 @@ struct snd_soc_platform bf5xx_i2s_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform);
 
+static int __init bfin_i2s_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_i2s_soc_platform);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_i2s_soc_platform);
+}
+module_exit(bfin_i2s_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index e020c160ee4..d1d95d2393f 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -132,7 +132,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	return ret;
 }
 
-static int bf5xx_i2s_startup(struct snd_pcm_substream *substream)
+static int bf5xx_i2s_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 
@@ -142,7 +143,8 @@ static int bf5xx_i2s_startup(struct snd_pcm_substream *substream)
 }
 
 static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -193,7 +195,8 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream)
+static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	bf5xx_i2s.counter--;
@@ -219,16 +222,14 @@ static int bf5xx_i2s_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void bf5xx_i2s_remove(struct platform_device *pdev,
-			   struct snd_soc_dai *dai)
+static void bf5xx_i2s_remove(struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
 	peripheral_free_list(&sport_req[sport_num][0]);
 }
 
 #ifdef CONFIG_PM
-static int bf5xx_i2s_suspend(struct platform_device *dev,
-			     struct snd_soc_dai *dai)
+static int bf5xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct sport_device *sport =
 		(struct sport_device *)dai->private_data;
@@ -289,7 +290,6 @@ static int bf5xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai bf5xx_i2s_dai = {
 	.name = "bf5xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = bf5xx_i2s_probe,
 	.remove = bf5xx_i2s_remove,
 	.suspend = bf5xx_i2s_suspend,
@@ -307,13 +307,24 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 	.ops = {
 		.startup   = bf5xx_i2s_startup,
 		.shutdown  = bf5xx_i2s_shutdown,
-		.hw_params = bf5xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = bf5xx_i2s_hw_params,
 		.set_fmt = bf5xx_i2s_set_dai_fmt,
 	},
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
+static int __init bfin_i2s_init(void)
+{
+	return snd_soc_register_dai(&bf5xx_i2s_dai);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&bf5xx_i2s_dai);
+}
+module_exit(bfin_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("I2S driver for ADI Blackfin");
diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h
index fcadcc081f7..2e63dea73e9 100644
--- a/sound/soc/blackfin/bf5xx-sport.h
+++ b/sound/soc/blackfin/bf5xx-sport.h
@@ -116,7 +116,7 @@ struct sport_device {
 	void *err_data;
 	unsigned char *tx_dma_buf;
 	unsigned char *rx_dma_buf;
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT
 	dma_addr_t tx_dma_phy;
 	dma_addr_t rx_dma_phy;
 	int tx_pos;/*pcm sample count*/
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index e15f67fd776..bc0cdded711 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -44,7 +44,7 @@
 #include "bf5xx-i2s-pcm.h"
 #include "bf5xx-i2s.h"
 
-static struct snd_soc_machine bf5xx_ssm2602;
+static struct snd_soc_card bf5xx_ssm2602;
 
 static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream)
 {
@@ -92,17 +92,17 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
 	 */
 
 	/* set codec DAI configuration */
-	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+	ret = snd_soc_dai_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 |
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
 
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -135,15 +135,15 @@ static struct ssm2602_setup_data bf5xx_ssm2602_setup = {
 	.i2c_address = 0x1b,
 };
 
-static struct snd_soc_machine bf5xx_ssm2602 = {
+static struct snd_soc_card bf5xx_ssm2602 = {
 	.name = "bf5xx_ssm2602",
+	.platform = &bf5xx_i2s_soc_platform,
 	.dai_link = &bf5xx_ssm2602_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
-	.machine = &bf5xx_ssm2602,
-	.platform = &bf5xx_i2s_soc_platform,
+	.card = &bf5xx_ssm2602,
 	.codec_dev = &soc_codec_dev_ssm2602,
 	.codec_data = &bf5xx_ssm2602_setup,
 };
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 38a0e3b620a..c41289b5f58 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1,31 +1,40 @@
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
-	depends on I2C
-	select SPI
-	select SPI_MASTER
-	select SND_SOC_AD73311
-	select SND_SOC_AK4535
-	select SND_SOC_CS4270
-	select SND_SOC_SSM2602
-	select SND_SOC_TLV320AIC23
-	select SND_SOC_TLV320AIC26
-	select SND_SOC_TLV320AIC3X
-	select SND_SOC_UDA1380
-	select SND_SOC_WM8510
-	select SND_SOC_WM8580
-	select SND_SOC_WM8731
-	select SND_SOC_WM8750
-	select SND_SOC_WM8753
-	select SND_SOC_WM8900
-	select SND_SOC_WM8903
-	select SND_SOC_WM8971
-	select SND_SOC_WM8990
+	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
+	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
+	select SND_SOC_AD73311 if I2C
+	select SND_SOC_AK4535 if I2C
+	select SND_SOC_CS4270 if I2C
+	select SND_SOC_PCM3008
+	select SND_SOC_SSM2602 if I2C
+	select SND_SOC_TLV320AIC23 if I2C
+	select SND_SOC_TLV320AIC26 if SPI_MASTER
+	select SND_SOC_TLV320AIC3X if I2C
+	select SND_SOC_TWL4030 if TWL4030_CORE
+	select SND_SOC_UDA134X
+	select SND_SOC_UDA1380 if I2C
+	select SND_SOC_WM8350 if MFD_WM8350
+	select SND_SOC_WM8510 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8580 if I2C
+	select SND_SOC_WM8728 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8731 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8750 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8753 if (I2C || SPI_MASTER)
+	select SND_SOC_WM8900 if I2C
+	select SND_SOC_WM8903 if I2C
+	select SND_SOC_WM8971 if I2C
+	select SND_SOC_WM8990 if I2C
+	select SND_SOC_WM9712 if SND_SOC_AC97_BUS
+	select SND_SOC_WM9713 if SND_SOC_AC97_BUS
         help
           Normally ASoC codec drivers are only built if a machine driver which
           uses them is also built since they are only usable with a machine
           driver.  Selecting this option will allow these drivers to be built
           without an explicit machine driver for test and development purposes.
 
+	  Support for the bus types used to access the codecs to be built must
+	  be selected separately.
+
           If unsure select "N".
 
 
@@ -60,6 +69,12 @@ config SND_SOC_CS4270_VD33_ERRATA
 	bool
 	depends on SND_SOC_CS4270
 
+config SND_SOC_L3
+       tristate
+
+config SND_SOC_PCM3008
+       tristate
+
 config SND_SOC_SSM2602
 	tristate
 
@@ -75,15 +90,29 @@ config SND_SOC_TLV320AIC3X
 	tristate
 	depends on I2C
 
+config SND_SOC_TWL4030
+	tristate
+	depends on TWL4030_CORE
+
+config SND_SOC_UDA134X
+       tristate
+       select SND_SOC_L3
+
 config SND_SOC_UDA1380
         tristate
 
+config SND_SOC_WM8350
+	tristate
+
 config SND_SOC_WM8510
 	tristate
 
 config SND_SOC_WM8580
 	tristate
 
+config SND_SOC_WM8728
+	tristate
+
 config SND_SOC_WM8731
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 90f0a585fc7..c4ddc9aa2bb 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,13 +3,19 @@ snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
+snd-soc-l3-objs := l3.o
+snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-twl4030-objs := twl4030.o
+snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
+snd-soc-wm8350-objs := wm8350.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8580-objs := wm8580.o
+snd-soc-wm8728-objs := wm8728.o
 snd-soc-wm8731-objs := wm8731.o
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
@@ -25,13 +31,19 @@ obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
+obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
+obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
+obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o
+obj-$(CONFIG_SND_SOC_WM8728)	+= snd-soc-wm8728.o
 obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index bd1ebdc6c86..fb53e6511af 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -24,7 +24,8 @@
 
 #define AC97_VERSION "0.6"
 
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -42,7 +43,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
 
 struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "AC97 Playback",
 		.channels_min = 1,
@@ -113,7 +114,7 @@ static int ac97_soc_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto bus_err;
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0)
 		goto bus_err;
 	return 0;
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 1397b8e06c0..73fdbb4d4a3 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -85,6 +85,9 @@ SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0),
 SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1),
 SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1),
 
+SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1),
+
 SOC_ENUM("Capture Source", ad1980_cap_src),
 
 SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
@@ -142,10 +145,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 
 struct snd_soc_dai ad1980_dai = {
 	.name = "AC97",
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 2,
-		.channels_max = 2,
+		.channels_max = 6,
 		.rates = SNDRV_PCM_RATE_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
 	.capture = {
@@ -192,6 +196,7 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 	struct snd_soc_codec *codec;
 	int ret = 0;
 	u16 vendor_id2;
+	u16 ext_status;
 
 	printk(KERN_INFO "AD1980 SoC Audio Codec\n");
 
@@ -234,7 +239,7 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 
 	ret = ad1980_reset(codec, 0);
 	if (ret < 0) {
-		printk(KERN_ERR "AC97 link error\n");
+		printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n");
 		goto reset_err;
 	}
 
@@ -253,12 +258,19 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 				"supported\n");
 	}
 
-	ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */
-	ac97_write(codec, AC97_PCM, 0x0000);	/* unmute PCM out volume */
-	ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */
+	/* unmute captures and playbacks volume */
+	ac97_write(codec, AC97_MASTER, 0x0000);
+	ac97_write(codec, AC97_PCM, 0x0000);
+	ac97_write(codec, AC97_REC_GAIN, 0x0000);
+	ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000);
+	ac97_write(codec, AC97_SURROUND_MASTER, 0x0000);
+
+	/*power on LFE/CENTER/Surround DACs*/
+	ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
+	ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
 
 	ad1980_add_controls(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ad1980: failed to register card\n");
 		goto reset_err;
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index 37af8607b00..b09289a1e55 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -8,14 +8,10 @@
  *  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
- *    25th Sep 2008   Initial version.
  */
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
@@ -68,7 +64,7 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ad73311: failed to register card\n");
 		goto register_err;
@@ -102,6 +98,18 @@ struct snd_soc_codec_device soc_codec_dev_ad73311 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
 
+static int __init ad73311_init(void)
+{
+	return snd_soc_register_dai(&ad73311_dai);
+}
+module_init(ad73311_init);
+
+static void __exit ad73311_exit(void)
+{
+	snd_soc_unregister_dai(&ad73311_dai);
+}
+module_exit(ad73311_exit);
+
 MODULE_DESCRIPTION("ASoC ad73311 driver");
 MODULE_AUTHOR("Cliff Cai ");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 2a89b5888e1..81300d8d42c 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -339,7 +339,8 @@ static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 }
 
 static int ak4535_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -451,8 +452,6 @@ struct snd_soc_dai ak4535_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
 		.hw_params = ak4535_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt = ak4535_set_dai_fmt,
 		.digital_mute = ak4535_mute,
 		.set_sysclk = ak4535_set_dai_sysclk,
@@ -513,7 +512,7 @@ static int ak4535_init(struct snd_soc_device *socdev)
 
 	ak4535_add_controls(codec);
 	ak4535_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ak4535: failed to register card\n");
 		goto card_err;
@@ -689,6 +688,18 @@ struct snd_soc_codec_device soc_codec_dev_ak4535 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
 
+static int __init ak4535_modinit(void)
+{
+	return snd_soc_register_dai(&ak4535_dai);
+}
+module_init(ak4535_modinit);
+
+static void __exit ak4535_exit(void)
+{
+	snd_soc_unregister_dai(&ak4535_dai);
+}
+module_exit(ak4535_exit);
+
 MODULE_DESCRIPTION("Soc AK4535 driver");
 MODULE_AUTHOR("Richard Purdie");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 0bbd94501d7..f1aa0c34421 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -360,13 +360,14 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 /*
  * Program the CS4270 with the given hardware parameters.
  *
- * The .dai_ops functions are used to provide board-specific data, like
+ * The .ops functions are used to provide board-specific data, like
  * input frequencies, to this driver.  This function takes that information,
  * combines it with the hardware parameters provided, and programs the
  * hardware accordingly.
  */
 static int cs4270_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -450,6 +451,19 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
+	/* Disable automatic volume control.  It's enabled by default, and
+	 * it causes volume change commands to be delayed, sometimes until
+	 * after playback has started.
+	 */
+
+	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
+	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
+	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
+	if (ret < 0) {
+		printk(KERN_ERR "I2C write failed\n");
+		return ret;
+	}
+
 	/* Thaw and power-up the codec */
 
 	ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
@@ -697,10 +711,10 @@ static int cs4270_probe(struct platform_device *pdev)
 	if (codec->control_data) {
 		/* Initialize codec ops */
 		cs4270_dai.ops.hw_params = cs4270_hw_params;
-		cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk;
-		cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt;
+		cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
+		cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
 #ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-		cs4270_dai.dai_ops.digital_mute = cs4270_mute;
+		cs4270_dai.ops.digital_mute = cs4270_mute;
 #endif
 	} else
 		printk(KERN_INFO "cs4270: no I2C device found, "
@@ -709,7 +723,7 @@ static int cs4270_probe(struct platform_device *pdev)
 	printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n");
 #endif
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "cs4270: failed to register card\n");
 		goto error_del_driver;
@@ -760,6 +774,18 @@ struct snd_soc_codec_device soc_codec_device_cs4270 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
 
+static int __init cs4270_init(void)
+{
+	return snd_soc_register_dai(&cs4270_dai);
+}
+module_init(cs4270_init);
+
+static void __exit cs4270_exit(void)
+{
+	snd_soc_unregister_dai(&cs4270_dai);
+}
+module_exit(cs4270_exit);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c
new file mode 100644
index 00000000000..5353af58862
--- /dev/null
+++ b/sound/soc/codecs/l3.c
@@ -0,0 +1,91 @@
+/*
+ * L3 code
+ *
+ *  Copyright (C) 2008, Christian Pellegrin <chripell@evolware.org>
+ *
+ * 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.
+ *
+ *
+ * based on:
+ *
+ * L3 bus algorithm module.
+ *
+ *  Copyright (C) 2001 Russell King, All Rights Reserved.
+ *
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/delay.h>
+
+#include <sound/l3.h>
+
+/*
+ * Send one byte of data to the chip.  Data is latched into the chip on
+ * the rising edge of the clock.
+ */
+static void sendbyte(struct l3_pins *adap, unsigned int byte)
+{
+	int i;
+
+	for (i = 0; i < 8; i++) {
+		adap->setclk(0);
+		udelay(adap->data_hold);
+		adap->setdat(byte & 1);
+		udelay(adap->data_setup);
+		adap->setclk(1);
+		udelay(adap->clock_high);
+		byte >>= 1;
+	}
+}
+
+/*
+ * Send a set of bytes to the chip.  We need to pulse the MODE line
+ * between each byte, but never at the start nor at the end of the
+ * transfer.
+ */
+static void sendbytes(struct l3_pins *adap, const u8 *buf,
+		      int len)
+{
+	int i;
+
+	for (i = 0; i < len; i++) {
+		if (i) {
+			udelay(adap->mode_hold);
+			adap->setmode(0);
+			udelay(adap->mode);
+		}
+		adap->setmode(1);
+		udelay(adap->mode_setup);
+		sendbyte(adap, buf[i]);
+	}
+}
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len)
+{
+	adap->setclk(1);
+	adap->setdat(1);
+	adap->setmode(1);
+	udelay(adap->mode);
+
+	adap->setmode(0);
+	udelay(adap->mode_setup);
+	sendbyte(adap, addr);
+	udelay(adap->mode_hold);
+
+	sendbytes(adap, data, len);
+
+	adap->setclk(1);
+	adap->setdat(1);
+	adap->setmode(0);
+
+	return len;
+}
+EXPORT_SYMBOL_GPL(l3_write);
+
+MODULE_DESCRIPTION("L3 bit-banging driver");
+MODULE_AUTHOR("Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
new file mode 100644
index 00000000000..9a3e67e5319
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.c
@@ -0,0 +1,212 @@
+/*
+ * ALSA Soc PCM3008 codec support
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on AC97 Soc codec, original copyright follow:
+ * Copyright 2005 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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ * Generic PCM3008 support.
+ */
+
+#include <linux/init.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "pcm3008.h"
+
+#define PCM3008_VERSION "0.2"
+
+#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |	\
+		       SNDRV_PCM_RATE_48000)
+
+struct snd_soc_dai pcm3008_dai = {
+	.name = "PCM3008 HiFi",
+	.playback = {
+		.stream_name = "PCM3008 Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = PCM3008_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "PCM3008 Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = PCM3008_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+};
+EXPORT_SYMBOL_GPL(pcm3008_dai);
+
+static void pcm3008_gpio_free(struct pcm3008_setup_data *setup)
+{
+	gpio_free(setup->dem0_pin);
+	gpio_free(setup->dem1_pin);
+	gpio_free(setup->pdad_pin);
+	gpio_free(setup->pdda_pin);
+}
+
+static int pcm3008_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+	int ret = 0;
+
+	printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->codec)
+		return -ENOMEM;
+
+	codec = socdev->codec;
+	mutex_init(&codec->mutex);
+
+	codec->name = "PCM3008";
+	codec->owner = THIS_MODULE;
+	codec->dai = &pcm3008_dai;
+	codec->num_dai = 1;
+	codec->write = NULL;
+	codec->read = NULL;
+	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) {
+		printk(KERN_ERR "pcm3008: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* Register Card. */
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "pcm3008: failed to register card\n");
+		goto card_err;
+	}
+
+	/* DEM1  DEM0  DE-EMPHASIS_MODE
+	 * Low   Low   De-emphasis 44.1 kHz ON
+	 * Low   High  De-emphasis OFF
+	 * High  Low   De-emphasis 48 kHz ON
+	 * High  High  De-emphasis 32 kHz ON
+	 */
+
+	/* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */
+	ret = gpio_request(setup->dem0_pin, "codec_dem0");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->dem0_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */
+	ret = gpio_request(setup->dem1_pin, "codec_dem1");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->dem1_pin, 0);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure PDAD GPIO. */
+	ret = gpio_request(setup->pdad_pin, "codec_pdad");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->pdad_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	/* Configure PDDA GPIO. */
+	ret = gpio_request(setup->pdda_pin, "codec_pdda");
+	if (ret == 0)
+		ret = gpio_direction_output(setup->pdda_pin, 1);
+	if (ret != 0)
+		goto gpio_err;
+
+	return ret;
+
+gpio_err:
+	pcm3008_gpio_free(setup);
+card_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	kfree(socdev->codec);
+
+	return ret;
+}
+
+static int pcm3008_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	if (!codec)
+		return 0;
+
+	pcm3008_gpio_free(setup);
+	snd_soc_free_pcms(socdev);
+	kfree(socdev->codec);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	gpio_set_value(setup->pdad_pin, 0);
+	gpio_set_value(setup->pdda_pin, 0);
+
+	return 0;
+}
+
+static int pcm3008_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct pcm3008_setup_data *setup = socdev->codec_data;
+
+	gpio_set_value(setup->pdad_pin, 1);
+	gpio_set_value(setup->pdda_pin, 1);
+
+	return 0;
+}
+#else
+#define pcm3008_soc_suspend NULL
+#define pcm3008_soc_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_pcm3008 = {
+	.probe = 	pcm3008_soc_probe,
+	.remove = 	pcm3008_soc_remove,
+	.suspend =	pcm3008_soc_suspend,
+	.resume =	pcm3008_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008);
+
+static int __init pcm3008_init(void)
+{
+	return snd_soc_register_dai(&pcm3008_dai);
+}
+module_init(pcm3008_init);
+
+static void __exit pcm3008_exit(void)
+{
+	snd_soc_unregister_dai(&pcm3008_dai);
+}
+module_exit(pcm3008_exit);
+
+MODULE_DESCRIPTION("Soc PCM3008 driver");
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h
new file mode 100644
index 00000000000..d04e87d3c06
--- /dev/null
+++ b/sound/soc/codecs/pcm3008.h
@@ -0,0 +1,25 @@
+/*
+ * PCM3008 ALSA SoC Layer
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech 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 __LINUX_SND_SOC_PCM3008_H
+#define __LINUX_SND_SOC_PCM3008_H
+
+struct pcm3008_setup_data {
+	unsigned dem0_pin;
+	unsigned dem1_pin;
+	unsigned pdad_pin;
+	unsigned pdda_pin;
+};
+
+extern struct snd_soc_codec_device soc_codec_dev_pcm3008;
+extern struct snd_soc_dai pcm3008_dai;
+
+#endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 44ef0dacd56..cac37361676 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -285,16 +285,23 @@ static inline int get_coeff(int mclk, int rate)
 }
 
 static int ssm2602_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	u16 srate;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
+	struct i2c_client *i2c = codec->control_data;
 	u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
 	int i = get_coeff(ssm2602->sysclk, params_rate(params));
 
+	if (substream == ssm2602->slave_substream) {
+		dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n");
+		return 0;
+	}
+
 	/*no match is found*/
 	if (i == ARRAY_SIZE(coeff_div))
 		return -EINVAL;
@@ -324,19 +331,26 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int ssm2602_startup(struct snd_pcm_substream *substream)
+static int ssm2602_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
+	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
 
 	/* The DAI has shared clocks so if we already have a playback or
 	 * capture going then constrain this substream to match it.
+	 * TODO: the ssm2602 allows pairs of non-matching PB/REC rates
 	 */
 	if (ssm2602->master_substream) {
 		master_runtime = ssm2602->master_substream->runtime;
+		dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
+			master_runtime->sample_bits,
+			master_runtime->rate);
+
 		snd_pcm_hw_constraint_minmax(substream->runtime,
 					     SNDRV_PCM_HW_PARAM_RATE,
 					     master_runtime->rate,
@@ -354,7 +368,8 @@ static int ssm2602_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
+static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -365,14 +380,21 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void ssm2602_shutdown(struct snd_pcm_substream *substream)
+static void ssm2602_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
+	struct ssm2602_priv *ssm2602 = codec->private_data;
 	/* deactivate */
 	if (!codec->active)
 		ssm2602_write(codec, SSM2602_ACTIVE, 0);
+
+	if (ssm2602->master_substream == substream)
+		ssm2602->master_substream = ssm2602->slave_substream;
+
+	ssm2602->slave_substream = NULL;
 }
 
 static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
@@ -432,10 +454,10 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		iface |= 0x0001;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
-		iface |= 0x0003;
+		iface |= 0x0013;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		iface |= 0x0013;
+		iface |= 0x0003;
 		break;
 	default:
 		return -EINVAL;
@@ -496,6 +518,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
 		SNDRV_PCM_RATE_96000)
 
+#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
 struct snd_soc_dai ssm2602_dai = {
 	.name = "SSM2602",
 	.playback = {
@@ -503,20 +528,18 @@ struct snd_soc_dai ssm2602_dai = {
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SSM2602_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE,},
+		.formats = SSM2602_FORMATS,},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SSM2602_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S32_LE,},
+		.formats = SSM2602_FORMATS,},
 	.ops = {
 		.startup = ssm2602_startup,
 		.prepare = ssm2602_pcm_prepare,
 		.hw_params = ssm2602_hw_params,
 		.shutdown = ssm2602_shutdown,
-	},
-	.dai_ops = {
 		.digital_mute = ssm2602_mute,
 		.set_sysclk = ssm2602_set_dai_sysclk,
 		.set_fmt = ssm2602_set_dai_fmt,
@@ -601,7 +624,7 @@ static int ssm2602_init(struct snd_soc_device *socdev)
 
 	ssm2602_add_controls(codec);
 	ssm2602_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		pr_err("ssm2602: failed to register card\n");
 		goto card_err;
@@ -770,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602);
 
+static int __init ssm2602_modinit(void)
+{
+	return snd_soc_register_dai(&ssm2602_dai);
+}
+module_init(ssm2602_modinit);
+
+static void __exit ssm2602_exit(void)
+{
+	snd_soc_unregister_dai(&ssm2602_dai);
+}
+module_exit(ssm2602_exit);
+
 MODULE_DESCRIPTION("ASoC ssm2602 driver");
 MODULE_AUTHOR("Cliff Cai");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 44308dac9e1..cfdea007c4c 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -37,12 +37,6 @@
 
 #define AIC23_VERSION "0.1"
 
-struct tlv320aic23_srate_reg_info {
-	u32 sample_rate;
-	u8 control;		/* SR3, SR2, SR1, SR0 and BOSR */
-	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
-};
-
 /*
  * AIC23 register cache
  */
@@ -261,20 +255,156 @@ static const struct snd_soc_dapm_route intercon[] = {
 
 };
 
-/* tlv320aic23 related */
-static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
-	{4000, 0x06, 1},	/*  4000 */
-	{8000, 0x06, 0},	/*  8000 */
-	{16000, 0x0C, 1},	/* 16000 */
-	{22050, 0x11, 1},	/* 22050 */
-	{24000, 0x00, 1},	/* 24000 */
-	{32000, 0x0C, 0},	/* 32000 */
-	{44100, 0x11, 0},	/* 44100 */
-	{48000, 0x00, 0},	/* 48000 */
-	{88200, 0x1F, 0},	/* 88200 */
-	{96000, 0x0E, 0},	/* 96000 */
+/* AIC23 driver data */
+struct aic23 {
+	struct snd_soc_codec codec;
+	int mclk;
+	int requested_adc;
+	int requested_dac;
+};
+
+/*
+ * Common Crystals used
+ * 11.2896 Mhz /128 = *88.2k  /192 = 58.8k
+ * 12.0000 Mhz /125 = *96k    /136 = 88.235K
+ * 12.2880 Mhz /128 = *96k    /192 = 64k
+ * 16.9344 Mhz /128 = 132.3k /192 = *88.2k
+ * 18.4320 Mhz /128 = 144k   /192 = *96k
+ */
+
+/*
+ * Normal BOSR 0-256/2 = 128, 1-384/2 = 192
+ * USB BOSR 0-250/2 = 125, 1-272/2 = 136
+ */
+static const int bosr_usb_divisor_table[] = {
+	128, 125, 192, 136
+};
+#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
+#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11)        | (1<<15))
+static const unsigned short sr_valid_mask[] = {
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 0*/
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 1*/
+	LOWER_GROUP,			/* Usb, bosr - 0*/
+	UPPER_GROUP,			/* Usb, bosr - 1*/
+};
+/*
+ * Every divisor is a factor of 11*12
+ */
+#define SR_MULT (11*12)
+#define A(x) (x) ? (SR_MULT/x) : 0
+static const unsigned char sr_adc_mult_table[] = {
+	A(2), A(2), A(12), A(12),  A(0), A(0), A(3), A(1),
+	A(2), A(2), A(11), A(11),  A(0), A(0), A(0), A(1)
+};
+static const unsigned char sr_dac_mult_table[] = {
+	A(2), A(12), A(2), A(12),  A(0), A(0), A(3), A(1),
+	A(2), A(11), A(2), A(11),  A(0), A(0), A(0), A(1)
 };
 
+static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+		int dac, int dac_l, int dac_h, int need_dac)
+{
+	if ((adc >= adc_l) && (adc <= adc_h) &&
+			(dac >= dac_l) && (dac <= dac_h)) {
+		int diff_adc = need_adc - adc;
+		int diff_dac = need_dac - dac;
+		return abs(diff_adc) + abs(diff_dac);
+	}
+	return UINT_MAX;
+}
+
+static int find_rate(int mclk, u32 need_adc, u32 need_dac)
+{
+	int i, j;
+	int best_i = -1;
+	int best_j = -1;
+	int best_div = 0;
+	unsigned best_score = UINT_MAX;
+	int adc_l, adc_h, dac_l, dac_h;
+
+	need_adc *= SR_MULT;
+	need_dac *= SR_MULT;
+	/*
+	 * rates given are +/- 1/32
+	 */
+	adc_l = need_adc - (need_adc >> 5);
+	adc_h = need_adc + (need_adc >> 5);
+	dac_l = need_dac - (need_dac >> 5);
+	dac_h = need_dac + (need_dac >> 5);
+	for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
+		int base = mclk / bosr_usb_divisor_table[i];
+		int mask = sr_valid_mask[i];
+		for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
+				j++, mask >>= 1) {
+			int adc;
+			int dac;
+			int score;
+			if ((mask & 1) == 0)
+				continue;
+			adc = base * sr_adc_mult_table[j];
+			dac = base * sr_dac_mult_table[j];
+			score = get_score(adc, adc_l, adc_h, need_adc,
+					dac, dac_l, dac_h, need_dac);
+			if (best_score > score) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 0;
+			}
+			score = get_score((adc >> 1), adc_l, adc_h, need_adc,
+					(dac >> 1), dac_l, dac_h, need_dac);
+			/* prefer to have a /2 */
+			if ((score != UINT_MAX) && (best_score >= score)) {
+				best_score = score;
+				best_i = i;
+				best_j = j;
+				best_div = 1;
+			}
+		}
+	}
+	return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
+}
+
+#ifdef DEBUG
+static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
+		u32 *sample_rate_adc, u32 *sample_rate_dac)
+{
+	int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE);
+	int sr = (src >> 2) & 0x0f;
+	int val = (mclk / bosr_usb_divisor_table[src & 3]);
+	int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
+	int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
+	if (src & TLV320AIC23_CLKIN_HALF) {
+		adc >>= 1;
+		dac >>= 1;
+	}
+	*sample_rate_adc = adc;
+	*sample_rate_dac = dac;
+}
+#endif
+
+static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
+		u32 sample_rate_adc, u32 sample_rate_dac)
+{
+	/* Search for the right sample rate */
+	int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
+	if (data < 0) {
+		printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
+				__func__, sample_rate_adc, sample_rate_dac);
+		return -EINVAL;
+	}
+	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+#ifdef DEBUG
+	{
+		u32 adc, dac;
+		get_current_sample_rates(codec, mclk, &adc, &dac);
+		printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
+			adc, dac, data);
+	}
+#endif
+	return 0;
+}
+
 static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
 {
 	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
@@ -288,32 +418,36 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
-	u16 iface_reg, data;
-	u8 count = 0;
+	u16 iface_reg;
+	int ret;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+	u32 sample_rate_adc = aic23->requested_adc;
+	u32 sample_rate_dac = aic23->requested_dac;
+	u32 sample_rate = params_rate(params);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		aic23->requested_dac = sample_rate_dac = sample_rate;
+		if (!sample_rate_adc)
+			sample_rate_adc = sample_rate;
+	} else {
+		aic23->requested_adc = sample_rate_adc = sample_rate;
+		if (!sample_rate_dac)
+			sample_rate_dac = sample_rate;
+	}
+	ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
+			sample_rate_dac);
+	if (ret < 0)
+		return ret;
 
 	iface_reg =
 	    tlv320aic23_read_reg_cache(codec,
 				       TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
-
-	/* Search for the right sample rate */
-	/* Verify what happens if the rate is not supported
-	 * now it goes to 96Khz */
-	while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
-	       (count < ARRAY_SIZE(srate_reg_info))) {
-		count++;
-	}
-
-	data =  (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
-		(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
-		TLV320AIC23_USB_CLK_ON;
-
-	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
-
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
@@ -332,7 +466,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -344,17 +479,23 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	/* deactivate */
 	if (!codec->active) {
 		udelay(50);
 		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
 	}
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		aic23->requested_dac = 0;
+	else
+		aic23->requested_adc = 0;
 }
 
 static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
@@ -400,7 +541,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	case SND_SOC_DAIFMT_I2S:
 		iface_reg |= TLV320AIC23_FOR_I2S;
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
 		iface_reg |= TLV320AIC23_FOR_DSP;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
@@ -422,12 +563,9 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				      int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-
-	switch (freq) {
-	case 12000000:
-		return 0;
-	}
-	return -EINVAL;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
+	aic23->mclk = freq;
+	return 0;
 }
 
 static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
@@ -478,12 +616,10 @@ struct snd_soc_dai tlv320aic23_dai = {
 		.prepare = tlv320aic23_pcm_prepare,
 		.hw_params = tlv320aic23_hw_params,
 		.shutdown = tlv320aic23_shutdown,
-		},
-	.dai_ops = {
-		    .digital_mute = tlv320aic23_mute,
-		    .set_fmt = tlv320aic23_set_dai_fmt,
-		    .set_sysclk = tlv320aic23_set_dai_sysclk,
-		    }
+		.digital_mute = tlv320aic23_mute,
+		.set_fmt = tlv320aic23_set_dai_fmt,
+		.set_sysclk = tlv320aic23_set_dai_sysclk,
+	}
 };
 EXPORT_SYMBOL_GPL(tlv320aic23_dai);
 
@@ -584,7 +720,7 @@ static int tlv320aic23_init(struct snd_soc_device *socdev)
 
 	tlv320aic23_add_controls(codec);
 	tlv320aic23_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "tlv320aic23: failed to register card\n");
 		goto card_err;
@@ -659,14 +795,15 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
+	struct aic23 *aic23;
 	int ret = 0;
 
 	printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
 
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
+	aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL);
+	if (aic23 == NULL)
 		return -ENOMEM;
-
+	codec = &aic23->codec;
 	socdev->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
@@ -687,6 +824,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->codec;
+	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	if (codec->control_data)
 		tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -697,7 +835,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)
 	i2c_del_driver(&tlv320aic23_i2c_driver);
 #endif
 	kfree(codec->reg_cache);
-	kfree(codec);
+	kfree(aic23);
 
 	return 0;
 }
@@ -709,6 +847,18 @@ struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
 
+static int __init tlv320aic23_modinit(void)
+{
+	return snd_soc_register_dai(&tlv320aic23_dai);
+}
+module_init(tlv320aic23_modinit);
+
+static void __exit tlv320aic23_exit(void)
+{
+	snd_soc_unregister_dai(&tlv320aic23_dai);
+}
+module_exit(tlv320aic23_exit);
+
 MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
 MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index bed8a9e63dd..29f2f1a017f 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -125,7 +125,8 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg,
  * Digital Audio Interface Operations
  */
 static int aic26_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params)
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -287,8 +288,6 @@ struct snd_soc_dai aic26_dai = {
 	},
 	.ops = {
 		.hw_params = aic26_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = aic26_mute,
 		.set_sysclk = aic26_set_sysclk,
 		.set_fmt = aic26_set_fmt,
@@ -360,7 +359,7 @@ static int aic26_probe(struct platform_device *pdev)
 
 	/* CODEC is setup, we can register the card now */
 	dev_dbg(&pdev->dev, "Registering card\n");
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "aic26: failed to register card\n");
 		goto card_err;
@@ -427,7 +426,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set);
 static int aic26_spi_probe(struct spi_device *spi)
 {
 	struct aic26 *aic26;
-	int rc, i, reg;
+	int ret, i, reg;
 
 	dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n");
 
@@ -457,6 +456,14 @@ static int aic26_spi_probe(struct spi_device *spi)
 	aic26->codec.reg_cache_size = AIC26_NUM_REGS;
 	aic26->codec.reg_cache = aic26->reg_cache;
 
+	aic26_dai.dev = &spi->dev;
+	ret = snd_soc_register_dai(&aic26_dai);
+	if (ret != 0) {
+		dev_err(&spi->dev, "Failed to register DAI: %d\n", ret);
+		kfree(aic26);
+		return ret;
+	}
+
 	/* Reset the codec to power on defaults */
 	aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00);
 
@@ -475,8 +482,8 @@ static int aic26_spi_probe(struct spi_device *spi)
 
 	/* Register the sysfs files for debugging */
 	/* Create SysFS files */
-	rc = device_create_file(&spi->dev, &dev_attr_keyclick);
-	if (rc)
+	ret = device_create_file(&spi->dev, &dev_attr_keyclick);
+	if (ret)
 		dev_info(&spi->dev, "error creating sysfs files\n");
 
 #if defined(CONFIG_SND_SOC_OF_SIMPLE)
@@ -493,6 +500,7 @@ static int aic26_spi_remove(struct spi_device *spi)
 {
 	struct aic26 *aic26 = dev_get_drvdata(&spi->dev);
 
+	snd_soc_unregister_dai(&aic26_dai);
 	kfree(aic26);
 
 	return 0;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index cff276ee261..b47a749c5ea 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -253,11 +253,17 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 
 	SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
 		     DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
-		     0x01, 0),
-	SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
-		     PGAR_2_RLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+	SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
+	SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
+	SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
+		     DACR1_2_LLOPM_VOL, 0, 0x7f, 1),
+	SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
+		     0, 0x7f, 1),
+	SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
+		     0, 0x7f, 1),
+	SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
+		     LINE2R_2_LLOPM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
 		     LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
 
 	SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
@@ -272,8 +278,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 		     DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
 	SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
 		     0x01, 0),
-	SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+	SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
 		     PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
+	SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
+		     0, 0x7f, 1),
+	SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,
+		     0, 0x7f, 1),
 	SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
 		     LINE2R_2_HPROUT_VOL, 0, 0x7f, 1),
 
@@ -281,8 +291,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 		     DACR1_2_HPRCOM_VOL, 0, 0x7f, 1),
 	SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
 		     0x01, 0),
-	SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
-		     PGAR_2_HPRCOM_VOL, 0, 0x7f, 1),
+	SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
+		     0, 0x7f, 1),
+	SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL,
+		     0, 0x7f, 1),
 	SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
 		     LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1),
 
@@ -333,7 +345,8 @@ SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]);
 
 /* Left DAC_L1 Mixer */
 static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", DACL1_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0),
@@ -341,7 +354,8 @@ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = {
 
 /* Right DAC_R1 Mixer */
 static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", DACR1_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0),
@@ -350,14 +364,18 @@ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = {
 /* Left PGA Mixer */
 static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = {
 	SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1),
+	SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1),
 	SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1),
 	SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1),
+	SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1),
 };
 
 /* Right PGA Mixer */
 static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
 	SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1),
+	SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1),
 	SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1),
+	SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1),
 	SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1),
 };
 
@@ -379,34 +397,42 @@ SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]);
 
 /* Left PGA Bypass Mixer */
 static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", PGAL_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPL Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPR Switch", PGAL_2_HPROUT_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPLCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPRCOM Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0),
 };
 
 /* Right PGA Bypass Mixer */
 static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", PGAR_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPL Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPR Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPLCOM Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPRCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
 };
 
 /* Left Line2 Bypass Mixer */
 static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPLCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
 };
 
 /* Right Line2 Bypass Mixer */
 static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = {
-	SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineL Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("LineR Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0),
 	SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0),
-	SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
+	SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
 };
 
 static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
@@ -439,22 +465,26 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	/* Mono Output */
 	SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
 
-	/* Left Inputs to Left ADC */
+	/* Inputs to Left ADC */
 	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0),
 	SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0,
 			   &aic3x_left_pga_mixer_controls[0],
 			   ARRAY_SIZE(aic3x_left_pga_mixer_controls)),
 	SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_left_line1_mux_controls),
+	SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0,
+			 &aic3x_left_line1_mux_controls),
 	SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_left_line2_mux_controls),
 
-	/* Right Inputs to Right ADC */
+	/* Inputs to Right ADC */
 	SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
 			 LINE1R_2_RADC_CTRL, 2, 0),
 	SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0,
 			   &aic3x_right_pga_mixer_controls[0],
 			   ARRAY_SIZE(aic3x_right_pga_mixer_controls)),
+	SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0,
+			 &aic3x_right_line1_mux_controls),
 	SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_right_line1_mux_controls),
 	SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
@@ -531,7 +561,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left DAC Mux", "DAC_L2", "Left DAC"},
 	{"Left DAC Mux", "DAC_L3", "Left DAC"},
 
-	{"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"},
+	{"Left DAC_L1 Mixer", "LineL Switch", "Left DAC Mux"},
+	{"Left DAC_L1 Mixer", "LineR Switch", "Left DAC Mux"},
 	{"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"},
 	{"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"},
 	{"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"},
@@ -557,7 +588,8 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Right DAC Mux", "DAC_R2", "Right DAC"},
 	{"Right DAC Mux", "DAC_R3", "Right DAC"},
 
-	{"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"},
+	{"Right DAC_R1 Mixer", "LineL Switch", "Right DAC Mux"},
+	{"Right DAC_R1 Mixer", "LineR Switch", "Right DAC Mux"},
 	{"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"},
 	{"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"},
 	{"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"},
@@ -592,8 +624,10 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left Line2L Mux", "differential", "LINE2L"},
 
 	{"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
+	{"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"},
 	{"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"},
 	{"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
+	{"Left PGA Mixer", "Mic3R Switch", "MIC3R"},
 
 	{"Left ADC", NULL, "Left PGA Mixer"},
 	{"Left ADC", NULL, "GPIO1 dmic modclk"},
@@ -605,18 +639,23 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Right Line2R Mux", "single-ended", "LINE2R"},
 	{"Right Line2R Mux", "differential", "LINE2R"},
 
+	{"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"},
 	{"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"},
 	{"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"},
+	{"Right PGA Mixer", "Mic3L Switch", "MIC3L"},
 	{"Right PGA Mixer", "Mic3R Switch", "MIC3R"},
 
 	{"Right ADC", NULL, "Right PGA Mixer"},
 	{"Right ADC", NULL, "GPIO1 dmic modclk"},
 
 	/* Left PGA Bypass */
-	{"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "LineL Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "LineR Switch", "Left PGA Mixer"},
 	{"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"},
-	{"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"},
-	{"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "HPL Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "HPR Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "HPLCOM Switch", "Left PGA Mixer"},
+	{"Left PGA Bypass Mixer", "HPRCOM Switch", "Left PGA Mixer"},
 
 	{"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"},
 	{"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"},
@@ -627,10 +666,13 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left HP Out", NULL, "Left PGA Bypass Mixer"},
 
 	/* Right PGA Bypass */
-	{"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "LineL Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "LineR Switch", "Right PGA Mixer"},
 	{"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"},
-	{"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"},
-	{"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "HPL Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "HPR Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "HPLCOM Switch", "Right PGA Mixer"},
+	{"Right PGA Bypass Mixer", "HPRCOM Switch", "Right PGA Mixer"},
 
 	{"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"},
 	{"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"},
@@ -643,10 +685,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Right HP Out", NULL, "Right PGA Bypass Mixer"},
 
 	/* Left Line2 Bypass */
-	{"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"},
+	{"Left Line2 Bypass Mixer", "LineL Switch", "Left Line2L Mux"},
+	{"Left Line2 Bypass Mixer", "LineR Switch", "Left Line2L Mux"},
 	{"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"},
 	{"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"},
-	{"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"},
+	{"Left Line2 Bypass Mixer", "HPLCOM Switch", "Left Line2L Mux"},
 
 	{"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"},
 	{"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"},
@@ -657,10 +700,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left HP Out", NULL, "Left Line2 Bypass Mixer"},
 
 	/* Right Line2 Bypass */
-	{"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"},
+	{"Right Line2 Bypass Mixer", "LineL Switch", "Right Line2R Mux"},
+	{"Right Line2 Bypass Mixer", "LineR Switch", "Right Line2R Mux"},
 	{"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"},
 	{"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"},
-	{"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"},
+	{"Right Line2 Bypass Mixer", "HPRCOM Switch", "Right Line2R Mux"},
 
 	{"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"},
 	{"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"},
@@ -694,7 +738,8 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int aic3x_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params)
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -846,6 +891,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
 	u8 iface_areg, iface_breg;
+	int delay = 0;
 
 	iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
 	iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
@@ -871,6 +917,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		       SND_SOC_DAIFMT_INV_MASK)) {
 	case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
 		break;
+	case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF):
+		delay = 1;
 	case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
 		iface_breg |= (0x01 << 6);
 		break;
@@ -887,6 +935,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	/* set iface */
 	aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg);
 	aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg);
+	aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
 
 	return 0;
 }
@@ -981,14 +1030,41 @@ int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
 }
 EXPORT_SYMBOL_GPL(aic3x_get_gpio);
 
+void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
+				 int headset_debounce, int button_debounce)
+{
+	u8 val;
+
+	val = ((detect & AIC3X_HEADSET_DETECT_MASK)
+		<< AIC3X_HEADSET_DETECT_SHIFT) |
+	      ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK)
+		<< AIC3X_HEADSET_DEBOUNCE_SHIFT) |
+	      ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK)
+		<< AIC3X_BUTTON_DEBOUNCE_SHIFT);
+
+	if (detect & AIC3X_HEADSET_DETECT_MASK)
+		val |= AIC3X_HEADSET_DETECT_ENABLED;
+
+	aic3x_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val);
+}
+EXPORT_SYMBOL_GPL(aic3x_set_headset_detection);
+
 int aic3x_headset_detected(struct snd_soc_codec *codec)
 {
 	u8 val;
-	aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val);
-	return (val >> 2) & 1;
+	aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
+	return (val >> 4) & 1;
 }
 EXPORT_SYMBOL_GPL(aic3x_headset_detected);
 
+int aic3x_button_pressed(struct snd_soc_codec *codec)
+{
+	u8 val;
+	aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
+	return (val >> 5) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_button_pressed);
+
 #define AIC3X_RATES	SNDRV_PCM_RATE_8000_96000
 #define AIC3X_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
@@ -1009,8 +1085,6 @@ struct snd_soc_dai aic3x_dai = {
 		.formats = AIC3X_FORMATS,},
 	.ops = {
 		.hw_params = aic3x_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = aic3x_mute,
 		.set_sysclk = aic3x_set_dai_sysclk,
 		.set_fmt = aic3x_set_dai_fmt,
@@ -1152,7 +1226,7 @@ static int aic3x_init(struct snd_soc_device *socdev)
 
 	aic3x_add_controls(codec);
 	aic3x_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "aic3x: failed to register card\n");
 		goto card_err;
@@ -1341,6 +1415,18 @@ struct snd_soc_codec_device soc_codec_dev_aic3x = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x);
 
+static int __init aic3x_modinit(void)
+{
+	return snd_soc_register_dai(&aic3x_dai);
+}
+module_init(aic3x_modinit);
+
+static void __exit aic3x_exit(void)
+{
+	snd_soc_unregister_dai(&aic3x_dai);
+}
+module_exit(aic3x_exit);
+
 MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 00a195aa02e..ac827e578c4 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -35,11 +35,15 @@
 #define AIC3X_ASD_INTF_CTRLA		8
 /* Audio serial data interface control register B */
 #define AIC3X_ASD_INTF_CTRLB		9
+/* Audio serial data interface control register C */
+#define AIC3X_ASD_INTF_CTRLC		10
 /* Audio overflow status and PLL R value programming register */
 #define AIC3X_OVRF_STATUS_AND_PLLR_REG	11
 /* Audio codec digital filter control register */
 #define AIC3X_CODEC_DFILT_CTRL		12
-
+/* Headset/button press detection register */
+#define AIC3X_HEADSET_DETECT_CTRL_A	13
+#define AIC3X_HEADSET_DETECT_CTRL_B	14
 /* ADC PGA Gain control registers */
 #define LADC_VOL			15
 #define RADC_VOL			16
@@ -48,7 +52,9 @@
 #define MIC3LR_2_RADC_CTRL		18
 /* Line1 Input control registers */
 #define LINE1L_2_LADC_CTRL		19
+#define LINE1R_2_LADC_CTRL		21
 #define LINE1R_2_RADC_CTRL		22
+#define LINE1L_2_RADC_CTRL		24
 /* Line2 Input control registers */
 #define LINE2L_2_LADC_CTRL		20
 #define LINE2R_2_RADC_CTRL		23
@@ -79,6 +85,8 @@
 #define LINE2L_2_HPLOUT_VOL		45
 #define LINE2R_2_HPROUT_VOL		62
 #define PGAL_2_HPLOUT_VOL		46
+#define PGAL_2_HPROUT_VOL		60
+#define PGAR_2_HPLOUT_VOL		49
 #define PGAR_2_HPROUT_VOL		63
 #define DACL1_2_HPLOUT_VOL		47
 #define DACR1_2_HPROUT_VOL		64
@@ -88,6 +96,8 @@
 #define LINE2L_2_HPLCOM_VOL		52
 #define LINE2R_2_HPRCOM_VOL		69
 #define PGAL_2_HPLCOM_VOL		53
+#define PGAR_2_HPLCOM_VOL		56
+#define PGAL_2_HPRCOM_VOL		67
 #define PGAR_2_HPRCOM_VOL		70
 #define DACL1_2_HPLCOM_VOL		54
 #define DACR1_2_HPRCOM_VOL		71
@@ -103,11 +113,17 @@
 #define MONOLOPM_CTRL			79
 /* Line Output Plus/Minus control registers */
 #define LINE2L_2_LLOPM_VOL		80
+#define LINE2L_2_RLOPM_VOL		87
+#define LINE2R_2_LLOPM_VOL		83
 #define LINE2R_2_RLOPM_VOL		90
 #define PGAL_2_LLOPM_VOL		81
+#define PGAL_2_RLOPM_VOL		88
+#define PGAR_2_LLOPM_VOL		84
 #define PGAR_2_RLOPM_VOL		91
 #define DACL1_2_LLOPM_VOL		82
+#define DACL1_2_RLOPM_VOL		89
 #define DACR1_2_RLOPM_VOL		92
+#define DACR1_2_LLOPM_VOL		85
 #define LLOPM_CTRL			86
 #define RLOPM_CTRL			93
 /* GPIO/IRQ registers */
@@ -221,7 +237,49 @@ enum {
 
 void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
 int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
+
+/* headset detection / button API */
+
+/* The AIC3x supports detection of stereo headsets (GND + left + right signal)
+ * and cellular headsets (GND + speaker output + microphone input).
+ * It is recommended to enable MIC bias for this function to work properly.
+ * For more information, please refer to the datasheet. */
+enum {
+	AIC3X_HEADSET_DETECT_OFF	= 0,
+	AIC3X_HEADSET_DETECT_STEREO	= 1,
+	AIC3X_HEADSET_DETECT_CELLULAR   = 2,
+	AIC3X_HEADSET_DETECT_BOTH	= 3
+};
+
+enum {
+	AIC3X_HEADSET_DEBOUNCE_16MS	= 0,
+	AIC3X_HEADSET_DEBOUNCE_32MS	= 1,
+	AIC3X_HEADSET_DEBOUNCE_64MS	= 2,
+	AIC3X_HEADSET_DEBOUNCE_128MS	= 3,
+	AIC3X_HEADSET_DEBOUNCE_256MS	= 4,
+	AIC3X_HEADSET_DEBOUNCE_512MS	= 5
+};
+
+enum {
+	AIC3X_BUTTON_DEBOUNCE_0MS	= 0,
+	AIC3X_BUTTON_DEBOUNCE_8MS	= 1,
+	AIC3X_BUTTON_DEBOUNCE_16MS	= 2,
+	AIC3X_BUTTON_DEBOUNCE_32MS	= 3
+};
+
+#define AIC3X_HEADSET_DETECT_ENABLED	0x80
+#define AIC3X_HEADSET_DETECT_SHIFT	5
+#define AIC3X_HEADSET_DETECT_MASK	3
+#define AIC3X_HEADSET_DEBOUNCE_SHIFT	2
+#define AIC3X_HEADSET_DEBOUNCE_MASK	7
+#define AIC3X_BUTTON_DEBOUNCE_SHIFT 	0
+#define AIC3X_BUTTON_DEBOUNCE_MASK	3
+
+/* see the enums above for valid parameters to this function */
+void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
+				 int headset_debounce, int button_debounce);
 int aic3x_headset_detected(struct snd_soc_codec *codec);
+int aic3x_button_pressed(struct snd_soc_codec *codec);
 
 struct aic3x_setup_data {
 	int i2c_bus;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
new file mode 100644
index 00000000000..51848880504
--- /dev/null
+++ b/sound/soc/codecs/twl4030.c
@@ -0,0 +1,1317 @@
+/*
+ * ALSA SoC TWL4030 codec driver
+ *
+ * Author:      Steve Sakoman, <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#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 <linux/i2c/twl4030.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 <sound/tlv.h>
+
+#include "twl4030.h"
+
+/*
+ * twl4030 register cache & default register settings
+ */
+static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
+	0x00, /* this register not used		*/
+	0x93, /* REG_CODEC_MODE		(0x1)	*/
+	0xc3, /* REG_OPTION		(0x2)	*/
+	0x00, /* REG_UNKNOWN		(0x3)	*/
+	0x00, /* REG_MICBIAS_CTL	(0x4)	*/
+	0x20, /* REG_ANAMICL		(0x5)	*/
+	0x00, /* REG_ANAMICR		(0x6)	*/
+	0x00, /* REG_AVADC_CTL		(0x7)	*/
+	0x00, /* REG_ADCMICSEL		(0x8)	*/
+	0x00, /* REG_DIGMIXING		(0x9)	*/
+	0x0c, /* REG_ATXL1PGA		(0xA)	*/
+	0x0c, /* REG_ATXR1PGA		(0xB)	*/
+	0x00, /* REG_AVTXL2PGA		(0xC)	*/
+	0x00, /* REG_AVTXR2PGA		(0xD)	*/
+	0x01, /* REG_AUDIO_IF		(0xE)	*/
+	0x00, /* REG_VOICE_IF		(0xF)	*/
+	0x00, /* REG_ARXR1PGA		(0x10)	*/
+	0x00, /* REG_ARXL1PGA		(0x11)	*/
+	0x6c, /* REG_ARXR2PGA		(0x12)	*/
+	0x6c, /* REG_ARXL2PGA		(0x13)	*/
+	0x00, /* REG_VRXPGA		(0x14)	*/
+	0x00, /* REG_VSTPGA		(0x15)	*/
+	0x00, /* REG_VRX2ARXPGA		(0x16)	*/
+	0x0c, /* REG_AVDAC_CTL		(0x17)	*/
+	0x00, /* REG_ARX2VTXPGA		(0x18)	*/
+	0x00, /* REG_ARXL1_APGA_CTL	(0x19)	*/
+	0x00, /* REG_ARXR1_APGA_CTL	(0x1A)	*/
+	0x4b, /* REG_ARXL2_APGA_CTL	(0x1B)	*/
+	0x4b, /* REG_ARXR2_APGA_CTL	(0x1C)	*/
+	0x00, /* REG_ATX2ARXPGA		(0x1D)	*/
+	0x00, /* REG_BT_IF		(0x1E)	*/
+	0x00, /* REG_BTPGA		(0x1F)	*/
+	0x00, /* REG_BTSTPGA		(0x20)	*/
+	0x00, /* REG_EAR_CTL		(0x21)	*/
+	0x24, /* REG_HS_SEL		(0x22)	*/
+	0x0a, /* REG_HS_GAIN_SET	(0x23)	*/
+	0x00, /* REG_HS_POPN_SET	(0x24)	*/
+	0x00, /* REG_PREDL_CTL		(0x25)	*/
+	0x00, /* REG_PREDR_CTL		(0x26)	*/
+	0x00, /* REG_PRECKL_CTL		(0x27)	*/
+	0x00, /* REG_PRECKR_CTL		(0x28)	*/
+	0x00, /* REG_HFL_CTL		(0x29)	*/
+	0x00, /* REG_HFR_CTL		(0x2A)	*/
+	0x00, /* REG_ALC_CTL		(0x2B)	*/
+	0x00, /* REG_ALC_SET1		(0x2C)	*/
+	0x00, /* REG_ALC_SET2		(0x2D)	*/
+	0x00, /* REG_BOOST_CTL		(0x2E)	*/
+	0x00, /* REG_SOFTVOL_CTL	(0x2F)	*/
+	0x00, /* REG_DTMF_FREQSEL	(0x30)	*/
+	0x00, /* REG_DTMF_TONEXT1H	(0x31)	*/
+	0x00, /* REG_DTMF_TONEXT1L	(0x32)	*/
+	0x00, /* REG_DTMF_TONEXT2H	(0x33)	*/
+	0x00, /* REG_DTMF_TONEXT2L	(0x34)	*/
+	0x00, /* REG_DTMF_TONOFF	(0x35)	*/
+	0x00, /* REG_DTMF_WANONOFF	(0x36)	*/
+	0x00, /* REG_I2S_RX_SCRAMBLE_H	(0x37)	*/
+	0x00, /* REG_I2S_RX_SCRAMBLE_M	(0x38)	*/
+	0x00, /* REG_I2S_RX_SCRAMBLE_L	(0x39)	*/
+	0x16, /* REG_APLL_CTL		(0x3A)	*/
+	0x00, /* REG_DTMF_CTL		(0x3B)	*/
+	0x00, /* REG_DTMF_PGA_CTL2	(0x3C)	*/
+	0x00, /* REG_DTMF_PGA_CTL1	(0x3D)	*/
+	0x00, /* REG_MISC_SET_1		(0x3E)	*/
+	0x00, /* REG_PCMBTMUX		(0x3F)	*/
+	0x00, /* not used		(0x40)	*/
+	0x00, /* not used		(0x41)	*/
+	0x00, /* not used		(0x42)	*/
+	0x00, /* REG_RX_PATH_SEL	(0x43)	*/
+	0x00, /* REG_VDL_APGA_CTL	(0x44)	*/
+	0x00, /* REG_VIBRA_CTL		(0x45)	*/
+	0x00, /* REG_VIBRA_SET		(0x46)	*/
+	0x00, /* REG_VIBRA_PWM_SET	(0x47)	*/
+	0x00, /* REG_ANAMIC_GAIN	(0x48)	*/
+	0x00, /* REG_MISC_SET_2		(0x49)	*/
+};
+
+/*
+ * read twl4030 register cache
+ */
+static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u8 *cache = codec->reg_cache;
+
+	return cache[reg];
+}
+
+/*
+ * write twl4030 register cache
+ */
+static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec,
+						u8 reg, u8 value)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= TWL4030_CACHEREGNUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the twl4030 register space
+ */
+static int twl4030_write(struct snd_soc_codec *codec,
+			unsigned int reg, unsigned int value)
+{
+	twl4030_write_reg_cache(codec, reg, value);
+	return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+}
+
+static void twl4030_clear_codecpdz(struct snd_soc_codec *codec)
+{
+	u8 mode;
+
+	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
+		mode & ~TWL4030_CODECPDZ);
+
+	/* REVISIT: this delay is present in TI sample drivers */
+	/* but there seems to be no TRM requirement for it     */
+	udelay(10);
+}
+
+static void twl4030_set_codecpdz(struct snd_soc_codec *codec)
+{
+	u8 mode;
+
+	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
+		mode | TWL4030_CODECPDZ);
+
+	/* REVISIT: this delay is present in TI sample drivers */
+	/* but there seems to be no TRM requirement for it     */
+	udelay(10);
+}
+
+static void twl4030_init_chip(struct snd_soc_codec *codec)
+{
+	int i;
+
+	/* clear CODECPDZ prior to setting register defaults */
+	twl4030_clear_codecpdz(codec);
+
+	/* set all audio section registers to reasonable defaults */
+	for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
+		twl4030_write(codec, i,	twl4030_reg[i]);
+
+}
+
+/* Earpiece */
+static const char *twl4030_earpiece_texts[] =
+		{"Off", "DACL1", "DACL2", "Invalid", "DACR1"};
+
+static const struct soc_enum twl4030_earpiece_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1,
+			ARRAY_SIZE(twl4030_earpiece_texts),
+			twl4030_earpiece_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
+SOC_DAPM_ENUM("Route", twl4030_earpiece_enum);
+
+/* PreDrive Left */
+static const char *twl4030_predrivel_texts[] =
+		{"Off", "DACL1", "DACL2", "Invalid", "DACR2"};
+
+static const struct soc_enum twl4030_predrivel_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1,
+			ARRAY_SIZE(twl4030_predrivel_texts),
+			twl4030_predrivel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
+SOC_DAPM_ENUM("Route", twl4030_predrivel_enum);
+
+/* PreDrive Right */
+static const char *twl4030_predriver_texts[] =
+		{"Off", "DACR1", "DACR2", "Invalid", "DACL2"};
+
+static const struct soc_enum twl4030_predriver_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1,
+			ARRAY_SIZE(twl4030_predriver_texts),
+			twl4030_predriver_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
+SOC_DAPM_ENUM("Route", twl4030_predriver_enum);
+
+/* Headset Left */
+static const char *twl4030_hsol_texts[] =
+		{"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_hsol_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
+			ARRAY_SIZE(twl4030_hsol_texts),
+			twl4030_hsol_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
+SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+
+/* Headset Right */
+static const char *twl4030_hsor_texts[] =
+		{"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_hsor_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
+			ARRAY_SIZE(twl4030_hsor_texts),
+			twl4030_hsor_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
+SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+
+/* Carkit Left */
+static const char *twl4030_carkitl_texts[] =
+		{"Off", "DACL1", "DACL2"};
+
+static const struct soc_enum twl4030_carkitl_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
+			ARRAY_SIZE(twl4030_carkitl_texts),
+			twl4030_carkitl_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+
+/* Carkit Right */
+static const char *twl4030_carkitr_texts[] =
+		{"Off", "DACR1", "DACR2"};
+
+static const struct soc_enum twl4030_carkitr_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
+			ARRAY_SIZE(twl4030_carkitr_texts),
+			twl4030_carkitr_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
+SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+
+/* Handsfree Left */
+static const char *twl4030_handsfreel_texts[] =
+		{"Voice", "DACL1", "DACL2", "DACR2"};
+
+static const struct soc_enum twl4030_handsfreel_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
+			ARRAY_SIZE(twl4030_handsfreel_texts),
+			twl4030_handsfreel_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
+
+/* Handsfree Right */
+static const char *twl4030_handsfreer_texts[] =
+		{"Voice", "DACR1", "DACR2", "DACL2"};
+
+static const struct soc_enum twl4030_handsfreer_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
+			ARRAY_SIZE(twl4030_handsfreer_texts),
+			twl4030_handsfreer_texts);
+
+static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
+SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
+
+static int outmixer_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	int ret = 0;
+	int val;
+
+	switch (e->reg) {
+	case TWL4030_REG_PREDL_CTL:
+	case TWL4030_REG_PREDR_CTL:
+	case TWL4030_REG_EAR_CTL:
+		val = w->value >> e->shift_l;
+		if (val == 3) {
+			printk(KERN_WARNING
+			"Invalid MUX setting for register 0x%02x (%d)\n",
+			      e->reg, val);
+			ret = -1;
+		}
+		break;
+	}
+
+	return ret;
+}
+
+static int handsfree_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
+	unsigned char hs_ctl;
+
+	hs_ctl = twl4030_read_reg_cache(w->codec, e->reg);
+
+	if (hs_ctl & TWL4030_HF_CTL_REF_EN) {
+		hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
+		twl4030_write(w->codec, e->reg, hs_ctl);
+		hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
+		twl4030_write(w->codec, e->reg, hs_ctl);
+		hs_ctl |= TWL4030_HF_CTL_HB_EN;
+		twl4030_write(w->codec, e->reg, hs_ctl);
+	} else {
+		hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN
+				| TWL4030_HF_CTL_HB_EN);
+		twl4030_write(w->codec, e->reg, hs_ctl);
+	}
+
+	return 0;
+}
+
+/*
+ * Some of the gain controls in TWL (mostly those which are associated with
+ * the outputs) are implemented in an interesting way:
+ * 0x0 : Power down (mute)
+ * 0x1 : 6dB
+ * 0x2 : 0 dB
+ * 0x3 : -6 dB
+ * Inverting not going to help with these.
+ * Custom volsw and volsw_2r get/put functions to handle these gain bits.
+ */
+#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\
+			       xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw_twl4030, \
+	.put = snd_soc_put_volsw_twl4030, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .shift = shift_left, .rshift = shift_right,\
+		 .max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\
+				 xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw_2r, \
+	.get = snd_soc_get_volsw_r2_twl4030,\
+	.put = snd_soc_put_volsw_r2_twl4030, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+		 .rshift = xshift, .max = xmax, .invert = xinvert} }
+#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \
+	SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \
+			       xinvert, tlv_array)
+
+static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	unsigned int rshift = mc->rshift;
+	int max = mc->max;
+	int mask = (1 << fls(max)) - 1;
+
+	ucontrol->value.integer.value[0] =
+		(snd_soc_read(codec, reg) >> shift) & mask;
+	if (ucontrol->value.integer.value[0])
+		ucontrol->value.integer.value[0] =
+			max + 1 - ucontrol->value.integer.value[0];
+
+	if (shift != rshift) {
+		ucontrol->value.integer.value[1] =
+			(snd_soc_read(codec, reg) >> rshift) & mask;
+		if (ucontrol->value.integer.value[1])
+			ucontrol->value.integer.value[1] =
+				max + 1 - ucontrol->value.integer.value[1];
+	}
+
+	return 0;
+}
+
+static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	unsigned int rshift = mc->rshift;
+	int max = mc->max;
+	int mask = (1 << fls(max)) - 1;
+	unsigned short val, val2, val_mask;
+
+	val = (ucontrol->value.integer.value[0] & mask);
+
+	val_mask = mask << shift;
+	if (val)
+		val = max + 1 - val;
+	val = val << shift;
+	if (shift != rshift) {
+		val2 = (ucontrol->value.integer.value[1] & mask);
+		val_mask |= mask << rshift;
+		if (val2)
+			val2 = max + 1 - val2;
+		val |= val2 << rshift;
+	}
+	return snd_soc_update_bits(codec, reg, val_mask, val);
+}
+
+static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	unsigned int shift = mc->shift;
+	int max = mc->max;
+	int mask = (1<<fls(max))-1;
+
+	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 (ucontrol->value.integer.value[0])
+		ucontrol->value.integer.value[0] =
+			max + 1 - ucontrol->value.integer.value[0];
+	if (ucontrol->value.integer.value[1])
+		ucontrol->value.integer.value[1] =
+			max + 1 - ucontrol->value.integer.value[1];
+
+	return 0;
+}
+
+static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	unsigned int shift = mc->shift;
+	int max = mc->max;
+	int mask = (1 << fls(max)) - 1;
+	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 (val)
+		val = max + 1 - val;
+	if (val2)
+		val2 = max + 1 - val2;
+
+	val = val << shift;
+	val2 = val2 << shift;
+
+	err = snd_soc_update_bits(codec, reg, val_mask, val);
+	if (err < 0)
+		return err;
+
+	err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+	return err;
+}
+
+static int twl4030_get_left_input(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	int result = 0;
+
+	/* one bit must be set a time */
+	reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN
+			| TWL4030_MAINMIC_EN;
+	if (reg != 0) {
+		result++;
+		while ((reg & 1) == 0) {
+			result++;
+			reg >>= 1;
+		}
+	}
+
+	ucontrol->value.integer.value[0] = result;
+	return 0;
+}
+
+static int twl4030_put_left_input(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	int value = ucontrol->value.integer.value[0];
+	u8 anamicl, micbias, avadc_ctl;
+
+	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN
+			| TWL4030_MAINMIC_EN);
+	micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL);
+	micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN);
+	avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL);
+
+	switch (value) {
+	case 1:
+		anamicl |= TWL4030_MAINMIC_EN;
+		micbias |= TWL4030_MICBIAS1_EN;
+		break;
+	case 2:
+		anamicl |= TWL4030_HSMIC_EN;
+		micbias |= TWL4030_HSMICBIAS_EN;
+		break;
+	case 3:
+		anamicl |= TWL4030_AUXL_EN;
+		break;
+	case 4:
+		anamicl |= TWL4030_CKMIC_EN;
+		break;
+	default:
+		break;
+	}
+
+	/* If some input is selected, enable amp and ADC */
+	if (value != 0) {
+		anamicl |= TWL4030_MICAMPL_EN;
+		avadc_ctl |= TWL4030_ADCL_EN;
+	} else {
+		anamicl &= ~TWL4030_MICAMPL_EN;
+		avadc_ctl &= ~TWL4030_ADCL_EN;
+	}
+
+	twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl);
+	twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias);
+	twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl);
+
+	return 1;
+}
+
+static int twl4030_get_right_input(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR);
+	int value = 0;
+
+	reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN;
+	switch (reg) {
+	case TWL4030_SUBMIC_EN:
+		value = 1;
+		break;
+	case TWL4030_AUXR_EN:
+		value = 2;
+		break;
+	default:
+		break;
+	}
+
+	ucontrol->value.integer.value[0] = value;
+	return 0;
+}
+
+static int twl4030_put_right_input(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = kcontrol->private_data;
+	int value = ucontrol->value.integer.value[0];
+	u8 anamicr, micbias, avadc_ctl;
+
+	anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR);
+	anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN);
+	micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL);
+	micbias &= ~TWL4030_MICBIAS2_EN;
+	avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL);
+
+	switch (value) {
+	case 1:
+		anamicr |= TWL4030_SUBMIC_EN;
+		micbias |= TWL4030_MICBIAS2_EN;
+		break;
+	case 2:
+		anamicr |= TWL4030_AUXR_EN;
+		break;
+	default:
+		break;
+	}
+
+	if (value != 0) {
+		anamicr |= TWL4030_MICAMPR_EN;
+		avadc_ctl |= TWL4030_ADCR_EN;
+	} else {
+		anamicr &= ~TWL4030_MICAMPR_EN;
+		avadc_ctl &= ~TWL4030_ADCR_EN;
+	}
+
+	twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr);
+	twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias);
+	twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl);
+
+	return 1;
+}
+
+static const char *twl4030_left_in_sel[] = {
+	"None",
+	"Main Mic",
+	"Headset Mic",
+	"Line In",
+	"Carkit Mic",
+};
+
+static const char *twl4030_right_in_sel[] = {
+	"None",
+	"Sub Mic",
+	"Line In",
+};
+
+static const struct soc_enum twl4030_left_input_mux =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel),
+		twl4030_left_in_sel);
+
+static const struct soc_enum twl4030_right_input_mux =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel),
+		twl4030_right_in_sel);
+
+/*
+ * FGAIN volume control:
+ * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
+ */
+static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1);
+
+/*
+ * CGAIN volume control:
+ * 0 dB to 12 dB in 6 dB steps
+ * value 2 and 3 means 12 dB
+ */
+static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
+
+/*
+ * Analog playback gain
+ * -24 dB to 12 dB in 2 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
+
+/*
+ * Gain controls tied to outputs
+ * -6 dB to 6 dB in 6 dB steps (mute instead of -12)
+ */
+static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
+
+/*
+ * Capture gain after the ADCs
+ * from 0 dB to 31 dB in 1 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
+
+/*
+ * Gain control for input amplifiers
+ * 0 dB to 30 dB in 6 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new twl4030_snd_controls[] = {
+	/* Common playback gain controls */
+	SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
+		TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
+		0, 0x3f, 0, digital_fine_tlv),
+	SOC_DOUBLE_R_TLV("DAC2 Digital Fine Playback Volume",
+		TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
+		0, 0x3f, 0, digital_fine_tlv),
+
+	SOC_DOUBLE_R_TLV("DAC1 Digital Coarse Playback Volume",
+		TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
+		6, 0x2, 0, digital_coarse_tlv),
+	SOC_DOUBLE_R_TLV("DAC2 Digital Coarse Playback Volume",
+		TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA,
+		6, 0x2, 0, digital_coarse_tlv),
+
+	SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume",
+		TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+		3, 0x12, 1, analog_tlv),
+	SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume",
+		TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+		3, 0x12, 1, analog_tlv),
+	SOC_DOUBLE_R("DAC1 Analog Playback Switch",
+		TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL,
+		1, 1, 0),
+	SOC_DOUBLE_R("DAC2 Analog Playback Switch",
+		TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
+		1, 1, 0),
+
+	/* Separate output gain controls */
+	SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
+		TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
+		4, 3, 0, output_tvl),
+
+	SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume",
+		TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl),
+
+	SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume",
+		TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL,
+		4, 3, 0, output_tvl),
+
+	SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
+		TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+
+	/* Common capture gain controls */
+	SOC_DOUBLE_R_TLV("Capture Volume",
+		TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA,
+		0, 0x1f, 0, digital_capture_tlv),
+
+	SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN,
+		0, 3, 5, 0, input_gain_tlv),
+
+	/* Input source controls */
+	SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux,
+		twl4030_get_left_input, twl4030_put_left_input),
+	SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux,
+		twl4030_get_right_input, twl4030_put_right_input),
+};
+
+/* add non dapm controls */
+static int twl4030_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&twl4030_snd_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("INL"),
+	SND_SOC_DAPM_INPUT("INR"),
+
+	SND_SOC_DAPM_OUTPUT("OUTL"),
+	SND_SOC_DAPM_OUTPUT("OUTR"),
+	SND_SOC_DAPM_OUTPUT("EARPIECE"),
+	SND_SOC_DAPM_OUTPUT("PREDRIVEL"),
+	SND_SOC_DAPM_OUTPUT("PREDRIVER"),
+	SND_SOC_DAPM_OUTPUT("HSOL"),
+	SND_SOC_DAPM_OUTPUT("HSOR"),
+	SND_SOC_DAPM_OUTPUT("CARKITL"),
+	SND_SOC_DAPM_OUTPUT("CARKITR"),
+	SND_SOC_DAPM_OUTPUT("HFL"),
+	SND_SOC_DAPM_OUTPUT("HFR"),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
+			TWL4030_REG_AVDAC_CTL, 0, 0),
+	SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
+			TWL4030_REG_AVDAC_CTL, 1, 0),
+	SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
+			TWL4030_REG_AVDAC_CTL, 2, 0),
+	SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
+			TWL4030_REG_AVDAC_CTL, 3, 0),
+
+	/* Analog PGAs */
+	SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
+			0, 0, NULL, 0),
+
+	/* Output MUX controls */
+	/* Earpiece */
+	SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_earpiece_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+	/* PreDrivL/R */
+	SND_SOC_DAPM_MUX_E("PredriveL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_predrivel_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+	SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_predriver_control, outmixer_event,
+		SND_SOC_DAPM_PRE_REG),
+	/* HeadsetL/R */
+	SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsol_control),
+	SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsor_control),
+	/* CarkitL/R */
+	SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_carkitl_control),
+	SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_carkitr_control),
+	/* HandsfreeL/R */
+	SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
+		&twl4030_dapm_handsfreel_control, handsfree_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
+		&twl4030_dapm_handsfreer_control, handsfree_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+
+	SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"ARXL1_APGA", NULL, "DAC Left1"},
+	{"ARXR1_APGA", NULL, "DAC Right1"},
+	{"ARXL2_APGA", NULL, "DAC Left2"},
+	{"ARXR2_APGA", NULL, "DAC Right2"},
+
+	/* Internal playback routings */
+	/* Earpiece */
+	{"Earpiece Mux", "DACL1", "ARXL1_APGA"},
+	{"Earpiece Mux", "DACL2", "ARXL2_APGA"},
+	{"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+	/* PreDrivL */
+	{"PredriveL Mux", "DACL1", "ARXL1_APGA"},
+	{"PredriveL Mux", "DACL2", "ARXL2_APGA"},
+	{"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+	/* PreDrivR */
+	{"PredriveR Mux", "DACR1", "ARXR1_APGA"},
+	{"PredriveR Mux", "DACR2", "ARXR2_APGA"},
+	{"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+	/* HeadsetL */
+	{"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
+	{"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+	/* HeadsetR */
+	{"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
+	{"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+	/* CarkitL */
+	{"CarkitL Mux", "DACL1", "ARXL1_APGA"},
+	{"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+	/* CarkitR */
+	{"CarkitR Mux", "DACR1", "ARXR1_APGA"},
+	{"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+	/* HandsfreeL */
+	{"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
+	{"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
+	{"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+	/* HandsfreeR */
+	{"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
+	{"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
+	{"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
+
+	/* outputs */
+	{"OUTL", NULL, "ARXL2_APGA"},
+	{"OUTR", NULL, "ARXR2_APGA"},
+	{"EARPIECE", NULL, "Earpiece Mux"},
+	{"PREDRIVEL", NULL, "PredriveL Mux"},
+	{"PREDRIVER", NULL, "PredriveR Mux"},
+	{"HSOL", NULL, "HeadsetL Mux"},
+	{"HSOR", NULL, "HeadsetR Mux"},
+	{"CARKITL", NULL, "CarkitL Mux"},
+	{"CARKITR", NULL, "CarkitR Mux"},
+	{"HFL", NULL, "HandsfreeL Mux"},
+	{"HFR", NULL, "HandsfreeR Mux"},
+
+	/* inputs */
+	{"ADCL", NULL, "INL"},
+	{"ADCR", NULL, "INR"},
+};
+
+static int twl4030_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets,
+				 ARRAY_SIZE(twl4030_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static void twl4030_power_up(struct snd_soc_codec *codec)
+{
+	u8 anamicl, regmisc1, byte, popn;
+	int i = 0;
+
+	/* set CODECPDZ to turn on codec */
+	twl4030_set_codecpdz(codec);
+
+	/* initiate offset cancellation */
+	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	twl4030_write(codec, TWL4030_REG_ANAMICL,
+		anamicl | TWL4030_CNCL_OFFSET_START);
+
+	/* wait for offset cancellation to complete */
+	do {
+		/* this takes a little while, so don't slam i2c */
+		udelay(2000);
+		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+				    TWL4030_REG_ANAMICL);
+	} while ((i++ < 100) &&
+		 ((byte & TWL4030_CNCL_OFFSET_START) ==
+		  TWL4030_CNCL_OFFSET_START));
+
+	/* anti-pop when changing analog gain */
+	regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+		regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
+
+	/* toggle CODECPDZ as per TRM */
+	twl4030_clear_codecpdz(codec);
+	twl4030_set_codecpdz(codec);
+
+	/* program anti-pop with bias ramp delay */
+	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+	popn &= TWL4030_RAMP_DELAY;
+	popn |=	TWL4030_RAMP_DELAY_645MS;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+	popn |=	TWL4030_VMID_EN;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+	/* enable anti-pop ramp */
+	popn |= TWL4030_RAMP_EN;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+}
+
+static void twl4030_power_down(struct snd_soc_codec *codec)
+{
+	u8 popn;
+
+	/* disable anti-pop ramp */
+	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+	popn &= ~TWL4030_RAMP_EN;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+	/* disable bias out */
+	popn &= ~TWL4030_VMID_EN;
+	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
+
+	/* power down */
+	twl4030_clear_codecpdz(codec);
+}
+
+static int twl4030_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		twl4030_power_up(codec);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		/* TODO: develop a twl4030_prepare function */
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* TODO: develop a twl4030_standby function */
+		twl4030_power_down(codec);
+		break;
+	case SND_SOC_BIAS_OFF:
+		twl4030_power_down(codec);
+		break;
+	}
+	codec->bias_level = level;
+
+	return 0;
+}
+
+static int twl4030_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u8 mode, old_mode, format, old_format;
+
+
+	/* bit rate */
+	old_mode = twl4030_read_reg_cache(codec,
+			TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
+	mode = old_mode & ~TWL4030_APLL_RATE;
+
+	switch (params_rate(params)) {
+	case 8000:
+		mode |= TWL4030_APLL_RATE_8000;
+		break;
+	case 11025:
+		mode |= TWL4030_APLL_RATE_11025;
+		break;
+	case 12000:
+		mode |= TWL4030_APLL_RATE_12000;
+		break;
+	case 16000:
+		mode |= TWL4030_APLL_RATE_16000;
+		break;
+	case 22050:
+		mode |= TWL4030_APLL_RATE_22050;
+		break;
+	case 24000:
+		mode |= TWL4030_APLL_RATE_24000;
+		break;
+	case 32000:
+		mode |= TWL4030_APLL_RATE_32000;
+		break;
+	case 44100:
+		mode |= TWL4030_APLL_RATE_44100;
+		break;
+	case 48000:
+		mode |= TWL4030_APLL_RATE_48000;
+		break;
+	default:
+		printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	if (mode != old_mode) {
+		/* change rate and set CODECPDZ */
+		twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+		twl4030_set_codecpdz(codec);
+	}
+
+	/* sample size */
+	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+	format = old_format;
+	format &= ~TWL4030_DATA_WIDTH;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		format |= TWL4030_DATA_WIDTH_16S_16W;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		format |= TWL4030_DATA_WIDTH_32S_24W;
+		break;
+	default:
+		printk(KERN_ERR "TWL4030 hw params: unknown format %d\n",
+			params_format(params));
+		return -EINVAL;
+	}
+
+	if (format != old_format) {
+
+		/* clear CODECPDZ before changing format (codec requirement) */
+		twl4030_clear_codecpdz(codec);
+
+		/* change format */
+		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+
+		/* set CODECPDZ afterwards */
+		twl4030_set_codecpdz(codec);
+	}
+	return 0;
+}
+
+static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 infreq;
+
+	switch (freq) {
+	case 19200000:
+		infreq = TWL4030_APLL_INFREQ_19200KHZ;
+		break;
+	case 26000000:
+		infreq = TWL4030_APLL_INFREQ_26000KHZ;
+		break;
+	case 38400000:
+		infreq = TWL4030_APLL_INFREQ_38400KHZ;
+		break;
+	default:
+		printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n",
+			freq);
+		return -EINVAL;
+	}
+
+	infreq |= TWL4030_APLL_EN;
+	twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+
+	return 0;
+}
+
+static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			     unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 old_format, format;
+
+	/* get format */
+	old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+	format = old_format;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		format &= ~(TWL4030_AIF_SLAVE_EN);
+		format &= ~(TWL4030_CLK256FS_EN);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		format |= TWL4030_AIF_SLAVE_EN;
+		format |= TWL4030_CLK256FS_EN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	format &= ~TWL4030_AIF_FORMAT;
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		format |= TWL4030_AIF_FORMAT_CODEC;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (format != old_format) {
+
+		/* clear CODECPDZ before changing format (codec requirement) */
+		twl4030_clear_codecpdz(codec);
+
+		/* change format */
+		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+
+		/* set CODECPDZ afterwards */
+		twl4030_set_codecpdz(codec);
+	}
+
+	return 0;
+}
+
+#define TWL4030_RATES	 (SNDRV_PCM_RATE_8000_48000)
+#define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
+
+struct snd_soc_dai twl4030_dai = {
+	.name = "twl4030",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = TWL4030_RATES,
+		.formats = TWL4030_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = TWL4030_RATES,
+		.formats = TWL4030_FORMATS,},
+	.ops = {
+		.hw_params = twl4030_hw_params,
+		.set_sysclk = twl4030_set_dai_sysclk,
+		.set_fmt = twl4030_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL_GPL(twl4030_dai);
+
+static int twl4030_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;
+
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int twl4030_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	twl4030_set_bias_level(codec, codec->suspend_bias_level);
+	return 0;
+}
+
+/*
+ * initialize the driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+
+static int twl4030_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	printk(KERN_INFO "TWL4030 Audio Codec init \n");
+
+	codec->name = "twl4030";
+	codec->owner = THIS_MODULE;
+	codec->read = twl4030_read_reg_cache;
+	codec->write = twl4030_write;
+	codec->set_bias_level = twl4030_set_bias_level;
+	codec->dai = &twl4030_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = sizeof(twl4030_reg);
+	codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
+					GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "twl4030: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	twl4030_init_chip(codec);
+
+	/* power on device */
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	twl4030_add_controls(codec);
+	twl4030_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "twl4030: 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 *twl4030_socdev;
+
+static int twl4030_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	twl4030_socdev = socdev;
+	twl4030_init(socdev);
+
+	return 0;
+}
+
+static int twl4030_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	printk(KERN_INFO "TWL4030 Audio Codec remove\n");
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_twl4030 = {
+	.probe = twl4030_probe,
+	.remove = twl4030_remove,
+	.suspend = twl4030_suspend,
+	.resume = twl4030_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
+
+static int __init twl4030_modinit(void)
+{
+	return snd_soc_register_dai(&twl4030_dai);
+}
+module_init(twl4030_modinit);
+
+static void __exit twl4030_exit(void)
+{
+	snd_soc_unregister_dai(&twl4030_dai);
+}
+module_exit(twl4030_exit);
+
+MODULE_DESCRIPTION("ASoC TWL4030 codec driver");
+MODULE_AUTHOR("Steve Sakoman");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
new file mode 100644
index 00000000000..54615c76802
--- /dev/null
+++ b/sound/soc/codecs/twl4030.h
@@ -0,0 +1,219 @@
+/*
+ * ALSA SoC TWL4030 codec driver
+ *
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __TWL4030_AUDIO_H__
+#define __TWL4030_AUDIO_H__
+
+#define TWL4030_REG_CODEC_MODE		0x1
+#define TWL4030_REG_OPTION		0x2
+#define TWL4030_REG_UNKNOWN		0x3
+#define TWL4030_REG_MICBIAS_CTL		0x4
+#define TWL4030_REG_ANAMICL		0x5
+#define TWL4030_REG_ANAMICR		0x6
+#define TWL4030_REG_AVADC_CTL		0x7
+#define TWL4030_REG_ADCMICSEL		0x8
+#define TWL4030_REG_DIGMIXING		0x9
+#define TWL4030_REG_ATXL1PGA		0xA
+#define TWL4030_REG_ATXR1PGA		0xB
+#define TWL4030_REG_AVTXL2PGA		0xC
+#define TWL4030_REG_AVTXR2PGA		0xD
+#define TWL4030_REG_AUDIO_IF		0xE
+#define TWL4030_REG_VOICE_IF		0xF
+#define TWL4030_REG_ARXR1PGA		0x10
+#define TWL4030_REG_ARXL1PGA		0x11
+#define TWL4030_REG_ARXR2PGA		0x12
+#define TWL4030_REG_ARXL2PGA		0x13
+#define TWL4030_REG_VRXPGA		0x14
+#define TWL4030_REG_VSTPGA		0x15
+#define TWL4030_REG_VRX2ARXPGA		0x16
+#define TWL4030_REG_AVDAC_CTL		0x17
+#define TWL4030_REG_ARX2VTXPGA		0x18
+#define TWL4030_REG_ARXL1_APGA_CTL	0x19
+#define TWL4030_REG_ARXR1_APGA_CTL	0x1A
+#define TWL4030_REG_ARXL2_APGA_CTL	0x1B
+#define TWL4030_REG_ARXR2_APGA_CTL	0x1C
+#define TWL4030_REG_ATX2ARXPGA		0x1D
+#define TWL4030_REG_BT_IF		0x1E
+#define TWL4030_REG_BTPGA		0x1F
+#define TWL4030_REG_BTSTPGA		0x20
+#define TWL4030_REG_EAR_CTL		0x21
+#define TWL4030_REG_HS_SEL		0x22
+#define TWL4030_REG_HS_GAIN_SET		0x23
+#define TWL4030_REG_HS_POPN_SET		0x24
+#define TWL4030_REG_PREDL_CTL		0x25
+#define TWL4030_REG_PREDR_CTL		0x26
+#define TWL4030_REG_PRECKL_CTL		0x27
+#define TWL4030_REG_PRECKR_CTL		0x28
+#define TWL4030_REG_HFL_CTL		0x29
+#define TWL4030_REG_HFR_CTL		0x2A
+#define TWL4030_REG_ALC_CTL		0x2B
+#define TWL4030_REG_ALC_SET1		0x2C
+#define TWL4030_REG_ALC_SET2		0x2D
+#define TWL4030_REG_BOOST_CTL		0x2E
+#define TWL4030_REG_SOFTVOL_CTL		0x2F
+#define TWL4030_REG_DTMF_FREQSEL	0x30
+#define TWL4030_REG_DTMF_TONEXT1H	0x31
+#define TWL4030_REG_DTMF_TONEXT1L	0x32
+#define TWL4030_REG_DTMF_TONEXT2H	0x33
+#define TWL4030_REG_DTMF_TONEXT2L	0x34
+#define TWL4030_REG_DTMF_TONOFF		0x35
+#define TWL4030_REG_DTMF_WANONOFF	0x36
+#define TWL4030_REG_I2S_RX_SCRAMBLE_H	0x37
+#define TWL4030_REG_I2S_RX_SCRAMBLE_M	0x38
+#define TWL4030_REG_I2S_RX_SCRAMBLE_L	0x39
+#define TWL4030_REG_APLL_CTL		0x3A
+#define TWL4030_REG_DTMF_CTL		0x3B
+#define TWL4030_REG_DTMF_PGA_CTL2	0x3C
+#define TWL4030_REG_DTMF_PGA_CTL1	0x3D
+#define TWL4030_REG_MISC_SET_1		0x3E
+#define TWL4030_REG_PCMBTMUX		0x3F
+#define TWL4030_REG_RX_PATH_SEL		0x43
+#define TWL4030_REG_VDL_APGA_CTL	0x44
+#define TWL4030_REG_VIBRA_CTL		0x45
+#define TWL4030_REG_VIBRA_SET		0x46
+#define TWL4030_REG_VIBRA_PWM_SET	0x47
+#define TWL4030_REG_ANAMIC_GAIN		0x48
+#define TWL4030_REG_MISC_SET_2		0x49
+
+#define TWL4030_CACHEREGNUM	(TWL4030_REG_MISC_SET_2 + 1)
+
+/* Bitfield Definitions */
+
+/* TWL4030_CODEC_MODE (0x01) Fields */
+
+#define TWL4030_APLL_RATE		0xF0
+#define TWL4030_APLL_RATE_8000		0x00
+#define TWL4030_APLL_RATE_11025		0x10
+#define TWL4030_APLL_RATE_12000		0x20
+#define TWL4030_APLL_RATE_16000		0x40
+#define TWL4030_APLL_RATE_22050		0x50
+#define TWL4030_APLL_RATE_24000		0x60
+#define TWL4030_APLL_RATE_32000		0x80
+#define TWL4030_APLL_RATE_44100		0x90
+#define TWL4030_APLL_RATE_48000		0xA0
+#define TWL4030_SEL_16K			0x04
+#define TWL4030_CODECPDZ		0x02
+#define TWL4030_OPT_MODE		0x01
+
+/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
+
+#define TWL4030_MICBIAS2_CTL		0x40
+#define TWL4030_MICBIAS1_CTL		0x20
+#define TWL4030_HSMICBIAS_EN		0x04
+#define TWL4030_MICBIAS2_EN		0x02
+#define TWL4030_MICBIAS1_EN		0x01
+
+/* ANAMICL (0x05) Fields */
+
+#define TWL4030_CNCL_OFFSET_START	0x80
+#define TWL4030_OFFSET_CNCL_SEL		0x60
+#define TWL4030_OFFSET_CNCL_SEL_ARX1	0x00
+#define TWL4030_OFFSET_CNCL_SEL_ARX2	0x20
+#define TWL4030_OFFSET_CNCL_SEL_VRX	0x40
+#define TWL4030_OFFSET_CNCL_SEL_ALL	0x60
+#define TWL4030_MICAMPL_EN		0x10
+#define TWL4030_CKMIC_EN		0x08
+#define TWL4030_AUXL_EN			0x04
+#define TWL4030_HSMIC_EN		0x02
+#define TWL4030_MAINMIC_EN		0x01
+
+/* ANAMICR (0x06) Fields */
+
+#define TWL4030_MICAMPR_EN		0x10
+#define TWL4030_AUXR_EN			0x04
+#define TWL4030_SUBMIC_EN		0x01
+
+/* AVADC_CTL (0x07) Fields */
+
+#define TWL4030_ADCL_EN			0x08
+#define TWL4030_AVADC_CLK_PRIORITY	0x04
+#define TWL4030_ADCR_EN			0x02
+
+/* AUDIO_IF (0x0E) Fields */
+
+#define TWL4030_AIF_SLAVE_EN		0x80
+#define TWL4030_DATA_WIDTH		0x60
+#define TWL4030_DATA_WIDTH_16S_16W	0x00
+#define TWL4030_DATA_WIDTH_32S_16W	0x40
+#define TWL4030_DATA_WIDTH_32S_24W	0x60
+#define TWL4030_AIF_FORMAT		0x18
+#define TWL4030_AIF_FORMAT_CODEC	0x00
+#define TWL4030_AIF_FORMAT_LEFT		0x08
+#define TWL4030_AIF_FORMAT_RIGHT	0x10
+#define TWL4030_AIF_FORMAT_TDM		0x18
+#define TWL4030_AIF_TRI_EN		0x04
+#define TWL4030_CLK256FS_EN		0x02
+#define TWL4030_AIF_EN			0x01
+
+/* HS_GAIN_SET (0x23) Fields */
+
+#define TWL4030_HSR_GAIN		0x0C
+#define TWL4030_HSR_GAIN_PWR_DOWN	0x00
+#define TWL4030_HSR_GAIN_PLUS_6DB	0x04
+#define TWL4030_HSR_GAIN_0DB		0x08
+#define TWL4030_HSR_GAIN_MINUS_6DB	0x0C
+#define TWL4030_HSL_GAIN		0x03
+#define TWL4030_HSL_GAIN_PWR_DOWN	0x00
+#define TWL4030_HSL_GAIN_PLUS_6DB	0x01
+#define TWL4030_HSL_GAIN_0DB		0x02
+#define TWL4030_HSL_GAIN_MINUS_6DB	0x03
+
+/* HS_POPN_SET (0x24) Fields */
+
+#define TWL4030_VMID_EN			0x40
+#define	TWL4030_EXTMUTE			0x20
+#define TWL4030_RAMP_DELAY		0x1C
+#define TWL4030_RAMP_DELAY_20MS		0x00
+#define TWL4030_RAMP_DELAY_40MS		0x04
+#define TWL4030_RAMP_DELAY_81MS		0x08
+#define TWL4030_RAMP_DELAY_161MS	0x0C
+#define TWL4030_RAMP_DELAY_323MS	0x10
+#define TWL4030_RAMP_DELAY_645MS	0x14
+#define TWL4030_RAMP_DELAY_1291MS	0x18
+#define TWL4030_RAMP_DELAY_2581MS	0x1C
+#define TWL4030_RAMP_EN			0x02
+
+/* HFL_CTL (0x29, 0x2A) Fields */
+#define TWL4030_HF_CTL_HB_EN		0x04
+#define TWL4030_HF_CTL_LOOP_EN		0x08
+#define TWL4030_HF_CTL_RAMP_EN		0x10
+#define TWL4030_HF_CTL_REF_EN		0x20
+
+/* APLL_CTL (0x3A) Fields */
+
+#define TWL4030_APLL_EN			0x10
+#define TWL4030_APLL_INFREQ		0x0F
+#define TWL4030_APLL_INFREQ_19200KHZ	0x05
+#define TWL4030_APLL_INFREQ_26000KHZ	0x06
+#define TWL4030_APLL_INFREQ_38400KHZ	0x0F
+
+/* REG_MISC_SET_1 (0x3E) Fields */
+
+#define TWL4030_CLK64_EN		0x80
+#define TWL4030_SCRAMBLE_EN		0x40
+#define TWL4030_FMLOOP_EN		0x20
+#define TWL4030_SMOOTH_ANAVOL_EN	0x02
+#define TWL4030_DIGMIC_LR_SWAP_EN	0x01
+
+extern struct snd_soc_dai twl4030_dai;
+extern struct snd_soc_codec_device soc_codec_dev_twl4030;
+
+#endif	/* End of __TWL4030_AUDIO_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
new file mode 100644
index 00000000000..a2c5064a774
--- /dev/null
+++ b/sound/soc/codecs/uda134x.c
@@ -0,0 +1,668 @@
+/*
+ * uda134x.c  --  UDA134X ALSA SoC Codec driver
+ *
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie
+ *
+ * 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/delay.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include <sound/uda134x.h>
+#include <sound/l3.h>
+
+#include "uda134x.h"
+
+
+#define POWER_OFF_ON_STANDBY 1
+/*
+  ALSA SOC usually puts the device in standby mode when it's not used
+  for sometime. If you define POWER_OFF_ON_STANDBY the driver will
+  turn off the ADC/DAC when this callback is invoked and turn it back
+  on when needed. Unfortunately this will result in a very light bump
+  (it can be audible only with good earphones). If this bothers you
+  just comment this line, you will have slightly higher power
+  consumption . Please note that sending the L3 command for ADC is
+  enough to make the bump, so it doesn't make difference if you
+  completely take off power from the codec.
+ */
+
+#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
+#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
+
+struct uda134x_priv {
+	int sysclk;
+	int dai_fmt;
+
+	struct snd_pcm_substream *master_substream;
+	struct snd_pcm_substream *slave_substream;
+};
+
+/* In-data addresses are hard-coded into the reg-cache values */
+static const char uda134x_reg[UDA134X_REGS_NUM] = {
+	/* Extended address registers */
+	0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
+	/* Status, data regs */
+	0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+};
+
+/*
+ * The codec has no support for reading its registers except for peak level...
+ */
+static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= UDA134X_REGS_NUM)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * Write the register cache
+ */
+static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec,
+	u8 reg, unsigned int value)
+{
+	u8 *cache = codec->reg_cache;
+
+	if (reg >= UDA134X_REGS_NUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * Write to the uda134x registers
+ *
+ */
+static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	int ret;
+	u8 addr;
+	u8 data = value;
+	struct uda134x_platform_data *pd = codec->control_data;
+
+	pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
+
+	if (reg >= UDA134X_REGS_NUM) {
+		printk(KERN_ERR "%s unkown register: reg: %d",
+		       __func__, reg);
+		return -EINVAL;
+	}
+
+	uda134x_write_reg_cache(codec, reg, value);
+
+	switch (reg) {
+	case UDA134X_STATUS0:
+	case UDA134X_STATUS1:
+		addr = UDA134X_STATUS_ADDR;
+		break;
+	case UDA134X_DATA000:
+	case UDA134X_DATA001:
+	case UDA134X_DATA010:
+		addr = UDA134X_DATA0_ADDR;
+		break;
+	case UDA134X_DATA1:
+		addr = UDA134X_DATA1_ADDR;
+		break;
+	default:
+		/* It's an extended address register */
+		addr =  (reg | UDA134X_EXTADDR_PREFIX);
+
+		ret = l3_write(&pd->l3,
+			       UDA134X_DATA0_ADDR, &addr, 1);
+		if (ret != 1)
+			return -EIO;
+
+		addr = UDA134X_DATA0_ADDR;
+		data = (value | UDA134X_EXTDATA_PREFIX);
+		break;
+	}
+
+	ret = l3_write(&pd->l3,
+		       addr, &data, 1);
+	if (ret != 1)
+		return -EIO;
+
+	return 0;
+}
+
+static inline void uda134x_reset(struct snd_soc_codec *codec)
+{
+	u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+	uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6));
+	msleep(1);
+	uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6));
+}
+
+static int uda134x_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010);
+
+	pr_debug("%s mute: %d\n", __func__, mute);
+
+	if (mute)
+		mute_reg |= (1<<2);
+	else
+		mute_reg &= ~(1<<2);
+
+	uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2));
+
+	return 0;
+}
+
+static int uda134x_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+	struct snd_pcm_runtime *master_runtime;
+
+	if (uda134x->master_substream) {
+		master_runtime = uda134x->master_substream->runtime;
+
+		pr_debug("%s constraining to %d bits at %d\n", __func__,
+			 master_runtime->sample_bits,
+			 master_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_RATE,
+					     master_runtime->rate,
+					     master_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+					     master_runtime->sample_bits,
+					     master_runtime->sample_bits);
+
+		uda134x->slave_substream = substream;
+	} else
+		uda134x->master_substream = substream;
+
+	return 0;
+}
+
+static void uda134x_shutdown(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	if (uda134x->master_substream == substream)
+		uda134x->master_substream = uda134x->slave_substream;
+
+	uda134x->slave_substream = NULL;
+}
+
+static int uda134x_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+	u8 hw_params;
+
+	if (substream == uda134x->slave_substream) {
+		pr_debug("%s ignoring hw_params for slave substream\n",
+			 __func__);
+		return 0;
+	}
+
+	hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0);
+	hw_params &= STATUS0_SYSCLK_MASK;
+	hw_params &= STATUS0_DAIFMT_MASK;
+
+	pr_debug("%s sysclk: %d, rate:%d\n", __func__,
+		 uda134x->sysclk, params_rate(params));
+
+	/* set SYSCLK / fs ratio */
+	switch (uda134x->sysclk / params_rate(params)) {
+	case 512:
+		break;
+	case 384:
+		hw_params |= (1<<4);
+		break;
+	case 256:
+		hw_params |= (1<<5);
+		break;
+	default:
+		printk(KERN_ERR "%s unsupported fs\n", __func__);
+		return -EINVAL;
+	}
+
+	pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__,
+		 uda134x->dai_fmt, params_format(params));
+
+	/* set DAI format and word length */
+	switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		switch (params_format(params)) {
+		case SNDRV_PCM_FORMAT_S16_LE:
+			hw_params |= (1<<1);
+			break;
+		case SNDRV_PCM_FORMAT_S18_3LE:
+			hw_params |= (1<<2);
+			break;
+		case SNDRV_PCM_FORMAT_S20_3LE:
+			hw_params |= ((1<<2) | (1<<1));
+			break;
+		default:
+			printk(KERN_ERR "%s unsupported format (right)\n",
+			       __func__);
+			return -EINVAL;
+		}
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		hw_params |= (1<<3);
+		break;
+	default:
+		printk(KERN_ERR "%s unsupported format\n", __func__);
+		return -EINVAL;
+	}
+
+	uda134x_write(codec, UDA134X_STATUS0, hw_params);
+
+	return 0;
+}
+
+static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+		 clk_id, freq, dir);
+
+	/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
+	   because the codec is slave. Of course limitations of the clock
+	   master (the IIS controller) apply.
+	   We'll error out on set_hw_params if it's not OK */
+	if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) {
+		uda134x->sysclk = freq;
+		return 0;
+	}
+
+	printk(KERN_ERR "%s unsupported sysclk\n", __func__);
+	return -EINVAL;
+}
+
+static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			       unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct uda134x_priv *uda134x = codec->private_data;
+
+	pr_debug("%s fmt: %08X\n", __func__, fmt);
+
+	/* codec supports only full slave mode */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) {
+		printk(KERN_ERR "%s unsupported slave mode\n", __func__);
+		return -EINVAL;
+	}
+
+	/* no support for clock inversion */
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) {
+		printk(KERN_ERR "%s unsupported clock inversion\n", __func__);
+		return -EINVAL;
+	}
+
+	/* We can't setup DAI format here as it depends on the word bit num */
+	/* so let's just store the value for later */
+	uda134x->dai_fmt = fmt;
+
+	return 0;
+}
+
+static int uda134x_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	u8 reg;
+	struct uda134x_platform_data *pd = codec->control_data;
+	int i;
+	u8 *cache = codec->reg_cache;
+
+	pr_debug("%s bias level %d\n", __func__, level);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* ADC, DAC on */
+		reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+		uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		/* power on */
+		if (pd->power) {
+			pd->power(1);
+			/* Sync reg_cache with the hardware */
+			for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++)
+				codec->write(codec, i, *cache++);
+		}
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* ADC, DAC power off */
+		reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+		uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* power off */
+		if (pd->power)
+			pd->power(0);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1",
+					    "Minimum2", "Maximum"};
+static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *uda134x_mixmode[] = {"Differential", "Analog1",
+					"Analog2", "Both"};
+
+static const struct soc_enum uda134x_mixer_enum[] = {
+SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting),
+SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph),
+SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode),
+};
+
+static const struct snd_kcontrol_new uda1341_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0),
+SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1),
+SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1),
+
+SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0),
+SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+SOC_ENUM("Input Mux", uda134x_mixer_enum[2]),
+
+SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0),
+SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1),
+SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0),
+
+SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0),
+SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0),
+SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0),
+SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0),
+SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0),
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new uda1340_snd_controls[] = {
+SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),
+
+SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0),
+SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0),
+
+SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]),
+SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
+
+SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
+};
+
+static int uda134x_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i, n;
+	const struct snd_kcontrol_new *ctrls;
+	struct uda134x_platform_data *pd = codec->control_data;
+
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1344:
+		n = ARRAY_SIZE(uda1340_snd_controls);
+		ctrls = uda1340_snd_controls;
+		break;
+	case UDA134X_UDA1341:
+		n = ARRAY_SIZE(uda1341_snd_controls);
+		ctrls = uda1341_snd_controls;
+		break;
+	default:
+		printk(KERN_ERR "%s unkown codec type: %d",
+		       __func__, pd->model);
+		return -EINVAL;
+	}
+
+	for (i = 0; i < n; i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&ctrls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+struct snd_soc_dai uda134x_dai = {
+	.name = "UDA134X",
+	/* playback capabilities */
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA134X_RATES,
+		.formats = UDA134X_FORMATS,
+	},
+	/* capture capabilities */
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA134X_RATES,
+		.formats = UDA134X_FORMATS,
+	},
+	/* pcm operations */
+	.ops = {
+		.startup = uda134x_startup,
+		.shutdown = uda134x_shutdown,
+		.hw_params = uda134x_hw_params,
+		.digital_mute = uda134x_mute,
+		.set_sysclk = uda134x_set_dai_sysclk,
+		.set_fmt = uda134x_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL(uda134x_dai);
+
+
+static int uda134x_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct uda134x_priv *uda134x;
+	void *codec_setup_data = socdev->codec_data;
+	int ret = -ENOMEM;
+	struct uda134x_platform_data *pd;
+
+	printk(KERN_INFO "UDA134X SoC Audio Codec\n");
+
+	if (!codec_setup_data) {
+		printk(KERN_ERR "UDA134X SoC codec: "
+		       "missing L3 bitbang function\n");
+		return -ENODEV;
+	}
+
+	pd = codec_setup_data;
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1341:
+	case UDA134X_UDA1344:
+		break;
+	default:
+		printk(KERN_ERR "UDA134X SoC codec: "
+		       "unsupported model %d\n",
+			pd->model);
+		return -EINVAL;
+	}
+
+	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->codec == NULL)
+		return ret;
+
+	codec = socdev->codec;
+
+	uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
+	if (uda134x == NULL)
+		goto priv_err;
+	codec->private_data = uda134x;
+
+	codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg),
+				   GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		goto reg_err;
+
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache_size = sizeof(uda134x_reg);
+	codec->reg_cache_step = 1;
+
+	codec->name = "UDA134X";
+	codec->owner = THIS_MODULE;
+	codec->dai = &uda134x_dai;
+	codec->num_dai = 1;
+	codec->read = uda134x_read_reg_cache;
+	codec->write = uda134x_write;
+#ifdef POWER_OFF_ON_STANDBY
+	codec->set_bias_level = uda134x_set_bias_level;
+#endif
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->control_data = codec_setup_data;
+
+	if (pd->power)
+		pd->power(1);
+
+	uda134x_reset(codec);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register pcms\n");
+		goto pcm_err;
+	}
+
+	ret = uda134x_add_controls(codec);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register controls\n");
+		goto pcm_err;
+	}
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "UDA134X: failed to register card\n");
+		goto card_err;
+	}
+
+	return 0;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+reg_err:
+	kfree(codec->private_data);
+priv_err:
+	kfree(codec);
+	return ret;
+}
+
+/* power down chip */
+static int uda134x_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	kfree(codec->private_data);
+	kfree(codec->reg_cache);
+	kfree(codec);
+
+	return 0;
+}
+
+#if defined(CONFIG_PM)
+static int uda134x_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;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int uda134x_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+	return 0;
+}
+#else
+#define uda134x_soc_suspend NULL
+#define uda134x_soc_resume NULL
+#endif /* CONFIG_PM */
+
+struct snd_soc_codec_device soc_codec_dev_uda134x = {
+	.probe =        uda134x_soc_probe,
+	.remove =       uda134x_soc_remove,
+	.suspend =      uda134x_soc_suspend,
+	.resume =       uda134x_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x);
+
+static int __init uda134x_init(void)
+{
+	return snd_soc_register_dai(&uda134x_dai);
+}
+module_init(uda134x_init);
+
+static void __exit uda134x_exit(void)
+{
+	snd_soc_unregister_dai(&uda134x_dai);
+}
+module_exit(uda134x_exit);
+
+MODULE_DESCRIPTION("UDA134X ALSA soc codec driver");
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h
new file mode 100644
index 00000000000..94f440490b3
--- /dev/null
+++ b/sound/soc/codecs/uda134x.h
@@ -0,0 +1,36 @@
+#ifndef _UDA134X_CODEC_H
+#define _UDA134X_CODEC_H
+
+#define UDA134X_L3ADDR	5
+#define UDA134X_DATA0_ADDR	((UDA134X_L3ADDR << 2) | 0)
+#define UDA134X_DATA1_ADDR	((UDA134X_L3ADDR << 2) | 1)
+#define UDA134X_STATUS_ADDR	((UDA134X_L3ADDR << 2) | 2)
+
+#define UDA134X_EXTADDR_PREFIX	0xC0
+#define UDA134X_EXTDATA_PREFIX	0xE0
+
+/* UDA134X registers */
+#define UDA134X_EA000	0
+#define UDA134X_EA001	1
+#define UDA134X_EA010	2
+#define UDA134X_EA011	3
+#define UDA134X_EA100	4
+#define UDA134X_EA101	5
+#define UDA134X_EA110	6
+#define UDA134X_EA111	7
+#define UDA134X_STATUS0 8
+#define UDA134X_STATUS1 9
+#define UDA134X_DATA000 10
+#define UDA134X_DATA001 11
+#define UDA134X_DATA010 12
+#define UDA134X_DATA1	13
+
+#define UDA134X_REGS_NUM 14
+
+#define STATUS0_DAIFMT_MASK (~(7<<1))
+#define STATUS0_SYSCLK_MASK (~(3<<4))
+
+extern struct snd_soc_dai uda134x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_uda134x;
+
+#endif
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index a69ee72a7af..e6bf0844fbf 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -407,7 +407,8 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * when the DAI is being clocked by the CPU DAI. It's up to the
  * machine and cpu DAI driver to do this before we are called.
  */
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
+static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -439,7 +440,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream)
 }
 
 static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -477,7 +479,8 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -560,8 +563,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.digital_mute = uda1380_mute,
 		.set_fmt = uda1380_set_dai_fmt,
 	},
@@ -579,8 +580,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.digital_mute = uda1380_mute,
 		.set_fmt = uda1380_set_dai_fmt,
 	},
@@ -598,8 +597,6 @@ struct snd_soc_dai uda1380_dai[] = {
 		.hw_params = uda1380_pcm_hw_params,
 		.shutdown = uda1380_pcm_shutdown,
 		.prepare = uda1380_pcm_prepare,
-	},
-	.dai_ops = {
 		.set_fmt = uda1380_set_dai_fmt,
 	},
 },
@@ -680,7 +677,7 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 	/* uda1380 init */
 	uda1380_add_controls(codec);
 	uda1380_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		pr_err("uda1380: failed to register card\n");
 		goto card_err;
@@ -844,6 +841,18 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
 
+static int __init uda1380_modinit(void)
+{
+	return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_init(uda1380_modinit);
+
+static void __exit uda1380_exit(void)
+{
+	snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai));
+}
+module_exit(uda1380_exit);
+
 MODULE_AUTHOR("Giorgio Padrin");
 MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
new file mode 100644
index 00000000000..e3989d406f5
--- /dev/null
+++ b/sound/soc/codecs/wm8350.c
@@ -0,0 +1,1583 @@
+/*
+ * wm8350.c -- WM8350 ALSA SoC audio driver
+ *
+ * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/wm8350/audio.h>
+#include <linux/mfd/wm8350/core.h>
+#include <linux/regulator/consumer.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 <sound/tlv.h>
+
+#include "wm8350.h"
+
+#define WM8350_OUTn_0dB 0x39
+
+#define WM8350_RAMP_NONE	0
+#define WM8350_RAMP_UP		1
+#define WM8350_RAMP_DOWN	2
+
+/* We only include the analogue supplies here; the digital supplies
+ * need to be available well before this driver can be probed.
+ */
+static const char *supply_names[] = {
+	"AVDD",
+	"HPVDD",
+};
+
+struct wm8350_output {
+	u16 active;
+	u16 left_vol;
+	u16 right_vol;
+	u16 ramp;
+	u16 mute;
+};
+
+struct wm8350_data {
+	struct snd_soc_codec codec;
+	struct wm8350_output out1;
+	struct wm8350_output out2;
+	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
+					    unsigned int reg)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350->reg_cache[reg];
+}
+
+static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
+				      unsigned int reg)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350_reg_read(wm8350, reg);
+}
+
+static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+			      unsigned int value)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350_reg_write(wm8350, reg, value);
+}
+
+/*
+ * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
+{
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_data->out1;
+	struct wm8350 *wm8350 = codec->control_data;
+	int left_complete = 0, right_complete = 0;
+	u16 reg, val;
+
+	/* left channel */
+	reg = wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME);
+	val = (reg & WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+
+	if (out1->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out1->left_vol) {
+			val++;
+			reg &= ~WM8350_OUT1L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else if (out1->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT1L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else
+		return 1;
+
+	/* right channel */
+	reg = wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME);
+	val = (reg & WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	if (out1->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out1->right_vol) {
+			val++;
+			reg &= ~WM8350_OUT1R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	} else if (out1->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT1R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	}
+
+	/* only hit the update bit if either volume has changed this step */
+	if (!left_complete || !right_complete)
+		wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, WM8350_OUT1_VU);
+
+	return left_complete & right_complete;
+}
+
+/*
+ * Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
+{
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out2 = &wm8350_data->out2;
+	struct wm8350 *wm8350 = codec->control_data;
+	int left_complete = 0, right_complete = 0;
+	u16 reg, val;
+
+	/* left channel */
+	reg = wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME);
+	val = (reg & WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	if (out2->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out2->left_vol) {
+			val++;
+			reg &= ~WM8350_OUT2L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else if (out2->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT2L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else
+		return 1;
+
+	/* right channel */
+	reg = wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME);
+	val = (reg & WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	if (out2->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out2->right_vol) {
+			val++;
+			reg &= ~WM8350_OUT2R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	} else if (out2->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT2R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	}
+
+	/* only hit the update bit if either volume has changed this step */
+	if (!left_complete || !right_complete)
+		wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, WM8350_OUT2_VU);
+
+	return left_complete & right_complete;
+}
+
+/*
+ * This work ramps both output PGAs at stream start/stop time to
+ * minimise pop associated with DAPM power switching.
+ * It's best to enable Zero Cross when ramping occurs to minimise any
+ * zipper noises.
+ */
+static void wm8350_pga_work(struct work_struct *work)
+{
+	struct snd_soc_codec *codec =
+	    container_of(work, struct snd_soc_codec, delayed_work.work);
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_data->out1,
+	    *out2 = &wm8350_data->out2;
+	int i, out1_complete, out2_complete;
+
+	/* do we need to ramp at all ? */
+	if (out1->ramp == WM8350_RAMP_NONE && out2->ramp == WM8350_RAMP_NONE)
+		return;
+
+	/* PGA volumes have 6 bits of resolution to ramp */
+	for (i = 0; i <= 63; i++) {
+		out1_complete = 1, out2_complete = 1;
+		if (out1->ramp != WM8350_RAMP_NONE)
+			out1_complete = wm8350_out1_ramp_step(codec);
+		if (out2->ramp != WM8350_RAMP_NONE)
+			out2_complete = wm8350_out2_ramp_step(codec);
+
+		/* ramp finished ? */
+		if (out1_complete && out2_complete)
+			break;
+
+		/* we need to delay longer on the up ramp */
+		if (out1->ramp == WM8350_RAMP_UP ||
+		    out2->ramp == WM8350_RAMP_UP) {
+			/* delay is longer over 0dB as increases are larger */
+			if (i >= WM8350_OUTn_0dB)
+				schedule_timeout_interruptible(msecs_to_jiffies
+							       (2));
+			else
+				schedule_timeout_interruptible(msecs_to_jiffies
+							       (1));
+		} else
+			udelay(50);	/* doesn't matter if we delay longer */
+	}
+
+	out1->ramp = WM8350_RAMP_NONE;
+	out2->ramp = WM8350_RAMP_NONE;
+}
+
+/*
+ * WM8350 Controls
+ */
+
+static int pga_event(struct snd_soc_dapm_widget *w,
+		     struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out;
+
+	switch (w->shift) {
+	case 0:
+	case 1:
+		out = &wm8350_data->out1;
+		break;
+	case 2:
+	case 3:
+		out = &wm8350_data->out2;
+		break;
+
+	default:
+		BUG();
+		return -1;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		out->ramp = WM8350_RAMP_UP;
+		out->active = 1;
+
+		if (!delayed_work_pending(&codec->delayed_work))
+			schedule_delayed_work(&codec->delayed_work,
+					      msecs_to_jiffies(1));
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		out->ramp = WM8350_RAMP_DOWN;
+		out->active = 0;
+
+		if (!delayed_work_pending(&codec->delayed_work))
+			schedule_delayed_work(&codec->delayed_work,
+					      msecs_to_jiffies(1));
+		break;
+	}
+
+	return 0;
+}
+
+static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8350_data *wm8350_priv = codec->private_data;
+	struct wm8350_output *out = NULL;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int ret;
+	unsigned int reg = mc->reg;
+	u16 val;
+
+	/* For OUT1 and OUT2 we shadow the values and only actually write
+	 * them out when active in order to ensure the amplifier comes on
+	 * as quietly as possible. */
+	switch (reg) {
+	case WM8350_LOUT1_VOLUME:
+		out = &wm8350_priv->out1;
+		break;
+	case WM8350_LOUT2_VOLUME:
+		out = &wm8350_priv->out2;
+		break;
+	default:
+		break;
+	}
+
+	if (out) {
+		out->left_vol = ucontrol->value.integer.value[0];
+		out->right_vol = ucontrol->value.integer.value[1];
+		if (!out->active)
+			return 1;
+	}
+
+	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+	if (ret < 0)
+		return ret;
+
+	/* now hit the volume update bits (always bit 8) */
+	val = wm8350_codec_read(codec, reg);
+	wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU);
+	return 1;
+}
+
+static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8350_data *wm8350_priv = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_priv->out1;
+	struct wm8350_output *out2 = &wm8350_priv->out2;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	unsigned int reg = mc->reg;
+
+	/* If these are cached registers use the cache */
+	switch (reg) {
+	case WM8350_LOUT1_VOLUME:
+		ucontrol->value.integer.value[0] = out1->left_vol;
+		ucontrol->value.integer.value[1] = out1->right_vol;
+		return 0;
+
+	case WM8350_LOUT2_VOLUME:
+		ucontrol->value.integer.value[0] = out2->left_vol;
+		ucontrol->value.integer.value[1] = out2->right_vol;
+		return 0;
+
+	default:
+		break;
+	}
+
+	return snd_soc_get_volsw_2r(kcontrol, ucontrol);
+}
+
+/* double control with volume update */
+#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+				xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+		SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+		SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw_2r, \
+	.get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+		 .rshift = xshift, .max = xmax, .invert = xinvert}, }
+
+static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
+static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
+static const char *wm8350_dacmutes[] = { "Fast", "Slow" };
+static const char *wm8350_dacfilter[] = { "Normal", "Sloping" };
+static const char *wm8350_adcfilter[] = { "None", "High Pass" };
+static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" };
+static const char *wm8350_lr[] = { "Left", "Right" };
+
+static const struct soc_enum wm8350_enum[] = {
+	SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 4, 4, wm8350_deemp),
+	SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol),
+	SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr),
+};
+
+static DECLARE_TLV_DB_LINEAR(pre_amp_tlv, -1200, 3525);
+static DECLARE_TLV_DB_LINEAR(out_pga_tlv, -5700, 600);
+static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1);
+static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1);
+static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1);
+
+static const unsigned int capture_sd_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 12, TLV_DB_SCALE_ITEM(-3600, 300, 1),
+	13, 15, TLV_DB_SCALE_ITEM(0, 0, 0),
+};
+
+static const struct snd_kcontrol_new wm8350_snd_controls[] = {
+	SOC_ENUM("Playback Deemphasis", wm8350_enum[0]),
+	SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]),
+	SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume",
+				WM8350_DAC_DIGITAL_VOLUME_L,
+				WM8350_DAC_DIGITAL_VOLUME_R,
+				0, 255, 0, dac_pcm_tlv),
+	SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
+	SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
+	SOC_ENUM("Playback PCM Filter", wm8350_enum[4]),
+	SOC_ENUM("Capture PCM Filter", wm8350_enum[5]),
+	SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]),
+	SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]),
+	SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume",
+				WM8350_ADC_DIGITAL_VOLUME_L,
+				WM8350_ADC_DIGITAL_VOLUME_R,
+				0, 255, 0, adc_pcm_tlv),
+	SOC_DOUBLE_TLV("Capture Sidetone Volume",
+		       WM8350_ADC_DIVIDER,
+		       8, 4, 15, 1, capture_sd_tlv),
+	SOC_WM8350_DOUBLE_R_TLV("Capture Volume",
+				WM8350_LEFT_INPUT_VOLUME,
+				WM8350_RIGHT_INPUT_VOLUME,
+				2, 63, 0, pre_amp_tlv),
+	SOC_DOUBLE_R("Capture ZC Switch",
+		     WM8350_LEFT_INPUT_VOLUME,
+		     WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0),
+	SOC_SINGLE_TLV("Left Input Left Sidetone Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Left Input Right Sidetone Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Left Input Bypass Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+		       9, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Left Sidetone Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       1, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Right Sidetone Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Bypass Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       13, 7, 0, out_mix_tlv),
+	SOC_SINGLE("Left Input Mixer +20dB Switch",
+		   WM8350_INPUT_MIXER_VOLUME_L, 0, 1, 0),
+	SOC_SINGLE("Right Input Mixer +20dB Switch",
+		   WM8350_INPUT_MIXER_VOLUME_R, 0, 1, 0),
+	SOC_SINGLE_TLV("Out4 Capture Volume",
+		       WM8350_INPUT_MIXER_VOLUME,
+		       1, 7, 0, out_mix_tlv),
+	SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume",
+				WM8350_LOUT1_VOLUME,
+				WM8350_ROUT1_VOLUME,
+				2, 63, 0, out_pga_tlv),
+	SOC_DOUBLE_R("Out1 Playback ZC Switch",
+		     WM8350_LOUT1_VOLUME,
+		     WM8350_ROUT1_VOLUME, 13, 1, 0),
+	SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume",
+				WM8350_LOUT2_VOLUME,
+				WM8350_ROUT2_VOLUME,
+				2, 63, 0, out_pga_tlv),
+	SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME,
+		     WM8350_ROUT2_VOLUME, 13, 1, 0),
+	SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0),
+	SOC_SINGLE_TLV("Out2 Beep Volume", WM8350_BEEP_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+
+	SOC_DOUBLE_R("Out1 Playback Switch",
+		     WM8350_LOUT1_VOLUME,
+		     WM8350_ROUT1_VOLUME,
+		     14, 1, 1),
+	SOC_DOUBLE_R("Out2 Playback Switch",
+		     WM8350_LOUT2_VOLUME,
+		     WM8350_ROUT2_VOLUME,
+		     14, 1, 1),
+};
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Playback Mixer */
+static const struct snd_kcontrol_new wm8350_left_play_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch",
+			WM8350_LEFT_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch",
+			WM8350_LEFT_MIXER_CONTROL, 2, 1, 0),
+	SOC_DAPM_SINGLE("Right Playback Switch",
+			WM8350_LEFT_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Left Sidetone Switch",
+			WM8350_LEFT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("Right Sidetone Switch",
+			WM8350_LEFT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Right Playback Mixer */
+static const struct snd_kcontrol_new wm8350_right_play_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 3, 1, 0),
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Sidetone Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("Right Sidetone Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Out4 Mixer */
+static const struct snd_kcontrol_new wm8350_out4_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Right Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Right Capture Switch",
+			WM8350_OUT4_MIXER_CONTROL, 9, 1, 0),
+	SOC_DAPM_SINGLE("Out3 Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 2, 1, 0),
+	SOC_DAPM_SINGLE("Right Mixer Switch",
+			WM8350_OUT4_MIXER_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("Left Mixer Switch",
+			WM8350_OUT4_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Out3 Mixer */
+static const struct snd_kcontrol_new wm8350_out3_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_OUT3_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Capture Switch",
+			WM8350_OUT3_MIXER_CONTROL, 8, 1, 0),
+	SOC_DAPM_SINGLE("Out4 Playback Switch",
+			WM8350_OUT3_MIXER_CONTROL, 3, 1, 0),
+	SOC_DAPM_SINGLE("Left Mixer Switch",
+			WM8350_OUT3_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Left Input Mixer */
+static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
+	SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_L, 1, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE("PGA Capture Switch",
+			WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Right Input Mixer */
+static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
+	SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_R, 5, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE("PGA Capture Switch",
+			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Left Mic Mixer */
+static const struct snd_kcontrol_new wm8350_left_mic_mixer_controls[] = {
+	SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 2, 1, 0),
+};
+
+/* Right Mic Mixer */
+static const struct snd_kcontrol_new wm8350_right_mic_mixer_controls[] = {
+	SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 9, 1, 0),
+	SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 8, 1, 0),
+	SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 10, 1, 0),
+};
+
+/* Beep Switch */
+static const struct snd_kcontrol_new wm8350_beep_switch_controls =
+SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1);
+
+/* Out4 Capture Mux */
+static const struct snd_kcontrol_new wm8350_out4_capture_controls =
+SOC_DAPM_ENUM("Route", wm8350_enum[8]);
+
+static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
+
+	SND_SOC_DAPM_PGA("IN3R PGA", WM8350_POWER_MGMT_2, 11, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IN3L PGA", WM8350_POWER_MGMT_2, 10, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("Right Out2 PGA", WM8350_POWER_MGMT_3, 3, 0, NULL,
+			   0, pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Left Out2 PGA", WM8350_POWER_MGMT_3, 2, 0, NULL, 0,
+			   pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Right Out1 PGA", WM8350_POWER_MGMT_3, 1, 0, NULL,
+			   0, pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Left Out1 PGA", WM8350_POWER_MGMT_3, 0, 0, NULL, 0,
+			   pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	SND_SOC_DAPM_MIXER("Right Capture Mixer", WM8350_POWER_MGMT_2,
+			   7, 0, &wm8350_right_capt_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_capt_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Capture Mixer", WM8350_POWER_MGMT_2,
+			   6, 0, &wm8350_left_capt_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_capt_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Out4 Mixer", WM8350_POWER_MGMT_2, 5, 0,
+			   &wm8350_out4_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_out4_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Out3 Mixer", WM8350_POWER_MGMT_2, 4, 0,
+			   &wm8350_out3_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_out3_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Right Playback Mixer", WM8350_POWER_MGMT_2, 1, 0,
+			   &wm8350_right_play_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_play_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Playback Mixer", WM8350_POWER_MGMT_2, 0, 0,
+			   &wm8350_left_play_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_play_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Mic Mixer", WM8350_POWER_MGMT_2, 8, 0,
+			   &wm8350_left_mic_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_mic_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Right Mic Mixer", WM8350_POWER_MGMT_2, 9, 0,
+			   &wm8350_right_mic_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_mic_mixer_controls)),
+
+	/* virtual mixer for Beep and Out2R */
+	SND_SOC_DAPM_MIXER("Out2 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("Beep", WM8350_POWER_MGMT_3, 7, 0,
+			    &wm8350_beep_switch_controls),
+
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
+			 WM8350_POWER_MGMT_4, 3, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture",
+			 WM8350_POWER_MGMT_4, 2, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback",
+			 WM8350_POWER_MGMT_4, 5, 0),
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback",
+			 WM8350_POWER_MGMT_4, 4, 0),
+
+	SND_SOC_DAPM_MICBIAS("Mic Bias", WM8350_POWER_MGMT_1, 4, 0),
+
+	SND_SOC_DAPM_MUX("Out4 Capture Channel", SND_SOC_NOPM, 0, 0,
+			 &wm8350_out4_capture_controls),
+
+	SND_SOC_DAPM_OUTPUT("OUT1R"),
+	SND_SOC_DAPM_OUTPUT("OUT1L"),
+	SND_SOC_DAPM_OUTPUT("OUT2R"),
+	SND_SOC_DAPM_OUTPUT("OUT2L"),
+	SND_SOC_DAPM_OUTPUT("OUT3"),
+	SND_SOC_DAPM_OUTPUT("OUT4"),
+
+	SND_SOC_DAPM_INPUT("IN1RN"),
+	SND_SOC_DAPM_INPUT("IN1RP"),
+	SND_SOC_DAPM_INPUT("IN2R"),
+	SND_SOC_DAPM_INPUT("IN1LP"),
+	SND_SOC_DAPM_INPUT("IN1LN"),
+	SND_SOC_DAPM_INPUT("IN2L"),
+	SND_SOC_DAPM_INPUT("IN3R"),
+	SND_SOC_DAPM_INPUT("IN3L"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* left playback mixer */
+	{"Left Playback Mixer", "Playback Switch", "Left DAC"},
+	{"Left Playback Mixer", "Left Bypass Switch", "IN3L PGA"},
+	{"Left Playback Mixer", "Right Playback Switch", "Right DAC"},
+	{"Left Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+	{"Left Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+	/* right playback mixer */
+	{"Right Playback Mixer", "Playback Switch", "Right DAC"},
+	{"Right Playback Mixer", "Right Bypass Switch", "IN3R PGA"},
+	{"Right Playback Mixer", "Left Playback Switch", "Left DAC"},
+	{"Right Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+	{"Right Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+	/* out4 playback mixer */
+	{"Out4 Mixer", "Right Playback Switch", "Right DAC"},
+	{"Out4 Mixer", "Left Playback Switch", "Left DAC"},
+	{"Out4 Mixer", "Right Capture Switch", "Right Capture Mixer"},
+	{"Out4 Mixer", "Out3 Playback Switch", "Out3 Mixer"},
+	{"Out4 Mixer", "Right Mixer Switch", "Right Playback Mixer"},
+	{"Out4 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+	{"OUT4", NULL, "Out4 Mixer"},
+
+	/* out3 playback mixer */
+	{"Out3 Mixer", "Left Playback Switch", "Left DAC"},
+	{"Out3 Mixer", "Left Capture Switch", "Left Capture Mixer"},
+	{"Out3 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+	{"Out3 Mixer", "Out4 Playback Switch", "Out4 Mixer"},
+	{"OUT3", NULL, "Out3 Mixer"},
+
+	/* out2 */
+	{"Right Out2 PGA", NULL, "Right Playback Mixer"},
+	{"Left Out2 PGA", NULL, "Left Playback Mixer"},
+	{"OUT2L", NULL, "Left Out2 PGA"},
+	{"OUT2R", NULL, "Right Out2 PGA"},
+
+	/* out1 */
+	{"Right Out1 PGA", NULL, "Right Playback Mixer"},
+	{"Left Out1 PGA", NULL, "Left Playback Mixer"},
+	{"OUT1L", NULL, "Left Out1 PGA"},
+	{"OUT1R", NULL, "Right Out1 PGA"},
+
+	/* ADCs */
+	{"Left ADC", NULL, "Left Capture Mixer"},
+	{"Right ADC", NULL, "Right Capture Mixer"},
+
+	/* Left capture mixer */
+	{"Left Capture Mixer", "L2 Capture Volume", "IN2L"},
+	{"Left Capture Mixer", "L3 Capture Volume", "IN3L PGA"},
+	{"Left Capture Mixer", "PGA Capture Switch", "Left Mic Mixer"},
+	{"Left Capture Mixer", NULL, "Out4 Capture Channel"},
+
+	/* Right capture mixer */
+	{"Right Capture Mixer", "L2 Capture Volume", "IN2R"},
+	{"Right Capture Mixer", "L3 Capture Volume", "IN3R PGA"},
+	{"Right Capture Mixer", "PGA Capture Switch", "Right Mic Mixer"},
+	{"Right Capture Mixer", NULL, "Out4 Capture Channel"},
+
+	/* L3 Inputs */
+	{"IN3L PGA", NULL, "IN3L"},
+	{"IN3R PGA", NULL, "IN3R"},
+
+	/* Left Mic mixer */
+	{"Left Mic Mixer", "INN Capture Switch", "IN1LN"},
+	{"Left Mic Mixer", "INP Capture Switch", "IN1LP"},
+	{"Left Mic Mixer", "IN2 Capture Switch", "IN2L"},
+
+	/* Right Mic mixer */
+	{"Right Mic Mixer", "INN Capture Switch", "IN1RN"},
+	{"Right Mic Mixer", "INP Capture Switch", "IN1RP"},
+	{"Right Mic Mixer", "IN2 Capture Switch", "IN2R"},
+
+	/* out 4 capture */
+	{"Out4 Capture Channel", NULL, "Out4 Mixer"},
+
+	/* Beep */
+	{"Beep", NULL, "IN3R PGA"},
+};
+
+static int wm8350_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&wm8350_snd_controls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static int wm8350_add_widgets(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec,
+					wm8350_dapm_widgets,
+					ARRAY_SIZE(wm8350_dapm_widgets));
+	if (ret != 0) {
+		dev_err(codec->dev, "dapm control register failed\n");
+		return ret;
+	}
+
+	/* set up audio paths */
+	ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	if (ret != 0) {
+		dev_err(codec->dev, "DAPM route register failed\n");
+		return ret;
+	}
+
+	return snd_soc_dapm_new_widgets(codec);
+}
+
+static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				 int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	u16 fll_4;
+
+	switch (clk_id) {
+	case WM8350_MCLK_SEL_MCLK:
+		wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+				  WM8350_MCLK_SEL);
+		break;
+	case WM8350_MCLK_SEL_PLL_MCLK:
+	case WM8350_MCLK_SEL_PLL_DAC:
+	case WM8350_MCLK_SEL_PLL_ADC:
+	case WM8350_MCLK_SEL_PLL_32K:
+		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+				WM8350_MCLK_SEL);
+		fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+		    ~WM8350_FLL_CLK_SRC_MASK;
+		wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
+		break;
+	}
+
+	/* MCLK direction */
+	if (dir == WM8350_MCLK_DIR_OUT)
+		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+				WM8350_MCLK_DIR);
+	else
+		wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+				  WM8350_MCLK_DIR);
+
+	return 0;
+}
+
+static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 val;
+
+	switch (div_id) {
+	case WM8350_ADC_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) &
+		    ~WM8350_ADC_CLKDIV_MASK;
+		wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div);
+		break;
+	case WM8350_DAC_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) &
+		    ~WM8350_DAC_CLKDIV_MASK;
+		wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
+		break;
+	case WM8350_BCLK_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_BCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_OPCLK_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_OPCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_SYS_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_MCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_DACLR_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+		    ~WM8350_DACLRC_RATE_MASK;
+		wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div);
+		break;
+	case WM8350_ADCLR_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+		    ~WM8350_ADCLRC_RATE_MASK;
+		wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+	    ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK);
+	u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) &
+	    ~WM8350_BCLK_MSTR;
+	u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+	    ~WM8350_DACLRC_ENA;
+	u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+	    ~WM8350_ADCLRC_ENA;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		master |= WM8350_BCLK_MSTR;
+		dac_lrc |= WM8350_DACLRC_ENA;
+		adc_lrc |= WM8350_ADCLRC_ENA;
+		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 |= 0x2 << 8;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x1 << 8;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x3 << 8;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= 0x3 << 8;	/* lg not sure which mode */
+		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 |= WM8350_AIF_LRCLK_INV | WM8350_AIF_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= WM8350_AIF_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= WM8350_AIF_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+	wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master);
+	wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
+	wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
+	return 0;
+}
+
+static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
+			      int cmd, struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) &
+	    WM8350_BCLK_MSTR;
+	int enabled = 0;
+
+	/* Check that the DACs or ADCs are enabled since they are
+	 * required for LRC in master mode. The DACs or ADCs need a
+	 * valid audio path i.e. pin -> ADC or DAC -> pin before
+	 * the LRC will be enabled in master mode. */
+	if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+		return 0;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+		    (WM8350_ADCR_ENA | WM8350_ADCL_ENA);
+	} else {
+		enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+		    (WM8350_DACR_ENA | WM8350_DACL_ENA);
+	}
+
+	if (!enabled) {
+		dev_err(codec->dev,
+		       "%s: invalid audio path - no clocks available\n",
+		       __func__);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+	    ~WM8350_AIF_WL_MASK;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x1 << 10;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x2 << 10;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= 0x3 << 10;
+		break;
+	}
+
+	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+	return 0;
+}
+
+static int wm8350_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+
+	if (mute)
+		wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+	else
+		wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+	return 0;
+}
+
+/* FLL divisors */
+struct _fll_div {
+	int div;		/* FLL_OUTDIV */
+	int n;
+	int k;
+	int ratio;		/* FLL_FRATIO */
+};
+
+/* The size in bits of the fll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static inline int fll_factors(struct _fll_div *fll_div, unsigned int input,
+			      unsigned int output)
+{
+	u64 Kpart;
+	unsigned int t1, t2, K, Nmod;
+
+	if (output >= 2815250 && output <= 3125000)
+		fll_div->div = 0x4;
+	else if (output >= 5625000 && output <= 6250000)
+		fll_div->div = 0x3;
+	else if (output >= 11250000 && output <= 12500000)
+		fll_div->div = 0x2;
+	else if (output >= 22500000 && output <= 25000000)
+		fll_div->div = 0x1;
+	else {
+		printk(KERN_ERR "wm8350: fll freq %d out of range\n", output);
+		return -EINVAL;
+	}
+
+	if (input > 48000)
+		fll_div->ratio = 1;
+	else
+		fll_div->ratio = 8;
+
+	t1 = output * (1 << (fll_div->div + 1));
+	t2 = input * fll_div->ratio;
+
+	fll_div->n = t1 / t2;
+	Nmod = t1 % t2;
+
+	if (Nmod) {
+		Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+		do_div(Kpart, t2);
+		K = Kpart & 0xFFFFFFFF;
+
+		/* Check if we need to round */
+		if ((K % 10) >= 5)
+			K += 5;
+
+		/* Move down to proper range now rounding is done */
+		K /= 10;
+		fll_div->k = K;
+	} else
+		fll_div->k = 0;
+
+	return 0;
+}
+
+static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
+			  int pll_id, unsigned int freq_in,
+			  unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	struct _fll_div fll_div;
+	int ret = 0;
+	u16 fll_1, fll_4;
+
+	/* power down FLL - we need to do this for reconfiguration */
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+			  WM8350_FLL_ENA | WM8350_FLL_OSC_ENA);
+
+	if (freq_out == 0 || freq_in == 0)
+		return ret;
+
+	ret = fll_factors(&fll_div, freq_in, freq_out);
+	if (ret < 0)
+		return ret;
+	dev_dbg(wm8350->dev,
+		"FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+		freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
+		fll_div.ratio);
+
+	/* set up N.K & dividers */
+	fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) &
+	    ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_1,
+			   fll_1 | (fll_div.div << 8) | 0x50);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_2,
+			   (fll_div.ratio << 11) | (fll_div.
+						    n & WM8350_FLL_N_MASK));
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
+	fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+	    ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_4,
+			   fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) |
+			   (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0));
+
+	/* power FLL on */
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA);
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA);
+
+	return 0;
+}
+
+static int wm8350_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_audio_platform_data *platform =
+		wm8350->codec.platform_data;
+	u16 pm1;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_VMID_50K |
+				 platform->codec_current_on << 14);
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1);
+		pm1 &= ~WM8350_VMID_MASK;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_VMID_50K);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
+						    priv->supplies);
+			if (ret != 0)
+				return ret;
+
+			/* Enable the system clock */
+			wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4,
+					WM8350_SYSCLK_ENA);
+
+			/* mute DAC & outputs */
+			wm8350_set_bits(wm8350, WM8350_DAC_MUTE,
+					WM8350_DAC_MUTE_ENA);
+
+			/* discharge cap memory */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+					 platform->dis_out1 |
+					 (platform->dis_out2 << 2) |
+					 (platform->dis_out3 << 4) |
+					 (platform->dis_out4 << 6));
+
+			/* wait for discharge */
+			schedule_timeout_interruptible(msecs_to_jiffies
+						       (platform->
+							cap_discharge_msecs));
+
+			/* enable antipop */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+					 (platform->vmid_s_curve << 8));
+
+			/* ramp up vmid */
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 (platform->
+					  codec_current_charge << 14) |
+					 WM8350_VMID_5K | WM8350_VMIDEN |
+					 WM8350_VBUFEN);
+
+			/* wait for vmid */
+			schedule_timeout_interruptible(msecs_to_jiffies
+						       (platform->
+							vmid_charge_msecs));
+
+			/* turn on vmid 300k  */
+			pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+			    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+			pm1 |= WM8350_VMID_300K |
+				(platform->codec_current_standby << 14);
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 pm1);
+
+
+			/* enable analogue bias */
+			pm1 |= WM8350_BIASEN;
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+			/* disable antipop */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+		} else {
+			/* turn on vmid 300k and reduce current */
+			pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+			    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 pm1 | WM8350_VMID_300K |
+					 (platform->
+					  codec_current_standby << 14));
+
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+
+		/* mute DAC & enable outputs */
+		wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+
+		wm8350_set_bits(wm8350, WM8350_POWER_MGMT_3,
+				WM8350_OUT1L_ENA | WM8350_OUT1R_ENA |
+				WM8350_OUT2L_ENA | WM8350_OUT2R_ENA);
+
+		/* enable anti pop S curve */
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+				 (platform->vmid_s_curve << 8));
+
+		/* turn off vmid  */
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~WM8350_VMIDEN;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+		/* wait */
+		schedule_timeout_interruptible(msecs_to_jiffies
+					       (platform->
+						vmid_discharge_msecs));
+
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+				 (platform->vmid_s_curve << 8) |
+				 platform->dis_out1 |
+				 (platform->dis_out2 << 2) |
+				 (platform->dis_out3 << 4) |
+				 (platform->dis_out4 << 6));
+
+		/* turn off VBuf and drain */
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~(WM8350_VBUFEN | WM8350_VMID_MASK);
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_OUTPUT_DRAIN_EN);
+
+		/* wait */
+		schedule_timeout_interruptible(msecs_to_jiffies
+					       (platform->drain_msecs));
+
+		pm1 &= ~WM8350_BIASEN;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+		/* disable anti-pop */
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+		wm8350_clear_bits(wm8350, WM8350_LOUT1_VOLUME,
+				  WM8350_OUT1L_ENA);
+		wm8350_clear_bits(wm8350, WM8350_ROUT1_VOLUME,
+				  WM8350_OUT1R_ENA);
+		wm8350_clear_bits(wm8350, WM8350_LOUT2_VOLUME,
+				  WM8350_OUT2L_ENA);
+		wm8350_clear_bits(wm8350, WM8350_ROUT2_VOLUME,
+				  WM8350_OUT2R_ENA);
+
+		/* disable clock gen */
+		wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+				  WM8350_SYSCLK_ENA);
+
+		regulator_bulk_disable(ARRAY_SIZE(priv->supplies),
+				       priv->supplies);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static int wm8350_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;
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8350_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		wm8350_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+	return 0;
+}
+
+static struct snd_soc_codec *wm8350_codec;
+
+static int wm8350_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct wm8350 *wm8350;
+	struct wm8350_data *priv;
+	int ret;
+	struct wm8350_output *out1;
+	struct wm8350_output *out2;
+
+	BUG_ON(!wm8350_codec);
+
+	socdev->codec = wm8350_codec;
+	codec = socdev->codec;
+	wm8350 = codec->control_data;
+	priv = codec->private_data;
+
+	/* Enable the codec */
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	/* Enable robust clocking mode in ADC */
+	wm8350_codec_write(codec, WM8350_SECURITY, 0xa7);
+	wm8350_codec_write(codec, 0xde, 0x13);
+	wm8350_codec_write(codec, WM8350_SECURITY, 0);
+
+	/* read OUT1 & OUT2 volumes */
+	out1 = &priv->out1;
+	out2 = &priv->out2;
+	out1->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME) &
+			  WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	out1->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME) &
+			   WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	out2->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME) &
+			  WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	out2->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME) &
+			   WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, 0);
+
+	/* Latch VU bits & mute */
+	wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME,
+			WM8350_OUT1_VU | WM8350_OUT1L_MUTE);
+	wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME,
+			WM8350_OUT2_VU | WM8350_OUT2L_MUTE);
+	wm8350_set_bits(wm8350, WM8350_ROUT1_VOLUME,
+			WM8350_OUT1_VU | WM8350_OUT1R_MUTE);
+	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
+			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
+
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to create pcms\n");
+		return ret;
+	}
+
+	wm8350_add_controls(codec);
+	wm8350_add_widgets(codec);
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to register card\n");
+		goto card_err;
+	}
+
+	return 0;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+	return ret;
+}
+
+static int wm8350_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	int ret;
+
+	/* cancel any work waiting to be queued. */
+	ret = cancel_delayed_work(&codec->delayed_work);
+
+	/* if there was any work waiting then we run it now and
+	 * wait for its completion */
+	if (ret) {
+		schedule_delayed_work(&codec->delayed_work, 0);
+		flush_scheduled_work();
+	}
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	return 0;
+}
+
+#define WM8350_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define WM8350_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8350_dai = {
+	.name = "WM8350",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8350_RATES,
+		.formats = WM8350_FORMATS,
+	},
+	.capture = {
+		 .stream_name = "Capture",
+		 .channels_min = 1,
+		 .channels_max = 2,
+		 .rates = WM8350_RATES,
+		 .formats = WM8350_FORMATS,
+	 },
+	.ops = {
+		 .hw_params = wm8350_pcm_hw_params,
+		 .digital_mute = wm8350_mute,
+		 .trigger = wm8350_pcm_trigger,
+		 .set_fmt = wm8350_set_dai_fmt,
+		 .set_sysclk = wm8350_set_dai_sysclk,
+		 .set_pll = wm8350_set_fll,
+		 .set_clkdiv = wm8350_set_clkdiv,
+	 },
+};
+EXPORT_SYMBOL_GPL(wm8350_dai);
+
+struct snd_soc_codec_device soc_codec_dev_wm8350 = {
+	.probe = 	wm8350_probe,
+	.remove = 	wm8350_remove,
+	.suspend = 	wm8350_suspend,
+	.resume =	wm8350_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350);
+
+static int wm8350_codec_probe(struct platform_device *pdev)
+{
+	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+	struct wm8350_data *priv;
+	struct snd_soc_codec *codec;
+	int ret, i;
+
+	if (wm8350->codec.platform_data == NULL) {
+		dev_err(&pdev->dev, "No audio platform data supplied\n");
+		return -EINVAL;
+	}
+
+	priv = kzalloc(sizeof(struct wm8350_data), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+		priv->supplies[i].supply = supply_names[i];
+
+	ret = regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies),
+				 priv->supplies);
+	if (ret != 0)
+		goto err_priv;
+
+	codec = &priv->codec;
+	wm8350->codec.codec = codec;
+
+	wm8350_dai.dev = &pdev->dev;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	codec->dev = &pdev->dev;
+	codec->name = "WM8350";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8350_codec_read;
+	codec->write = wm8350_codec_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8350_set_bias_level;
+	codec->dai = &wm8350_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8350_MAX_REGISTER;
+	codec->private_data = priv;
+	codec->control_data = wm8350;
+
+	/* Put the codec into reset if it wasn't already */
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work);
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0)
+		goto err_supply;
+
+	wm8350_codec = codec;
+
+	ret = snd_soc_register_dai(&wm8350_dai);
+	if (ret != 0)
+		goto err_codec;
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err_supply:
+	regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+err_priv:
+	kfree(priv);
+	wm8350_codec = NULL;
+	return ret;
+}
+
+static int __devexit wm8350_codec_remove(struct platform_device *pdev)
+{
+	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = wm8350->codec.codec;
+	struct wm8350_data *priv = codec->private_data;
+
+	snd_soc_unregister_dai(&wm8350_dai);
+	snd_soc_unregister_codec(codec);
+	regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+	kfree(priv);
+	wm8350_codec = NULL;
+	return 0;
+}
+
+static struct platform_driver wm8350_codec_driver = {
+	.driver = {
+		   .name = "wm8350-codec",
+		   .owner = THIS_MODULE,
+		   },
+	.probe = wm8350_codec_probe,
+	.remove = __devexit_p(wm8350_codec_remove),
+};
+
+static __init int wm8350_init(void)
+{
+	return platform_driver_register(&wm8350_codec_driver);
+}
+module_init(wm8350_init);
+
+static __exit void wm8350_exit(void)
+{
+	platform_driver_unregister(&wm8350_codec_driver);
+}
+module_exit(wm8350_exit);
+
+MODULE_DESCRIPTION("ASoC WM8350 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8350-codec");
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
new file mode 100644
index 00000000000..cc2887aa6c3
--- /dev/null
+++ b/sound/soc/codecs/wm8350.h
@@ -0,0 +1,20 @@
+/*
+ * wm8350.h - WM8903 audio codec interface
+ *
+ * Copyright 2008 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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#ifndef _WM8350_H
+#define _WM8350_H
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm8350_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8350;
+
+#endif
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index d8ca2da8d63..40f8238df71 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -463,7 +463,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -585,8 +586,6 @@ struct snd_soc_dai wm8510_dai = {
 		.formats = WM8510_FORMATS,},
 	.ops = {
 		.hw_params = wm8510_pcm_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = wm8510_mute,
 		.set_fmt = wm8510_set_dai_fmt,
 		.set_clkdiv = wm8510_set_dai_clkdiv,
@@ -659,7 +658,7 @@ static int wm8510_init(struct snd_soc_device *socdev)
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm8510_add_controls(codec);
 	wm8510_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8510: failed to register card\n");
 		goto card_err;
@@ -890,6 +889,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8510 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
 
+static int __init wm8510_modinit(void)
+{
+	return snd_soc_register_dai(&wm8510_dai);
+}
+module_init(wm8510_modinit);
+
+static void __exit wm8510_exit(void)
+{
+	snd_soc_unregister_dai(&wm8510_dai);
+}
+module_exit(wm8510_exit);
+
 MODULE_DESCRIPTION("ASoC WM8510 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 627ebfb4209..d004e584529 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -548,13 +548,13 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai_link *dai = rtd->dai;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
-	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id);
+	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
 
 	paifb &= ~WM8580_AIF_LENGTH_MASK;
 	/* bit size */
@@ -574,7 +574,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb);
+	wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb);
 	return 0;
 }
 
@@ -798,8 +798,6 @@ struct snd_soc_dai wm8580_dai[] = {
 		},
 		.ops = {
 			 .hw_params = wm8580_paif_hw_params,
-		 },
-		.dai_ops = {
 			 .set_fmt = wm8580_set_paif_dai_fmt,
 			 .set_clkdiv = wm8580_set_dai_clkdiv,
 			 .set_pll = wm8580_set_dai_pll,
@@ -818,8 +816,6 @@ struct snd_soc_dai wm8580_dai[] = {
 		},
 		.ops = {
 			 .hw_params = wm8580_paif_hw_params,
-		 },
-		.dai_ops = {
 			 .set_fmt = wm8580_set_paif_dai_fmt,
 			 .set_clkdiv = wm8580_set_dai_clkdiv,
 			 .set_pll = wm8580_set_dai_pll,
@@ -873,7 +869,7 @@ static int wm8580_init(struct snd_soc_device *socdev)
 	wm8580_add_controls(codec);
 	wm8580_add_widgets(codec);
 
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8580: failed to register card\n");
 		goto card_err;
@@ -900,85 +896,85 @@ static struct snd_soc_device *wm8580_socdev;
  *    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 wm8580_i2c_driver;
-static struct i2c_client client_template;
-
-static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+static int wm8580_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8580_socdev;
-	struct wm8580_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) {
-		dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr);
-		goto err;
-	}
-
 	ret = wm8580_init(socdev);
-	if (ret < 0) {
+	if (ret < 0)
 		dev_err(&i2c->dev, "failed to initialise WM8580\n");
-		goto err;
-	}
-
-	return ret;
-
-err:
-	kfree(codec);
-	kfree(i2c);
 	return ret;
 }
 
-static int wm8580_i2c_detach(struct i2c_client *client)
+static int wm8580_i2c_remove(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 wm8580_i2c_attach(struct i2c_adapter *adap)
-{
-	return i2c_probe(adap, &addr_data, wm8580_codec_probe);
-}
+static const struct i2c_device_id wm8580_i2c_id[] = {
+	{ "wm8580", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
 
-/* corgi i2c codec control layer */
 static struct i2c_driver wm8580_i2c_driver = {
 	.driver = {
 		.name = "WM8580 I2C Codec",
 		.owner = THIS_MODULE,
 	},
-	.attach_adapter = wm8580_i2c_attach,
-	.detach_client =  wm8580_i2c_detach,
-	.command =        NULL,
+	.probe =    wm8580_i2c_probe,
+	.remove =   wm8580_i2c_remove,
+	.id_table = wm8580_i2c_id,
 };
 
-static struct i2c_client client_template = {
-	.name =   "WM8580",
-	.driver = &wm8580_i2c_driver,
-};
+static int wm8580_add_i2c_device(struct platform_device *pdev,
+				 const struct wm8580_setup_data *setup)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+	int ret;
+
+	ret = i2c_add_driver(&wm8580_i2c_driver);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "can't add i2c driver\n");
+		return ret;
+	}
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = setup->i2c_address;
+	strlcpy(info.type, "wm8580", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(setup->i2c_bus);
+	if (!adapter) {
+		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+			setup->i2c_bus);
+		goto err_driver;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		goto err_driver;
+	}
+
+	return 0;
+
+err_driver:
+	i2c_del_driver(&wm8580_i2c_driver);
+	return -ENODEV;
+}
 #endif
 
 static int wm8580_probe(struct platform_device *pdev)
@@ -1011,11 +1007,8 @@ static int wm8580_probe(struct platform_device *pdev)
 
 #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(&wm8580_i2c_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add i2c driver");
+		ret = wm8580_add_i2c_device(pdev, setup);
 	}
 #else
 		/* Add other interfaces here */
@@ -1034,6 +1027,7 @@ static int wm8580_remove(struct platform_device *pdev)
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8580_i2c_driver);
 #endif
 	kfree(codec->private_data);
@@ -1048,6 +1042,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
 
+static int __init wm8580_modinit(void)
+{
+	return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_init(wm8580_modinit);
+
+static void __exit wm8580_exit(void)
+{
+	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+}
+module_exit(wm8580_exit);
+
 MODULE_DESCRIPTION("ASoC WM8580 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h
index 589ddaba21d..09e4422f6f2 100644
--- a/sound/soc/codecs/wm8580.h
+++ b/sound/soc/codecs/wm8580.h
@@ -29,6 +29,7 @@
 #define WM8580_CLKSRC_NONE 5
 
 struct wm8580_setup_data {
+	int i2c_bus;
 	unsigned short i2c_address;
 };
 
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
new file mode 100644
index 00000000000..80b11983e13
--- /dev/null
+++ b/sound/soc/codecs/wm8728.c
@@ -0,0 +1,585 @@
+/*
+ * wm8728.c  --  WM8728 ALSA SoC Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.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 <sound/tlv.h>
+
+#include "wm8728.h"
+
+struct snd_soc_codec_device soc_codec_dev_wm8728;
+
+/*
+ * We can't read the WM8728 register space so we cache them instead.
+ * Note that the defaults here aren't the physical defaults, we latch
+ * the volume update bits, mute the output and enable infinite zero
+ * detect.
+ */
+static const u16 wm8728_reg_defaults[] = {
+	0x1ff,
+	0x1ff,
+	0x001,
+	0x100,
+};
+
+static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	return cache[reg];
+}
+
+static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
+	u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	cache[reg] = value;
+}
+
+/*
+ * write to the WM8728 register space
+ */
+static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[2];
+
+	/* data is
+	 *   D15..D9 WM8728 register offset
+	 *   D8...D0 register data
+	 */
+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+	data[1] = value & 0x00ff;
+
+	wm8728_write_reg_cache(codec, reg, value);
+
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new wm8728_snd_controls[] = {
+
+SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL,
+		 0, 255, 0, wm8728_tlv),
+
+SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0),
+};
+
+static int wm8728_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&wm8728_snd_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/*
+ * DAPM controls.
+ */
+static const struct snd_soc_dapm_widget wm8728_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("VOUTL"),
+SND_SOC_DAPM_OUTPUT("VOUTR"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"VOUTL", NULL, "DAC"},
+	{"VOUTR", NULL, "DAC"},
+};
+
+static int wm8728_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets,
+				  ARRAY_SIZE(wm8728_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(codec);
+
+	return 0;
+}
+
+static int wm8728_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+
+	if (mute)
+		wm8728_write(codec, WM8728_DACCTL, mute_reg | 1);
+	else
+		wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1);
+
+	return 0;
+}
+
+static int wm8728_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+
+	dac &= ~0x18;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		dac |= 0x10;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		dac |= 0x08;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8728_write(codec, WM8728_DACCTL, dac);
+
+	return 0;
+}
+
+static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL);
+
+	/* Currently only I2S is supported by the driver, though the
+	 * hardware is more flexible.
+	 */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* The hardware only support full slave mode */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		iface &= ~0x22;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |=  0x20;
+		iface &= ~0x02;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x02;
+		iface &= ~0x20;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= 0x22;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8728_write(codec, WM8728_IFCTL, iface);
+	return 0;
+}
+
+static int wm8728_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u16 reg;
+	int i;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Power everything up... */
+			reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+			wm8728_write(codec, WM8728_DACCTL, reg & ~0x4);
+
+			/* ..then sync in the register cache. */
+			for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++)
+				wm8728_write(codec, i,
+					     wm8728_read_reg_cache(codec, i));
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		reg = wm8728_read_reg_cache(codec, WM8728_DACCTL);
+		wm8728_write(codec, WM8728_DACCTL, reg | 0x4);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8728_RATES (SNDRV_PCM_RATE_8000_192000)
+
+#define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8728_dai = {
+	.name = "WM8728",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = WM8728_RATES,
+		.formats = WM8728_FORMATS,
+	},
+	.ops = {
+		 .hw_params = wm8728_hw_params,
+		 .digital_mute = wm8728_mute,
+		 .set_fmt = wm8728_set_dai_fmt,
+	}
+};
+EXPORT_SYMBOL_GPL(wm8728_dai);
+
+static int wm8728_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;
+
+	wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int wm8728_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8728_set_bias_level(codec, codec->suspend_bias_level);
+
+	return 0;
+}
+
+/*
+ * initialise the WM8728 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8728_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	codec->name = "WM8728";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8728_read_reg_cache;
+	codec->write = wm8728_write;
+	codec->set_bias_level = wm8728_set_bias_level;
+	codec->dai = &wm8728_dai;
+	codec->num_dai = 1;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults);
+	codec->reg_cache = kmemdup(wm8728_reg_defaults,
+				   sizeof(wm8728_reg_defaults),
+				   GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8728: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	wm8728_add_controls(codec);
+	wm8728_add_widgets(codec);
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8728: 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 *wm8728_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8728 2 wire address is determined by GPIO5
+ * state during powerup.
+ *    low  = 0x1a
+ *    high = 0x1b
+ */
+
+static int wm8728_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct snd_soc_device *socdev = wm8728_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = wm8728_init(socdev);
+	if (ret < 0)
+		pr_err("failed to initialise WM8728\n");
+
+	return ret;
+}
+
+static int wm8728_i2c_remove(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	kfree(codec->reg_cache);
+	return 0;
+}
+
+static const struct i2c_device_id wm8728_i2c_id[] = {
+	{ "wm8728", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id);
+
+static struct i2c_driver wm8728_i2c_driver = {
+	.driver = {
+		.name = "WM8728 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8728_i2c_probe,
+	.remove =   wm8728_i2c_remove,
+	.id_table = wm8728_i2c_id,
+};
+
+static int wm8728_add_i2c_device(struct platform_device *pdev,
+				 const struct wm8728_setup_data *setup)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+	int ret;
+
+	ret = i2c_add_driver(&wm8728_i2c_driver);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "can't add i2c driver\n");
+		return ret;
+	}
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = setup->i2c_address;
+	strlcpy(info.type, "wm8728", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(setup->i2c_bus);
+	if (!adapter) {
+		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
+			setup->i2c_bus);
+		goto err_driver;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		goto err_driver;
+	}
+
+	return 0;
+
+err_driver:
+	i2c_del_driver(&wm8728_i2c_driver);
+	return -ENODEV;
+}
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8728_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_device *socdev = wm8728_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	codec->control_data = spi;
+
+	ret = wm8728_init(socdev);
+	if (ret < 0)
+		dev_err(&spi->dev, "failed to initialise WM8728\n");
+
+	return ret;
+}
+
+static int __devexit wm8728_spi_remove(struct spi_device *spi)
+{
+	return 0;
+}
+
+static struct spi_driver wm8728_spi_driver = {
+	.driver = {
+		.name	= "wm8728",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8728_spi_probe,
+	.remove		= __devexit_p(wm8728_spi_remove),
+};
+
+static int wm8728_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
+static int wm8728_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct wm8728_setup_data *setup;
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	setup = socdev->codec_data;
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	wm8728_socdev = socdev;
+	ret = -ENODEV;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	if (setup->i2c_address) {
+		codec->hw_write = (hw_write_t)i2c_master_send;
+		ret = wm8728_add_i2c_device(pdev, setup);
+	}
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	if (setup->spi) {
+		codec->hw_write = (hw_write_t)wm8728_spi_write;
+		ret = spi_register_driver(&wm8728_spi_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add spi driver");
+	}
+#endif
+
+	if (ret != 0)
+		kfree(codec);
+
+	return ret;
+}
+
+/* power down chip */
+static int wm8728_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)
+		wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_unregister_device(codec->control_data);
+	i2c_del_driver(&wm8728_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8728_spi_driver);
+#endif
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8728 = {
+	.probe = 	wm8728_probe,
+	.remove = 	wm8728_remove,
+	.suspend = 	wm8728_suspend,
+	.resume =	wm8728_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728);
+
+static int __init wm8728_modinit(void)
+{
+	return snd_soc_register_dai(&wm8728_dai);
+}
+module_init(wm8728_modinit);
+
+static void __exit wm8728_exit(void)
+{
+	snd_soc_unregister_dai(&wm8728_dai);
+}
+module_exit(wm8728_exit);
+
+MODULE_DESCRIPTION("ASoC WM8728 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8728.h b/sound/soc/codecs/wm8728.h
new file mode 100644
index 00000000000..d269c132474
--- /dev/null
+++ b/sound/soc/codecs/wm8728.h
@@ -0,0 +1,30 @@
+/*
+ * wm8728.h  --  WM8728 ASoC codec driver
+ *
+ * Copyright 2008 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8728_H
+#define _WM8728_H
+
+#define WM8728_DACLVOL   0x00
+#define WM8728_DACRVOL   0x01
+#define WM8728_DACCTL    0x02
+#define WM8728_IFCTL     0x03
+
+struct wm8728_setup_data {
+	int            spi;
+	int            i2c_bus;
+	unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8728_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8728;
+
+#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7f8a7e36b33..c444b9f2701 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -264,7 +264,8 @@ static inline int get_coeff(int mclk, int rate)
 }
 
 static int wm8731_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -293,7 +294,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
+static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -305,7 +307,8 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void wm8731_shutdown(struct snd_pcm_substream *substream)
+static void wm8731_shutdown(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -461,8 +464,6 @@ struct snd_soc_dai wm8731_dai = {
 		.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,
@@ -544,7 +545,7 @@ static int wm8731_init(struct snd_soc_device *socdev)
 
 	wm8731_add_controls(codec);
 	wm8731_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8731: failed to register card\n");
 		goto card_err;
@@ -792,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 
+static int __init wm8731_modinit(void)
+{
+	return snd_soc_register_dai(&wm8731_dai);
+}
+module_init(wm8731_modinit);
+
+static void __exit wm8731_exit(void)
+{
+	snd_soc_unregister_dai(&wm8731_dai);
+}
+module_exit(wm8731_exit);
+
 MODULE_DESCRIPTION("ASoC WM8731 driver");
 MODULE_AUTHOR("Richard Purdie");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 9b7296ee5b0..5997fa68e0d 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -614,7 +614,8 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -709,8 +710,6 @@ struct snd_soc_dai wm8750_dai = {
 		.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,
@@ -819,7 +818,7 @@ static int wm8750_init(struct snd_soc_device *socdev)
 
 	wm8750_add_controls(codec);
 	wm8750_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8750: failed to register card\n");
 		goto card_err;
@@ -1086,6 +1085,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750);
 
+static int __init wm8750_modinit(void)
+{
+	return snd_soc_register_dai(&wm8750_dai);
+}
+module_init(wm8750_modinit);
+
+static void __exit wm8750_exit(void)
+{
+	snd_soc_unregister_dai(&wm8750_dai);
+}
+module_exit(wm8750_exit);
+
 MODULE_DESCRIPTION("ASoC WM8750 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index d426eaa2218..6c21b50c937 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -922,7 +922,8 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1155,7 +1156,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1323,16 +1325,15 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
+		.formats = WM8753_FORMATS},
 	.capture = { /* dummy for fast DAI switching */
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
+		.formats = WM8753_FORMATS},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode1h_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1356,8 +1357,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_pcm_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_pcm_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode1v_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1385,8 +1385,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_pcm_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_pcm_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode2_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1410,8 +1409,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode3_4_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1439,8 +1437,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
 	.ops = {
-		.hw_params = wm8753_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8753_i2s_hw_params,
 		.digital_mute = wm8753_mute,
 		.set_fmt = wm8753_mode3_4_set_dai_fmt,
 		.set_clkdiv = wm8753_set_dai_clkdiv,
@@ -1608,7 +1605,7 @@ static int wm8753_init(struct snd_soc_device *socdev)
 
 	wm8753_add_controls(codec);
 	wm8753_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8753: failed to register card\n");
 		goto card_err;
@@ -1877,6 +1874,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
 
+static int __init wm8753_modinit(void)
+{
+	return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_init(wm8753_modinit);
+
+static void __exit wm8753_exit(void)
+{
+	snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+}
+module_exit(wm8753_exit);
+
 MODULE_DESCRIPTION("ASoC WM8753 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 3b326c9b558..6767de10ded 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -138,6 +138,10 @@
 struct snd_soc_codec_device soc_codec_dev_wm8900;
 
 struct wm8900_priv {
+	struct snd_soc_codec codec;
+
+	u16 reg_cache[WM8900_MAXREG];
+
 	u32 fll_in; /* FLL input frequency */
 	u32 fll_out; /* FLL output frequency */
 };
@@ -727,7 +731,8 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec)
 }
 
 static int wm8900_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1117,8 +1122,6 @@ struct snd_soc_dai wm8900_dai = {
 	 },
 	.ops = {
 		.hw_params = wm8900_hw_params,
-	 },
-	.dai_ops = {
 		 .set_clkdiv = wm8900_set_dai_clkdiv,
 		 .set_pll = wm8900_set_dai_pll,
 		 .set_fmt = wm8900_set_dai_fmt,
@@ -1283,16 +1286,28 @@ static int wm8900_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialise the WM8900 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8900_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8900_codec;
+
+static int wm8900_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
 {
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret = 0;
+	struct wm8900_priv *wm8900;
+	struct snd_soc_codec *codec;
 	unsigned int reg;
-	struct i2c_client *i2c_client = socdev->codec->control_data;
+	int ret;
+
+	wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
+	if (wm8900 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8900->codec;
+	codec->private_data = wm8900;
+	codec->reg_cache = &wm8900->reg_cache[0];
+	codec->reg_cache_size = WM8900_MAXREG;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
 	codec->name = "WM8900";
 	codec->owner = THIS_MODULE;
@@ -1300,33 +1315,28 @@ static int wm8900_init(struct snd_soc_device *socdev)
 	codec->write = wm8900_write;
 	codec->dai = &wm8900_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = WM8900_MAXREG;
-	codec->reg_cache = kmemdup(wm8900_reg_defaults,
-				   sizeof(wm8900_reg_defaults), GFP_KERNEL);
-
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+	codec->control_data = i2c;
+	codec->set_bias_level = wm8900_set_bias_level;
+	codec->dev = &i2c->dev;
 
 	reg = wm8900_read(codec, WM8900_REG_ID);
 	if (reg != 0x8900) {
-		dev_err(&i2c_client->dev, "Device is not a WM8900 - ID %x\n",
-			reg);
-		return -ENODEV;
-	}
-
-	codec->private_data = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL);
-	if (codec->private_data == NULL) {
-		ret = -ENOMEM;
-		goto priv_err;
+		dev_err(&i2c->dev, "Device is not a WM8900 - ID %x\n", reg);
+		ret = -ENODEV;
+		goto err;
 	}
 
 	/* Read back from the chip */
 	reg = wm8900_chip_read(codec, WM8900_REG_POWER1);
 	reg = (reg >> 12) & 0xf;
-	dev_info(&i2c_client->dev, "WM8900 revision %d\n", reg);
+	dev_info(&i2c->dev, "WM8900 revision %d\n", reg);
 
 	wm8900_reset(codec);
 
+	/* Turn the chip on */
+	wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
 	/* Latch the volume update bits */
 	wm8900_write(codec, WM8900_REG_LINVOL,
 		     wm8900_read(codec, WM8900_REG_LINVOL) | 0x100);
@@ -1352,160 +1362,98 @@ static int wm8900_init(struct snd_soc_device *socdev)
 	/* Set the DAC and mixer output bias */
 	wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81);
 
-	/* Register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-	if (ret < 0) {
-		dev_err(&i2c_client->dev, "Failed to register new PCMs\n");
-		goto pcm_err;
-	}
-
-	/* Turn the chip on */
-	codec->bias_level = SND_SOC_BIAS_OFF;
-	wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	wm8900_add_controls(codec);
-	wm8900_add_widgets(codec);
-
-	ret = snd_soc_register_card(socdev);
-	if (ret < 0) {
-		dev_err(&i2c_client->dev, "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);
-priv_err:
-	kfree(codec->private_data);
-	return ret;
-}
+	wm8900_dai.dev = &i2c->dev;
 
-static struct snd_soc_device *wm8900_socdev;
+	wm8900_codec = codec;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
-
-/* Magic definition of all other variables and things */
-I2C_CLIENT_INSMOD;
-
-static struct i2c_driver wm8900_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 wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind)
-{
-	struct snd_soc_device *socdev = wm8900_socdev;
-	struct wm8900_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;
-
-	dev_err(&adap->dev, "Probe on %x\n", addr);
-
-	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) {
-		dev_err(&adap->dev,
-			"failed to attach codec at addr %x\n", addr);
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
 		goto err;
 	}
 
-	ret = wm8900_init(socdev);
-	if (ret < 0) {
-		dev_err(&adap->dev, "failed to initialise WM8900\n");
-		goto err;
+	ret = snd_soc_register_dai(&wm8900_dai);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
 	}
+
 	return ret;
 
+err_codec:
+	snd_soc_unregister_codec(codec);
 err:
-	kfree(codec);
-	kfree(i2c);
+	kfree(wm8900);
+	wm8900_codec = NULL;
 	return ret;
 }
 
-static int wm8900_i2c_detach(struct i2c_client *client)
+static int wm8900_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	i2c_detach_client(client);
-	kfree(codec->reg_cache);
-	kfree(client);
+	snd_soc_unregister_dai(&wm8900_dai);
+	snd_soc_unregister_codec(wm8900_codec);
+
+	wm8900_set_bias_level(wm8900_codec, SND_SOC_BIAS_OFF);
+
+	wm8900_dai.dev = NULL;
+	kfree(wm8900_codec->private_data);
+	wm8900_codec = NULL;
+
 	return 0;
 }
 
-static int wm8900_i2c_attach(struct i2c_adapter *adap)
-{
-	return i2c_probe(adap, &addr_data, wm8900_codec_probe);
-}
+static const struct i2c_device_id wm8900_i2c_id[] = {
+	{ "wm8900", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id);
 
-/* corgi i2c codec control layer */
 static struct i2c_driver wm8900_i2c_driver = {
 	.driver = {
-		.name = "WM8900 I2C codec",
+		.name = "WM8900",
 		.owner = THIS_MODULE,
 	},
-	.attach_adapter = wm8900_i2c_attach,
-	.detach_client =  wm8900_i2c_detach,
-	.command =        NULL,
-};
-
-static struct i2c_client client_template = {
-	.name =   "WM8900",
-	.driver = &wm8900_i2c_driver,
+	.probe = wm8900_i2c_probe,
+	.remove = wm8900_i2c_remove,
+	.id_table = wm8900_i2c_id,
 };
-#endif
 
 static int wm8900_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8900_setup_data *setup;
 	struct snd_soc_codec *codec;
 	int ret = 0;
 
-	dev_info(&pdev->dev, "WM8900 Audio Codec\n");
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
+	if (!wm8900_codec) {
+		dev_err(&pdev->dev, "I2C client not yet instantiated\n");
+		return -ENODEV;
+	}
 
+	codec = wm8900_codec;
 	socdev->codec = codec;
 
-	codec->set_bias_level = wm8900_set_bias_level;
+	/* Register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register new PCMs\n");
+		goto pcm_err;
+	}
 
-	wm8900_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(&wm8900_i2c_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add i2c driver");
+	wm8900_add_controls(codec);
+	wm8900_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "Failed to register card\n");
+		goto card_err;
 	}
-#else
-#error Non-I2C interfaces not yet supported
-#endif
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
 	return ret;
 }
 
@@ -1513,17 +1461,9 @@ static int wm8900_probe(struct platform_device *pdev)
 static int wm8900_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)
-		wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_del_driver(&wm8900_i2c_driver);
-#endif
-	kfree(codec);
 
 	return 0;
 }
@@ -1536,6 +1476,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8900 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900);
 
+static int __init wm8900_modinit(void)
+{
+	return i2c_add_driver(&wm8900_i2c_driver);
+}
+module_init(wm8900_modinit);
+
+static void __exit wm8900_exit(void)
+{
+	i2c_del_driver(&wm8900_i2c_driver);
+}
+module_exit(wm8900_exit);
+
 MODULE_DESCRIPTION("ASoC WM8900 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h
index ba450d99e90..fd15007d10c 100644
--- a/sound/soc/codecs/wm8900.h
+++ b/sound/soc/codecs/wm8900.h
@@ -52,12 +52,6 @@
 #define WM8900_DAC_CLKDIV_5_5 0x14
 #define WM8900_DAC_CLKDIV_6   0x18
 
-#define WM8900_
-
-struct wm8900_setup_data {
-	unsigned short i2c_address;
-};
-
 extern struct snd_soc_dai wm8900_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8900;
 
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce40d787760..bde74546db4 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -33,19 +33,6 @@
 
 #include "wm8903.h"
 
-struct wm8903_priv {
-	int sysclk;
-
-	/* Reference counts */
-	int charge_pump_users;
-	int class_w_users;
-	int playback_active;
-	int capture_active;
-
-	struct snd_pcm_substream *master_substream;
-	struct snd_pcm_substream *slave_substream;
-};
-
 /* Register defaults at reset */
 static u16 wm8903_reg_defaults[] = {
 	0x8903,     /* R0   - SW Reset and ID */
@@ -223,6 +210,23 @@ static u16 wm8903_reg_defaults[] = {
 	0x0000,     /* R172 - Analogue Output Bias 0 */
 };
 
+struct wm8903_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[ARRAY_SIZE(wm8903_reg_defaults)];
+
+	int sysclk;
+
+	/* Reference counts */
+	int charge_pump_users;
+	int class_w_users;
+	int playback_active;
+	int capture_active;
+
+	struct snd_pcm_substream *master_substream;
+	struct snd_pcm_substream *slave_substream;
+};
+
+
 static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec,
 						 unsigned int reg)
 {
@@ -360,6 +364,8 @@ static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache)
 static void wm8903_reset(struct snd_soc_codec *codec)
 {
 	wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0);
+	memcpy(codec->reg_cache, wm8903_reg_defaults,
+	       sizeof(wm8903_reg_defaults));
 }
 
 #define WM8903_OUTPUT_SHORT 0x8
@@ -392,6 +398,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
 		break;
 	default:
 		BUG();
+		return -EINVAL;  /* Spurious warning from some compilers */
 	}
 
 	switch (w->shift) {
@@ -403,6 +410,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
 		break;
 	default:
 		BUG();
+		return -EINVAL;  /* Spurious warning from some compilers */
 	}
 
 	if (event & SND_SOC_DAPM_PRE_PMU) {
@@ -773,14 +781,14 @@ static const struct snd_kcontrol_new left_output_mixer[] = {
 SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
 SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
-SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 0, 1, 0),
 };
 
 static const struct snd_kcontrol_new right_output_mixer[] = {
 SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 2, 1, 0),
 SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
-SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0),
+SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 0, 1, 0),
 };
 
 static const struct snd_kcontrol_new left_speaker_mixer[] = {
@@ -788,7 +796,7 @@ SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 2, 1, 0),
 SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 1, 1, 0),
 SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0,
-		1, 1, 0),
+		0, 1, 0),
 };
 
 static const struct snd_kcontrol_new right_speaker_mixer[] = {
@@ -797,7 +805,7 @@ SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 2, 1, 0),
 SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
 		1, 1, 0),
 SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0,
-		1, 1, 0),
+		0, 1, 0),
 };
 
 static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = {
@@ -989,6 +997,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			wm8903_write(codec, WM8903_CLOCK_RATES_2,
+				     WM8903_CLK_SYS_ENA);
+
 			wm8903_run_sequence(codec, 0);
 			wm8903_sync_reg_cache(codec, codec->reg_cache);
 
@@ -1019,6 +1030,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_OFF:
 		wm8903_run_sequence(codec, 32);
+		reg = wm8903_read(codec, WM8903_CLOCK_RATES_2);
+		reg &= ~WM8903_CLK_SYS_ENA;
+		wm8903_write(codec, WM8903_CLOCK_RATES_2, reg);
 		break;
 	}
 
@@ -1257,7 +1271,8 @@ static struct {
 	{ 0,      0 },
 };
 
-static int wm8903_startup(struct snd_pcm_substream *substream)
+static int wm8903_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1298,7 +1313,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void wm8903_shutdown(struct snd_pcm_substream *substream)
+static void wm8903_shutdown(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1317,7 +1333,8 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream)
 }
 
 static int wm8903_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1515,8 +1532,6 @@ struct snd_soc_dai wm8903_dai = {
 		 .startup = wm8903_startup,
 		 .shutdown = wm8903_shutdown,
 		 .hw_params = wm8903_hw_params,
-	},
-	.dai_ops = {
 		 .digital_mute = wm8903_digital_mute,
 		 .set_fmt = wm8903_set_dai_fmt,
 		 .set_sysclk = wm8903_set_dai_sysclk
@@ -1560,39 +1575,48 @@ static int wm8903_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialise the WM8903 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8903_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8903_codec;
+
+static int wm8903_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
 {
-	struct snd_soc_codec *codec = socdev->codec;
-	struct i2c_client *i2c = codec->control_data;
-	int ret = 0;
+	struct wm8903_priv *wm8903;
+	struct snd_soc_codec *codec;
+	int ret;
 	u16 val;
 
-	val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID);
-	if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) {
-		dev_err(&i2c->dev,
-			"Device with ID register %x is not a WM8903\n", val);
-		return -ENODEV;
-	}
+	wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
+	if (wm8903 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8903->codec;
 
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->dev = &i2c->dev;
 	codec->name = "WM8903";
 	codec->owner = THIS_MODULE;
 	codec->read = wm8903_read;
 	codec->write = wm8903_write;
+	codec->hw_write = (hw_write_t)i2c_master_send;
 	codec->bias_level = SND_SOC_BIAS_OFF;
 	codec->set_bias_level = wm8903_set_bias_level;
 	codec->dai = &wm8903_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = ARRAY_SIZE(wm8903_reg_defaults);
-	codec->reg_cache = kmemdup(wm8903_reg_defaults,
-				   sizeof(wm8903_reg_defaults),
-				   GFP_KERNEL);
-	if (codec->reg_cache == NULL) {
-		dev_err(&i2c->dev, "Failed to allocate register cache\n");
-		return -ENOMEM;
+	codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache);
+	codec->reg_cache = &wm8903->reg_cache[0];
+	codec->private_data = wm8903;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID);
+	if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) {
+		dev_err(&i2c->dev,
+			"Device with ID register %x is not a WM8903\n", val);
+		return -ENODEV;
 	}
 
 	val = wm8903_read(codec, WM8903_REVISION_NUMBER);
@@ -1601,16 +1625,6 @@ static int wm8903_init(struct snd_soc_device *socdev)
 
 	wm8903_reset(codec);
 
-	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
-	if (ret < 0) {
-		dev_err(&i2c->dev, "failed to create pcms\n");
-		goto pcm_err;
-	}
-
-	/* SYSCLK is required for pretty much anything */
-	wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA);
-
 	/* power on device */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -1645,47 +1659,45 @@ static int wm8903_init(struct snd_soc_device *socdev)
 	val |= WM8903_DAC_MUTEMODE;
 	wm8903_write(codec, WM8903_DAC_DIGITAL_1, val);
 
-	wm8903_add_controls(codec);
-	wm8903_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
-	if (ret < 0) {
-		dev_err(&i2c->dev, "wm8903: failed to register card\n");
-		goto card_err;
+	wm8903_dai.dev = &i2c->dev;
+	wm8903_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&wm8903_dai);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
 	}
 
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-pcm_err:
-	kfree(codec->reg_cache);
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	wm8903_codec = NULL;
+	kfree(wm8903);
 	return ret;
 }
 
-static struct snd_soc_device *wm8903_socdev;
-
-static int wm8903_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static int wm8903_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_device *socdev = wm8903_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
 
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
+	snd_soc_unregister_dai(&wm8903_dai);
+	snd_soc_unregister_codec(codec);
 
-	ret = wm8903_init(socdev);
-	if (ret < 0)
-		dev_err(&i2c->dev, "Device initialisation failed\n");
+	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
-	return ret;
-}
+	kfree(codec->private_data);
+
+	wm8903_codec = NULL;
+	wm8903_dai.dev = NULL;
 
-static int wm8903_i2c_remove(struct i2c_client *client)
-{
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
 	return 0;
 }
 
@@ -1709,75 +1721,37 @@ static struct i2c_driver wm8903_i2c_driver = {
 static int wm8903_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8903_setup_data *setup;
-	struct snd_soc_codec *codec;
-	struct wm8903_priv *wm8903;
-	struct i2c_board_info board_info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *i2c_client;
 	int ret = 0;
 
-	setup = socdev->codec_data;
-
-	if (!setup->i2c_address) {
-		dev_err(&pdev->dev, "No codec address provided\n");
-		return -ENODEV;
+	if (!wm8903_codec) {
+		dev_err(&pdev->dev, "I2C device not yet probed\n");
+		goto err;
 	}
 
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
+	socdev->codec = wm8903_codec;
 
-	wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL);
-	if (wm8903 == NULL) {
-		ret = -ENOMEM;
-		goto err_codec;
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to create pcms\n");
+		goto err;
 	}
 
-	codec->private_data = wm8903;
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-
-	wm8903_socdev = socdev;
+	wm8903_add_controls(socdev->codec);
+	wm8903_add_widgets(socdev->codec);
 
-	codec->hw_write = (hw_write_t)i2c_master_send;
-	ret = i2c_add_driver(&wm8903_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		goto err_priv;
-	} else {
-		memset(&board_info, 0, sizeof(board_info));
-		strlcpy(board_info.type, "wm8903", I2C_NAME_SIZE);
-		board_info.addr = setup->i2c_address;
-
-		adapter = i2c_get_adapter(setup->i2c_bus);
-		if (!adapter) {
-			dev_err(&pdev->dev, "Can't get I2C bus %d\n",
-				setup->i2c_bus);
-			ret = -ENODEV;
-			goto err_adapter;
-		}
-
-		i2c_client = i2c_new_device(adapter, &board_info);
-		i2c_put_adapter(adapter);
-		if (i2c_client == NULL) {
-			dev_err(&pdev->dev,
-				"I2C driver registration failed\n");
-			ret = -ENODEV;
-			goto err_adapter;
-		}
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "wm8903: failed to register card\n");
+		goto card_err;
 	}
 
 	return ret;
 
-err_adapter:
-	i2c_del_driver(&wm8903_i2c_driver);
-err_priv:
-	kfree(codec->private_data);
-err_codec:
-	kfree(codec);
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+err:
 	return ret;
 }
 
@@ -1792,10 +1766,6 @@ static int wm8903_remove(struct platform_device *pdev)
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-	i2c_unregister_device(socdev->codec->control_data);
-	i2c_del_driver(&wm8903_i2c_driver);
-	kfree(codec->private_data);
-	kfree(codec);
 
 	return 0;
 }
@@ -1808,6 +1778,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8903 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903);
 
+static int __init wm8903_modinit(void)
+{
+	return i2c_add_driver(&wm8903_i2c_driver);
+}
+module_init(wm8903_modinit);
+
+static void __exit wm8903_exit(void)
+{
+	i2c_del_driver(&wm8903_i2c_driver);
+}
+module_exit(wm8903_exit);
+
 MODULE_DESCRIPTION("ASoC WM8903 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.cm>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index cec622f2f66..0ea27e2b996 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -18,11 +18,6 @@
 extern struct snd_soc_dai wm8903_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8903;
 
-struct wm8903_setup_data {
-	int i2c_bus;
-	int i2c_address;
-};
-
 #define WM8903_MCLK_DIV_2 1
 #define WM8903_CLK_SYS    2
 #define WM8903_BCLK       3
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index f41a578ddd4..88ead7f8dd9 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -541,7 +541,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -634,8 +635,6 @@ struct snd_soc_dai wm8971_dai = {
 		.formats = WM8971_FORMATS,},
 	.ops = {
 		.hw_params = wm8971_pcm_hw_params,
-	},
-	.dai_ops = {
 		.digital_mute = wm8971_mute,
 		.set_fmt = wm8971_set_dai_fmt,
 		.set_sysclk = wm8971_set_dai_sysclk,
@@ -748,7 +747,7 @@ static int wm8971_init(struct snd_soc_device *socdev)
 
 	wm8971_add_controls(codec);
 	wm8971_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8971: failed to register card\n");
 		goto card_err;
@@ -936,6 +935,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8971 = {
 
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971);
 
+static int __init wm8971_modinit(void)
+{
+	return snd_soc_register_dai(&wm8971_dai);
+}
+module_init(wm8971_modinit);
+
+static void __exit wm8971_exit(void)
+{
+	snd_soc_unregister_dai(&wm8971_dai);
+}
+module_exit(wm8971_exit);
+
 MODULE_DESCRIPTION("ASoC WM8971 driver");
 MODULE_AUTHOR("Lab126");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 572d22b0880..5b5afc14447 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -106,6 +106,7 @@ static const u16 wm8990_reg[] = {
 	0x0008,     /* R60 - PLL1 */
 	0x0031,     /* R61 - PLL2 */
 	0x0026,     /* R62 - PLL3 */
+	0x0000,	    /* R63 - Driver internal */
 };
 
 /*
@@ -126,10 +127,9 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
 
-	/* Reset register is uncached */
-	if (reg == 0)
+	/* Reset register and reserved registers are uncached */
+	if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
 		return;
 
 	cache[reg] = value;
@@ -1172,7 +1172,8 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
  * Set PCM DAI bit size and sample rate.
  */
 static int wm8990_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
@@ -1222,8 +1223,14 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		break;
+
 	case SND_SOC_BIAS_PREPARE:
+		/* VMID=2*50k */
+		val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+			~WM8990_VMID_MODE_MASK;
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2);
 		break;
+
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->bias_level == SND_SOC_BIAS_OFF) {
 			/* Enable all output discharge bits */
@@ -1272,10 +1279,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 
 			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
 			wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
-		} else {
-			/* ON -> standby */
 
+			/* Enable workaround for ADC clocking issue. */
+			wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2);
+			wm8990_write(codec, WM8990_EXT_CTL1, 0xa003);
+			wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0);
 		}
+
+		/* VMID=2*250k */
+		val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) &
+			~WM8990_VMID_MODE_MASK;
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4);
 		break;
 
 	case SND_SOC_BIAS_OFF:
@@ -1349,8 +1363,7 @@ struct snd_soc_dai wm8990_dai = {
 		.rates = WM8990_RATES,
 		.formats = WM8990_FORMATS,},
 	.ops = {
-		.hw_params = wm8990_hw_params,},
-	.dai_ops = {
+		.hw_params = wm8990_hw_params,
 		.digital_mute = wm8990_mute,
 		.set_fmt = wm8990_set_dai_fmt,
 		.set_clkdiv = wm8990_set_dai_clkdiv,
@@ -1449,7 +1462,7 @@ static int wm8990_init(struct snd_soc_device *socdev)
 
 	wm8990_add_controls(codec);
 	wm8990_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm8990: failed to register card\n");
 		goto card_err;
@@ -1630,6 +1643,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8990 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
 
+static int __init wm8990_modinit(void)
+{
+	return snd_soc_register_dai(&wm8990_dai);
+}
+module_init(wm8990_modinit);
+
+static void __exit wm8990_exit(void)
+{
+	snd_soc_unregister_dai(&wm8990_dai);
+}
+module_exit(wm8990_exit);
+
 MODULE_DESCRIPTION("ASoC WM8990 driver");
 MODULE_AUTHOR("Liam Girdwood");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h
index 0e192f3b078..7114ddc88b4 100644
--- a/sound/soc/codecs/wm8990.h
+++ b/sound/soc/codecs/wm8990.h
@@ -80,8 +80,8 @@
 #define WM8990_PLL3                             0x3E
 #define WM8990_INTDRIVBITS			0x3F
 
-#define WM8990_REGISTER_COUNT                   60
-#define WM8990_MAX_REGISTER                     0x3F
+#define WM8990_EXT_ACCESS_ENA			0x75
+#define WM8990_EXT_CTL1				0x7a
 
 /*
  * Field Definitions.
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index ffb471e420e..af83d629078 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -487,7 +487,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 	return 0;
 }
 
-static int ac97_prepare(struct snd_pcm_substream *substream)
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -507,7 +508,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
 	return ac97_write(codec, reg, runtime->rate);
 }
 
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -533,7 +535,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97_BUS,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "HiFi Playback",
 		.channels_min = 1,
@@ -688,7 +690,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 
 	ret = wm9712_reset(codec, 0);
 	if (ret < 0) {
-		printk(KERN_ERR "AC97 link error\n");
+		printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n");
 		goto reset_err;
 	}
 
@@ -698,7 +700,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm9712_add_controls(codec);
 	wm9712_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "wm9712: failed to register card\n");
 		goto reset_err;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 945b32ed988..f3ca8aaf013 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -928,11 +928,10 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
 }
 
 static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
 
 	switch (params_format(params)) {
@@ -954,11 +953,10 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
+static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = dai->codec;
 	u16 status;
 
 	/* Gracefully shut down the voice interface. */
@@ -969,12 +967,11 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream)
 	ac97_write(codec, AC97_EXTENDED_MID, status);
 }
 
-static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
+static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
+	struct snd_soc_codec *codec = dai->codec;
 	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;
 
@@ -989,12 +986,11 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream)
 	return ac97_write(codec, reg, runtime->rate);
 }
 
-static int ac97_aux_prepare(struct snd_pcm_substream *substream)
+static int ac97_aux_prepare(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
 {
+	struct snd_soc_codec *codec = dai->codec;
 	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);
@@ -1028,7 +1024,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
-	.type = SND_SOC_DAI_AC97_BUS,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "HiFi Playback",
 		.channels_min = 1,
@@ -1042,8 +1038,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
-		.prepare = ac97_hifi_prepare,},
-	.dai_ops = {
+		.prepare = ac97_hifi_prepare,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,},
 	},
@@ -1056,8 +1051,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
-		.prepare = ac97_aux_prepare,},
-	.dai_ops = {
+		.prepare = ac97_aux_prepare,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,},
 	},
@@ -1077,8 +1071,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.formats = WM9713_PCM_FORMATS,},
 	.ops = {
 		.hw_params = wm9713_pcm_hw_params,
-		.shutdown = wm9713_voiceshutdown,},
-	.dai_ops = {
+		.shutdown = wm9713_voiceshutdown,
 		.set_clkdiv = wm9713_set_dai_clkdiv,
 		.set_pll = wm9713_set_dai_pll,
 		.set_fmt = wm9713_set_dai_fmt,
@@ -1097,6 +1090,8 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
 	}
 
 	soc_ac97_ops.reset(codec->ac97);
+	if (soc_ac97_ops.warm_reset)
+		soc_ac97_ops.warm_reset(codec->ac97);
 	if (ac97_read(codec, 0) != wm9713_reg[0])
 		return -EIO;
 	return 0;
@@ -1240,7 +1235,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	wm9713_reset(codec, 0);
 	ret = wm9713_reset(codec, 1);
 	if (ret < 0) {
-		printk(KERN_ERR "AC97 link error\n");
+		printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n");
 		goto reset_err;
 	}
 
@@ -1252,7 +1247,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 
 	wm9713_add_controls(codec);
 	wm9713_add_widgets(codec);
-	ret = snd_soc_register_card(socdev);
+	ret = snd_soc_init_card(socdev);
 	if (ret < 0)
 		goto reset_err;
 	return 0;
@@ -1288,7 +1283,6 @@ static int wm9713_soc_remove(struct platform_device *pdev)
 	snd_soc_free_ac97_codec(codec);
 	kfree(codec->private_data);
 	kfree(codec->reg_cache);
-	kfree(codec->dai);
 	kfree(codec);
 	return 0;
 }
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 8f7e3383490..b502741692d 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -17,3 +17,13 @@ config SND_DAVINCI_SOC_EVM
 	help
 	  Say Y if you want to add support for SoC audio on TI
 	  DaVinci EVM platform.
+
+config SND_DAVINCI_SOC_SFFSDR
+	tristate "SoC Audio support for SFFSDR"
+	depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR
+	select SND_DAVINCI_SOC_I2S
+	select SND_SOC_PCM3008
+	select SFFSDR_FPGA
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Lyrtech SFFSDR board.
diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile
index ca772e5b463..ca8bae1fc3f 100644
--- a/sound/soc/davinci/Makefile
+++ b/sound/soc/davinci/Makefile
@@ -7,5 +7,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o
 
 # DAVINCI Machine Support
 snd-soc-evm-objs := davinci-evm.o
+snd-soc-sffsdr-objs := davinci-sffsdr.o
 
 obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o
+obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9e6062cd6b5..01b948bb55a 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -28,6 +28,8 @@
 
 #define EVM_CODEC_CLOCK 22579200
 
+#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
+		SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
 static int evm_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *params)
 {
@@ -37,14 +39,12 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 
 	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-					 SND_SOC_DAIFMT_CBM_CFM);
+	ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT);
 	if (ret < 0)
 		return ret;
 
 	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
-				       SND_SOC_DAIFMT_IB_NF);
+	ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
 	if (ret < 0)
 		return ret;
 
@@ -128,8 +128,9 @@ static struct snd_soc_dai_link evm_dai = {
 };
 
 /* davinci-evm audio machine driver */
-static struct snd_soc_machine snd_soc_machine_evm = {
+static struct snd_soc_card snd_soc_card_evm = {
 	.name = "DaVinci EVM",
+	.platform = &davinci_soc_platform,
 	.dai_link = &evm_dai,
 	.num_links = 1,
 };
@@ -142,8 +143,7 @@ static struct aic3x_setup_data evm_aic3x_setup = {
 
 /* evm audio subsystem */
 static struct snd_soc_device evm_snd_devdata = {
-	.machine = &snd_soc_machine_evm,
-	.platform = &davinci_soc_platform,
+	.card = &snd_soc_card_evm,
 	.codec_dev = &soc_codec_dev_aic3x,
 	.codec_data = &evm_aic3x_setup,
 };
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index abb5fedb0b1..0fee779e3c7 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -59,6 +59,7 @@
 #define DAVINCI_MCBSP_PCR_CLKXP		(1 << 1)
 #define DAVINCI_MCBSP_PCR_FSRP		(1 << 2)
 #define DAVINCI_MCBSP_PCR_FSXP		(1 << 3)
+#define DAVINCI_MCBSP_PCR_SCLKME	(1 << 7)
 #define DAVINCI_MCBSP_PCR_CLKRM		(1 << 8)
 #define DAVINCI_MCBSP_PCR_CLKXM		(1 << 9)
 #define DAVINCI_MCBSP_PCR_FSRM		(1 << 10)
@@ -110,16 +111,59 @@ static void davinci_mcbsp_start(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_platform *platform = socdev->card->platform;
 	u32 w;
+	int ret;
 
 	/* Start the sample generator and enable transmitter/receiver */
 	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
 	MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* Stop the DMA to avoid data loss */
+		/* while the transmitter is out of reset to handle XSYNCERR */
+		if (platform->pcm_ops->trigger) {
+			ret = platform->pcm_ops->trigger(substream,
+				SNDRV_PCM_TRIGGER_STOP);
+			if (ret < 0)
+				printk(KERN_DEBUG "Playback DMA stop failed\n");
+		}
+
+		/* Enable the transmitter */
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
 		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
-	else
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+		/* wait for any unexpected frame sync error to occur */
+		udelay(100);
+
+		/* Disable the transmitter to clear any outstanding XSYNCERR */
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+		/* Restart the DMA */
+		if (platform->pcm_ops->trigger) {
+			ret = platform->pcm_ops->trigger(substream,
+				SNDRV_PCM_TRIGGER_START);
+			if (ret < 0)
+				printk(KERN_DEBUG "Playback DMA start failed\n");
+		}
+		/* Enable the transmitter */
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+
+	} else {
+
+		/* Enable the reciever */
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
 		MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+	}
+
 
 	/* Start frame sync */
 	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
@@ -144,7 +188,8 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream)
 	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
 }
 
-static int davinci_i2s_startup(struct snd_pcm_substream *substream)
+static int davinci_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -155,61 +200,138 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
+#define DEFAULT_BITPERSAMPLE	16
+
 static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 				   unsigned int fmt)
 {
 	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
-	u32 w;
+	unsigned int pcr;
+	unsigned int srgr;
+	unsigned int rcr;
+	unsigned int xcr;
+	srgr = DAVINCI_MCBSP_SRGR_FSGM |
+		DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) |
+		DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1);
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG,
-					DAVINCI_MCBSP_PCR_FSXM |
-					DAVINCI_MCBSP_PCR_FSRM |
-					DAVINCI_MCBSP_PCR_CLKXM |
-					DAVINCI_MCBSP_PCR_CLKRM);
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG,
-					DAVINCI_MCBSP_SRGR_FSGM);
+		/* cpu is master */
+		pcr = DAVINCI_MCBSP_PCR_FSXM |
+			DAVINCI_MCBSP_PCR_FSRM |
+			DAVINCI_MCBSP_PCR_CLKXM |
+			DAVINCI_MCBSP_PCR_CLKRM;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		/* McBSP CLKR pin is the input for the Sample Rate Generator.
+		 * McBSP FSR and FSX are driven by the Sample Rate Generator. */
+		pcr = DAVINCI_MCBSP_PCR_SCLKME |
+			DAVINCI_MCBSP_PCR_FSXM |
+			DAVINCI_MCBSP_PCR_FSRM;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0);
+		/* codec is master */
+		pcr = 0;
 		break;
 	default:
+		printk(KERN_ERR "%s:bad master\n", __func__);
 		return -EINVAL;
 	}
 
-	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
-	case SND_SOC_DAIFMT_IB_NF:
-		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
-		MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
-			       DAVINCI_MCBSP_PCR_CLKRP, 1);
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+	rcr = DAVINCI_MCBSP_RCR_RFRLEN1(1);
+	xcr = DAVINCI_MCBSP_XCR_XFIG | DAVINCI_MCBSP_XCR_XFRLEN1(1);
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_B:
 		break;
-	case SND_SOC_DAIFMT_NB_IF:
-		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
-		MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP |
-			       DAVINCI_MCBSP_PCR_FSRP, 1);
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+	case SND_SOC_DAIFMT_I2S:
+		/* Davinci doesn't support TRUE I2S, but some codecs will have
+		 * the left and right channels contiguous. This allows
+		 * dsp_a mode to be used with an inverted normal frame clk.
+		 * If your codec is master and does not have contiguous
+		 * channels, then you will have sound on only one channel.
+		 * Try using a different mode, or codec as slave.
+		 *
+		 * The TLV320AIC33 is an example of a codec where this works.
+		 * It has a variable bit clock frequency allowing it to have
+		 * valid data on every bit clock.
+		 *
+		 * The TLV320AIC23 is an example of a codec where this does not
+		 * work. It has a fixed bit clock frequency with progressively
+		 * more empty bit clock slots between channels as the sample
+		 * rate is lowered.
+		 */
+		fmt ^= SND_SOC_DAIFMT_NB_IF;
+	case SND_SOC_DAIFMT_DSP_A:
+		rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1);
+		xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
+		break;
+	default:
+		printk(KERN_ERR "%s:bad format\n", __func__);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		/* CLKRP Receive clock polarity,
+		 *	1 - sampled on rising edge of CLKR
+		 *	valid on rising edge
+		 * CLKXP Transmit clock polarity,
+		 *	1 - clocked on falling edge of CLKX
+		 *	valid on rising edge
+		 * FSRP  Receive frame sync pol, 0 - active high
+		 * FSXP  Transmit frame sync pol, 0 - active high
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP);
 		break;
 	case SND_SOC_DAIFMT_IB_IF:
-		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG);
-		MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP |
-			       DAVINCI_MCBSP_PCR_CLKRP |
-			       DAVINCI_MCBSP_PCR_FSXP |
-			       DAVINCI_MCBSP_PCR_FSRP, 1);
-		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w);
+		/* CLKRP Receive clock polarity,
+		 *	0 - sampled on falling edge of CLKR
+		 *	valid on falling edge
+		 * CLKXP Transmit clock polarity,
+		 *	0 - clocked on rising edge of CLKX
+		 *	valid on falling edge
+		 * FSRP  Receive frame sync pol, 1 - active low
+		 * FSXP  Transmit frame sync pol, 1 - active low
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
 		break;
-	case SND_SOC_DAIFMT_NB_NF:
+	case SND_SOC_DAIFMT_NB_IF:
+		/* CLKRP Receive clock polarity,
+		 *	1 - sampled on rising edge of CLKR
+		 *	valid on rising edge
+		 * CLKXP Transmit clock polarity,
+		 *	1 - clocked on falling edge of CLKX
+		 *	valid on rising edge
+		 * FSRP  Receive frame sync pol, 1 - active low
+		 * FSXP  Transmit frame sync pol, 1 - active low
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP |
+			DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		/* CLKRP Receive clock polarity,
+		 *	0 - sampled on falling edge of CLKR
+		 *	valid on falling edge
+		 * CLKXP Transmit clock polarity,
+		 *	0 - clocked on rising edge of CLKX
+		 *	valid on falling edge
+		 * FSRP  Receive frame sync pol, 0 - active high
+		 * FSXP  Transmit frame sync pol, 0 - active high
+		 */
 		break;
 	default:
 		return -EINVAL;
 	}
-
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr);
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr);
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr);
 	return 0;
 }
 
 static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
@@ -219,25 +341,20 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 	u32 w;
 
 	/* general line settings */
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
-				DAVINCI_MCBSP_SPCR_RINTM(3) |
-				DAVINCI_MCBSP_SPCR_XINTM(3) |
-				DAVINCI_MCBSP_SPCR_FREE);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG,
-				DAVINCI_MCBSP_RCR_RFRLEN1(1) |
-				DAVINCI_MCBSP_RCR_RDATDLY(1));
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG,
-				DAVINCI_MCBSP_XCR_XFRLEN1(1) |
-				DAVINCI_MCBSP_XCR_XDATDLY(1) |
-				DAVINCI_MCBSP_XCR_XFIG);
+	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+	} else {
+		w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w);
+	}
 
 	i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
-	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
+	w = DAVINCI_MCBSP_SRGR_FSGM;
 	MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
 
 	i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
-	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
 	MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1);
 	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w);
 
@@ -260,20 +377,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
-	MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
-		       DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
+			       DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w);
 
-	w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
-	MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
-		       DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
-	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
+	} else {
+		w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG);
+		MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
+			       DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1);
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w);
 
+	}
 	return 0;
 }
 
-static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -299,8 +420,8 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
 	struct davinci_mcbsp_dev *dev;
 	struct resource *mem, *ioarea;
 	struct evm_snd_platform_data *pdata;
@@ -361,8 +482,8 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 			       struct snd_soc_dai *dai)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
 	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
 	struct resource *mem;
 
@@ -381,7 +502,6 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 struct snd_soc_dai davinci_i2s_dai = {
 	.name = "davinci-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = davinci_i2s_probe,
 	.remove = davinci_i2s_remove,
 	.playback = {
@@ -397,13 +517,24 @@ struct snd_soc_dai davinci_i2s_dai = {
 	.ops = {
 		.startup = davinci_i2s_startup,
 		.trigger = davinci_i2s_trigger,
-		.hw_params = davinci_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = davinci_i2s_hw_params,
 		.set_fmt = davinci_i2s_set_dai_fmt,
 	},
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
+static int __init davinci_i2s_init(void)
+{
+	return snd_soc_register_dai(&davinci_i2s_dai);
+}
+module_init(davinci_i2s_init);
+
+static void __exit davinci_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&davinci_i2s_dai);
+}
+module_exit(davinci_i2s_exit);
+
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 76feaa65737..74abc9b4f1c 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -14,6 +14,7 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/dma-mapping.h>
+#include <linux/kernel.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -24,13 +25,6 @@
 
 #include "davinci-pcm.h"
 
-#define DAVINCI_PCM_DEBUG 0
-#if DAVINCI_PCM_DEBUG
-#define DPRINTK(x...) printk(KERN_DEBUG x)
-#else
-#define DPRINTK(x...)
-#endif
-
 static struct snd_pcm_hardware davinci_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
@@ -78,8 +72,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	dma_offset = prtd->period * period_size;
 	dma_pos = runtime->dma_addr + dma_offset;
 
-	DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x "
-		"period_size=%x\n", lch, dma_pos, period_size);
+	pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
+		"dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size);
 
 	data_type = prtd->params->data_type;
 	count = period_size / data_type;
@@ -112,7 +106,7 @@ static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
 	struct snd_pcm_substream *substream = data;
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
 
-	DPRINTK("lch=%d, status=0x%x\n", lch, ch_status);
+	pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status);
 
 	if (unlikely(ch_status != DMA_COMPLETE))
 		return;
@@ -316,8 +310,8 @@ static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
 					   &buf->addr, GFP_KERNEL);
 
-	DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
-		(void *) buf->area, (void *) buf->addr, size);
+	pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, "
+		"size=%d\n", (void *) buf->area, (void *) buf->addr, size);
 
 	if (!buf->area)
 		return -ENOMEM;
@@ -384,6 +378,18 @@ struct snd_soc_platform davinci_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(davinci_soc_platform);
 
+static int __init davinci_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&davinci_soc_platform);
+}
+module_init(davinci_soc_platform_init);
+
+static void __exit davinci_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&davinci_soc_platform);
+}
+module_exit(davinci_soc_platform_exit);
+
 MODULE_AUTHOR("Vladimir Barinov");
 MODULE_DESCRIPTION("TI DAVINCI PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
new file mode 100644
index 00000000000..f67579d5276
--- /dev/null
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -0,0 +1,157 @@
+/*
+ * ASoC driver for Lyrtech SFFSDR board.
+ *
+ * Author:	Hugo Villeneuve
+ * Copyright (C) 2008 Lyrtech inc
+ *
+ * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow:
+ * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/dma.h>
+#include <asm/plat-sffsdr/sffsdr-fpga.h>
+
+#include <mach/mcbsp.h>
+#include <mach/edma.h>
+
+#include "../codecs/pcm3008.h"
+#include "davinci-pcm.h"
+#include "davinci-i2s.h"
+
+static int sffsdr_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int fs;
+	int ret = 0;
+
+	/* Set cpu DAI configuration:
+	 * CLKX and CLKR are the inputs for the Sample Rate Generator.
+	 * FSX and FSR are outputs, driven by the sample Rate Generator. */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_RIGHT_J |
+				  SND_SOC_DAIFMT_CBM_CFS |
+				  SND_SOC_DAIFMT_IB_NF);
+	if (ret < 0)
+		return ret;
+
+	/* Fsref can be 32000, 44100 or 48000. */
+	fs = params_rate(params);
+
+	pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs);
+
+	return sffsdr_fpga_set_codec_fs(fs);
+}
+
+static struct snd_soc_ops sffsdr_ops = {
+	.hw_params = sffsdr_hw_params,
+};
+
+/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sffsdr_dai = {
+	.name = "PCM3008", /* Codec name */
+	.stream_name = "PCM3008 HiFi",
+	.cpu_dai = &davinci_i2s_dai,
+	.codec_dai = &pcm3008_dai,
+	.ops = &sffsdr_ops,
+};
+
+/* davinci-sffsdr audio machine driver */
+static struct snd_soc_card snd_soc_sffsdr = {
+	.name = "DaVinci SFFSDR",
+	.platform = &davinci_soc_platform,
+	.dai_link = &sffsdr_dai,
+	.num_links = 1,
+};
+
+/* sffsdr audio private data */
+static struct pcm3008_setup_data sffsdr_pcm3008_setup = {
+	.dem0_pin = GPIO(45),
+	.dem1_pin = GPIO(46),
+	.pdad_pin = GPIO(47),
+	.pdda_pin = GPIO(38),
+};
+
+/* sffsdr audio subsystem */
+static struct snd_soc_device sffsdr_snd_devdata = {
+	.card = &snd_soc_sffsdr,
+	.codec_dev = &soc_codec_dev_pcm3008,
+	.codec_data = &sffsdr_pcm3008_setup,
+};
+
+static struct resource sffsdr_snd_resources[] = {
+	{
+		.start = DAVINCI_MCBSP_BASE,
+		.end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+		.flags = IORESOURCE_MEM,
+	},
+};
+
+static struct evm_snd_platform_data sffsdr_snd_data = {
+	.tx_dma_ch	= DAVINCI_DMA_MCBSP_TX,
+	.rx_dma_ch	= DAVINCI_DMA_MCBSP_RX,
+};
+
+static struct platform_device *sffsdr_snd_device;
+
+static int __init sffsdr_init(void)
+{
+	int ret;
+
+	sffsdr_snd_device = platform_device_alloc("soc-audio", 0);
+	if (!sffsdr_snd_device) {
+		printk(KERN_ERR "platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata);
+	sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev;
+	sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data;
+
+	ret = platform_device_add_resources(sffsdr_snd_device,
+					    sffsdr_snd_resources,
+					    ARRAY_SIZE(sffsdr_snd_resources));
+	if (ret) {
+		printk(KERN_ERR "platform device add ressources failed\n");
+		goto error;
+	}
+
+	ret = platform_device_add(sffsdr_snd_device);
+	if (ret)
+		goto error;
+
+	return ret;
+
+error:
+	platform_device_put(sffsdr_snd_device);
+	return ret;
+}
+
+static void __exit sffsdr_exit(void)
+{
+	platform_device_unregister(sffsdr_snd_device);
+}
+
+module_init(sffsdr_init);
+module_exit(sffsdr_exit);
+
+MODULE_AUTHOR("Hugo Villeneuve");
+MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 8d73edc5610..95c12b26fe3 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -20,7 +20,7 @@ config SND_SOC_MPC8610_HPCD
 
 config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
-	depends on SND_SOC && PPC_MPC52xx && PPC_BESTCOMM
+	depends on PPC_MPC52xx && PPC_BESTCOMM
 	select SND_SOC_OF_SIMPLE
 	select PPC_BESTCOMM_GEN_BD
 	help
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index d2d3da9729f..64993eda567 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -284,7 +284,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
  * fsl_dma_new: initialize this PCM driver.
  *
  * This function is called when the codec driver calls snd_soc_new_pcms(),
- * once for each .dai_link in the machine driver's snd_soc_machine
+ * once for each .dai_link in the machine driver's snd_soc_card
  * structure.
  */
 static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
@@ -853,6 +853,18 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info)
 }
 EXPORT_SYMBOL_GPL(fsl_dma_configure);
 
+static int __init fsl_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&fsl_soc_platform);
+}
+module_init(fsl_soc_platform_init);
+
+static void __exit fsl_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&fsl_soc_platform);
+}
+module_exit(fsl_soc_platform_exit);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale Elo DMA ASoC PCM module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 157a7895ffa..c6d6eb71dc1 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -266,7 +266,8 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id)
  * If this is the first stream open, then grab the IRQ and program most of
  * the SSI registers.
  */
-static int fsl_ssi_startup(struct snd_pcm_substream *substream)
+static int fsl_ssi_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -411,7 +412,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream)
  * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
  * clock master.
  */
-static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
+static int fsl_ssi_prepare(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -441,7 +443,8 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream)
  * The DMA channel is in external master start and pause mode, which
  * means the SSI completely controls the flow of data.
  */
-static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -490,7 +493,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd)
  *
  * Shutdown the SSI if there are no other substreams open.
  */
-static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
+static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
@@ -578,8 +582,6 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 		.prepare = fsl_ssi_prepare,
 		.shutdown = fsl_ssi_shutdown,
 		.trigger = fsl_ssi_trigger,
-	},
-	.dai_ops = {
 		.set_sysclk = fsl_ssi_set_sysclk,
 		.set_fmt = fsl_ssi_set_fmt,
 	},
@@ -671,6 +673,14 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	fsl_ssi_dai->private_data = ssi_private;
 	fsl_ssi_dai->name = ssi_private->name;
 	fsl_ssi_dai->id = ssi_info->id;
+	fsl_ssi_dai->dev = ssi_info->dev;
+
+	ret = snd_soc_register_dai(fsl_ssi_dai);
+	if (ret != 0) {
+		dev_err(ssi_info->dev, "failed to register DAI: %d\n", ret);
+		kfree(fsl_ssi_dai);
+		return NULL;
+	}
 
 	return fsl_ssi_dai;
 }
@@ -688,6 +698,8 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
 
 	device_remove_file(ssi_private->dev, &ssi_private->dev_attr);
 
+	snd_soc_unregister_dai(&ssi_private->cpu_dai);
+
 	kfree(ssi_private);
 }
 EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 94a02eaa482..9eb1ce185bd 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -187,7 +187,8 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
  * If this is the first stream open, then grab the IRQ and program most of
  * the PSC registers.
  */
-static int psc_i2s_startup(struct snd_pcm_substream *substream)
+static int psc_i2s_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -220,7 +221,8 @@ static int psc_i2s_startup(struct snd_pcm_substream *substream)
 }
 
 static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -256,7 +258,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
+static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
 {
 	snd_pcm_set_runtime_buffer(substream, NULL);
 	return 0;
@@ -268,7 +271,8 @@ static int psc_i2s_hw_free(struct snd_pcm_substream *substream)
  * This function is called by ALSA to start, stop, pause, and resume the DMA
  * transfer of data.
  */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -383,7 +387,8 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
  *
  * Shutdown the PSC if there are no other substreams open.
  */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream)
+static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
@@ -464,7 +469,6 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
 static struct snd_soc_dai psc_i2s_dai_template = {
-	.type = SND_SOC_DAI_I2S,
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
@@ -483,8 +487,6 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.hw_free = psc_i2s_hw_free,
 		.shutdown = psc_i2s_shutdown,
 		.trigger = psc_i2s_trigger,
-	},
-	.dai_ops = {
 		.set_sysclk = psc_i2s_set_sysclk,
 		.set_fmt = psc_i2s_set_fmt,
 	},
@@ -826,6 +828,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 	if (rc)
 		dev_info(psc_i2s->dev, "error creating sysfs files\n");
 
+	snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
+
 	/* Tell the ASoC OF helpers about it */
 	of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
 				     &psc_i2s->dai);
@@ -839,6 +843,8 @@ static int __devexit psc_i2s_of_remove(struct of_device *op)
 
 	dev_dbg(&op->dev, "psc_i2s_remove()\n");
 
+	snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
+
 	bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
 	bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 94f89debde1..bcec3f60bad 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -29,7 +29,7 @@
 struct mpc8610_hpcd_data {
 	struct snd_soc_device sound_devdata;
 	struct snd_soc_dai_link dai;
-	struct snd_soc_machine machine;
+	struct snd_soc_card machine;
 	unsigned int dai_format;
 	unsigned int codec_clk_direction;
 	unsigned int cpu_clk_direction;
@@ -185,7 +185,7 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
 /**
  * mpc8610_hpcd_machine: ASoC machine data
  */
-static struct snd_soc_machine mpc8610_hpcd_machine = {
+static struct snd_soc_card mpc8610_hpcd_machine = {
 	.probe = mpc8610_hpcd_machine_probe,
 	.remove = mpc8610_hpcd_machine_remove,
 	.name = "MPC8610 HPCD",
@@ -465,9 +465,9 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 		goto error;
 	}
 
-	machine_data->sound_devdata.machine = &mpc8610_hpcd_machine;
+	machine_data->sound_devdata.card = &mpc8610_hpcd_machine;
 	machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270;
-	machine_data->sound_devdata.platform = &fsl_soc_platform;
+	machine_data->machine.platform = &fsl_soc_platform;
 
 	sound_device->dev.platform_data = machine_data;
 
diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c
index 0382fdac51c..8bc5cd9e972 100644
--- a/sound/soc/fsl/soc-of-simple.c
+++ b/sound/soc/fsl/soc-of-simple.c
@@ -31,7 +31,7 @@ struct of_snd_soc_device {
 	int id;
 	struct list_head list;
 	struct snd_soc_device device;
-	struct snd_soc_machine machine;
+	struct snd_soc_card card;
 	struct snd_soc_dai_link dai_link;
 	struct platform_device *pdev;
 	struct device_node *platform_node;
@@ -58,9 +58,9 @@ of_snd_soc_get_device(struct device_node *codec_node)
 	/* Initialize the structure and add it to the global list */
 	of_soc->codec_node = codec_node;
 	of_soc->id = of_snd_soc_next_index++;
-	of_soc->machine.dai_link = &of_soc->dai_link;
-	of_soc->machine.num_links = 1;
-	of_soc->device.machine = &of_soc->machine;
+	of_soc->card.dai_link = &of_soc->dai_link;
+	of_soc->card.num_links = 1;
+	of_soc->device.card = &of_soc->card;
 	of_soc->dai_link.ops = &of_snd_soc_ops;
 	list_add(&of_soc->list, &of_snd_soc_device_list);
 
@@ -158,8 +158,8 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform,
 
 	of_soc->platform_node = node;
 	of_soc->dai_link.cpu_dai = cpu_dai;
-	of_soc->device.platform = platform;
-	of_soc->machine.name = of_soc->dai_link.cpu_dai->name;
+	of_soc->card.platform = platform;
+	of_soc->card.name = of_soc->dai_link.cpu_dai->name;
 
 	/* Now try to register the SoC device */
 	of_snd_soc_register_device(of_soc);
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 8b7766b998d..a7b1d77b210 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,6 +1,6 @@
 config SND_OMAP_SOC
 	tristate "SoC Audio for the Texas Instruments OMAP chips"
-	depends on ARCH_OMAP && SND_SOC
+	depends on ARCH_OMAP
 
 config SND_OMAP_SOC_MCBSP
 	tristate
@@ -21,3 +21,36 @@ config SND_OMAP_SOC_OSK5912
 	select SND_SOC_TLV320AIC23
 	help
 	  Say Y if you want to add support for SoC audio on osk5912.
+
+config SND_OMAP_SOC_OVERO
+	tristate "SoC Audio support for Gumstix Overo"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the Gumstix Overo.
+
+config SND_OMAP_SOC_OMAP2EVM
+	tristate "SoC Audio support for OMAP2EVM board"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the omap2evm board.
+
+config SND_OMAP_SOC_SDP3430
+	tristate "SoC Audio support for Texas Instruments SDP3430"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on Texas Instruments
+	  SDP3430.
+
+config SND_OMAP_SOC_OMAP3_PANDORA
+	tristate "SoC Audio support for OMAP3 Pandora"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index e09d1f297f6..76fedd96e36 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -8,6 +8,14 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
 snd-soc-osk5912-objs := osk5912.o
+snd-soc-overo-objs := overo.o
+snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-sdp3430-objs := sdp3430.o
+snd-soc-omap3pandora-objs := omap3pandora.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
+obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
+obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index fae3ad36e0b..25593fee912 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -70,9 +70,13 @@ static void n810_ext_control(struct snd_soc_codec *codec)
 
 static int n810_startup(struct snd_pcm_substream *substream)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->socdev->codec;
 
+	snd_pcm_hw_constraint_minmax(runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+
 	n810_ext_control(codec);
 	return clk_enable(sys_clkout2);
 }
@@ -282,8 +286,9 @@ static struct snd_soc_dai_link n810_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_n810 = {
+static struct snd_soc_card snd_soc_n810 = {
 	.name = "N810",
+	.platform = &omap_soc_platform,
 	.dai_link = &n810_dai,
 	.num_links = 1,
 };
@@ -298,8 +303,7 @@ static struct aic3x_setup_data n810_aic33_setup = {
 
 /* Audio subsystem */
 static struct snd_soc_device n810_snd_devdata = {
-	.machine = &snd_soc_machine_n810,
-	.platform = &omap_soc_platform,
+	.card = &snd_soc_n810,
 	.codec_dev = &soc_codec_dev_aic3x,
 	.codec_data = &n810_aic33_setup,
 };
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8485a8a9d0f..ec5e18a7875 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -36,9 +36,7 @@
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
 
-#define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_44100 | \
-				 SNDRV_PCM_RATE_48000 | \
-				 SNDRV_PCM_RATE_KNOT)
+#define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_8000_96000)
 
 struct omap_mcbsp_data {
 	unsigned int			bus_id;
@@ -140,7 +138,8 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
 static const unsigned long omap34xx_mcbsp_port[][2] = {};
 #endif
 
-static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -153,7 +152,8 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
 	return err;
 }
 
-static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -165,7 +165,8 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
 	}
 }
 
-static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -194,14 +195,15 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
-				    struct snd_pcm_hw_params *params)
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
-	int wlen;
+	int wlen, channels;
 	unsigned long port;
 
 	if (cpu_class_is_omap1()) {
@@ -230,12 +232,17 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		return 0;
 	}
 
-	switch (params_channels(params)) {
+	channels = params_channels(params);
+	switch (channels) {
 	case 2:
-		/* Set 1 word per (McBPSP) frame and use dual-phase frames */
-		regs->rcr2	|= RFRLEN2(1 - 1) | RPHASE;
+		/* Use dual-phase frames */
+		regs->rcr2	|= RPHASE;
+		regs->xcr2	|= XPHASE;
+	case 1:
+		/* Set 1 word per (McBSP) frame */
+		regs->rcr2	|= RFRLEN2(1 - 1);
 		regs->rcr1	|= RFRLEN1(1 - 1);
-		regs->xcr2	|= XFRLEN2(1 - 1) | XPHASE;
+		regs->xcr2	|= XFRLEN2(1 - 1);
 		regs->xcr1	|= XFRLEN1(1 - 1);
 		break;
 	default:
@@ -263,9 +270,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		regs->srgr2	|= FPER(wlen * 2 - 1);
 		regs->srgr1	|= FWID(wlen - 1);
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
-		regs->srgr2	|= FPER(wlen * 2 - 1);
-		regs->srgr1	|= FWID(wlen * 2 - 2);
+	case SND_SOC_DAIFMT_DSP_B:
+		regs->srgr2	|= FPER(wlen * channels - 1);
+		regs->srgr1	|= FWID(wlen * channels - 2);
 		break;
 	}
 
@@ -302,7 +309,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		regs->rcr2	|= RDATDLY(1);
 		regs->xcr2	|= XDATDLY(1);
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
+	case SND_SOC_DAIFMT_DSP_B:
 		/* 0-bit data delay */
 		regs->rcr2      |= RDATDLY(0);
 		regs->xcr2      |= XDATDLY(0);
@@ -452,17 +459,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 
 #define OMAP_MCBSP_DAI_BUILDER(link_id)				\
 {								\
-	.name = "omap-mcbsp-dai-(link_id)",			\
+	.name = "omap-mcbsp-dai-"#link_id,			\
 	.id = (link_id),					\
-	.type = SND_SOC_DAI_I2S,				\
 	.playback = {						\
-		.channels_min = 2,				\
+		.channels_min = 1,				\
 		.channels_max = 2,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
 	.capture = {						\
-		.channels_min = 2,				\
+		.channels_min = 1,				\
 		.channels_max = 2,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
@@ -472,8 +478,6 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		.shutdown = omap_mcbsp_dai_shutdown,		\
 		.trigger = omap_mcbsp_dai_trigger,		\
 		.hw_params = omap_mcbsp_dai_hw_params,		\
-	},							\
-	.dai_ops = {						\
 		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
 		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
 		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
@@ -495,6 +499,19 @@ struct snd_soc_dai omap_mcbsp_dai[] = {
 
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
+static int __init snd_omap_mcbsp_init(void)
+{
+	return snd_soc_register_dais(omap_mcbsp_dai,
+				     ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_init(snd_omap_mcbsp_init);
+
+static void __exit snd_omap_mcbsp_exit(void)
+{
+	snd_soc_unregister_dais(omap_mcbsp_dai, ARRAY_SIZE(omap_mcbsp_dai));
+}
+module_exit(snd_omap_mcbsp_exit);
+
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index acd68efb2b7..b0362dfd5b7 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	prtd->dma_data = dma_data;
 	err = omap_request_dma(dma_data->dma_req, dma_data->name,
 			       omap_pcm_dma_irq, substream, &prtd->dma_ch);
-	if (!err & !cpu_is_omap1510()) {
+	if (!err && !cpu_is_omap1510()) {
 		/*
 		 * Link channel with itself so DMA doesn't need any
 		 * reprogramming while looping the buffer
@@ -354,6 +354,18 @@ struct snd_soc_platform omap_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(omap_soc_platform);
 
+static int __init omap_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&omap_soc_platform);
+}
+module_init(omap_soc_platform_init);
+
+static void __exit omap_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&omap_soc_platform);
+}
+module_exit(omap_soc_platform_exit);
+
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
new file mode 100644
index 00000000000..0c2322dcf02
--- /dev/null
+++ b/sound/soc/omap/omap2evm.c
@@ -0,0 +1,151 @@
+/*
+ * omap2evm.c  --  SoC audio machine driver for omap2evm board
+ *
+ * Author: Arun KS <arunks@mistralsolutions.com>
+ *
+ * Based on sound/soc/omap/overo.c by Steve Sakoman
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap2evm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap2evm_ops = {
+	.hw_params = omap2evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap2evm_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &omap2evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap2evm = {
+	.name = "omap2evm",
+	.platform = &omap_soc_platform,
+	.dai_link = &omap2evm_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap2evm_snd_devdata = {
+	.card = &snd_soc_omap2evm,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap2evm_snd_device;
+
+static int __init omap2evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap2evm()) {
+		pr_debug("Not omap2evm!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "omap2evm SoC init\n");
+
+	omap2evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap2evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata);
+	omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev;
+	*(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(omap2evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap2evm_snd_device);
+
+	return ret;
+}
+module_init(omap2evm_soc_init);
+
+static void __exit omap2evm_soc_exit(void)
+{
+	platform_device_unregister(omap2evm_snd_device);
+}
+module_exit(omap2evm_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC omap2evm");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
new file mode 100644
index 00000000000..fd24a4acd2f
--- /dev/null
+++ b/sound/soc/omap/omap3beagle.c
@@ -0,0 +1,149 @@
+/*
+ * omap3beagle.c  --  SoC audio for OMAP3 Beagle
+ *
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3beagle_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_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap3beagle_ops = {
+	.hw_params = omap3beagle_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3beagle_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &omap3beagle_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3beagle = {
+	.name = "omap3beagle",
+	.platform = &omap_soc_platform,
+	.dai_link = &omap3beagle_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3beagle_snd_devdata = {
+	.card = &snd_soc_omap3beagle,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3beagle_snd_device;
+
+static int __init omap3beagle_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3_beagle()) {
+		pr_debug("Not OMAP3 Beagle!\n");
+		return -ENODEV;
+	}
+	pr_info("OMAP3 Beagle SoC init\n");
+
+	omap3beagle_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!omap3beagle_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(omap3beagle_snd_device, &omap3beagle_snd_devdata);
+	omap3beagle_snd_devdata.dev = &omap3beagle_snd_device->dev;
+	*(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(omap3beagle_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(omap3beagle_snd_device);
+
+	return ret;
+}
+
+static void __exit omap3beagle_soc_exit(void)
+{
+	platform_device_unregister(omap3beagle_snd_device);
+}
+
+module_init(omap3beagle_soc_init);
+module_exit(omap3beagle_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
new file mode 100644
index 00000000000..bd91594496b
--- /dev/null
+++ b/sound/soc/omap/omap3pandora.c
@@ -0,0 +1,311 @@
+/*
+ * omap3pandora.c  --  SoC audio for Pandora Handheld Console
+ *
+ * Author: Gražvydas Ignotas <notasas@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+#define OMAP3_PANDORA_DAC_POWER_GPIO	118
+#define OMAP3_PANDORA_AMP_POWER_GPIO	14
+
+#define PREFIX "ASoC omap3pandora: "
+
+static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
+	struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set codec system clock\n");
+		return ret;
+	}
+
+	/* Set McBSP clock to external */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set cpu system clock\n");
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set SRG clock divider\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int omap3pandora_out_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_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+					  SND_SOC_DAIFMT_I2S |
+					  SND_SOC_DAIFMT_IB_NF |
+					  SND_SOC_DAIFMT_CBS_CFS);
+}
+
+static int omap3pandora_in_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_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+					  SND_SOC_DAIFMT_I2S |
+					  SND_SOC_DAIFMT_NB_NF |
+					  SND_SOC_DAIFMT_CBS_CFS);
+}
+
+static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
+		gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1);
+	} else {
+		gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+		mdelay(1);
+		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+	}
+
+	return 0;
+}
+
+/*
+ * Audio paths on Pandora board:
+ *
+ *  |O| ---> PCM DAC +-> AMP -> Headphone Jack
+ *  |M|         A    +--------> Line Out
+ *  |A| <~~clk~~+
+ *  |P| <--- TWL4030 <--------- Line In and MICs
+ */
+static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
+			   0, 0, NULL, 0, omap3pandora_hp_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+};
+
+static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_MIC("Mic (external)", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+	{"Headphone Amplifier", NULL, "PCM DAC"},
+	{"Line Out", NULL, "PCM DAC"},
+	{"Headphone Jack", NULL, "Headphone Amplifier"},
+};
+
+static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
+	{"INL", NULL, "Line In"},
+	{"INR", NULL, "Line In"},
+	{"INL", NULL, "Mic (Internal)"},
+	{"INR", NULL, "Mic (external)"},
+};
+
+static int omap3pandora_out_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
+				ARRAY_SIZE(omap3pandora_out_dapm_widgets));
+	if (ret < 0)
+		return ret;
+
+	snd_soc_dapm_add_routes(codec, omap3pandora_out_map,
+		ARRAY_SIZE(omap3pandora_out_map));
+
+	return snd_soc_dapm_sync(codec);
+}
+
+static int omap3pandora_in_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets,
+				ARRAY_SIZE(omap3pandora_in_dapm_widgets));
+	if (ret < 0)
+		return ret;
+
+	snd_soc_dapm_add_routes(codec, omap3pandora_in_map,
+		ARRAY_SIZE(omap3pandora_in_map));
+
+	return snd_soc_dapm_sync(codec);
+}
+
+static struct snd_soc_ops omap3pandora_out_ops = {
+	.hw_params = omap3pandora_out_hw_params,
+};
+
+static struct snd_soc_ops omap3pandora_in_ops = {
+	.hw_params = omap3pandora_in_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3pandora_dai[] = {
+	{
+		.name = "PCM1773",
+		.stream_name = "HiFi Out",
+		.cpu_dai = &omap_mcbsp_dai[0],
+		.codec_dai = &twl4030_dai,
+		.ops = &omap3pandora_out_ops,
+		.init = omap3pandora_out_init,
+	}, {
+		.name = "TWL4030",
+		.stream_name = "Line/Mic In",
+		.cpu_dai = &omap_mcbsp_dai[1],
+		.codec_dai = &twl4030_dai,
+		.ops = &omap3pandora_in_ops,
+		.init = omap3pandora_in_init,
+	}
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_omap3pandora = {
+	.name = "omap3pandora",
+	.platform = &omap_soc_platform,
+	.dai_link = omap3pandora_dai,
+	.num_links = ARRAY_SIZE(omap3pandora_dai),
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3pandora_snd_data = {
+	.card = &snd_soc_card_omap3pandora,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3pandora_snd_device;
+
+static int __init omap3pandora_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3_pandora()) {
+		pr_debug(PREFIX "Not OMAP3 Pandora\n");
+		return -ENODEV;
+	}
+	pr_info("OMAP3 Pandora SoC init\n");
+
+	ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
+	if (ret) {
+		pr_err(PREFIX "Failed to get DAC power GPIO\n");
+		return ret;
+	}
+
+	ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+	if (ret) {
+		pr_err(PREFIX "Failed to set DAC power GPIO direction\n");
+		goto fail0;
+	}
+
+	ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power");
+	if (ret) {
+		pr_err(PREFIX "Failed to get amp power GPIO\n");
+		goto fail0;
+	}
+
+	ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+	if (ret) {
+		pr_err(PREFIX "Failed to set amp power GPIO direction\n");
+		goto fail1;
+	}
+
+	omap3pandora_snd_device = platform_device_alloc("soc-audio", -1);
+	if (omap3pandora_snd_device == NULL) {
+		pr_err(PREFIX "Platform device allocation failed\n");
+		ret = -ENOMEM;
+		goto fail1;
+	}
+
+	platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data);
+	omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev;
+	*(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */
+	*(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */
+
+	ret = platform_device_add(omap3pandora_snd_device);
+	if (ret) {
+		pr_err(PREFIX "Unable to add platform device\n");
+		goto fail2;
+	}
+
+	return 0;
+
+fail2:
+	platform_device_put(omap3pandora_snd_device);
+fail1:
+	gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+fail0:
+	gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+	return ret;
+}
+module_init(omap3pandora_soc_init);
+
+static void __exit omap3pandora_soc_exit(void)
+{
+	platform_device_unregister(omap3pandora_snd_device);
+	gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+	gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+}
+module_exit(omap3pandora_soc_exit);
+
+MODULE_AUTHOR("Grazvydas Ignotas <notasas@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index 0fe73379689..cd41a948df7 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -61,7 +61,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 
 	/* Set codec DAI configuration */
 	err = snd_soc_dai_set_fmt(codec_dai,
-				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_DSP_B |
 				  SND_SOC_DAIFMT_NB_IF |
 				  SND_SOC_DAIFMT_CBM_CFM);
 	if (err < 0) {
@@ -71,7 +71,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 
 	/* Set cpu DAI configuration */
 	err = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_DSP_A |
+				  SND_SOC_DAIFMT_DSP_B |
 				  SND_SOC_DAIFMT_NB_IF |
 				  SND_SOC_DAIFMT_CBM_CFM);
 	if (err < 0) {
@@ -143,16 +143,16 @@ static struct snd_soc_dai_link osk_dai = {
 };
 
 /* Audio machine driver */
-static struct snd_soc_machine snd_soc_machine_osk = {
+static struct snd_soc_card snd_soc_card_osk = {
 	.name = "OSK5912",
+	.platform = &omap_soc_platform,
 	.dai_link = &osk_dai,
 	.num_links = 1,
 };
 
 /* Audio subsystem */
 static struct snd_soc_device osk_snd_devdata = {
-	.machine = &snd_soc_machine_osk,
-	.platform = &omap_soc_platform,
+	.card = &snd_soc_card_osk,
 	.codec_dev = &soc_codec_dev_tlv320aic23,
 };
 
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
new file mode 100644
index 00000000000..a72dc4e159e
--- /dev/null
+++ b/sound/soc/omap/overo.c
@@ -0,0 +1,148 @@
+/*
+ * overo.c  --  SoC audio for Gumstix Overo
+ *
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int overo_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_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops overo_ops = {
+	.hw_params = overo_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link overo_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &overo_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_overo = {
+	.name = "overo",
+	.platform = &omap_soc_platform,
+	.dai_link = &overo_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device overo_snd_devdata = {
+	.card = &snd_soc_card_overo,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *overo_snd_device;
+
+static int __init overo_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_overo()) {
+		pr_debug("Not Overo!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "overo SoC init\n");
+
+	overo_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!overo_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(overo_snd_device, &overo_snd_devdata);
+	overo_snd_devdata.dev = &overo_snd_device->dev;
+	*(unsigned int *)overo_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(overo_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(overo_snd_device);
+
+	return ret;
+}
+module_init(overo_soc_init);
+
+static void __exit overo_soc_exit(void)
+{
+	platform_device_unregister(overo_snd_device);
+}
+module_exit(overo_soc_exit);
+
+MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>");
+MODULE_DESCRIPTION("ALSA SoC overo");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
new file mode 100644
index 00000000000..ad97836818b
--- /dev/null
+++ b/sound/soc/omap/sdp3430.c
@@ -0,0 +1,152 @@
+/*
+ * sdp3430.c  --  SoC audio for TI OMAP3430 SDP
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.com>
+ *
+ * Based on:
+ * Author: Steve Sakoman <steve@sakoman.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int sdp3430_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_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops sdp3430_ops = {
+	.hw_params = sdp3430_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link sdp3430_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai,
+	.ops = &sdp3430_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_sdp3430 = {
+	.name = "SDP3430",
+	.platform = &omap_soc_platform,
+	.dai_link = &sdp3430_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device sdp3430_snd_devdata = {
+	.machine = &snd_soc_machine_sdp3430,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *sdp3430_snd_device;
+
+static int __init sdp3430_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap_3430sdp()) {
+		pr_debug("Not SDP3430!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "SDP3430 SoC init\n");
+
+	sdp3430_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!sdp3430_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata);
+	sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev;
+	*(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(sdp3430_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(sdp3430_snd_device);
+
+	return ret;
+}
+module_init(sdp3430_soc_init);
+
+static void __exit sdp3430_soc_exit(void)
+{
+	platform_device_unregister(sdp3430_snd_device);
+}
+module_exit(sdp3430_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC SDP3430");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f8c1cdd940a..f82e1069947 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -21,6 +21,9 @@ config SND_PXA2XX_SOC_AC97
 config SND_PXA2XX_SOC_I2S
 	tristate
 
+config SND_PXA_SOC_SSP
+	tristate
+
 config SND_PXA2XX_SOC_CORGI
 	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
 	depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
@@ -75,3 +78,22 @@ config SND_PXA2XX_SOC_EM_X270
 	help
 	  Say Y if you want to add support for SoC audio on
 	  CompuLab EM-x270.
+
+config SND_PXA2XX_SOC_PALM27X
+	bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
+	depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5)
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  Palm T|X, T5 or LifeDrive handheld computer.
+
+config SND_SOC_ZYLONITE
+	tristate "SoC Audio support for Marvell Zylonite"
+	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
+	select SND_PXA2XX_SOC_AC97
+	select SND_PXA_SOC_SSP
+	select SND_SOC_WM9713
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Zylonite reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 5bc8edf9dca..08a9f279772 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -2,10 +2,12 @@
 snd-soc-pxa2xx-objs := pxa2xx-pcm.o
 snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
 snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
+snd-soc-pxa-ssp-objs := pxa-ssp.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
 obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
 obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
+obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
 
 # PXA Machine Support
 snd-soc-corgi-objs := corgi.o
@@ -14,6 +16,8 @@ snd-soc-tosa-objs := tosa.o
 snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
+snd-soc-palm27x-objs := palm27x.o
+snd-soc-zylonite-objs := zylonite.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -21,3 +25,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
+obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 2718eaf7895..1ba25a55952 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -108,15 +108,11 @@ static int corgi_startup(struct snd_pcm_substream *substream)
 }
 
 /* we need to unmute the HP at shutdown as the mute burns power on corgi */
-static int corgi_shutdown(struct snd_pcm_substream *substream)
+static void 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 */
 	gpio_set_value(CORGI_GPIO_MUTE_L, 1);
 	gpio_set_value(CORGI_GPIO_MUTE_R, 1);
-	return 0;
 }
 
 static int corgi_hw_params(struct snd_pcm_substream *substream,
@@ -314,8 +310,9 @@ static struct snd_soc_dai_link corgi_dai = {
 };
 
 /* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &corgi_dai,
 	.num_links = 1,
 };
@@ -328,8 +325,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
 
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
-	.machine = &snd_soc_machine_corgi,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_corgi,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &corgi_wm8731_setup,
 };
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 6781c5be242..2e3386dfa0f 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -29,7 +29,7 @@
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_machine e800;
+static struct snd_soc_card e800;
 
 static struct snd_soc_dai_link e800_dai[] = {
 {
@@ -40,15 +40,15 @@ static struct snd_soc_dai_link e800_dai[] = {
 },
 };
 
-static struct snd_soc_machine e800 = {
+static struct snd_soc_card e800 = {
 	.name = "Toshiba e800",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = e800_dai,
 	.num_links = ARRAY_SIZE(e800_dai),
 };
 
 static struct snd_soc_device e800_snd_devdata = {
-	.machine = &e800,
-	.platform = &pxa2xx_soc_platform,
+	.card = &e800,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index e6ff6929ab4..fe4a729ea64 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -23,7 +23,6 @@
 #include <linux/moduleparam.h>
 #include <linux/device.h>
 
-#include <sound/driver.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
@@ -53,15 +52,15 @@ static struct snd_soc_dai_link em_x270_dai[] = {
 	},
 };
 
-static struct snd_soc_machine em_x270 = {
+static struct snd_soc_card em_x270 = {
 	.name = "EM-X270",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = em_x270_dai,
 	.num_links = ARRAY_SIZE(em_x270_dai),
 };
 
 static struct snd_soc_device em_x270_snd_devdata = {
-	.machine = &em_x270,
-	.platform = &pxa2xx_soc_platform,
+	.card = &em_x270,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
new file mode 100644
index 00000000000..4a9cf3083af
--- /dev/null
+++ b/sound/soc/pxa/palm27x.c
@@ -0,0 +1,269 @@
+/*
+ * linux/sound/soc/pxa/palm27x.c
+ *
+ * SoC Audio driver for Palm T|X, T5 and LifeDrive
+ *
+ * based on tosa.c
+ *
+ * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <linux/interrupt.h>
+#include <linux/irq.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+#include <mach/palmasoc.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int palm27x_jack_func = 1;
+static int palm27x_spk_func = 1;
+static int palm27x_ep_gpio = -1;
+
+static void palm27x_ext_control(struct snd_soc_codec *codec)
+{
+	if (!palm27x_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Speaker");
+	else
+		snd_soc_dapm_disable_pin(codec, "Speaker");
+
+	if (!palm27x_jack_func)
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+	snd_soc_dapm_sync(codec);
+}
+
+static int palm27x_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 */
+	palm27x_ext_control(codec);
+	return 0;
+}
+
+static struct snd_soc_ops palm27x_ops = {
+	.startup = palm27x_startup,
+};
+
+static irqreturn_t palm27x_interrupt(int irq, void *v)
+{
+	palm27x_spk_func = gpio_get_value(palm27x_ep_gpio);
+	palm27x_jack_func = !palm27x_spk_func;
+	return IRQ_HANDLED;
+}
+
+static int palm27x_get_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_jack_func;
+	return 0;
+}
+
+static int palm27x_set_jack(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_jack_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+static int palm27x_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = palm27x_spk_func;
+	return 0;
+}
+
+static int palm27x_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (palm27x_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	palm27x_spk_func = ucontrol->value.integer.value[0];
+	palm27x_ext_control(codec);
+	return 1;
+}
+
+/* PalmTX machine dapm widgets */
+static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* PalmTX audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* headphone connected to HPOUTL, HPOUTR */
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+
+	/* ext speaker connected to ROUT2, LOUT2 */
+	{"Speaker", NULL, "LOUT2"},
+	{"Speaker", NULL, "ROUT2"},
+};
+
+static const char *jack_function[] = {"Headphone", "Off"};
+static const char *spk_function[] = {"On", "Off"};
+static const struct soc_enum palm27x_enum[] = {
+	SOC_ENUM_SINGLE_EXT(2, jack_function),
+	SOC_ENUM_SINGLE_EXT(2, spk_function),
+};
+
+static const struct snd_kcontrol_new palm27x_controls[] = {
+	SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack,
+		palm27x_set_jack),
+	SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk,
+		palm27x_set_spk),
+};
+
+static int palm27x_ac97_init(struct snd_soc_codec *codec)
+{
+	int i, err;
+
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONOOUT");
+
+	/* add palm27x specific controls */
+	for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&palm27x_controls[i],
+						codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	/* add palm27x specific widgets */
+	snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
+				ARRAY_SIZE(palm27x_dapm_widgets));
+
+	/* set up palm27x specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static struct snd_soc_dai_link palm27x_dai[] = {
+{
+	.name = "AC97 HiFi",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.init = palm27x_ac97_init,
+	.ops = &palm27x_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 = &palm27x_ops,
+},
+};
+
+static struct snd_soc_card palm27x_asoc = {
+	.name = "Palm/PXA27x",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = palm27x_dai,
+	.num_links = ARRAY_SIZE(palm27x_dai),
+};
+
+static struct snd_soc_device palm27x_snd_devdata = {
+	.card = &palm27x_asoc,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *palm27x_snd_device;
+
+static int __init palm27x_asoc_init(void)
+{
+	int ret;
+
+	if (!(machine_is_palmtx() || machine_is_palmt5() ||
+		machine_is_palmld()))
+		return -ENODEV;
+
+	ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_input(palm27x_ep_gpio);
+	if (ret)
+		goto err_alloc;
+
+	if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt,
+			IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+			"Headphone jack", NULL))
+		goto err_alloc;
+
+	palm27x_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!palm27x_snd_device) {
+		ret = -ENOMEM;
+		goto err_dev;
+	}
+
+	platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata);
+	palm27x_snd_devdata.dev = &palm27x_snd_device->dev;
+	ret = platform_device_add(palm27x_snd_device);
+
+	if (ret != 0)
+		goto put_device;
+
+	return 0;
+
+put_device:
+	platform_device_put(palm27x_snd_device);
+err_dev:
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+err_alloc:
+	gpio_free(palm27x_ep_gpio);
+
+	return ret;
+}
+
+static void __exit palm27x_asoc_exit(void)
+{
+	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
+	gpio_free(palm27x_ep_gpio);
+	platform_device_unregister(palm27x_snd_device);
+}
+
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+{
+	palm27x_ep_gpio = data->jack_gpio;
+}
+
+module_init(palm27x_asoc_init);
+module_exit(palm27x_asoc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 4d9930c5278..6e9827189ff 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -276,8 +276,9 @@ static struct snd_soc_dai_link poodle_dai = {
 };
 
 /* poodle audio machine driver */
-static struct snd_soc_machine snd_soc_machine_poodle = {
+static struct snd_soc_card snd_soc_poodle = {
 	.name = "Poodle",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &poodle_dai,
 	.num_links = 1,
 };
@@ -290,8 +291,7 @@ static struct wm8731_setup_data poodle_wm8731_setup = {
 
 /* poodle audio subsystem */
 static struct snd_soc_device poodle_snd_devdata = {
-	.machine = &snd_soc_machine_poodle,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_poodle,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &poodle_wm8731_setup,
 };
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
new file mode 100644
index 00000000000..73cb6b4c2f2
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -0,0 +1,931 @@
+#define DEBUG
+/*
+ * pxa-ssp.c  --  ALSA Soc Audio Layer
+ *
+ * Copyright 2005,2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         Mark Brown <broonie@opensource.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.
+ *
+ * TODO:
+ *  o Test network mode for > 16bit sample size
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+
+#include <mach/hardware.h>
+#include <mach/pxa-regs.h>
+#include <mach/regs-ssp.h>
+#include <mach/audio.h>
+#include <mach/ssp.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa-ssp.h"
+
+/*
+ * SSP audio private data
+ */
+struct ssp_priv {
+	struct ssp_dev dev;
+	unsigned int sysclk;
+	int dai_fmt;
+#ifdef CONFIG_PM
+	struct ssp_state state;
+#endif
+};
+
+#define PXA2xx_SSP1_BASE	0x41000000
+#define PXA27x_SSP2_BASE	0x41700000
+#define PXA27x_SSP3_BASE	0x41900000
+#define PXA3xx_SSP4_BASE	0x41a00000
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = {
+	.name			= "SSP1 PCM Mono out",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(14),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = {
+	.name			= "SSP1 PCM Mono in",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(13),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = {
+	.name			= "SSP1 PCM Stereo out",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(14),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = {
+	.name			= "SSP1 PCM Stereo in",
+	.dev_addr		= PXA2xx_SSP1_BASE + SSDR,
+	.drcmr			= &DRCMR(13),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = {
+	.name			= "SSP2 PCM Mono out",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(16),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = {
+	.name			= "SSP2 PCM Mono in",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(15),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = {
+	.name			= "SSP2 PCM Stereo out",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(16),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = {
+	.name			= "SSP2 PCM Stereo in",
+	.dev_addr		= PXA27x_SSP2_BASE + SSDR,
+	.drcmr			= &DRCMR(15),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = {
+	.name			= "SSP3 PCM Mono out",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = {
+	.name			= "SSP3 PCM Mono in",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = {
+	.name			= "SSP3 PCM Stereo out",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = {
+	.name			= "SSP3 PCM Stereo in",
+	.dev_addr		= PXA27x_SSP3_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = {
+	.name			= "SSP4 PCM Mono out",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = {
+	.name			= "SSP4 PCM Mono in",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH2,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = {
+	.name			= "SSP4 PCM Stereo out",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(67),
+	.dcmd			= DCMD_INCSRCADDR | DCMD_FLOWTRG |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = {
+	.name			= "SSP4 PCM Stereo in",
+	.dev_addr		= PXA3xx_SSP4_BASE + SSDR,
+	.drcmr			= &DRCMR(66),
+	.dcmd			= DCMD_INCTRGADDR | DCMD_FLOWSRC |
+				  DCMD_BURST16 | DCMD_WIDTH4,
+};
+
+static void dump_registers(struct ssp_device *ssp)
+{
+	dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
+		 ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1),
+		 ssp_read_reg(ssp, SSTO));
+
+	dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n",
+		 ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR),
+		 ssp_read_reg(ssp, SSACD));
+}
+
+static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
+	{
+		&pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in,
+		&pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in,
+		&pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in,
+		&pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in,
+	},
+	{
+		&pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in,
+		&pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in,
+	},
+};
+
+static int pxa_ssp_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	int ret = 0;
+
+	if (!cpu_dai->active) {
+		ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ);
+		if (ret < 0)
+			return ret;
+		ssp_disable(&priv->dev);
+	}
+	return ret;
+}
+
+static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active) {
+		ssp_disable(&priv->dev);
+		ssp_exit(&priv->dev);
+	}
+}
+
+#ifdef CONFIG_PM
+
+static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	ssp_save_state(&priv->dev, &priv->state);
+	clk_disable(priv->dev.ssp->clk);
+	return 0;
+}
+
+static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+
+	if (!cpu_dai->active)
+		return 0;
+
+	clk_enable(priv->dev.ssp->clk);
+	ssp_restore_state(&priv->dev, &priv->state);
+	ssp_enable(&priv->dev);
+
+	return 0;
+}
+
+#else
+#define pxa_ssp_suspend	NULL
+#define pxa_ssp_resume	NULL
+#endif
+
+/**
+ * ssp_set_clkdiv - set SSP clock divider
+ * @div: serial clock rate divider
+ */
+static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+{
+	struct ssp_device *ssp = dev->ssp;
+	u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+
+	ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+	int clk_id, unsigned int freq, int dir)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+
+	dev_dbg(&ssp->pdev->dev,
+		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
+		cpu_dai->id, clk_id, freq);
+
+	switch (clk_id) {
+	case PXA_SSP_CLK_NET_PLL:
+		sscr0 |= SSCR0_MOD;
+		break;
+	case PXA_SSP_CLK_PLL:
+		/* Internal PLL is fixed */
+		if (cpu_is_pxa25x())
+			priv->sysclk = 1843200;
+		else
+			priv->sysclk = 13000000;
+		break;
+	case PXA_SSP_CLK_EXT:
+		priv->sysclk = freq;
+		sscr0 |= SSCR0_ECS;
+		break;
+	case PXA_SSP_CLK_NET:
+		priv->sysclk = freq;
+		sscr0 |= SSCR0_NCS | SSCR0_MOD;
+		break;
+	case PXA_SSP_CLK_AUDIO:
+		priv->sysclk = 0;
+		ssp_set_scr(&priv->dev, 1);
+		sscr0 |= SSCR0_ADC;
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	/* The SSP clock must be disabled when changing SSP clock mode
+	 * on PXA2xx.  On PXA3xx it must be enabled when doing so. */
+	if (!cpu_is_pxa3xx())
+		clk_disable(priv->dev.ssp->clk);
+	val = ssp_read_reg(ssp, SSCR0) | sscr0;
+	ssp_write_reg(ssp, SSCR0, val);
+	if (!cpu_is_pxa3xx())
+		clk_enable(priv->dev.ssp->clk);
+
+	return 0;
+}
+
+/*
+ * Set the SSP clock dividers.
+ */
+static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
+	int div_id, int div)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	switch (div_id) {
+	case PXA_SSP_AUDIO_DIV_ACDS:
+		val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div);
+		ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_AUDIO_DIV_SCDB:
+		val = ssp_read_reg(ssp, SSACD);
+		val &= ~SSACD_SCDB;
+#if defined(CONFIG_PXA3xx)
+		if (cpu_is_pxa3xx())
+			val &= ~SSACD_SCDX8;
+#endif
+		switch (div) {
+		case PXA_SSP_CLK_SCDB_1:
+			val |= SSACD_SCDB;
+			break;
+		case PXA_SSP_CLK_SCDB_4:
+			break;
+#if defined(CONFIG_PXA3xx)
+		case PXA_SSP_CLK_SCDB_8:
+			if (cpu_is_pxa3xx())
+				val |= SSACD_SCDX8;
+			else
+				return -EINVAL;
+			break;
+#endif
+		default:
+			return -EINVAL;
+		}
+		ssp_write_reg(ssp, SSACD, val);
+		break;
+	case PXA_SSP_DIV_SCR:
+		ssp_set_scr(&priv->dev, div);
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+/*
+ * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
+ */
+static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
+	int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70;
+
+#if defined(CONFIG_PXA3xx)
+	if (cpu_is_pxa3xx())
+		ssp_write_reg(ssp, SSACDD, 0);
+#endif
+
+	switch (freq_out) {
+	case 5622000:
+		break;
+	case 11345000:
+		ssacd |= (0x1 << 4);
+		break;
+	case 12235000:
+		ssacd |= (0x2 << 4);
+		break;
+	case 14857000:
+		ssacd |= (0x3 << 4);
+		break;
+	case 32842000:
+		ssacd |= (0x4 << 4);
+		break;
+	case 48000000:
+		ssacd |= (0x5 << 4);
+		break;
+	case 0:
+		/* Disable */
+		break;
+
+	default:
+#ifdef CONFIG_PXA3xx
+		/* PXA3xx has a clock ditherer which can be used to generate
+		 * a wider range of frequencies - calculate a value for it.
+		 */
+		if (cpu_is_pxa3xx()) {
+			u32 val;
+			u64 tmp = 19968;
+			tmp *= 1000000;
+			do_div(tmp, freq_out);
+			val = tmp;
+
+			val = (val << 16) | 64;;
+			ssp_write_reg(ssp, SSACDD, val);
+
+			ssacd |= (0x6 << 4);
+
+			dev_dbg(&ssp->pdev->dev,
+				"Using SSACDD %x to supply %dHz\n",
+				val, freq_out);
+			break;
+		}
+#endif
+
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSACD, ssacd);
+
+	return 0;
+}
+
+/*
+ * Set the active slots in TDM/Network mode
+ */
+static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+	unsigned int mask, int slots)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr0;
+
+	sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7);
+
+	/* set number of active slots */
+	sscr0 |= SSCR0_SlotsPerFrm(slots);
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	/* set active slot mask */
+	ssp_write_reg(ssp, SSTSA, mask);
+	ssp_write_reg(ssp, SSRSA, mask);
+	return 0;
+}
+
+/*
+ * Tristate the SSP DAI lines
+ */
+static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
+	int tristate)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr1;
+
+	sscr1 = ssp_read_reg(ssp, SSCR1);
+	if (tristate)
+		sscr1 &= ~SSCR1_TTE;
+	else
+		sscr1 |= SSCR1_TTE;
+	ssp_write_reg(ssp, SSCR1, sscr1);
+
+	return 0;
+}
+
+/*
+ * Set up the SSP DAI format.
+ * The SSP Port must be inactive before calling this function as the
+ * physical interface format is changed.
+ */
+static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	u32 sscr0;
+	u32 sscr1;
+	u32 sspsp;
+
+	/* reset port settings */
+	sscr0 = ssp_read_reg(ssp, SSCR0) &
+		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+	sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
+	sspsp = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		sscr1 |= SSCR1_SCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSCR0, sscr0);
+	ssp_write_reg(ssp, SSCR1, sscr1);
+	ssp_write_reg(ssp, SSPSP, sspsp);
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			sspsp |= SSPSP_FSRT;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			sspsp |= SSPSP_SFRMP;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A:
+		sspsp |= SSPSP_FSRT;
+	case SND_SOC_DAIFMT_DSP_B:
+		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr1 |= SSCR1_TRAIL | SSCR1_RWOT;
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			sspsp |= SSPSP_SFRMP;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	ssp_write_reg(ssp, SSCR0, sscr0);
+	ssp_write_reg(ssp, SSCR1, sscr1);
+	ssp_write_reg(ssp, SSPSP, sspsp);
+
+	dump_registers(ssp);
+
+	/* Since we are configuring the timings for the format by hand
+	 * we have to defer some things until hw_params() where we
+	 * know parameters like the sample size.
+	 */
+	priv->dai_fmt = fmt;
+
+	return 0;
+}
+
+/*
+ * Set the SSP audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int dma = 0, chn = params_channels(params);
+	u32 sscr0;
+	u32 sspsp;
+	int width = snd_pcm_format_physical_width(params_format(params));
+
+	/* select correct DMA params */
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		dma = 1; /* capture DMA offset is 1,3 */
+	if (chn == 2)
+		dma += 2; /* stereo DMA offset is 2, mono is 0 */
+	cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
+
+	dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
+
+	/* we can only change the settings if the port is not in use */
+	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
+		return 0;
+
+	/* clear selected SSP bits */
+	sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	/* bit size */
+	sscr0 = ssp_read_reg(ssp, SSCR0);
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+#ifdef CONFIG_PXA3xx
+		if (cpu_is_pxa3xx())
+			sscr0 |= SSCR0_FPCKE;
+#endif
+		sscr0 |= SSCR0_DataSize(16);
+		if (params_channels(params) > 1)
+			sscr0 |= SSCR0_EDSS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
+		/* we must be in network mode (2 slots) for 24 bit stereo */
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
+		/* we must be in network mode (2 slots) for 32 bit stereo */
+		break;
+	}
+	ssp_write_reg(ssp, SSCR0, sscr0);
+
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* Cleared when the DAI format is set */
+		sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+		ssp_write_reg(ssp, SSPSP, sspsp);
+		break;
+	default:
+		break;
+	}
+
+	/* We always use a network mode so we always require TDM slots
+	 * - complain loudly and fail if they've not been set up yet.
+	 */
+	if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+		dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
+		return -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return 0;
+}
+
+static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int ret = 0;
+	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_device *ssp = priv->dev.ssp;
+	int val;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_RESUME:
+		ssp_enable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= SSCR1_TSRE;
+		else
+			val |= SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		val = ssp_read_reg(ssp, SSSR);
+		ssp_write_reg(ssp, SSSR, val);
+		break;
+	case SNDRV_PCM_TRIGGER_START:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val |= SSCR1_TSRE;
+		else
+			val |= SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		ssp_enable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val &= ~SSCR1_TSRE;
+		else
+			val &= ~SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		ssp_disable(&priv->dev);
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		val = ssp_read_reg(ssp, SSCR1);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			val &= ~SSCR1_TSRE;
+		else
+			val &= ~SSCR1_RSRE;
+		ssp_write_reg(ssp, SSCR1, val);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	dump_registers(ssp);
+
+	return ret;
+}
+
+static int pxa_ssp_probe(struct platform_device *pdev,
+			    struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv;
+	int ret;
+
+	priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->dev.ssp = ssp_request(dai->id, "SoC audio");
+	if (priv->dev.ssp == NULL) {
+		ret = -ENODEV;
+		goto err_priv;
+	}
+
+	dai->private_data = priv;
+
+	return 0;
+
+err_priv:
+	kfree(priv);
+	return ret;
+}
+
+static void pxa_ssp_remove(struct platform_device *pdev,
+			      struct snd_soc_dai *dai)
+{
+	struct ssp_priv *priv = dai->private_data;
+	ssp_free(priv->dev.ssp);
+}
+
+#define PXA_SSP_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 PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			    SNDRV_PCM_FMTBIT_S24_LE |	\
+			    SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai pxa_ssp_dai[] = {
+	{
+		.name = "pxa2xx-ssp1",
+		.id = 0,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			 .channels_min = 1,
+			 .channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{	.name = "pxa2xx-ssp2",
+		.id = 1,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{
+		.name = "pxa2xx-ssp3",
+		.id = 2,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+	{
+		.name = "pxa2xx-ssp4",
+		.id = 3,
+		.probe = pxa_ssp_probe,
+		.remove = pxa_ssp_remove,
+		.suspend = pxa_ssp_suspend,
+		.resume = pxa_ssp_resume,
+		.playback = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		},
+		.capture = {
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = PXA_SSP_RATES,
+			.formats = PXA_SSP_FORMATS,
+		 },
+		.ops = {
+			.startup = pxa_ssp_startup,
+			.shutdown = pxa_ssp_shutdown,
+			.trigger = pxa_ssp_trigger,
+			.hw_params = pxa_ssp_hw_params,
+			.set_sysclk = pxa_ssp_set_dai_sysclk,
+			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
+			.set_pll = pxa_ssp_set_dai_pll,
+			.set_fmt = pxa_ssp_set_dai_fmt,
+			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
+			.set_tristate = pxa_ssp_set_dai_tristate,
+		},
+	},
+};
+EXPORT_SYMBOL_GPL(pxa_ssp_dai);
+
+static int __init pxa_ssp_init(void)
+{
+	return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_init(pxa_ssp_init);
+
+static void __exit pxa_ssp_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+}
+module_exit(pxa_ssp_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h
new file mode 100644
index 00000000000..91deadd5567
--- /dev/null
+++ b/sound/soc/pxa/pxa-ssp.h
@@ -0,0 +1,47 @@
+/*
+ * ASoC PXA SSP port support
+ *
+ * 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 _PXA_SSP_H
+#define _PXA_SSP_H
+
+/* pxa DAI SSP IDs */
+#define PXA_DAI_SSP1			0
+#define PXA_DAI_SSP2			1
+#define PXA_DAI_SSP3			2
+#define PXA_DAI_SSP4			3
+
+/* SSP clock sources */
+#define PXA_SSP_CLK_PLL	0
+#define PXA_SSP_CLK_EXT	1
+#define PXA_SSP_CLK_NET	2
+#define PXA_SSP_CLK_AUDIO	3
+#define PXA_SSP_CLK_NET_PLL	4
+
+/* SSP audio dividers */
+#define PXA_SSP_AUDIO_DIV_ACDS		0
+#define PXA_SSP_AUDIO_DIV_SCDB		1
+#define PXA_SSP_DIV_SCR				2
+
+/* SSP ACDS audio dividers values */
+#define PXA_SSP_CLK_AUDIO_DIV_1		0
+#define PXA_SSP_CLK_AUDIO_DIV_2		1
+#define PXA_SSP_CLK_AUDIO_DIV_4		2
+#define PXA_SSP_CLK_AUDIO_DIV_8		3
+#define PXA_SSP_CLK_AUDIO_DIV_16	4
+#define PXA_SSP_CLK_AUDIO_DIV_32	5
+
+/* SSP divider bypass */
+#define PXA_SSP_CLK_SCDB_4		0
+#define PXA_SSP_CLK_SCDB_1		1
+#define PXA_SSP_CLK_SCDB_8		2
+
+#define PXA_SSP_PLL_OUT  0
+
+extern struct snd_soc_dai pxa_ssp_dai[4];
+
+#endif
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index a7a3a9c5c6f..780db6757ad 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -87,14 +87,12 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
 };
 
 #ifdef CONFIG_PM
-static int pxa2xx_ac97_suspend(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_suspend();
 }
 
-static int pxa2xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_resume();
 }
@@ -117,7 +115,8 @@ static void pxa2xx_ac97_remove(struct platform_device *pdev,
 }
 
 static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -131,7 +130,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -145,7 +145,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				     struct snd_pcm_hw_params *params,
+				     struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -170,7 +171,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = pxa2xx_ac97_probe,
 	.remove = pxa2xx_ac97_remove,
 	.suspend = pxa2xx_ac97_suspend,
@@ -193,7 +194,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-aux",
 	.id = 1,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.playback = {
 		.stream_name = "AC97 Aux Playback",
 		.channels_min = 1,
@@ -212,7 +213,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-mic",
 	.id = 2,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.capture = {
 		.stream_name = "AC97 Mic Capture",
 		.channels_min = 1,
@@ -227,6 +228,18 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __init pxa_ac97_init(void)
+{
+	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_init(pxa_ac97_init);
+
+static void __exit pxa_ac97_exit(void)
+{
+	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+}
+module_exit(pxa_ac97_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index e758034db5c..517991fb109 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -121,7 +121,8 @@ static struct pxa2xx_gpio gpio_bus[] = {
 	},
 };
 
-static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
+static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -187,7 +188,8 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 }
 
 static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -248,7 +250,8 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			      struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -269,7 +272,8 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
 	return ret;
 }
 
-static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
+static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
 {
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		SACR1 |= SACR1_DRPL;
@@ -289,8 +293,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
 }
 
 #ifdef CONFIG_PM
-static int pxa2xx_i2s_suspend(struct platform_device *dev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -307,8 +310,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev,
 	return 0;
 }
 
-static int pxa2xx_i2s_resume(struct platform_device *pdev,
-	struct snd_soc_dai *dai)
+static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -336,7 +338,6 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.suspend = pxa2xx_i2s_suspend,
 	.resume = pxa2xx_i2s_resume,
 	.playback = {
@@ -353,8 +354,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.startup = pxa2xx_i2s_startup,
 		.shutdown = pxa2xx_i2s_shutdown,
 		.trigger = pxa2xx_i2s_trigger,
-		.hw_params = pxa2xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = pxa2xx_i2s_hw_params,
 		.set_fmt = pxa2xx_i2s_set_dai_fmt,
 		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
 	},
@@ -364,12 +364,23 @@ EXPORT_SYMBOL_GPL(pxa_i2s_dai);
 
 static int pxa2xx_i2s_probe(struct platform_device *dev)
 {
+	int ret;
+
 	clk_i2s = clk_get(&dev->dev, "I2SCLK");
-	return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0;
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(clk_i2s);
+
+	pxa_i2s_dai.dev = &dev->dev;
+	ret = snd_soc_register_dai(&pxa_i2s_dai);
+	if (ret != 0)
+		clk_put(clk_i2s);
+
+	return ret;
 }
 
 static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
 {
+	snd_soc_unregister_dai(&pxa_i2s_dai);
 	clk_put(clk_i2s);
 	clk_i2s = ERR_PTR(-ENOENT);
 	return 0;
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index afcd892cd2f..c670d08e7c9 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -69,7 +69,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-struct snd_pcm_ops pxa2xx_pcm_ops = {
+static struct snd_pcm_ops pxa2xx_pcm_ops = {
 	.open		= __pxa2xx_pcm_open,
 	.close		= __pxa2xx_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
@@ -118,6 +118,18 @@ struct snd_soc_platform pxa2xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
 
+static int __init pxa2xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&pxa2xx_soc_platform);
+}
+module_init(pxa2xx_soc_platform_init);
+
+static void __exit pxa2xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&pxa2xx_soc_platform);
+}
+module_exit(pxa2xx_soc_platform_exit);
+
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d307b6757e9..a3b9e6bdf97 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -319,8 +319,9 @@ static struct snd_soc_dai_link spitz_dai = {
 };
 
 /* spitz audio machine driver */
-static struct snd_soc_machine snd_soc_machine_spitz = {
+static struct snd_soc_card snd_soc_spitz = {
 	.name = "Spitz",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = &spitz_dai,
 	.num_links = 1,
 };
@@ -333,8 +334,7 @@ static struct wm8750_setup_data spitz_wm8750_setup = {
 
 /* spitz audio subsystem */
 static struct snd_soc_device spitz_snd_devdata = {
-	.machine = &snd_soc_machine_spitz,
-	.platform = &pxa2xx_soc_platform,
+	.card = &snd_soc_spitz,
 	.codec_dev = &soc_codec_dev_wm8750,
 	.codec_data = &spitz_wm8750_setup,
 };
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index afefe41b8c4..c77194f74c9 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -38,7 +38,7 @@
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_machine tosa;
+static struct snd_soc_card tosa;
 
 #define TOSA_HP        0
 #define TOSA_MIC_INT   1
@@ -230,15 +230,37 @@ static struct snd_soc_dai_link tosa_dai[] = {
 },
 };
 
-static struct snd_soc_machine tosa = {
+static int tosa_probe(struct platform_device *dev)
+{
+	int ret;
+
+	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
+	if (ret)
+		return ret;
+	ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
+	if (ret)
+		gpio_free(TOSA_GPIO_L_MUTE);
+
+	return ret;
+}
+
+static int tosa_remove(struct platform_device *dev)
+{
+	gpio_free(TOSA_GPIO_L_MUTE);
+	return 0;
+}
+
+static struct snd_soc_card tosa = {
 	.name = "Tosa",
+	.platform = &pxa2xx_soc_platform,
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
+	.probe = tosa_probe,
+	.remove = tosa_remove,
 };
 
 static struct snd_soc_device tosa_snd_devdata = {
-	.machine = &tosa,
-	.platform = &pxa2xx_soc_platform,
+	.card = &tosa,
 	.codec_dev = &soc_codec_dev_wm9712,
 };
 
@@ -251,11 +273,6 @@ static int __init tosa_init(void)
 	if (!machine_is_tosa())
 		return -ENODEV;
 
-	ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack");
-	if (ret)
-		return ret;
-	gpio_direction_output(TOSA_GPIO_L_MUTE, 0);
-
 	tosa_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!tosa_snd_device) {
 		ret = -ENOMEM;
@@ -272,15 +289,12 @@ static int __init tosa_init(void)
 	platform_device_put(tosa_snd_device);
 
 err_alloc:
-	gpio_free(TOSA_GPIO_L_MUTE);
-
 	return ret;
 }
 
 static void __exit tosa_exit(void)
 {
 	platform_device_unregister(tosa_snd_device);
-	gpio_free(TOSA_GPIO_L_MUTE);
 }
 
 module_init(tosa_init);
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
new file mode 100644
index 00000000000..f8e9ecd589d
--- /dev/null
+++ b/sound/soc/pxa/zylonite.c
@@ -0,0 +1,219 @@
+/*
+ * zylonite.c  --  SoC audio for Zylonite
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.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/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm9713.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+#include "pxa-ssp.h"
+
+static struct snd_soc_card zylonite;
+
+static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_MIC("Headset Microphone", NULL),
+	SND_SOC_DAPM_MIC("Handset Microphone", NULL),
+	SND_SOC_DAPM_SPK("Multiactor", NULL),
+	SND_SOC_DAPM_SPK("Headset Earpiece", NULL),
+};
+
+/* Currently supported audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* Headphone output connected to HPL/HPR */
+	{ "Headphone", NULL,  "HPL" },
+	{ "Headphone", NULL,  "HPR" },
+
+	/* On-board earpiece */
+	{ "Headset Earpiece", NULL, "OUT3" },
+
+	/* Headphone mic */
+	{ "MIC2A", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "Headset Microphone" },
+
+	/* On-board mic */
+	{ "MIC1", NULL, "Mic Bias" },
+	{ "Mic Bias", NULL, "Handset Microphone" },
+
+	/* Multiactor differentially connected over SPKL/SPKR */
+	{ "Multiactor", NULL, "SPKL" },
+	{ "Multiactor", NULL, "SPKR" },
+};
+
+static int zylonite_wm9713_init(struct snd_soc_codec *codec)
+{
+	/* Currently we only support use of the AC97 clock here.  If
+	 * CLK_POUT is selected by SW15 then the clock API will need
+	 * to be used to request and enable it here.
+	 */
+
+	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
+				  ARRAY_SIZE(zylonite_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* Static setup for now */
+	snd_soc_dapm_enable_pin(codec, "Headphone");
+	snd_soc_dapm_enable_pin(codec, "Headset Earpiece");
+
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static int zylonite_voice_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_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int pll_out = 0;
+	unsigned int acds = 0;
+	unsigned int wm9713_div = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+		wm9713_div = 12;
+		pll_out = 2048000;
+		break;
+	case 16000:
+		wm9713_div = 6;
+		pll_out = 4096000;
+		break;
+	case 48000:
+	default:
+		wm9713_div = 2;
+		pll_out = 12288000;
+		acds = 1;
+		break;
+	}
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai,
+				       params_channels(params),
+				       params_channels(params));
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+	if (ret < 0)
+		return ret;
+
+	/* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
+	 * to be set instead.
+	 */
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+				     WM9713_PCMDIV(wm9713_div));
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops zylonite_voice_ops = {
+	.hw_params = zylonite_voice_hw_params,
+};
+
+static struct snd_soc_dai_link zylonite_dai[] = {
+{
+	.name = "AC97",
+	.stream_name = "AC97 HiFi",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+	.init = zylonite_wm9713_init,
+},
+{
+	.name = "AC97 Aux",
+	.stream_name = "AC97 Aux",
+	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+},
+{
+	.name = "WM9713 Voice",
+	.stream_name = "WM9713 Voice",
+	.cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3],
+	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
+	.ops = &zylonite_voice_ops,
+},
+};
+
+static struct snd_soc_card zylonite = {
+	.name = "Zylonite",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = zylonite_dai,
+	.num_links = ARRAY_SIZE(zylonite_dai),
+};
+
+static struct snd_soc_device zylonite_snd_ac97_devdata = {
+	.card = &zylonite,
+	.codec_dev = &soc_codec_dev_wm9713,
+};
+
+static struct platform_device *zylonite_snd_ac97_device;
+
+static int __init zylonite_init(void)
+{
+	int ret;
+
+	zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+	if (!zylonite_snd_ac97_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(zylonite_snd_ac97_device,
+			     &zylonite_snd_ac97_devdata);
+	zylonite_snd_ac97_devdata.dev = &zylonite_snd_ac97_device->dev;
+
+	ret = platform_device_add(zylonite_snd_ac97_device);
+	if (ret != 0)
+		platform_device_put(zylonite_snd_ac97_device);
+
+	return ret;
+}
+
+static void __exit zylonite_exit(void)
+{
+	platform_device_unregister(zylonite_snd_ac97_device);
+}
+
+module_init(zylonite_init);
+module_exit(zylonite_exit);
+
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index b9f2353effe..fcd03acf10f 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650
 	  Say Y if you want to add support for SoC audio on ln2440sbc
 	  with the ALC650.
 
+config SND_S3C24XX_SOC_S3C24XX_UDA134X
+	tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
+       	depends on SND_S3C24XX_SOC
+       	select SND_S3C24XX_SOC_I2S
+       	select SND_SOC_UDA134X
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 0aa5fb0b970..96b3f3f617d 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
+snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
 obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
+obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c
index 4eab2c19c45..12c71482d25 100644
--- a/sound/soc/s3c24xx/ln2440sbc_alc650.c
+++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c
@@ -27,7 +27,7 @@
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
 
-static struct snd_soc_machine ln2440sbc;
+static struct snd_soc_card ln2440sbc;
 
 static struct snd_soc_dai_link ln2440sbc_dai[] = {
 {
@@ -38,15 +38,15 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = {
 },
 };
 
-static struct snd_soc_machine ln2440sbc = {
+static struct snd_soc_card ln2440sbc = {
 	.name = "LN2440SBC",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = ln2440sbc_dai,
 	.num_links = ARRAY_SIZE(ln2440sbc_dai),
 };
 
 static struct snd_soc_device ln2440sbc_snd_ac97_devdata = {
-	.machine = &ln2440sbc,
-	.platform = &s3c24xx_soc_platform,
+	.card = &ln2440sbc,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 87ddfefcc2f..45bb12e8ea4 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -59,7 +59,7 @@
 #define NEO_CAPTURE_HEADSET		7
 #define NEO_CAPTURE_BLUETOOTH		8
 
-static struct snd_soc_machine neo1973;
+static struct snd_soc_card neo1973;
 static struct i2c_client *i2c;
 
 static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
@@ -548,7 +548,6 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 static struct snd_soc_dai bt_dai = {
 	.name = "Bluetooth",
 	.id = 0,
-	.type = SND_SOC_DAI_PCM,
 	.playback = {
 		.channels_min = 1,
 		.channels_max = 1,
@@ -579,8 +578,9 @@ static struct snd_soc_dai_link neo1973_dai[] = {
 },
 };
 
-static struct snd_soc_machine neo1973 = {
+static struct snd_soc_card neo1973 = {
 	.name = "neo1973",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = neo1973_dai,
 	.num_links = ARRAY_SIZE(neo1973_dai),
 };
@@ -591,8 +591,7 @@ static struct wm8753_setup_data neo1973_wm8753_setup = {
 };
 
 static struct snd_soc_device neo1973_snd_devdata = {
-	.machine = &neo1973,
-	.platform = &s3c24xx_soc_platform,
+	.card = &neo1973,
 	.codec_dev = &soc_codec_dev_wm8753,
 	.codec_data = &neo1973_wm8753_setup,
 };
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index ded7d995a92..f3fc0aba0aa 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -343,7 +343,8 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 }
 
 static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	u32 iismod;
@@ -373,7 +374,8 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	unsigned long irqs;
@@ -647,8 +649,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 }
 
 #ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct platform_device *dev,
-			      struct snd_soc_dai *dai)
+static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 	u32 iismod;
@@ -663,25 +664,24 @@ static int s3c2412_i2s_suspend(struct platform_device *dev,
 		iismod = readl(i2s->regs + S3C2412_IISMOD);
 
 		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__);
+			pr_warning("%s: RXDMA active?\n", __func__);
 
 		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__);
+			pr_warning("%s: TXDMA active?\n", __func__);
 
 		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			dev_warn(&dev->dev, "%s: IIS active\n", __func__);
+			pr_warning("%s: IIS active\n", __func__);
 	}
 
 	return 0;
 }
 
-static int s3c2412_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *dai)
+static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 
-	dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n",
-		 dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
+		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
 
 	if (dai->active) {
 		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
@@ -711,7 +711,6 @@ static int s3c2412_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai s3c2412_i2s_dai = {
 	.name	= "s3c2412-i2s",
 	.id	= 0,
-	.type	= SND_SOC_DAI_I2S,
 	.probe	= s3c2412_i2s_probe,
 	.suspend = s3c2412_i2s_suspend,
 	.resume = s3c2412_i2s_resume,
@@ -730,8 +729,6 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 	.ops = {
 		.trigger	= s3c2412_i2s_trigger,
 		.hw_params	= s3c2412_i2s_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt	= s3c2412_i2s_set_fmt,
 		.set_clkdiv	= s3c2412_i2s_set_clkdiv,
 		.set_sysclk	= s3c2412_i2s_set_sysclk,
@@ -739,6 +736,19 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
+static int __init s3c2412_i2s_init(void)
+{
+	return snd_soc_register_dai(&s3c2412_i2s_dai);
+}
+module_init(s3c2412_i2s_init);
+
+static void __exit s3c2412_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c2412_i2s_dai);
+}
+module_exit(s3c2412_i2s_exit);
+
+
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 19c5c3cf5d8..1bfce40bb2e 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -271,7 +271,8 @@ static void s3c2443_ac97_remove(struct platform_device *pdev,
 }
 
 static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -284,7 +285,8 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
 {
 	u32 ac_glbctrl;
 
@@ -313,7 +315,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *params)
+				      struct snd_pcm_hw_params *params,
+				      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
@@ -327,7 +330,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 }
 
 static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
-	int cmd)
+				    int cmd, struct snd_soc_dai *dai)
 {
 	u32 ac_glbctrl;
 
@@ -356,7 +359,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "s3c2443-ac97",
 	.id = 0,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.probe = s3c2443_ac97_probe,
 	.remove = s3c2443_ac97_remove,
 	.playback = {
@@ -378,7 +381,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97-mic",
 	.id = 1,
-	.type = SND_SOC_DAI_AC97,
+	.ac97_control = 1,
 	.capture = {
 		.stream_name = "AC97 Mic Capture",
 		.channels_min = 1,
@@ -393,6 +396,21 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
+static int __init s3c2443_ac97_init(void)
+{
+	return snd_soc_register_dais(s3c2443_ac97_dai,
+				     ARRAY_SIZE(s3c2443_ac97_dai));
+}
+module_init(s3c2443_ac97_init);
+
+static void __exit s3c2443_ac97_exit(void)
+{
+	snd_soc_unregister_dais(s3c2443_ac97_dai,
+				ARRAY_SIZE(s3c2443_ac97_dai));
+}
+module_exit(s3c2443_ac97_exit);
+
+
 MODULE_AUTHOR("Graeme Gregory");
 MODULE_DESCRIPTION("AC97 driver for the Samsung s3c2443 chip");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index ba4476b55fb..6f4d439b57a 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -243,7 +243,8 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 }
 
 static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	u32 iismod;
@@ -261,10 +262,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
+		iismod &= ~S3C2410_IISMOD_16BIT;
+		((struct s3c24xx_pcm_dma_params *)
+		  rtd->dai->cpu_dai->dma_data)->dma_size = 1;
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		iismod |= S3C2410_IISMOD_16BIT;
+		((struct s3c24xx_pcm_dma_params *)
+		  rtd->dai->cpu_dai->dma_data)->dma_size = 2;
 		break;
+	default:
+		return -EINVAL;
 	}
 
 	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -272,7 +280,8 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	int ret = 0;
 
@@ -410,8 +419,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
 }
 
 #ifdef CONFIG_PM
-static int s3c24xx_i2s_suspend(struct platform_device *pdev,
-		struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 
@@ -425,8 +433,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int s3c24xx_i2s_resume(struct platform_device *pdev,
-		struct snd_soc_dai *cpu_dai)
+static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 	clk_enable(s3c24xx_i2s.iis_clk);
@@ -452,7 +459,6 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev,
 struct snd_soc_dai s3c24xx_i2s_dai = {
 	.name = "s3c24xx-i2s",
 	.id = 0,
-	.type = SND_SOC_DAI_I2S,
 	.probe = s3c24xx_i2s_probe,
 	.suspend = s3c24xx_i2s_suspend,
 	.resume = s3c24xx_i2s_resume,
@@ -468,8 +474,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = {
 		.trigger = s3c24xx_i2s_trigger,
-		.hw_params = s3c24xx_i2s_hw_params,},
-	.dai_ops = {
+		.hw_params = s3c24xx_i2s_hw_params,
 		.set_fmt = s3c24xx_i2s_set_fmt,
 		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
 		.set_sysclk = s3c24xx_i2s_set_sysclk,
@@ -477,6 +482,18 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
+static int __init s3c24xx_i2s_init(void)
+{
+	return snd_soc_register_dai(&s3c24xx_i2s_dai);
+}
+module_init(s3c24xx_i2s_init);
+
+static void __exit s3c24xx_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c24xx_i2s_dai);
+}
+module_exit(s3c24xx_i2s_exit);
+
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("s3c24xx I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index e13e614bada..7c64d31d067 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -465,6 +465,18 @@ struct snd_soc_platform s3c24xx_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
 
+static int __init s3c24xx_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&s3c24xx_soc_platform);
+}
+module_init(s3c24xx_soc_platform_init);
+
+static void __exit s3c24xx_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&s3c24xx_soc_platform);
+}
+module_exit(s3c24xx_soc_platform_exit);
+
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
new file mode 100644
index 00000000000..a0a4d1832a1
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -0,0 +1,373 @@
+/*
+ * Modifications by Christian Pellegrin <chripell@evolware.org>
+ *
+ * s3c24xx_uda134x.c  --  S3C24XX_UDA134X ALSA SoC Audio board driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * 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/clk.h>
+#include <linux/mutex.h>
+#include <linux/gpio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/s3c24xx_uda134x.h>
+#include <sound/uda134x.h>
+
+#include <asm/plat-s3c24xx/regs-iis.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+#include "../codecs/uda134x.h"
+
+
+/* #define ENFORCE_RATES 1 */
+/*
+  Unfortunately the S3C24XX in master mode has a limited capacity of
+  generating the clock for the codec. If you define this only rates
+  that are really available will be enforced. But be careful, most
+  user level application just want the usual sampling frequencies (8,
+  11.025, 22.050, 44.1 kHz) and anyway resampling is a costly
+  operation for embedded systems. So if you aren't very lucky or your
+  hardware engineer wasn't very forward-looking it's better to leave
+  this undefined. If you do so an approximate value for the requested
+  sampling rate in the range -/+ 5% will be chosen. If this in not
+  possible an error will be returned.
+*/
+
+static struct clk *xtal;
+static struct clk *pclk;
+/* this is need because we don't have a place where to keep the
+ * pointers to the clocks in each substream. We get the clocks only
+ * when we are actually using them so we don't block stuff like
+ * frequency change or oscillator power-off */
+static int clk_users;
+static DEFINE_MUTEX(clk_lock);
+
+static unsigned int rates[33 * 2];
+#ifdef ENFORCE_RATES
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+	.count	= ARRAY_SIZE(rates),
+	.list	= rates,
+	.mask	= 0,
+};
+#endif
+
+static struct platform_device *s3c24xx_uda134x_snd_device;
+
+static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
+{
+	int ret = 0;
+#ifdef ENFORCE_RATES
+	struct snd_pcm_runtime *runtime = substream->runtime;;
+#endif
+
+	mutex_lock(&clk_lock);
+	pr_debug("%s %d\n", __func__, clk_users);
+	if (clk_users == 0) {
+		xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal");
+		if (!xtal) {
+			printk(KERN_ERR "%s cannot get xtal\n", __func__);
+			ret = -EBUSY;
+		} else {
+			pclk = clk_get(&s3c24xx_uda134x_snd_device->dev,
+				       "pclk");
+			if (!pclk) {
+				printk(KERN_ERR "%s cannot get pclk\n",
+				       __func__);
+				clk_put(xtal);
+				ret = -EBUSY;
+			}
+		}
+		if (!ret) {
+			int i, j;
+
+			for (i = 0; i < 2; i++) {
+				int fs = i ? 256 : 384;
+
+				rates[i*33] = clk_get_rate(xtal) / fs;
+				for (j = 1; j < 33; j++)
+					rates[i*33 + j] = clk_get_rate(pclk) /
+						(j * fs);
+			}
+		}
+	}
+	clk_users += 1;
+	mutex_unlock(&clk_lock);
+	if (!ret) {
+#ifdef ENFORCE_RATES
+		ret = snd_pcm_hw_constraint_list(runtime, 0,
+						 SNDRV_PCM_HW_PARAM_RATE,
+						 &hw_constraints_rates);
+		if (ret < 0)
+			printk(KERN_ERR "%s cannot set constraints\n",
+			       __func__);
+#endif
+	}
+	return ret;
+}
+
+static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream)
+{
+	mutex_lock(&clk_lock);
+	pr_debug("%s %d\n", __func__, clk_users);
+	clk_users -= 1;
+	if (clk_users == 0) {
+		clk_put(xtal);
+		xtal = NULL;
+		clk_put(pclk);
+		pclk = NULL;
+	}
+	mutex_unlock(&clk_lock);
+}
+
+static int s3c24xx_uda134x_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_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	unsigned int clk = 0;
+	int ret = 0;
+	int clk_source, fs_mode;
+	unsigned long rate = params_rate(params);
+	long err, cerr;
+	unsigned int div;
+	int i, bi;
+
+	err = 999999;
+	bi = 0;
+	for (i = 0; i < 2*33; i++) {
+		cerr = rates[i] - rate;
+		if (cerr < 0)
+			cerr = -cerr;
+		if (cerr < err) {
+			err = cerr;
+			bi = i;
+		}
+	}
+	if (bi / 33 == 1)
+		fs_mode = S3C2410_IISMOD_256FS;
+	else
+		fs_mode = S3C2410_IISMOD_384FS;
+	if (bi % 33 == 0) {
+		clk_source = S3C24XX_CLKSRC_MPLL;
+		div = 1;
+	} else {
+		clk_source = S3C24XX_CLKSRC_PCLK;
+		div = bi % 33;
+	}
+	pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi);
+
+	clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate;
+	pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__,
+		 fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS",
+		 clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK",
+		 div, clk, err);
+
+	if ((err * 100 / rate) > 5) {
+		printk(KERN_ERR "S3C24XX_UDA134X: effective frequency "
+		       "too different from desired (%ld%%)\n",
+		       err * 100 / rate);
+		return -EINVAL;
+	}
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk,
+			SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+			S3C2410_IISMOD_32FS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+			S3C24XX_PRESCALE(div, div));
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops s3c24xx_uda134x_ops = {
+	.startup = s3c24xx_uda134x_startup,
+	.shutdown = s3c24xx_uda134x_shutdown,
+	.hw_params = s3c24xx_uda134x_hw_params,
+};
+
+static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = {
+	.name = "UDA134X",
+	.stream_name = "UDA134X",
+	.codec_dai = &uda134x_dai,
+	.cpu_dai = &s3c24xx_i2s_dai,
+	.ops = &s3c24xx_uda134x_ops,
+};
+
+static struct snd_soc_card snd_soc_s3c24xx_uda134x = {
+	.name = "S3C24XX_UDA134X",
+	.platform = &s3c24xx_soc_platform,
+	.dai_link = &s3c24xx_uda134x_dai_link,
+	.num_links = 1,
+};
+
+static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins;
+
+static void setdat(int v)
+{
+	gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0);
+}
+
+static void setclk(int v)
+{
+	gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0);
+}
+
+static void setmode(int v)
+{
+	gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0);
+}
+
+static struct uda134x_platform_data s3c24xx_uda134x = {
+	.l3 = {
+		.setdat = setdat,
+		.setclk = setclk,
+		.setmode = setmode,
+		.data_hold = 1,
+		.data_setup = 1,
+		.clock_high = 1,
+		.mode_hold = 1,
+		.mode = 1,
+		.mode_setup = 1,
+	},
+};
+
+static struct snd_soc_device s3c24xx_uda134x_snd_devdata = {
+	.card = &snd_soc_s3c24xx_uda134x,
+	.codec_dev = &soc_codec_dev_uda134x,
+	.codec_data = &s3c24xx_uda134x,
+};
+
+static int s3c24xx_uda134x_setup_pin(int pin, char *fun)
+{
+	if (gpio_request(pin, "s3c24xx_uda134x") < 0) {
+		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+		       "l3 %s pin already in use", fun);
+		return -EBUSY;
+	}
+	gpio_direction_output(pin, 0);
+	return 0;
+}
+
+static int s3c24xx_uda134x_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n");
+
+	s3c24xx_uda134x_l3_pins = pdev->dev.platform_data;
+	if (s3c24xx_uda134x_l3_pins == NULL) {
+		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+		       "unable to find platform data\n");
+		return -ENODEV;
+	}
+	s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power;
+	s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model;
+
+	if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data,
+				      "data") < 0)
+		return -EBUSY;
+	if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk,
+				      "clk") < 0) {
+		gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+		return -EBUSY;
+	}
+	if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode,
+				      "mode") < 0) {
+		gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+		gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+		return -EBUSY;
+	}
+
+	s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!s3c24xx_uda134x_snd_device) {
+		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: "
+		       "Unable to register\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(s3c24xx_uda134x_snd_device,
+			     &s3c24xx_uda134x_snd_devdata);
+	s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev;
+	ret = platform_device_add(s3c24xx_uda134x_snd_device);
+	if (ret) {
+		printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n");
+		platform_device_put(s3c24xx_uda134x_snd_device);
+	}
+
+	return ret;
+}
+
+static int s3c24xx_uda134x_remove(struct platform_device *pdev)
+{
+	platform_device_unregister(s3c24xx_uda134x_snd_device);
+	gpio_free(s3c24xx_uda134x_l3_pins->l3_data);
+	gpio_free(s3c24xx_uda134x_l3_pins->l3_clk);
+	gpio_free(s3c24xx_uda134x_l3_pins->l3_mode);
+	return 0;
+}
+
+static struct platform_driver s3c24xx_uda134x_driver = {
+	.probe  = s3c24xx_uda134x_probe,
+	.remove = s3c24xx_uda134x_remove,
+	.driver = {
+		.name = "s3c24xx_uda134x",
+		.owner = THIS_MODULE,
+	},
+};
+
+static int __init s3c24xx_uda134x_init(void)
+{
+	return platform_driver_register(&s3c24xx_uda134x_driver);
+}
+
+static void __exit s3c24xx_uda134x_exit(void)
+{
+	platform_driver_unregister(&s3c24xx_uda134x_driver);
+}
+
+
+module_init(s3c24xx_uda134x_init);
+module_exit(s3c24xx_uda134x_exit);
+
+MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>");
+MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
index 8515d6ff03f..a2a4f5323c1 100644
--- a/sound/soc/s3c24xx/smdk2443_wm9710.c
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -23,7 +23,7 @@
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-ac97.h"
 
-static struct snd_soc_machine smdk2443;
+static struct snd_soc_card smdk2443;
 
 static struct snd_soc_dai_link smdk2443_dai[] = {
 {
@@ -34,15 +34,15 @@ static struct snd_soc_dai_link smdk2443_dai[] = {
 },
 };
 
-static struct snd_soc_machine smdk2443 = {
+static struct snd_soc_card smdk2443 = {
 	.name = "SMDK2443",
+	.platform = &s3c24xx_soc_platform,
 	.dai_link = smdk2443_dai,
 	.num_links = ARRAY_SIZE(smdk2443_dai),
 };
 
 static struct snd_soc_device smdk2443_snd_ac97_devdata = {
-	.machine = &smdk2443,
-	.platform = &s3c24xx_soc_platform,
+	.card = &smdk2443,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 9faa12622d0..0dad3a0bb92 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -348,6 +348,18 @@ struct snd_soc_platform sh7760_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(sh7760_soc_platform);
 
+static int __init sh7760_soc_platform_init(void)
+{
+	return snd_soc_register_platform(&sh7760_soc_platform);
+}
+module_init(sh7760_soc_platform_init);
+
+static void __exit sh7760_soc_platform_exit(void)
+{
+	snd_soc_unregister_platform(&sh7760_soc_platform);
+}
+module_exit(sh7760_soc_platform_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index df7bc345c32..eab31838bad 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -236,7 +236,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 static int hac_hw_params(struct snd_pcm_substream *substream,
-			 struct snd_pcm_hw_params *params)
+			 struct snd_pcm_hw_params *params,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id];
@@ -270,7 +271,7 @@ struct snd_soc_dai sh4_hac_dai[] = {
 {
 	.name			= "HAC0",
 	.id			= 0,
-	.type			= SND_SOC_DAI_AC97,
+	.ac97_control		= 1,
 	.playback = {
 		.rates		= AC97_RATES,
 		.formats	= AC97_FMTS,
@@ -290,8 +291,8 @@ struct snd_soc_dai sh4_hac_dai[] = {
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
 	.name			= "HAC1",
+	.ac97_control		= 1,
 	.id			= 1,
-	.type			= SND_SOC_DAI_AC97,
 	.playback = {
 		.rates		= AC97_RATES,
 		.formats	= AC97_FMTS,
@@ -313,6 +314,18 @@ struct snd_soc_dai sh4_hac_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_hac_dai);
 
+static int __init sh4_hac_init(void)
+{
+	return snd_soc_register_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_init(sh4_hac_init);
+
+static void __exit sh4_hac_exit(void)
+{
+	snd_soc_unregister_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai));
+}
+module_exit(sh4_hac_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 92bfaf4774a..ce7f95b59de 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -38,15 +38,15 @@ static struct snd_soc_dai_link sh7760_ac97_dai = {
 	.ops = NULL,
 };
 
-static struct snd_soc_machine sh7760_ac97_soc_machine  = {
+static struct snd_soc_card sh7760_ac97_soc_machine  = {
 	.name = "SH7760 AC97",
+	.platform = &sh7760_soc_platform,
 	.dai_link = &sh7760_ac97_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device sh7760_ac97_snd_devdata = {
-	.machine = &sh7760_ac97_soc_machine,
-	.platform = &sh7760_soc_platform,
+	.card = &sh7760_ac97_soc_machine,
 	.codec_dev = &soc_codec_dev_ac97,
 };
 
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 55c3464163a..d1e5390fdde 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -89,7 +89,8 @@ struct ssi_priv {
  * track usage of the SSI; it is simplex-only so prevent attempts of
  * concurrent playback + capture. FIXME: any locking required?
  */
-static int ssi_startup(struct snd_pcm_substream *substream)
+static int ssi_startup(struct snd_pcm_substream *substream,
+		       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -101,7 +102,8 @@ static int ssi_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static void ssi_shutdown(struct snd_pcm_substream *substream)
+static void ssi_shutdown(struct snd_pcm_substream *substream,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -109,7 +111,8 @@ static void ssi_shutdown(struct snd_pcm_substream *substream)
 	ssi->inuse = 0;
 }
 
-static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
+static int ssi_trigger(struct snd_pcm_substream *substream, int cmd,
+		       struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -129,7 +132,8 @@ static int ssi_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static int ssi_hw_params(struct snd_pcm_substream *substream,
-			 struct snd_pcm_hw_params *params)
+			 struct snd_pcm_hw_params *params,
+			 struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id];
@@ -336,7 +340,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI0",
 	.id			= 0,
-	.type			= SND_SOC_DAI_I2S,
 	.playback = {
 		.rates		= SSI_RATES,
 		.formats	= SSI_FMTS,
@@ -354,8 +357,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.shutdown	= ssi_shutdown,
 		.trigger	= ssi_trigger,
 		.hw_params	= ssi_hw_params,
-	},
-	.dai_ops = {
 		.set_sysclk	= ssi_set_sysclk,
 		.set_clkdiv	= ssi_set_clkdiv,
 		.set_fmt	= ssi_set_fmt,
@@ -365,7 +366,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI1",
 	.id			= 1,
-	.type			= SND_SOC_DAI_I2S,
 	.playback = {
 		.rates		= SSI_RATES,
 		.formats	= SSI_FMTS,
@@ -383,8 +383,6 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.shutdown	= ssi_shutdown,
 		.trigger	= ssi_trigger,
 		.hw_params	= ssi_hw_params,
-	},
-	.dai_ops = {
 		.set_sysclk	= ssi_set_sysclk,
 		.set_clkdiv	= ssi_set_clkdiv,
 		.set_fmt	= ssi_set_fmt,
@@ -394,6 +392,18 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 };
 EXPORT_SYMBOL_GPL(sh4_ssi_dai);
 
+static int __init sh4_ssi_init(void)
+{
+	return snd_soc_register_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_init(sh4_ssi_init);
+
+static void __exit sh4_ssi_exit(void)
+{
+	snd_soc_unregister_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai));
+}
+module_exit(sh4_ssi_exit);
+
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver");
 MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 16c7453f494..b098c0b4c58 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -26,6 +26,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/bitops.h>
+#include <linux/debugfs.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -34,18 +35,23 @@
 #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);
 
+#ifdef CONFIG_DEBUG_FS
+static struct dentry *debugfs_root;
+#endif
+
+static DEFINE_MUTEX(client_mutex);
+static LIST_HEAD(card_list);
+static LIST_HEAD(dai_list);
+static LIST_HEAD(platform_list);
+static LIST_HEAD(codec_list);
+
+static int snd_soc_register_card(struct snd_soc_card *card);
+static int snd_soc_unregister_card(struct snd_soc_card *card);
+
 /*
  * 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.
@@ -107,20 +113,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 }
 #endif
 
-static inline const char *get_dai_name(int type)
-{
-	switch (type) {
-	case SND_SOC_DAI_AC97_BUS:
-	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
@@ -130,9 +122,10 @@ 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_soc_card *card = socdev->card;
 	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_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret = 0;
@@ -141,7 +134,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 
 	/* startup the audio subsystem */
 	if (cpu_dai->ops.startup) {
-		ret = cpu_dai->ops.startup(substream);
+		ret = cpu_dai->ops.startup(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open interface %s\n",
 				cpu_dai->name);
@@ -158,7 +151,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	}
 
 	if (codec_dai->ops.startup) {
-		ret = codec_dai->ops.startup(substream);
+		ret = codec_dai->ops.startup(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open codec %s\n",
 				codec_dai->name);
@@ -228,12 +221,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		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);
+	pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
+	pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
+	pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
+		 runtime->hw.channels_max);
+	pr_debug("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;
@@ -255,7 +248,7 @@ codec_dai_err:
 
 platform_err:
 	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream);
+		cpu_dai->ops.shutdown(substream, cpu_dai);
 out:
 	mutex_unlock(&pcm_mutex);
 	return ret;
@@ -268,8 +261,9 @@ out:
  */
 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_card *card = container_of(work, struct snd_soc_card,
+						 delayed_work.work);
+	struct snd_soc_device *socdev = card->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_dai *codec_dai;
 	int i;
@@ -278,18 +272,18 @@ static void close_delayed_work(struct work_struct *work)
 	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");
+		pr_debug("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) {
 
 			/* Reduce power if no longer active */
 			if (codec->active == 0) {
-				dbg("pop wq D1 %s %s\n", codec->name,
-					codec_dai->playback.stream_name);
+				pr_debug("pop wq D1 %s %s\n", codec->name,
+					 codec_dai->playback.stream_name);
 				snd_soc_dapm_set_bias_level(socdev,
 					SND_SOC_BIAS_PREPARE);
 			}
@@ -301,8 +295,8 @@ static void close_delayed_work(struct work_struct *work)
 
 			/* Fall into standby if no longer active */
 			if (codec->active == 0) {
-				dbg("pop wq D3 %s %s\n", codec->name,
-					codec_dai->playback.stream_name);
+				pr_debug("pop wq D3 %s %s\n", codec->name,
+					 codec_dai->playback.stream_name);
 				snd_soc_dapm_set_bias_level(socdev,
 					SND_SOC_BIAS_STANDBY);
 			}
@@ -320,8 +314,9 @@ 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_card *card = socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -346,10 +341,10 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 		snd_soc_dai_digital_mute(codec_dai, 1);
 
 	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream);
+		cpu_dai->ops.shutdown(substream, cpu_dai);
 
 	if (codec_dai->ops.shutdown)
-		codec_dai->ops.shutdown(substream);
+		codec_dai->ops.shutdown(substream, codec_dai);
 
 	if (machine->ops && machine->ops->shutdown)
 		machine->ops->shutdown(substream);
@@ -361,7 +356,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	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,
+		schedule_delayed_work(&card->delayed_work,
 			msecs_to_jiffies(pmdown_time));
 	} else {
 		/* capture streams can be powered down now */
@@ -387,8 +382,9 @@ 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_card *card = socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -413,7 +409,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	}
 
 	if (codec_dai->ops.prepare) {
-		ret = codec_dai->ops.prepare(substream);
+		ret = codec_dai->ops.prepare(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: codec DAI prepare error\n");
 			goto out;
@@ -421,58 +417,49 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	}
 
 	if (cpu_dai->ops.prepare) {
-		ret = cpu_dai->ops.prepare(substream);
+		ret = cpu_dai->ops.prepare(substream, cpu_dai);
 		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);
-			snd_soc_dai_digital_mute(codec_dai, 0);
-		}
-	} else {
-		/* no delayed work - do we need to power up codec */
-		if (codec->bias_level != SND_SOC_BIAS_ON) {
+	/* cancel any delayed stream shutdown that is pending */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+	    codec_dai->pop_wait) {
+		codec_dai->pop_wait = 0;
+		cancel_delayed_work(&card->delayed_work);
+	}
 
-			snd_soc_dapm_set_bias_level(socdev,
-						    SND_SOC_BIAS_PREPARE);
+	/* do we need to power up codec */
+	if (codec->bias_level != SND_SOC_BIAS_ON) {
+		snd_soc_dapm_set_bias_level(socdev,
+					    SND_SOC_BIAS_PREPARE);
 
-			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-				snd_soc_dapm_stream_event(codec,
+		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,
+		else
+			snd_soc_dapm_stream_event(codec,
 					codec_dai->capture.stream_name,
 					SND_SOC_DAPM_STREAM_START);
 
-			snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
-			snd_soc_dai_digital_mute(codec_dai, 0);
+		snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
+		snd_soc_dai_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,
+	} 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,
+		else
+			snd_soc_dapm_stream_event(codec,
 					codec_dai->capture.stream_name,
 					SND_SOC_DAPM_STREAM_START);
 
-			snd_soc_dai_digital_mute(codec_dai, 0);
-		}
+		snd_soc_dai_digital_mute(codec_dai, 0);
 	}
 
 out:
@@ -491,7 +478,8 @@ static int soc_pcm_hw_params(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_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret = 0;
@@ -507,7 +495,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (codec_dai->ops.hw_params) {
-		ret = codec_dai->ops.hw_params(substream, params);
+		ret = codec_dai->ops.hw_params(substream, params, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
 				codec_dai->name);
@@ -516,7 +504,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (cpu_dai->ops.hw_params) {
-		ret = cpu_dai->ops.hw_params(substream, params);
+		ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
@@ -539,11 +527,11 @@ out:
 
 platform_err:
 	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream);
+		cpu_dai->ops.hw_free(substream, cpu_dai);
 
 interface_err:
 	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream);
+		codec_dai->ops.hw_free(substream, codec_dai);
 
 codec_err:
 	if (machine->ops && machine->ops->hw_free)
@@ -561,7 +549,8 @@ 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_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
@@ -582,10 +571,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 
 	/* now free hw params for the DAI's  */
 	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream);
+		codec_dai->ops.hw_free(substream, codec_dai);
 
 	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream);
+		cpu_dai->ops.hw_free(substream, cpu_dai);
 
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -595,14 +584,15 @@ 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_card *card= socdev->card;
 	struct snd_soc_dai_link *machine = rtd->dai;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret;
 
 	if (codec_dai->ops.trigger) {
-		ret = codec_dai->ops.trigger(substream, cmd);
+		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -614,7 +604,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	}
 
 	if (cpu_dai->ops.trigger) {
-		ret = cpu_dai->ops.trigger(substream, cmd);
+		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -636,8 +626,8 @@ static struct snd_pcm_ops soc_pcm_ops = {
 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_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
 	int i;
@@ -653,29 +643,29 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot);
 
 	/* mute any active DAC's */
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
-		if (dai->dai_ops.digital_mute && dai->playback.active)
-			dai->dai_ops.digital_mute(dai, 1);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+		if (dai->ops.digital_mute && dai->playback.active)
+			dai->ops.digital_mute(dai, 1);
 	}
 
 	/* suspend all pcms */
-	for (i = 0; i < machine->num_links; i++)
-		snd_pcm_suspend_all(machine->dai_link[i].pcm);
+	for (i = 0; i < card->num_links; i++)
+		snd_pcm_suspend_all(card->dai_link[i].pcm);
 
-	if (machine->suspend_pre)
-		machine->suspend_pre(pdev, state);
+	if (card->suspend_pre)
+		card->suspend_pre(pdev, state);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_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);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
+		if (cpu_dai->suspend && !cpu_dai->ac97_control)
+			cpu_dai->suspend(cpu_dai);
 		if (platform->suspend)
-			platform->suspend(pdev, cpu_dai);
+			platform->suspend(cpu_dai);
 	}
 
 	/* close any waiting streams and save state */
-	run_delayed_work(&socdev->delayed_work);
+	run_delayed_work(&card->delayed_work);
 	codec->suspend_bias_level = codec->bias_level;
 
 	for (i = 0; i < codec->num_dai; i++) {
@@ -692,14 +682,14 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	if (codec_dev->suspend)
 		codec_dev->suspend(pdev, state);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_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);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+		if (cpu_dai->suspend && cpu_dai->ac97_control)
+			cpu_dai->suspend(cpu_dai);
 	}
 
-	if (machine->suspend_post)
-		machine->suspend_post(pdev, state);
+	if (card->suspend_post)
+		card->suspend_post(pdev, state);
 
 	return 0;
 }
@@ -709,11 +699,11 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
  */
 static void soc_resume_deferred(struct work_struct *work)
 {
-	struct snd_soc_device *socdev = container_of(work,
-						     struct snd_soc_device,
-						     deferred_resume_work);
-	struct snd_soc_machine *machine = socdev->machine;
-	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_card *card = container_of(work,
+						 struct snd_soc_card,
+						 deferred_resume_work);
+	struct snd_soc_device *socdev = card->socdev;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct platform_device *pdev = to_platform_device(socdev->dev);
@@ -723,15 +713,15 @@ static void soc_resume_deferred(struct work_struct *work)
 	 * so userspace apps are blocked from touching us
 	 */
 
-	dev_info(socdev->dev, "starting resume work\n");
+	dev_dbg(socdev->dev, "starting resume work\n");
 
-	if (machine->resume_pre)
-		machine->resume_pre(pdev);
+	if (card->resume_pre)
+		card->resume_pre(pdev);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_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);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+		if (cpu_dai->resume && cpu_dai->ac97_control)
+			cpu_dai->resume(cpu_dai);
 	}
 
 	if (codec_dev->resume)
@@ -749,24 +739,24 @@ static void soc_resume_deferred(struct work_struct *work)
 	}
 
 	/* unmute any active DACs */
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_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 < card->num_links; i++) {
+		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
+		if (dai->ops.digital_mute && dai->playback.active)
+			dai->ops.digital_mute(dai, 0);
 	}
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_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);
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
+		if (cpu_dai->resume && !cpu_dai->ac97_control)
+			cpu_dai->resume(cpu_dai);
 		if (platform->resume)
-			platform->resume(pdev, cpu_dai);
+			platform->resume(cpu_dai);
 	}
 
-	if (machine->resume_post)
-		machine->resume_post(pdev);
+	if (card->resume_post)
+		card->resume_post(pdev);
 
-	dev_info(socdev->dev, "resume work completed\n");
+	dev_dbg(socdev->dev, "resume work completed\n");
 
 	/* userspace can access us now we are back as we were before */
 	snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0);
@@ -776,11 +766,12 @@ static void soc_resume_deferred(struct work_struct *work)
 static int soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = socdev->card;
 
-	dev_info(socdev->dev, "scheduling resume work\n");
+	dev_dbg(socdev->dev, "scheduling resume work\n");
 
-	if (!schedule_work(&socdev->deferred_resume_work))
-		dev_err(socdev->dev, "work item may be lost\n");
+	if (!schedule_work(&card->deferred_resume_work))
+		dev_err(socdev->dev, "resume work item may be lost\n");
 
 	return 0;
 }
@@ -790,23 +781,83 @@ static int soc_resume(struct platform_device *pdev)
 #define soc_resume	NULL
 #endif
 
-/* probes a new socdev */
-static int soc_probe(struct platform_device *pdev)
+static void snd_soc_instantiate_card(struct snd_soc_card *card)
 {
-	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;
+	struct platform_device *pdev = container_of(card->dev,
+						    struct platform_device,
+						    dev);
+	struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
+	struct snd_soc_platform *platform;
+	struct snd_soc_dai *dai;
+	int i, found, ret, ac97;
+
+	if (card->instantiated)
+		return;
+
+	found = 0;
+	list_for_each_entry(platform, &platform_list, list)
+		if (card->platform == platform) {
+			found = 1;
+			break;
+		}
+	if (!found) {
+		dev_dbg(card->dev, "Platform %s not registered\n",
+			card->platform->name);
+		return;
+	}
+
+	ac97 = 0;
+	for (i = 0; i < card->num_links; i++) {
+		found = 0;
+		list_for_each_entry(dai, &dai_list, list)
+			if (card->dai_link[i].cpu_dai == dai) {
+				found = 1;
+				break;
+			}
+		if (!found) {
+			dev_dbg(card->dev, "DAI %s not registered\n",
+				card->dai_link[i].cpu_dai->name);
+			return;
+		}
 
-	if (machine->probe) {
-		ret = machine->probe(pdev);
+		if (card->dai_link[i].cpu_dai->ac97_control)
+			ac97 = 1;
+	}
+
+	/* If we have AC97 in the system then don't wait for the
+	 * codec.  This will need revisiting if we have to handle
+	 * systems with mixed AC97 and non-AC97 parts.  Only check for
+	 * DAIs currently; we can't do this per link since some AC97
+	 * codecs have non-AC97 DAIs.
+	 */
+	if (!ac97)
+		for (i = 0; i < card->num_links; i++) {
+			found = 0;
+			list_for_each_entry(dai, &dai_list, list)
+				if (card->dai_link[i].codec_dai == dai) {
+					found = 1;
+					break;
+				}
+			if (!found) {
+				dev_dbg(card->dev, "DAI %s not registered\n",
+					card->dai_link[i].codec_dai->name);
+				return;
+			}
+		}
+
+	/* Note that we do not current check for codec components */
+
+	dev_dbg(card->dev, "All components present, instantiating\n");
+
+	/* Found everything, bring it up */
+	if (card->probe) {
+		ret = card->probe(pdev);
 		if (ret < 0)
-			return ret;
+			return;
 	}
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->probe) {
 			ret = cpu_dai->probe(pdev, cpu_dai);
 			if (ret < 0)
@@ -827,13 +878,15 @@ static int soc_probe(struct platform_device *pdev)
 	}
 
 	/* DAPM stream work */
-	INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
+	INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
 #ifdef CONFIG_PM
 	/* deferred resume work */
-	INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred);
+	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
 #endif
 
-	return 0;
+	card->instantiated = 1;
+
+	return;
 
 platform_err:
 	if (codec_dev->remove)
@@ -841,15 +894,45 @@ platform_err:
 
 cpu_dai_err:
 	for (i--; i >= 0; i--) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->remove)
 			cpu_dai->remove(pdev, cpu_dai);
 	}
 
-	if (machine->remove)
-		machine->remove(pdev);
+	if (card->remove)
+		card->remove(pdev);
+}
 
-	return ret;
+/*
+ * Attempt to initialise any uninitalised cards.  Must be called with
+ * client_mutex.
+ */
+static void snd_soc_instantiate_cards(void)
+{
+	struct snd_soc_card *card;
+	list_for_each_entry(card, &card_list, list)
+		snd_soc_instantiate_card(card);
+}
+
+/* probes a new socdev */
+static int soc_probe(struct platform_device *pdev)
+{
+	int ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_card *card = socdev->card;
+
+	/* Bodge while we push things out of socdev */
+	card->socdev = socdev;
+
+	/* Bodge while we unpick instantiation */
+	card->dev = &pdev->dev;
+	ret = snd_soc_register_card(card);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to register card\n");
+		return ret;
+	}
+
+	return 0;
 }
 
 /* removes a socdev */
@@ -857,11 +940,11 @@ 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_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
-	run_delayed_work(&socdev->delayed_work);
+	run_delayed_work(&card->delayed_work);
 
 	if (platform->remove)
 		platform->remove(pdev);
@@ -869,14 +952,16 @@ static int soc_remove(struct platform_device *pdev)
 	if (codec_dev->remove)
 		codec_dev->remove(pdev);
 
-	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	for (i = 0; i < card->num_links; i++) {
+		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
 		if (cpu_dai->remove)
 			cpu_dai->remove(pdev, cpu_dai);
 	}
 
-	if (machine->remove)
-		machine->remove(pdev);
+	if (card->remove)
+		card->remove(pdev);
+
+	snd_soc_unregister_card(card);
 
 	return 0;
 }
@@ -898,6 +983,8 @@ 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_card *card = socdev->card;
+	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *codec_dai = dai_link->codec_dai;
 	struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
 	struct snd_soc_pcm_runtime *rtd;
@@ -914,8 +1001,8 @@ static int soc_new_pcm(struct snd_soc_device *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);
+	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
+		num);
 
 	if (codec_dai->playback.channels_min)
 		playback = 1;
@@ -933,13 +1020,13 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 
 	dai_link->pcm = pcm;
 	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;
+	soc_pcm_ops.mmap = platform->pcm_ops->mmap;
+	soc_pcm_ops.pointer = platform->pcm_ops->pointer;
+	soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
+	soc_pcm_ops.copy = platform->pcm_ops->copy;
+	soc_pcm_ops.silence = platform->pcm_ops->silence;
+	soc_pcm_ops.ack = platform->pcm_ops->ack;
+	soc_pcm_ops.page = platform->pcm_ops->page;
 
 	if (playback)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
@@ -947,24 +1034,22 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	if (capture)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
 
-	ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
+	ret = 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;
+	pcm->private_free = 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)
+static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
 {
-	struct snd_soc_device *devdata = dev_get_drvdata(dev);
 	struct snd_soc_codec *codec = devdata->codec;
 	int i, step = 1, count = 0;
 
@@ -1001,8 +1086,110 @@ static ssize_t codec_reg_show(struct device *dev,
 
 	return count;
 }
+static ssize_t codec_reg_show(struct device *dev,
+	struct device_attribute *attr, char *buf)
+{
+	struct snd_soc_device *devdata = dev_get_drvdata(dev);
+	return soc_codec_reg_show(devdata, buf);
+}
+
 static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
 
+#ifdef CONFIG_DEBUG_FS
+static int codec_reg_open_file(struct inode *inode, struct file *file)
+{
+	file->private_data = inode->i_private;
+	return 0;
+}
+
+static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
+			       size_t count, loff_t *ppos)
+{
+	ssize_t ret;
+	struct snd_soc_codec *codec = file->private_data;
+	struct device *card_dev = codec->card->dev;
+	struct snd_soc_device *devdata = card_dev->driver_data;
+	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+	ret = soc_codec_reg_show(devdata, buf);
+	if (ret >= 0)
+		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+	kfree(buf);
+	return ret;
+}
+
+static ssize_t codec_reg_write_file(struct file *file,
+		const char __user *user_buf, size_t count, loff_t *ppos)
+{
+	char buf[32];
+	int buf_size;
+	char *start = buf;
+	unsigned long reg, value;
+	int step = 1;
+	struct snd_soc_codec *codec = file->private_data;
+
+	buf_size = min(count, (sizeof(buf)-1));
+	if (copy_from_user(buf, user_buf, buf_size))
+		return -EFAULT;
+	buf[buf_size] = 0;
+
+	if (codec->reg_cache_step)
+		step = codec->reg_cache_step;
+
+	while (*start == ' ')
+		start++;
+	reg = simple_strtoul(start, &start, 16);
+	if ((reg >= codec->reg_cache_size) || (reg % step))
+		return -EINVAL;
+	while (*start == ' ')
+		start++;
+	if (strict_strtoul(start, 16, &value))
+		return -EINVAL;
+	codec->write(codec, reg, value);
+	return buf_size;
+}
+
+static const struct file_operations codec_reg_fops = {
+	.open = codec_reg_open_file,
+	.read = codec_reg_read_file,
+	.write = codec_reg_write_file,
+};
+
+static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+{
+	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
+						 debugfs_root, codec,
+						 &codec_reg_fops);
+	if (!codec->debugfs_reg)
+		printk(KERN_WARNING
+		       "ASoC: Failed to create codec register debugfs file\n");
+
+	codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
+						     debugfs_root,
+						     &codec->pop_time);
+	if (!codec->debugfs_pop_time)
+		printk(KERN_WARNING
+		       "Failed to create pop time debugfs file\n");
+}
+
+static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
+{
+	debugfs_remove(codec->debugfs_pop_time);
+	debugfs_remove(codec->debugfs_reg);
+}
+
+#else
+
+static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+{
+}
+
+static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
+{
+}
+#endif
+
 /**
  * snd_soc_new_ac97_codec - initailise AC97 device
  * @codec: audio codec
@@ -1121,7 +1308,7 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits);
 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;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0, i;
 
 	mutex_lock(&codec->mutex);
@@ -1140,11 +1327,11 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 	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);
+	for (i = 0; i < card->num_links; i++) {
+		ret = soc_new_pcm(socdev, &card->dai_link[i], i);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't create pcm %s\n",
-				machine->dai_link[i].stream_name);
+				card->dai_link[i].stream_name);
 			mutex_unlock(&codec->mutex);
 			return ret;
 		}
@@ -1156,7 +1343,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
 
 /**
- * snd_soc_register_card - register sound card
+ * snd_soc_init_card - register sound card
  * @socdev: the SoC audio device
  *
  * Register a SoC sound card. Also registers an AC97 device if the
@@ -1164,29 +1351,28 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
  *
  * Returns 0 for success, else error.
  */
-int snd_soc_register_card(struct snd_soc_device *socdev)
+int snd_soc_init_card(struct snd_soc_device *socdev)
 {
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0, i, ac97 = 0, err = 0;
 
-	for (i = 0; i < machine->num_links; i++) {
-		if (socdev->machine->dai_link[i].init) {
-			err = socdev->machine->dai_link[i].init(codec);
+	for (i = 0; i < card->num_links; i++) {
+		if (card->dai_link[i].init) {
+			err = card->dai_link[i].init(codec);
 			if (err < 0) {
 				printk(KERN_ERR "asoc: failed to init %s\n",
-					socdev->machine->dai_link[i].stream_name);
+					card->dai_link[i].stream_name);
 				continue;
 			}
 		}
-		if (socdev->machine->dai_link[i].codec_dai->type ==
-			SND_SOC_DAI_AC97_BUS)
+		if (card->dai_link[i].codec_dai->ac97_control)
 			ac97 = 1;
 	}
 	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
-		 "%s", machine->name);
+		 "%s",  card->name);
 	snprintf(codec->card->longname, sizeof(codec->card->longname),
-		 "%s (%s)", machine->name, codec->name);
+		 "%s (%s)", card->name, codec->name);
 
 	ret = snd_card_register(codec->card);
 	if (ret < 0) {
@@ -1216,12 +1402,13 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 	if (err < 0)
 		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
+	soc_init_codec_debugfs(socdev->codec);
 	mutex_unlock(&codec->mutex);
 
 out:
 	return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_register_card);
+EXPORT_SYMBOL_GPL(snd_soc_init_card);
 
 /**
  * snd_soc_free_pcms - free sound card and pcms
@@ -1239,10 +1426,11 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev)
 #endif
 
 	mutex_lock(&codec->mutex);
+	soc_cleanup_codec_debugfs(socdev->codec);
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
-		if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
+		if (codec_dai->ac97_control && codec->ac97) {
 			soc_ac97_dev_unregister(codec);
 			goto free_card;
 		}
@@ -1756,8 +1944,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
 	else
 		return -EINVAL;
 }
@@ -1776,8 +1964,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 	int div_id, int div)
 {
-	if (dai->dai_ops.set_clkdiv)
-		return dai->dai_ops.set_clkdiv(dai, div_id, div);
+	if (dai->ops.set_clkdiv)
+		return dai->ops.set_clkdiv(dai, div_id, div);
 	else
 		return -EINVAL;
 }
@@ -1795,8 +1983,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	if (dai->dai_ops.set_pll)
-		return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out);
+	if (dai->ops.set_pll)
+		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
 	else
 		return -EINVAL;
 }
@@ -1805,15 +1993,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
 /**
  * snd_soc_dai_set_fmt - configure DAI hardware audio format.
  * @dai: DAI
- * @clk_id: DAI specific clock ID
  * @fmt: SND_SOC_DAIFMT_ format value.
  *
  * Configures the DAI hardware format and clocking.
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->dai_ops.set_fmt)
-		return dai->dai_ops.set_fmt(dai, fmt);
+	if (dai->ops.set_fmt)
+		return dai->ops.set_fmt(dai, fmt);
 	else
 		return -EINVAL;
 }
@@ -1831,8 +2018,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int mask, int slots)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_tdm_slot(dai, mask, slots);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_tdm_slot(dai, mask, slots);
 	else
 		return -EINVAL;
 }
@@ -1847,8 +2034,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-	if (dai->dai_ops.set_sysclk)
-		return dai->dai_ops.set_tristate(dai, tristate);
+	if (dai->ops.set_sysclk)
+		return dai->ops.set_tristate(dai, tristate);
 	else
 		return -EINVAL;
 }
@@ -1863,21 +2050,242 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-	if (dai->dai_ops.digital_mute)
-		return dai->dai_ops.digital_mute(dai, mute);
+	if (dai->ops.digital_mute)
+		return dai->ops.digital_mute(dai, mute);
 	else
 		return -EINVAL;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
 
-static int __devinit snd_soc_init(void)
+/**
+ * snd_soc_register_card - Register a card with the ASoC core
+ *
+ * @param card Card to register
+ *
+ * Note that currently this is an internal only function: it will be
+ * exposed to machine drivers after further backporting of ASoC v2
+ * registration APIs.
+ */
+static int snd_soc_register_card(struct snd_soc_card *card)
+{
+	if (!card->name || !card->dev)
+		return -EINVAL;
+
+	INIT_LIST_HEAD(&card->list);
+	card->instantiated = 0;
+
+	mutex_lock(&client_mutex);
+	list_add(&card->list, &card_list);
+	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(card->dev, "Registered card '%s'\n", card->name);
+
+	return 0;
+}
+
+/**
+ * snd_soc_unregister_card - Unregister a card with the ASoC core
+ *
+ * @param card Card to unregister
+ *
+ * Note that currently this is an internal only function: it will be
+ * exposed to machine drivers after further backporting of ASoC v2
+ * registration APIs.
+ */
+static int snd_soc_unregister_card(struct snd_soc_card *card)
+{
+	mutex_lock(&client_mutex);
+	list_del(&card->list);
+	mutex_unlock(&client_mutex);
+
+	dev_dbg(card->dev, "Unregistered card '%s'\n", card->name);
+
+	return 0;
+}
+
+/**
+ * snd_soc_register_dai - Register a DAI with the ASoC core
+ *
+ * @param dai DAI to register
+ */
+int snd_soc_register_dai(struct snd_soc_dai *dai)
+{
+	if (!dai->name)
+		return -EINVAL;
+
+	/* The device should become mandatory over time */
+	if (!dai->dev)
+		printk(KERN_WARNING "No device for DAI %s\n", dai->name);
+
+	INIT_LIST_HEAD(&dai->list);
+
+	mutex_lock(&client_mutex);
+	list_add(&dai->list, &dai_list);
+	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Registered DAI '%s'\n", dai->name);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_dai);
+
+/**
+ * snd_soc_unregister_dai - Unregister a DAI from the ASoC core
+ *
+ * @param dai DAI to unregister
+ */
+void snd_soc_unregister_dai(struct snd_soc_dai *dai)
+{
+	mutex_lock(&client_mutex);
+	list_del(&dai->list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Unregistered DAI '%s'\n", dai->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);
+
+/**
+ * snd_soc_register_dais - Register multiple DAIs with the ASoC core
+ *
+ * @param dai Array of DAIs to register
+ * @param count Number of DAIs
+ */
+int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count)
+{
+	int i, ret;
+
+	for (i = 0; i < count; i++) {
+		ret = snd_soc_register_dai(&dai[i]);
+		if (ret != 0)
+			goto err;
+	}
+
+	return 0;
+
+err:
+	for (i--; i >= 0; i--)
+		snd_soc_unregister_dai(&dai[i]);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_dais);
+
+/**
+ * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
+ *
+ * @param dai Array of DAIs to unregister
+ * @param count Number of DAIs
+ */
+void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
+{
+	int i;
+
+	for (i = 0; i < count; i++)
+		snd_soc_unregister_dai(&dai[i]);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);
+
+/**
+ * snd_soc_register_platform - Register a platform with the ASoC core
+ *
+ * @param platform platform to register
+ */
+int snd_soc_register_platform(struct snd_soc_platform *platform)
+{
+	if (!platform->name)
+		return -EINVAL;
+
+	INIT_LIST_HEAD(&platform->list);
+
+	mutex_lock(&client_mutex);
+	list_add(&platform->list, &platform_list);
+	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Registered platform '%s'\n", platform->name);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_platform);
+
+/**
+ * snd_soc_unregister_platform - Unregister a platform from the ASoC core
+ *
+ * @param platform platform to unregister
+ */
+void snd_soc_unregister_platform(struct snd_soc_platform *platform)
 {
-	printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
+	mutex_lock(&client_mutex);
+	list_del(&platform->list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Unregistered platform '%s'\n", platform->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
+
+/**
+ * snd_soc_register_codec - Register a codec with the ASoC core
+ *
+ * @param codec codec to register
+ */
+int snd_soc_register_codec(struct snd_soc_codec *codec)
+{
+	if (!codec->name)
+		return -EINVAL;
+
+	/* The device should become mandatory over time */
+	if (!codec->dev)
+		printk(KERN_WARNING "No device for codec %s\n", codec->name);
+
+	INIT_LIST_HEAD(&codec->list);
+
+	mutex_lock(&client_mutex);
+	list_add(&codec->list, &codec_list);
+	snd_soc_instantiate_cards();
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Registered codec '%s'\n", codec->name);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_register_codec);
+
+/**
+ * snd_soc_unregister_codec - Unregister a codec from the ASoC core
+ *
+ * @param codec codec to unregister
+ */
+void snd_soc_unregister_codec(struct snd_soc_codec *codec)
+{
+	mutex_lock(&client_mutex);
+	list_del(&codec->list);
+	mutex_unlock(&client_mutex);
+
+	pr_debug("Unregistered codec '%s'\n", codec->name);
+}
+EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);
+
+static int __init snd_soc_init(void)
+{
+#ifdef CONFIG_DEBUG_FS
+	debugfs_root = debugfs_create_dir("asoc", NULL);
+	if (IS_ERR(debugfs_root) || !debugfs_root) {
+		printk(KERN_WARNING
+		       "ASoC: Failed to create debugfs directory\n");
+		debugfs_root = NULL;
+	}
+#endif
+
 	return platform_driver_register(&soc_driver);
 }
 
-static void snd_soc_exit(void)
+static void __exit snd_soc_exit(void)
 {
+#ifdef CONFIG_DEBUG_FS
+	debugfs_remove_recursive(debugfs_root);
+#endif
 	platform_driver_unregister(&soc_driver);
 }
 
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7351db9606e..8863eddbac0 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -37,7 +37,6 @@
 #include <linux/bitops.h>
 #include <linux/platform_device.h>
 #include <linux/jiffies.h>
-#include <linux/debugfs.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -67,17 +66,13 @@ static int dapm_status = 1;
 module_param(dapm_status, int, 0);
 MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
 
-static struct dentry *asoc_debugfs;
-
-static u32 pop_time;
-
-static void pop_wait(void)
+static void pop_wait(u32 pop_time)
 {
 	if (pop_time)
 		schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time));
 }
 
-static void pop_dbg(const char *fmt, ...)
+static void pop_dbg(u32 pop_time, const char *fmt, ...)
 {
 	va_list args;
 
@@ -85,7 +80,7 @@ static void pop_dbg(const char *fmt, ...)
 
 	if (pop_time) {
 		vprintk(fmt, args);
-		pop_wait();
+		pop_wait(pop_time);
 	}
 
 	va_end(args);
@@ -230,10 +225,11 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
 
 	change = old != new;
 	if (change) {
-		pop_dbg("pop test %s : %s in %d ms\n", widget->name,
-			widget->power ? "on" : "off", pop_time);
+		pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n",
+			widget->name, widget->power ? "on" : "off",
+			codec->pop_time);
 		snd_soc_write(codec, widget->reg, new);
-		pop_wait();
+		pop_wait(codec->pop_time);
 	}
 	pr_debug("reg %x old %x new %x change %d\n", widget->reg,
 		 old, new, change);
@@ -293,7 +289,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
 	struct snd_soc_dapm_widget *w)
 {
 	int i, ret = 0;
-	char name[32];
+	size_t name_len;
 	struct snd_soc_dapm_path *path;
 
 	/* add kcontrol */
@@ -307,11 +303,16 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
 				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);
+			name_len = 2 + strlen(w->name)
+				+ strlen(w->kcontrols[i].name);
+			path->long_name = kmalloc(name_len, GFP_KERNEL);
 			if (path->long_name == NULL)
 				return -ENOMEM;
 
+			snprintf(path->long_name, name_len, "%s %s",
+				 w->name, w->kcontrols[i].name);
+			path->long_name[name_len - 1] = '\0';
+
 			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
 				path->long_name);
 			ret = snd_ctl_add(codec->card, path->kcontrol);
@@ -821,23 +822,9 @@ 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)
 		return 0;
-
-	ret = device_create_file(dev, &dev_attr_dapm_widget);
-	if (ret != 0)
-		return ret;
-
-	asoc_debugfs = debugfs_create_dir("asoc", NULL);
-	if (!IS_ERR(asoc_debugfs) && asoc_debugfs)
-		debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs,
-				   &pop_time);
-	else
-		asoc_debugfs = NULL;
-
-	return 0;
+	return device_create_file(dev, &dev_attr_dapm_widget);
 }
 
 static void snd_soc_dapm_sys_remove(struct device *dev)
@@ -845,9 +832,6 @@ static void snd_soc_dapm_sys_remove(struct device *dev)
 	if (dapm_status) {
 		device_remove_file(dev, &dev_attr_dapm_widget);
 	}
-
-	if (asoc_debugfs)
-		debugfs_remove_recursive(asoc_debugfs);
 }
 
 /* free all dapm widgets and resources */
@@ -1007,28 +991,6 @@ err:
 }
 
 /**
- * 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.
- *
- * This function has been deprecated in favour of snd_soc_dapm_add_routes().
- *
- * 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)
-{
-	return snd_soc_dapm_add_route(codec, sink, control, source);
-}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input);
-
-/**
  * snd_soc_dapm_add_routes - Add routes between DAPM widgets
  * @codec: codec
  * @route: audio routes
@@ -1358,8 +1320,12 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
 
 	for (i = 0; i < num; i++) {
 		ret = snd_soc_dapm_new_control(codec, widget);
-		if (ret < 0)
+		if (ret < 0) {
+			printk(KERN_ERR
+			       "ASoC: Failed to create DAPM control %s: %d\n",
+			       widget->name, ret);
 			return ret;
+		}
 		widget++;
 	}
 	return 0;
@@ -1440,11 +1406,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
 				enum snd_soc_bias_level level)
 {
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_card *card = socdev->card;
 	int ret = 0;
 
-	if (machine->set_bias_level)
-		ret = machine->set_bias_level(machine, level);
+	if (card->set_bias_level)
+		ret = card->set_bias_level(card, level);
 	if (ret == 0 && codec->set_bias_level)
 		ret = codec->set_bias_level(codec, level);