Merge commit '85082fd7cbe3173198aac0eb5e85ab1edcc6352c' into test-build

Manual fixup of:

	arch/powerpc/Kconfig

92 files changed, 10356 insertions, 1164 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 18f28ac4bfe8..f743530add8f 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -2,15 +2,8 @@
 # SoC audio configuration
 #
 
-menu "System on Chip audio support"
-	depends on SND!=n
-
-config SND_SOC_AC97_BUS
-	bool
-
-config SND_SOC
+menuconfig SND_SOC
 	tristate "ALSA for SoC audio support"
-	depends on SND
 	select SND_PCM
 	---help---
 
@@ -23,8 +16,15 @@ config SND_SOC
 	  This ASoC audio support can also be built as a module.  If so, the module
 	  will be called snd-soc-core.
 
+if SND_SOC
+
+config SND_SOC_AC97_BUS
+	bool
+
 # All the supported Soc's
+source "sound/soc/at32/Kconfig"
 source "sound/soc/at91/Kconfig"
+source "sound/soc/au1x/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
 source "sound/soc/sh/Kconfig"
@@ -35,4 +35,5 @@ source "sound/soc/omap/Kconfig"
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
 
-endmenu
+endif	# SND_SOC
+
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 782db2127108..933a66d30804 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,5 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
-obj-$(CONFIG_SND_SOC)	+= codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/
+obj-$(CONFIG_SND_SOC)	+= codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
+obj-$(CONFIG_SND_SOC)	+= omap/ au1x/
diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig
new file mode 100644
index 000000000000..b0765e86c085
--- /dev/null
+++ b/sound/soc/at32/Kconfig
@@ -0,0 +1,34 @@
+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
new file mode 100644
index 000000000000..c03e55ececeb
--- /dev/null
+++ b/sound/soc/at32/Makefile
@@ -0,0 +1,11 @@
+# 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
new file mode 100644
index 000000000000..435f1daf177c
--- /dev/null
+++ b/sound/soc/at32/at32-pcm.c
@@ -0,0 +1,491 @@
+/* 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, 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, 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
new file mode 100644
index 000000000000..2a52430417da
--- /dev/null
+++ b/sound/soc/at32/at32-pcm.h
@@ -0,0 +1,79 @@
+/* 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
new file mode 100644
index 000000000000..4ef6492c902e
--- /dev/null
+++ b/sound/soc/at32/at32-ssc.c
@@ -0,0 +1,849 @@
+/* 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
new file mode 100644
index 000000000000..3c052dbbe460
--- /dev/null
+++ b/sound/soc/at32/at32-ssc.h
@@ -0,0 +1,59 @@
+/* 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/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c
new file mode 100644
index 000000000000..fee5f8e58957
--- /dev/null
+++ b/sound/soc/at32/playpaq_wm8510.c
@@ -0,0 +1,522 @@
+/* sound/soc/at32/playpaq_wm8510.c
+ * ASoC machine driver for PlayPaq using WM8510 codec
+ *
+ * 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.
+ *
+ * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c
+ *
+ * NOTE: If you don't have the AT32 enhanced portmux configured (which
+ * isn't currently in the mainline or Atmel patched kernel), you will
+ * need to set the MCLK pin (PA30) to peripheral A in your board initialization
+ * code.  Something like:
+ *	at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0);
+ *
+ */
+
+/* #define DEBUG */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/errno.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/arch/at32ap700x.h>
+#include <asm/arch/portmux.h>
+
+#include "../codecs/wm8510.h"
+#include "at32-pcm.h"
+#include "at32-ssc.h"
+
+
+/*-------------------------------------------------------------------------*\
+ * constants
+\*-------------------------------------------------------------------------*/
+#define MCLK_PIN		GPIO_PIN_PA(30)
+#define MCLK_PERIPH		GPIO_PERIPH_A
+
+
+/*-------------------------------------------------------------------------*\
+ * data types
+\*-------------------------------------------------------------------------*/
+/* SSC clocking data */
+struct ssc_clock_data {
+	/* CMR div */
+	unsigned int cmr_div;
+
+	/* Frame period (as needed by xCMR.PERIOD) */
+	unsigned int period;
+
+	/* The SSC clock rate these settings where calculated for */
+	unsigned long ssc_rate;
+};
+
+
+/*-------------------------------------------------------------------------*\
+ * module data
+\*-------------------------------------------------------------------------*/
+static struct clk *_gclk0;
+static struct clk *_pll0;
+
+#define CODEC_CLK (_gclk0)
+
+
+/*-------------------------------------------------------------------------*\
+ * Sound SOC operations
+\*-------------------------------------------------------------------------*/
+#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
+static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *cpu_dai)
+{
+	struct at32_ssc_info *ssc_p = cpu_dai->private_data;
+	struct ssc_device *ssc = ssc_p->ssc;
+	struct ssc_clock_data cd;
+	unsigned int rate, width_bits, channels;
+	unsigned int bitrate, ssc_div;
+	unsigned actual_rate;
+
+
+	/*
+	 * Figure out required bitrate
+	 */
+	rate = params_rate(params);
+	channels = params_channels(params);
+	width_bits = snd_pcm_format_physical_width(params_format(params));
+	bitrate = rate * width_bits * channels;
+
+
+	/*
+	 * Figure out required SSC divider and period for required bitrate
+	 */
+	cd.ssc_rate = clk_get_rate(ssc->clk);
+	ssc_div = cd.ssc_rate / bitrate;
+	cd.cmr_div = ssc_div / 2;
+	if (ssc_div & 1) {
+		/* round cmr_div up */
+		cd.cmr_div++;
+	}
+	cd.period = width_bits - 1;
+
+
+	/*
+	 * Find actual rate, compare to requested rate
+	 */
+	actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
+	pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n",
+		 rate, actual_rate);
+
+
+	return cd;
+}
+#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
+
+
+
+static int playpaq_wm8510_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 at32_ssc_info *ssc_p = cpu_dai->private_data;
+	struct ssc_device *ssc = ssc_p->ssc;
+	unsigned int pll_out = 0, bclk = 0, mclk_div = 0;
+	int ret;
+
+
+	/* Due to difficulties with getting the correct clocks from the AT32's
+	 * PLL0, we're going to let the CODEC be in charge of all the clocks
+	 */
+#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
+	const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+#else
+	struct ssc_clock_data cd;
+	const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+#endif
+
+	if (ssc == NULL) {
+		pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n");
+		return -EINVAL;
+	}
+
+
+	/*
+	 * Figure out PLL and BCLK dividers for WM8510
+	 */
+	switch (params_rate(params)) {
+	case 48000:
+		pll_out = 12288000;
+		mclk_div = WM8510_MCLKDIV_1;
+		bclk = WM8510_BCLKDIV_8;
+		break;
+
+	case 44100:
+		pll_out = 11289600;
+		mclk_div = WM8510_MCLKDIV_1;
+		bclk = WM8510_BCLKDIV_8;
+		break;
+
+	case 22050:
+		pll_out = 11289600;
+		mclk_div = WM8510_MCLKDIV_2;
+		bclk = WM8510_BCLKDIV_8;
+		break;
+
+	case 16000:
+		pll_out = 12288000;
+		mclk_div = WM8510_MCLKDIV_3;
+		bclk = WM8510_BCLKDIV_8;
+		break;
+
+	case 11025:
+		pll_out = 11289600;
+		mclk_div = WM8510_MCLKDIV_4;
+		bclk = WM8510_BCLKDIV_8;
+		break;
+
+	case 8000:
+		pll_out = 12288000;
+		mclk_div = WM8510_MCLKDIV_6;
+		bclk = WM8510_BCLKDIV_8;
+		break;
+
+	default:
+		pr_warning("playpaq_wm8510: Unsupported sample rate %d\n",
+			   params_rate(params));
+		return -EINVAL;
+	}
+
+
+	/*
+	 * set CPU and CODEC DAI configuration
+	 */
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0) {
+		pr_warning("playpaq_wm8510: "
+			   "Failed to set CODEC DAI format (%d)\n",
+			   ret);
+		return ret;
+	}
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0) {
+		pr_warning("playpaq_wm8510: "
+			   "Failed to set CPU DAI format (%d)\n",
+			   ret);
+		return ret;
+	}
+
+
+	/*
+	 * Set CPU clock configuration
+	 */
+#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
+	cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai);
+	pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n",
+		 cd.cmr_div, cd.period);
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div);
+	if (ret < 0) {
+		pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n",
+			   ret);
+		return ret;
+	}
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD,
+					  cd.period);
+	if (ret < 0) {
+		pr_warning("playpaq_wm8510: "
+			   "Failed to set CPU transmit period (%d)\n",
+			   ret);
+		return ret;
+	}
+#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
+
+
+	/*
+	 * Set CODEC clock configuration
+	 */
+	pr_debug("playpaq_wm8510: "
+		 "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n",
+		 clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div);
+
+
+#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk);
+	if (ret < 0) {
+		pr_warning
+		    ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n",
+		     ret);
+		return ret;
+	}
+#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
+
+
+	ret = snd_soc_dai_set_pll(codec_dai, 0,
+					 clk_get_rate(CODEC_CLK), pll_out);
+	if (ret < 0) {
+		pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
+			   ret);
+		return ret;
+	}
+
+
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div);
+	if (ret < 0) {
+		pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n",
+			   ret);
+		return ret;
+	}
+
+
+	return 0;
+}
+
+
+
+static struct snd_soc_ops playpaq_wm8510_ops = {
+	.hw_params = playpaq_wm8510_hw_params,
+};
+
+
+
+static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+
+
+static const char *intercon[][3] = {
+	/* speaker connected to SPKOUT */
+	{"Ext Spk", NULL, "SPKOUTP"},
+	{"Ext Spk", NULL, "SPKOUTN"},
+
+	{"Mic Bias", NULL, "Int Mic"},
+	{"MICN", NULL, "Mic Bias"},
+	{"MICP", NULL, "Mic Bias"},
+
+	/* Terminator */
+	{NULL, NULL, NULL},
+};
+
+
+
+static int playpaq_wm8510_init(struct snd_soc_codec *codec)
+{
+	int i;
+
+	/*
+	 * Add DAPM widgets
+	 */
+	for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
+		snd_soc_dapm_new_control(codec, &playpaq_dapm_widgets[i]);
+
+
+
+	/*
+	 * Setup audio path interconnects
+	 */
+	for (i = 0; intercon[i][0] != NULL; i++) {
+		snd_soc_dapm_connect_input(codec,
+					   intercon[i][0],
+					   intercon[i][1], intercon[i][2]);
+	}
+
+
+	/* always connected pins */
+	snd_soc_dapm_enable_pin(codec, "Int Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	snd_soc_dapm_sync(codec);
+
+
+
+	/* Make CSB show PLL rate */
+	snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV,
+				       WM8510_OPCLKDIV_1 | 4);
+
+	return 0;
+}
+
+
+
+static struct snd_soc_dai_link playpaq_wm8510_dai = {
+	.name = "WM8510",
+	.stream_name = "WM8510 PCM",
+	.cpu_dai = &at32_ssc_dai[0],
+	.codec_dai = &wm8510_dai,
+	.init = playpaq_wm8510_init,
+	.ops = &playpaq_wm8510_ops,
+};
+
+
+
+static struct snd_soc_machine snd_soc_machine_playpaq = {
+	.name = "LRS_PlayPaq_WM8510",
+	.dai_link = &playpaq_wm8510_dai,
+	.num_links = 1,
+};
+
+
+
+static struct wm8510_setup_data playpaq_wm8510_setup = {
+	.i2c_address = 0x1a,
+};
+
+
+
+static struct snd_soc_device playpaq_wm8510_snd_devdata = {
+	.machine = &snd_soc_machine_playpaq,
+	.platform = &at32_soc_platform,
+	.codec_dev = &soc_codec_dev_wm8510,
+	.codec_data = &playpaq_wm8510_setup,
+};
+
+static struct platform_device *playpaq_snd_device;
+
+
+static int __init playpaq_asoc_init(void)
+{
+	int ret = 0;
+	struct at32_ssc_info *ssc_p = playpaq_wm8510_dai.cpu_dai->private_data;
+	struct ssc_device *ssc = NULL;
+
+
+	/*
+	 * Request SSC device
+	 */
+	ssc = ssc_request(0);
+	if (IS_ERR(ssc)) {
+		ret = PTR_ERR(ssc);
+		ssc = NULL;
+		goto err_ssc;
+	}
+	ssc_p->ssc = ssc;
+
+
+	/*
+	 * Configure MCLK for WM8510
+	 */
+	_gclk0 = clk_get(NULL, "gclk0");
+	if (IS_ERR(_gclk0)) {
+		_gclk0 = NULL;
+		goto err_gclk0;
+	}
+	_pll0 = clk_get(NULL, "pll0");
+	if (IS_ERR(_pll0)) {
+		_pll0 = NULL;
+		goto err_pll0;
+	}
+	if (clk_set_parent(_gclk0, _pll0)) {
+		pr_warning("snd-soc-playpaq: "
+			   "Failed to set PLL0 as parent for DAC clock\n");
+		goto err_set_clk;
+	}
+	clk_set_rate(CODEC_CLK, 12000000);
+	clk_enable(CODEC_CLK);
+
+#if defined CONFIG_AT32_ENHANCED_PORTMUX
+	at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0);
+#endif
+
+
+	/*
+	 * Create and register platform device
+	 */
+	playpaq_snd_device = platform_device_alloc("soc-audio", 0);
+	if (playpaq_snd_device == NULL) {
+		ret = -ENOMEM;
+		goto err_device_alloc;
+	}
+
+	platform_set_drvdata(playpaq_snd_device, &playpaq_wm8510_snd_devdata);
+	playpaq_wm8510_snd_devdata.dev = &playpaq_snd_device->dev;
+
+	ret = platform_device_add(playpaq_snd_device);
+	if (ret) {
+		pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n",
+			   ret);
+		goto err_device_add;
+	}
+
+	return 0;
+
+
+err_device_add:
+	if (playpaq_snd_device != NULL) {
+		platform_device_put(playpaq_snd_device);
+		playpaq_snd_device = NULL;
+	}
+err_device_alloc:
+err_set_clk:
+	if (_pll0 != NULL) {
+		clk_put(_pll0);
+		_pll0 = NULL;
+	}
+err_pll0:
+	if (_gclk0 != NULL) {
+		clk_put(_gclk0);
+		_gclk0 = NULL;
+	}
+err_gclk0:
+	if (ssc != NULL) {
+		ssc_free(ssc);
+		ssc = NULL;
+	}
+err_ssc:
+	return ret;
+}
+
+
+static void __exit playpaq_asoc_exit(void)
+{
+	struct at32_ssc_info *ssc_p = playpaq_wm8510_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;
+	}
+
+	if (_gclk0 != NULL) {
+		clk_put(_gclk0);
+		_gclk0 = NULL;
+	}
+	if (_pll0 != NULL) {
+		clk_put(_pll0);
+		_pll0 = NULL;
+	}
+
+#if defined CONFIG_AT32_ENHANCED_PORTMUX
+	at32_free_pin(MCLK_PIN);
+#endif
+
+	platform_device_unregister(playpaq_snd_device);
+	playpaq_snd_device = NULL;
+}
+
+module_init(playpaq_asoc_init);
+module_exit(playpaq_asoc_exit);
+
+MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
+MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig
index 5cb93fd3a407..905186502e00 100644
--- a/sound/soc/at91/Kconfig
+++ b/sound/soc/at91/Kconfig
@@ -1,6 +1,6 @@
 config SND_AT91_SOC
 	tristate "SoC Audio for the Atmel AT91 System-on-Chip"
-	depends on ARCH_AT91 && SND_SOC
+	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
diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c
index ccac6bd2889c..d47492b2b6e5 100644
--- a/sound/soc/at91/at91-pcm.c
+++ b/sound/soc/at91/at91-pcm.c
@@ -318,7 +318,7 @@ static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
 static u64 at91_pcm_dmamask = 0xffffffff;
 
 static int at91_pcm_new(struct snd_card *card,
-	struct snd_soc_codec_dai *dai, struct snd_pcm *pcm)
+	struct snd_soc_dai *dai, struct snd_pcm *pcm)
 {
 	int ret = 0;
 
@@ -367,7 +367,7 @@ static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 #ifdef CONFIG_PM
 static int at91_pcm_suspend(struct platform_device *pdev,
-	struct snd_soc_cpu_dai *dai)
+	struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = dai->runtime;
 	struct at91_runtime_data *prtd;
@@ -392,7 +392,7 @@ static int at91_pcm_suspend(struct platform_device *pdev,
 }
 
 static int at91_pcm_resume(struct platform_device *pdev,
-	struct snd_soc_cpu_dai *dai)
+	struct snd_soc_dai *dai)
 {
 	struct snd_pcm_runtime *runtime = dai->runtime;
 	struct at91_runtime_data *prtd;
diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c
index bc35d00a38f8..090e607f8692 100644
--- a/sound/soc/at91/at91-ssc.c
+++ b/sound/soc/at91/at91-ssc.c
@@ -41,7 +41,7 @@
 #define	DBG(x...)
 #endif
 
-#if defined(CONFIG_ARCH_AT91SAM9260)
+#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20)
 #define NUM_SSC_DEVICES		1
 #else
 #define NUM_SSC_DEVICES		3
@@ -281,7 +281,7 @@ static void at91_ssc_shutdown(struct snd_pcm_substream *substream)
 /*
  * Record the SSC system clock rate.
  */
-static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	/*
@@ -303,7 +303,7 @@ static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
 /*
  * Record the DAI format for use in hw_params().
  */
-static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+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];
@@ -315,7 +315,7 @@ static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
 /*
  * Record SSC clock dividers for use in hw_params().
  */
-static int at91_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+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];
@@ -634,7 +634,7 @@ static int at91_ssc_prepare(struct snd_pcm_substream *substream)
 
 #ifdef CONFIG_PM
 static int at91_ssc_suspend(struct platform_device *pdev,
-	struct snd_soc_cpu_dai *cpu_dai)
+	struct snd_soc_dai *cpu_dai)
 {
 	struct at91_ssc_info *ssc_p;
 
@@ -662,7 +662,7 @@ static int at91_ssc_suspend(struct platform_device *pdev,
 }
 
 static int at91_ssc_resume(struct platform_device *pdev,
-	struct snd_soc_cpu_dai *cpu_dai)
+	struct snd_soc_dai *cpu_dai)
 {
 	struct at91_ssc_info *ssc_p;
 
@@ -700,7 +700,7 @@ static int at91_ssc_resume(struct platform_device *pdev,
 #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_cpu_dai at91_ssc_dai[NUM_SSC_DEVICES] = {
+struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = {
 	{	.name = "at91-ssc0",
 		.id = 0,
 		.type = SND_SOC_DAI_PCM,
diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h
index b188f973df9f..6b7bf382d06f 100644
--- a/sound/soc/at91/at91-ssc.h
+++ b/sound/soc/at91/at91-ssc.h
@@ -21,7 +21,7 @@
 #define AT91SSC_TCMR_PERIOD	1 /* BCLK divider for transmit FS */
 #define AT91SSC_RCMR_PERIOD	2 /* BCLK divider for receive FS */
 
-extern struct snd_soc_cpu_dai at91_ssc_dai[];
+extern struct snd_soc_dai at91_ssc_dai[];
 
 #endif /* _AT91_SSC_H */
 
diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c
index 1347dcf3f80b..d532de954241 100644
--- a/sound/soc/at91/eti_b1_wm8731.c
+++ b/sound/soc/at91/eti_b1_wm8731.c
@@ -53,18 +53,18 @@ static struct clk *pllb_clk;
 static int eti_b1_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	int ret;
 
 	/* cpu clock is the AT91 master clock sent to the SSC */
-	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
+	ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
 		60000000, SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
 
 	/* codec system clock is supplied by PCK1, set to 12MHz */
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
 		12000000, SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -87,8 +87,8 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	int ret;
 
 #ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
@@ -96,13 +96,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream,
 	int cmr_div, period;
 
 	/* 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_CBS_CFS);
 	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_CBS_CFS);
 	if (ret < 0)
 		return ret;
@@ -141,17 +141,17 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* set the MCK divider for BCLK */
-	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
 	if (ret < 0)
 		return ret;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		/* set the BCLK divider for DACLRC */
-		ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
 						AT91SSC_TCMR_PERIOD, period);
 	} else {
 		/* set the BCLK divider for ADCLRC */
-		ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
 						AT91SSC_RCMR_PERIOD, period);
 	}
 	if (ret < 0)
@@ -163,13 +163,13 @@ static int eti_b1_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;
@@ -191,7 +191,7 @@ static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = {
 	SND_SOC_DAPM_SPK("Ext Spk", NULL),
 };
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 
 	/* speaker connected to LHPOUT */
 	{"Ext Spk", NULL, "LHPOUT"},
@@ -199,9 +199,6 @@ static const char *intercon[][3] = {
 	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
 	{"MICIN", NULL, "Mic Bias"},
 	{"Mic Bias", NULL, "Int Mic"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 /*
@@ -209,30 +206,24 @@ static const char *intercon[][3] = {
  */
 static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
 {
-	int i;
-
 	DBG("eti_b1_wm8731_init() called\n");
 
 	/* Add specific widgets */
-	for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) {
-		snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]);
-	}
+	snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets,
+				  ARRAY_SIZE(eti_b1_dapm_widgets));
 
 	/* Set up specific audio path interconnects */
-	for(i = 0; intercon[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, intercon[i][0],
-			intercon[i][1], intercon[i][2]);
-	}
+	snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon));
 
 	/* not connected */
-	snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
-	snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
 
 	/* always connected */
-	snd_soc_dapm_set_endpoint(codec, "Int Mic", 1);
-	snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+	snd_soc_dapm_enable_pin(codec, "Int Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 
 	return 0;
 }
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
new file mode 100644
index 000000000000..410a893aa66b
--- /dev/null
+++ b/sound/soc/au1x/Kconfig
@@ -0,0 +1,32 @@
+##
+## Au1200/Au1550 PSC + DBDMA
+##
+config SND_SOC_AU1XPSC
+	tristate "SoC Audio for Au1200/Au1250/Au1550"
+	depends on SOC_AU1200 || SOC_AU1550
+	help
+	  This option enables support for the Programmable Serial
+	  Controllers in AC97 and I2S mode, and the Descriptor-Based DMA
+	  Controller (DBDMA) as found on the Au1200/Au1250/Au1550 SoC.
+
+config SND_SOC_AU1XPSC_I2S
+	tristate
+
+config SND_SOC_AU1XPSC_AC97
+	tristate
+	select AC97_BUS
+	select SND_AC97_CODEC
+	select SND_SOC_AC97_BUS
+
+
+##
+## Boards
+##
+config SND_SOC_SAMPLE_PSC_AC97
+	tristate "Sample Au12x0/Au1550 PSC AC97 sound machine"
+	depends on SND_SOC_AU1XPSC
+	select SND_SOC_AU1XPSC_AC97
+	select SND_SOC_AC97_CODEC
+	help
+	  This is a sample AC97 sound machine for use in Au12x0/Au1550
+	  based systems which have audio on PSC1 (e.g. Db1200 demoboard).
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
new file mode 100644
index 000000000000..6c6950b8003a
--- /dev/null
+++ b/sound/soc/au1x/Makefile
@@ -0,0 +1,13 @@
+# Au1200/Au1550 PSC audio
+snd-soc-au1xpsc-dbdma-objs := dbdma2.o
+snd-soc-au1xpsc-i2s-objs := psc-i2s.o
+snd-soc-au1xpsc-ac97-objs := psc-ac97.o
+
+obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o
+obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
+obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
+
+# Boards
+snd-soc-sample-ac97-objs := sample-ac97.o
+
+obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
new file mode 100644
index 000000000000..1466d9328800
--- /dev/null
+++ b/sound/soc/au1x/dbdma2.c
@@ -0,0 +1,421 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * 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.
+ *
+ * DMA glue for Au1x-PSC audio.
+ *
+ * NOTE: all of these drivers can only work with a SINGLE instance
+ *	 of a PSC. Multiple independent audio devices are impossible
+ *	 with ASoC v1.
+ */
+
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+/*#define PCM_DEBUG*/
+
+#define MSG(x...)	printk(KERN_INFO "au1xpsc_pcm: " x)
+#ifdef PCM_DEBUG
+#define DBG		MSG
+#else
+#define DBG(x...)	do {} while (0)
+#endif
+
+struct au1xpsc_audio_dmadata {
+	/* DDMA control data */
+	unsigned int ddma_id;		/* DDMA direction ID for this PSC */
+	u32 ddma_chan;			/* DDMA context */
+
+	/* PCM context (for irq handlers) */
+	struct snd_pcm_substream *substream;
+	unsigned long curr_period;	/* current segment DDMA is working on */
+	unsigned long q_period;		/* queue period(s) */
+	unsigned long dma_area;		/* address of queued DMA area */
+	unsigned long dma_area_s;	/* start address of DMA area */
+	unsigned long pos;		/* current byte position being played */
+	unsigned long periods;		/* number of SG segments in total */
+	unsigned long period_bytes;	/* size in bytes of one SG segment */
+
+	/* runtime data */
+	int msbits;
+};
+
+/* instance data. There can be only one, MacLeod!!!! */
+static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
+
+/*
+ * These settings are somewhat okay, at least on my machine audio plays
+ * almost skip-free. Especially the 64kB buffer seems to help a LOT.
+ */
+#define AU1XPSC_PERIOD_MIN_BYTES	1024
+#define AU1XPSC_BUFFER_MIN_BYTES	65536
+
+#define AU1XPSC_PCM_FMTS					\
+	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_U8 |	\
+	 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |	\
+	 SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |	\
+	 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |	\
+	 SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE |	\
+	 0)
+
+/* PCM hardware DMA capabilities - platform specific */
+static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
+	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+			    SNDRV_PCM_INFO_INTERLEAVED,
+	.formats	  = AU1XPSC_PCM_FMTS,
+	.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
+	.period_bytes_max = 4096 * 1024 - 1,
+	.periods_min	  = 2,
+	.periods_max	  = 4096,	/* 2 to as-much-as-you-like */
+	.buffer_bytes_max = 4096 * 1024 - 1,
+	.fifo_size	  = 16,		/* fifo entries of AC97/I2S PSC */
+};
+
+static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd)
+{
+	au1xxx_dbdma_put_source_flags(cd->ddma_chan,
+				(void *)phys_to_virt(cd->dma_area),
+				cd->period_bytes, DDMA_FLAGS_IE);
+
+	/* update next-to-queue period */
+	++cd->q_period;
+	cd->dma_area += cd->period_bytes;
+	if (cd->q_period >= cd->periods) {
+		cd->q_period = 0;
+		cd->dma_area = cd->dma_area_s;
+	}
+}
+
+static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd)
+{
+	au1xxx_dbdma_put_dest_flags(cd->ddma_chan,
+				(void *)phys_to_virt(cd->dma_area),
+				cd->period_bytes, DDMA_FLAGS_IE);
+
+	/* update next-to-queue period */
+	++cd->q_period;
+	cd->dma_area += cd->period_bytes;
+	if (cd->q_period >= cd->periods) {
+		cd->q_period = 0;
+		cd->dma_area = cd->dma_area_s;
+	}
+}
+
+static void au1x_pcm_dmatx_cb(int irq, void *dev_id)
+{
+	struct au1xpsc_audio_dmadata *cd = dev_id;
+
+	cd->pos += cd->period_bytes;
+	if (++cd->curr_period >= cd->periods) {
+		cd->pos = 0;
+		cd->curr_period = 0;
+	}
+	snd_pcm_period_elapsed(cd->substream);
+	au1x_pcm_queue_tx(cd);
+}
+
+static void au1x_pcm_dmarx_cb(int irq, void *dev_id)
+{
+	struct au1xpsc_audio_dmadata *cd = dev_id;
+
+	cd->pos += cd->period_bytes;
+	if (++cd->curr_period >= cd->periods) {
+		cd->pos = 0;
+		cd->curr_period = 0;
+	}
+	snd_pcm_period_elapsed(cd->substream);
+	au1x_pcm_queue_rx(cd);
+}
+
+static void au1x_pcm_dbdma_free(struct au1xpsc_audio_dmadata *pcd)
+{
+	if (pcd->ddma_chan) {
+		au1xxx_dbdma_stop(pcd->ddma_chan);
+		au1xxx_dbdma_reset(pcd->ddma_chan);
+		au1xxx_dbdma_chan_free(pcd->ddma_chan);
+		pcd->ddma_chan = 0;
+		pcd->msbits = 0;
+	}
+}
+
+/* in case of missing DMA ring or changed TX-source / RX-dest bit widths,
+ * allocate (or reallocate) a 2-descriptor DMA ring with bit depth according
+ * to ALSA-supplied sample depth.  This is due to limitations in the dbdma api
+ * (cannot adjust source/dest widths of already allocated descriptor ring).
+ */
+static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
+				 int stype, int msbits)
+{
+	/* DMA only in 8/16/32 bit widths */
+	if (msbits == 24)
+		msbits = 32;
+
+	/* check current config: correct bits and descriptors allocated? */
+	if ((pcd->ddma_chan) && (msbits == pcd->msbits))
+		goto out;	/* all ok! */
+
+	au1x_pcm_dbdma_free(pcd);
+
+	if (stype == PCM_RX)
+		pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
+					DSCR_CMD0_ALWAYS,
+					au1x_pcm_dmarx_cb, (void *)pcd);
+	else
+		pcd->ddma_chan = au1xxx_dbdma_chan_alloc(DSCR_CMD0_ALWAYS,
+					pcd->ddma_id,
+					au1x_pcm_dmatx_cb, (void *)pcd);
+
+	if (!pcd->ddma_chan)
+		return -ENOMEM;;
+
+	au1xxx_dbdma_set_devwidth(pcd->ddma_chan, msbits);
+	au1xxx_dbdma_ring_alloc(pcd->ddma_chan, 2);
+
+	pcd->msbits = msbits;
+
+	au1xxx_dbdma_stop(pcd->ddma_chan);
+	au1xxx_dbdma_reset(pcd->ddma_chan);
+
+out:
+	return 0;
+}
+
+static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct au1xpsc_audio_dmadata *pcd;
+	int stype, ret;
+
+	ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	if (ret < 0)
+		goto out;
+
+	stype = SUBSTREAM_TYPE(substream);
+	pcd = au1xpsc_audio_pcmdma[stype];
+
+	DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
+	    "runtime->min_align %d\n",
+		(unsigned long)runtime->dma_area,
+		(unsigned long)runtime->dma_addr, runtime->dma_bytes,
+		runtime->min_align);
+
+	DBG("bits %d  frags %d  frag_bytes %d  is_rx %d\n", params->msbits,
+		params_periods(params), params_period_bytes(params), stype);
+
+	ret = au1x_pcm_dbdma_realloc(pcd, stype, params->msbits);
+	if (ret) {
+		MSG("DDMA channel (re)alloc failed!\n");
+		goto out;
+	}
+
+	pcd->substream = substream;
+	pcd->period_bytes = params_period_bytes(params);
+	pcd->periods = params_periods(params);
+	pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr;
+	pcd->q_period = 0;
+	pcd->curr_period = 0;
+	pcd->pos = 0;
+
+	ret = 0;
+out:
+	return ret;
+}
+
+static int au1xpsc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_lib_free_pages(substream);
+	return 0;
+}
+
+static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct au1xpsc_audio_dmadata *pcd =
+			au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)];
+
+	au1xxx_dbdma_reset(pcd->ddma_chan);
+
+	if (SUBSTREAM_TYPE(substream) == PCM_RX) {
+		au1x_pcm_queue_rx(pcd);
+		au1x_pcm_queue_rx(pcd);
+	} else {
+		au1x_pcm_queue_tx(pcd);
+		au1x_pcm_queue_tx(pcd);
+	}
+
+	return 0;
+}
+
+static int au1xpsc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	u32 c = au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->ddma_chan;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		au1xxx_dbdma_start(c);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		au1xxx_dbdma_stop(c);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static snd_pcm_uframes_t
+au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	return bytes_to_frames(substream->runtime,
+		au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]->pos);
+}
+
+static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
+{
+	snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
+	return 0;
+}
+
+static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
+{
+	au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[SUBSTREAM_TYPE(substream)]);
+	return 0;
+}
+
+struct snd_pcm_ops au1xpsc_pcm_ops = {
+	.open		= au1xpsc_pcm_open,
+	.close		= au1xpsc_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= au1xpsc_pcm_hw_params,
+	.hw_free	= au1xpsc_pcm_hw_free,
+	.prepare	= au1xpsc_pcm_prepare,
+	.trigger	= au1xpsc_pcm_trigger,
+	.pointer	= au1xpsc_pcm_pointer,
+};
+
+static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int au1xpsc_pcm_new(struct snd_card *card,
+			   struct snd_soc_dai *dai,
+			   struct snd_pcm *pcm)
+{
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+		card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1);
+
+	return 0;
+}
+
+static int au1xpsc_pcm_probe(struct platform_device *pdev)
+{
+	struct resource *r;
+	int ret;
+
+	if (au1xpsc_audio_pcmdma[PCM_TX] || au1xpsc_audio_pcmdma[PCM_RX])
+		return -EBUSY;
+
+	/* TX DMA */
+	au1xpsc_audio_pcmdma[PCM_TX]
+		= kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
+	if (!au1xpsc_audio_pcmdma[PCM_TX])
+		return -ENOMEM;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!r) {
+		ret = -ENODEV;
+		goto out1;
+	}
+	(au1xpsc_audio_pcmdma[PCM_TX])->ddma_id = r->start;
+
+	/* RX DMA */
+	au1xpsc_audio_pcmdma[PCM_RX]
+		= kzalloc(sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
+	if (!au1xpsc_audio_pcmdma[PCM_RX])
+		return -ENOMEM;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!r) {
+		ret = -ENODEV;
+		goto out2;
+	}
+	(au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start;
+
+	return 0;
+
+out2:
+	kfree(au1xpsc_audio_pcmdma[PCM_RX]);
+	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
+out1:
+	kfree(au1xpsc_audio_pcmdma[PCM_TX]);
+	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
+	return ret;
+}
+
+static int au1xpsc_pcm_remove(struct platform_device *pdev)
+{
+	int i;
+
+	for (i = 0; i < 2; i++) {
+		if (au1xpsc_audio_pcmdma[i]) {
+			au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]);
+			kfree(au1xpsc_audio_pcmdma[i]);
+			au1xpsc_audio_pcmdma[i] = NULL;
+		}
+	}
+
+	return 0;
+}
+
+/* au1xpsc audio platform */
+struct snd_soc_platform au1xpsc_soc_platform = {
+	.name		= "au1xpsc-pcm-dbdma",
+	.probe		= au1xpsc_pcm_probe,
+	.remove		= au1xpsc_pcm_remove,
+	.pcm_ops 	= &au1xpsc_pcm_ops,
+	.pcm_new	= au1xpsc_pcm_new,
+	.pcm_free	= au1xpsc_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(au1xpsc_soc_platform);
+
+static int __init au1xpsc_audio_dbdma_init(void)
+{
+	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
+	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
+	return 0;
+}
+
+static void __exit au1xpsc_audio_dbdma_exit(void)
+{
+}
+
+module_init(au1xpsc_audio_dbdma_init);
+module_exit(au1xpsc_audio_dbdma_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
new file mode 100644
index 000000000000..57facbad6825
--- /dev/null
+++ b/sound/soc/au1x/psc-ac97.c
@@ -0,0 +1,387 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * 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.
+ *
+ * Au1xxx-PSC AC97 glue.
+ *
+ * NOTE: all of these drivers can only work with a SINGLE instance
+ *	 of a PSC. Multiple independent audio devices are impossible
+ *	 with ASoC v1.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+#define AC97_DIR	\
+	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AC97_RATES	\
+	SNDRV_PCM_RATE_8000_48000
+
+#define AC97_FMTS	\
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
+
+#define AC97PCR_START(stype)	\
+	((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
+#define AC97PCR_STOP(stype)	\
+	((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
+#define AC97PCR_CLRFIFO(stype)	\
+	((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+
+/* instance data. There can be only one, MacLeod!!!! */
+static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
+
+/* AC97 controller reads codec register */
+static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
+					unsigned short reg)
+{
+	/* FIXME */
+	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+	unsigned short data, tmo;
+
+	au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata));
+	au_sync();
+
+	tmo = 1000;
+	while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
+		udelay(2);
+
+	if (!tmo)
+		data = 0xffff;
+	else
+		data = au_readl(AC97_CDC(pscdata)) & 0xffff;
+
+	au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+	au_sync();
+
+	return data;
+}
+
+/* AC97 controller writes to codec register */
+static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+				unsigned short val)
+{
+	/* FIXME */
+	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+	unsigned int tmo;
+
+	au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata));
+	au_sync();
+	tmo = 1000;
+	while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo)
+		au_sync();
+
+	au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+	au_sync();
+}
+
+/* AC97 controller asserts a warm reset */
+static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+	/* FIXME */
+	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+
+	au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata));
+	au_sync();
+	msleep(10);
+	au_writel(0, AC97_RST(pscdata));
+	au_sync();
+}
+
+static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+	/* FIXME */
+	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+	int i;
+
+	/* disable PSC during cold reset */
+	au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+	au_sync();
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata));
+	au_sync();
+
+	/* issue cold reset */
+	au_writel(PSC_AC97RST_RST, AC97_RST(pscdata));
+	au_sync();
+	msleep(500);
+	au_writel(0, AC97_RST(pscdata));
+	au_sync();
+
+	/* enable PSC */
+	au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+	au_sync();
+
+	/* wait for PSC to indicate it's ready */
+	i = 100000;
+	while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
+		au_sync();
+
+	if (i == 0) {
+		printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n");
+		return;
+	}
+
+	/* enable the ac97 function */
+	au_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+	au_sync();
+
+	/* wait for AC97 core to become ready */
+	i = 100000;
+	while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
+		au_sync();
+	if (i == 0)
+		printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n");
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+	.read		= au1xpsc_ac97_read,
+	.write		= au1xpsc_ac97_write,
+	.reset		= au1xpsc_ac97_cold_reset,
+	.warm_reset	= au1xpsc_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params)
+{
+	/* FIXME */
+	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+	unsigned long r, stat;
+	int chans, stype = SUBSTREAM_TYPE(substream);
+
+	chans = params_channels(params);
+
+	r = au_readl(AC97_CFG(pscdata));
+	stat = au_readl(AC97_STAT(pscdata));
+
+	/* already active? */
+	if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) {
+		/* reject parameters not currently set up */
+		if ((PSC_AC97CFG_GET_LEN(r) != params->msbits) ||
+		    (pscdata->rate != params_rate(params)))
+			return -EINVAL;
+	} else {
+		/* disable AC97 device controller first */
+		au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+		au_sync();
+
+		/* set sample bitdepth: REG[24:21]=(BITS-2)/2 */
+		r &= ~PSC_AC97CFG_LEN_MASK;
+		r |= PSC_AC97CFG_SET_LEN(params->msbits);
+
+		/* channels: enable slots for front L/R channel */
+		if (stype == PCM_TX) {
+			r &= ~PSC_AC97CFG_TXSLOT_MASK;
+			r |= PSC_AC97CFG_TXSLOT_ENA(3);
+			r |= PSC_AC97CFG_TXSLOT_ENA(4);
+		} else {
+			r &= ~PSC_AC97CFG_RXSLOT_MASK;
+			r |= PSC_AC97CFG_RXSLOT_ENA(3);
+			r |= PSC_AC97CFG_RXSLOT_ENA(4);
+		}
+
+		/* finally enable the AC97 controller again */
+		au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+		au_sync();
+
+		pscdata->cfg = r;
+		pscdata->rate = params_rate(params);
+	}
+
+	return 0;
+}
+
+static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
+				int cmd)
+{
+	/* FIXME */
+	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
+	int ret, stype = SUBSTREAM_TYPE(substream);
+
+	ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		au_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
+		au_sync();
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
+		au_sync();
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	return ret;
+}
+
+static int au1xpsc_ac97_probe(struct platform_device *pdev,
+			      struct snd_soc_dai *dai)
+{
+	int ret;
+	struct resource *r;
+	unsigned long sel;
+
+	if (au1xpsc_ac97_workdata)
+		return -EBUSY;
+
+	au1xpsc_ac97_workdata =
+		kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
+	if (!au1xpsc_ac97_workdata)
+		return -ENOMEM;
+
+	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!r) {
+		ret = -ENODEV;
+		goto out0;
+	}
+
+	ret = -EBUSY;
+	au1xpsc_ac97_workdata->ioarea =
+		request_mem_region(r->start, r->end - r->start + 1,
+					"au1xpsc_ac97");
+	if (!au1xpsc_ac97_workdata->ioarea)
+		goto out0;
+
+	au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff);
+	if (!au1xpsc_ac97_workdata->mmio)
+		goto out1;
+
+	/* configuration: max dma trigger threshold, enable ac97 */
+	 au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
+				      PSC_AC97CFG_TT_FIFO8 |
+				      PSC_AC97CFG_DE_ENABLE;
+
+	/* preserve PSC clock source set up by platform (dev.platform_data
+	 * is already occupied by soc layer)
+	 */
+	sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK;
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+	au_sync();
+	au_writel(0, PSC_SEL(au1xpsc_ac97_workdata));
+	au_sync();
+	au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata));
+	au_sync();
+	/* next up: cold reset.  Dont check for PSC-ready now since
+	 * there may not be any codec clock yet.
+	 */
+
+	return 0;
+
+out1:
+	release_resource(au1xpsc_ac97_workdata->ioarea);
+	kfree(au1xpsc_ac97_workdata->ioarea);
+out0:
+	kfree(au1xpsc_ac97_workdata);
+	au1xpsc_ac97_workdata = NULL;
+	return ret;
+}
+
+static void au1xpsc_ac97_remove(struct platform_device *pdev,
+				struct snd_soc_dai *dai)
+{
+	/* disable PSC completely */
+	au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+	au_sync();
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+	au_sync();
+
+	iounmap(au1xpsc_ac97_workdata->mmio);
+	release_resource(au1xpsc_ac97_workdata->ioarea);
+	kfree(au1xpsc_ac97_workdata->ioarea);
+	kfree(au1xpsc_ac97_workdata);
+	au1xpsc_ac97_workdata = NULL;
+}
+
+static int au1xpsc_ac97_suspend(struct platform_device *pdev,
+				struct snd_soc_dai *dai)
+{
+	/* save interesting registers and disable PSC */
+	au1xpsc_ac97_workdata->pm[0] =
+			au_readl(PSC_SEL(au1xpsc_ac97_workdata));
+
+	au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+	au_sync();
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+	au_sync();
+
+	return 0;
+}
+
+static int au1xpsc_ac97_resume(struct platform_device *pdev,
+			       struct snd_soc_dai *dai)
+{
+	/* restore PSC clock config */
+	au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
+			PSC_SEL(au1xpsc_ac97_workdata));
+	au_sync();
+
+	/* after this point the ac97 core will cold-reset the codec.
+	 * During cold-reset the PSC is reinitialized and the last
+	 * configuration set up in hw_params() is restored.
+	 */
+	return 0;
+}
+
+struct snd_soc_dai au1xpsc_ac97_dai = {
+	.name			= "au1xpsc_ac97",
+	.type			= SND_SOC_DAI_AC97,
+	.probe			= au1xpsc_ac97_probe,
+	.remove			= au1xpsc_ac97_remove,
+	.suspend		= au1xpsc_ac97_suspend,
+	.resume			= au1xpsc_ac97_resume,
+	.playback = {
+		.rates		= AC97_RATES,
+		.formats	= AC97_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 2,
+	},
+	.capture = {
+		.rates		= AC97_RATES,
+		.formats	= AC97_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 2,
+	},
+	.ops = {
+		.trigger	= au1xpsc_ac97_trigger,
+		.hw_params	= au1xpsc_ac97_hw_params,
+	},
+};
+EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
+
+static int __init au1xpsc_ac97_init(void)
+{
+	au1xpsc_ac97_workdata = NULL;
+	return 0;
+}
+
+static void __exit au1xpsc_ac97_exit(void)
+{
+}
+
+module_init(au1xpsc_ac97_init);
+module_exit(au1xpsc_ac97_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
new file mode 100644
index 000000000000..ba4b5c199f21
--- /dev/null
+++ b/sound/soc/au1x/psc-i2s.c
@@ -0,0 +1,414 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * 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.
+ *
+ * Au1xxx-PSC I2S glue.
+ *
+ * NOTE: all of these drivers can only work with a SINGLE instance
+ *	 of a PSC. Multiple independent audio devices are impossible
+ *	 with ASoC v1.
+ * NOTE: so far only PSC slave mode (bit- and frameclock) is supported.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+
+#include "psc.h"
+
+/* supported I2S DAI hardware formats */
+#define AU1XPSC_I2S_DAIFMT \
+	(SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J |	\
+	 SND_SOC_DAIFMT_NB_NF)
+
+/* supported I2S direction */
+#define AU1XPSC_I2S_DIR \
+	(SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AU1XPSC_I2S_RATES \
+	SNDRV_PCM_RATE_8000_192000
+
+#define AU1XPSC_I2S_FMTS \
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+#define I2SSTAT_BUSY(stype)	\
+	((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
+#define I2SPCR_START(stype)	\
+	((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
+#define I2SPCR_STOP(stype)	\
+	((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
+#define I2SPCR_CLRFIFO(stype)	\
+	((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
+
+
+/* instance data. There can be only one, MacLeod!!!! */
+static struct au1xpsc_audio_data *au1xpsc_i2s_workdata;
+
+static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+			       unsigned int fmt)
+{
+	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
+	unsigned long ct;
+	int ret;
+
+	ret = -EINVAL;
+
+	ct = pscdata->cfg;
+
+	ct &= ~(PSC_I2SCFG_XM | PSC_I2SCFG_MLJ);	/* left-justified */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		ct |= PSC_I2SCFG_XM;	/* enable I2S mode */
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		ct |= PSC_I2SCFG_MLJ;	/* LSB (right-) justified */
+		break;
+	default:
+		goto out;
+	}
+
+	ct &= ~(PSC_I2SCFG_BI | PSC_I2SCFG_WI);		/* IB-IF */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		ct |= PSC_I2SCFG_BI | PSC_I2SCFG_WI;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		ct |= PSC_I2SCFG_BI;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		ct |= PSC_I2SCFG_WI;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		break;
+	default:
+		goto out;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:	/* CODEC master */
+		ct |= PSC_I2SCFG_MS;	/* PSC I2S slave mode */
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:	/* CODEC slave */
+		ct &= ~PSC_I2SCFG_MS;	/* PSC I2S Master mode */
+		break;
+	default:
+		goto out;
+	}
+
+	pscdata->cfg = ct;
+	ret = 0;
+out:
+	return ret;
+}
+
+static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
+
+	int cfgbits;
+	unsigned long stat;
+
+	/* check if the PSC is already streaming data */
+	stat = au_readl(I2S_STAT(pscdata));
+	if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) {
+		/* reject parameters not currently set up in hardware */
+		cfgbits = au_readl(I2S_CFG(pscdata));
+		if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) ||
+		    (params_rate(params) != pscdata->rate))
+			return -EINVAL;
+	} else {
+		/* set sample bitdepth */
+		pscdata->cfg &= ~(0x1f << 4);
+		pscdata->cfg |= PSC_I2SCFG_SET_LEN(params->msbits);
+		/* remember current rate for other stream */
+		pscdata->rate = params_rate(params);
+	}
+	return 0;
+}
+
+/* Configure PSC late:  on my devel systems the codec  is I2S master and
+ * supplies the i2sbitclock __AND__ i2sMclk (!) to the PSC unit.  ASoC
+ * uses aggressive PM and  switches the codec off  when it is not in use
+ * which also means the PSC unit doesn't get any clocks and is therefore
+ * dead. That's why this chunk here gets called from the trigger callback
+ * because I can be reasonably certain the codec is driving the clocks.
+ */
+static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata)
+{
+	unsigned long tmo;
+
+	/* bring PSC out of sleep, and configure I2S unit */
+	au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+	au_sync();
+
+	tmo = 1000000;
+	while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo)
+		tmo--;
+
+	if (!tmo)
+		goto psc_err;
+
+	au_writel(0, I2S_CFG(pscdata));
+	au_sync();
+	au_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata));
+	au_sync();
+
+	/* wait for I2S controller to become ready */
+	tmo = 1000000;
+	while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo)
+		tmo--;
+
+	if (tmo)
+		return 0;
+
+psc_err:
+	au_writel(0, I2S_CFG(pscdata));
+	au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+	au_sync();
+	return -ETIMEDOUT;
+}
+
+static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype)
+{
+	unsigned long tmo, stat;
+	int ret;
+
+	ret = 0;
+
+	/* if both TX and RX are idle, configure the PSC  */
+	stat = au_readl(I2S_STAT(pscdata));
+	if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
+		ret = au1xpsc_i2s_configure(pscdata);
+		if (ret)
+			goto out;
+	}
+
+	au_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata));
+	au_sync();
+	au_writel(I2SPCR_START(stype), I2S_PCR(pscdata));
+	au_sync();
+
+	/* wait for start confirmation */
+	tmo = 1000000;
+	while (!(au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
+		tmo--;
+
+	if (!tmo) {
+		au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
+		au_sync();
+		ret = -ETIMEDOUT;
+	}
+out:
+	return ret;
+}
+
+static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
+{
+	unsigned long tmo, stat;
+
+	au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
+	au_sync();
+
+	/* wait for stop confirmation */
+	tmo = 1000000;
+	while ((au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
+		tmo--;
+
+	/* if both TX and RX are idle, disable PSC */
+	stat = au_readl(I2S_STAT(pscdata));
+	if (!(stat & (PSC_I2SSTAT_RB | PSC_I2SSTAT_RB))) {
+		au_writel(0, I2S_CFG(pscdata));
+		au_sync();
+		au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+		au_sync();
+	}
+	return 0;
+}
+
+static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
+	int ret, stype = SUBSTREAM_TYPE(substream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		ret = au1xpsc_i2s_start(pscdata, stype);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		ret = au1xpsc_i2s_stop(pscdata, stype);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	return ret;
+}
+
+static int au1xpsc_i2s_probe(struct platform_device *pdev,
+			     struct snd_soc_dai *dai)
+{
+	struct resource *r;
+	unsigned long sel;
+	int ret;
+
+	if (au1xpsc_i2s_workdata)
+		return -EBUSY;
+
+	au1xpsc_i2s_workdata =
+		kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
+	if (!au1xpsc_i2s_workdata)
+		return -ENOMEM;
+
+	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!r) {
+		ret = -ENODEV;
+		goto out0;
+	}
+
+	ret = -EBUSY;
+	au1xpsc_i2s_workdata->ioarea =
+		request_mem_region(r->start, r->end - r->start + 1,
+					"au1xpsc_i2s");
+	if (!au1xpsc_i2s_workdata->ioarea)
+		goto out0;
+
+	au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff);
+	if (!au1xpsc_i2s_workdata->mmio)
+		goto out1;
+
+	/* preserve PSC clock source set up by platform (dev.platform_data
+	 * is already occupied by soc layer)
+	 */
+	sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK;
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+	au_sync();
+	au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata));
+	au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+	au_sync();
+
+	/* preconfigure: set max rx/tx fifo depths */
+	au1xpsc_i2s_workdata->cfg |=
+			PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8;
+
+	/* don't wait for I2S core to become ready now; clocks may not
+	 * be running yet; depending on clock input for PSC a wait might
+	 * time out.
+	 */
+
+	return 0;
+
+out1:
+	release_resource(au1xpsc_i2s_workdata->ioarea);
+	kfree(au1xpsc_i2s_workdata->ioarea);
+out0:
+	kfree(au1xpsc_i2s_workdata);
+	au1xpsc_i2s_workdata = NULL;
+	return ret;
+}
+
+static void au1xpsc_i2s_remove(struct platform_device *pdev,
+			       struct snd_soc_dai *dai)
+{
+	au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+	au_sync();
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+	au_sync();
+
+	iounmap(au1xpsc_i2s_workdata->mmio);
+	release_resource(au1xpsc_i2s_workdata->ioarea);
+	kfree(au1xpsc_i2s_workdata->ioarea);
+	kfree(au1xpsc_i2s_workdata);
+	au1xpsc_i2s_workdata = NULL;
+}
+
+static int au1xpsc_i2s_suspend(struct platform_device *pdev,
+			       struct snd_soc_dai *cpu_dai)
+{
+	/* save interesting register and disable PSC */
+	au1xpsc_i2s_workdata->pm[0] =
+		au_readl(PSC_SEL(au1xpsc_i2s_workdata));
+
+	au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+	au_sync();
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+	au_sync();
+
+	return 0;
+}
+
+static int au1xpsc_i2s_resume(struct platform_device *pdev,
+			      struct snd_soc_dai *cpu_dai)
+{
+	/* select I2S mode and PSC clock */
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+	au_sync();
+	au_writel(0, PSC_SEL(au1xpsc_i2s_workdata));
+	au_sync();
+	au_writel(au1xpsc_i2s_workdata->pm[0],
+			PSC_SEL(au1xpsc_i2s_workdata));
+	au_sync();
+
+	return 0;
+}
+
+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,
+	.resume			= au1xpsc_i2s_resume,
+	.playback = {
+		.rates		= AU1XPSC_I2S_RATES,
+		.formats	= AU1XPSC_I2S_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 8,	/* 2 without external help */
+	},
+	.capture = {
+		.rates		= AU1XPSC_I2S_RATES,
+		.formats	= AU1XPSC_I2S_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 8,	/* 2 without external help */
+	},
+	.ops = {
+		.trigger	= au1xpsc_i2s_trigger,
+		.hw_params	= au1xpsc_i2s_hw_params,
+	},
+	.dai_ops = {
+		.set_fmt	= au1xpsc_i2s_set_fmt,
+	},
+};
+EXPORT_SYMBOL(au1xpsc_i2s_dai);
+
+static int __init au1xpsc_i2s_init(void)
+{
+	au1xpsc_i2s_workdata = NULL;
+	return 0;
+}
+
+static void __exit au1xpsc_i2s_exit(void)
+{
+}
+
+module_init(au1xpsc_i2s_init);
+module_exit(au1xpsc_i2s_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
new file mode 100644
index 000000000000..8fdb1a04a07b
--- /dev/null
+++ b/sound/soc/au1x/psc.h
@@ -0,0 +1,53 @@
+/*
+ * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ *
+ * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ * 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: all of these drivers can only work with a SINGLE instance
+ *	 of a PSC. Multiple independent audio devices are impossible
+ *	 with ASoC v1.
+ */
+
+#ifndef _AU1X_PCM_H
+#define _AU1X_PCM_H
+
+extern struct snd_soc_dai au1xpsc_ac97_dai;
+extern struct snd_soc_dai au1xpsc_i2s_dai;
+extern struct snd_soc_platform au1xpsc_soc_platform;
+extern struct snd_ac97_bus_ops soc_ac97_ops;
+
+struct au1xpsc_audio_data {
+	void __iomem *mmio;
+
+	unsigned long cfg;
+	unsigned long rate;
+
+	unsigned long pm[2];
+	struct resource *ioarea;
+};
+
+#define PCM_TX	0
+#define PCM_RX	1
+
+#define SUBSTREAM_TYPE(substream) \
+	((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
+
+/* easy access macros */
+#define PSC_CTRL(x)	((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET)
+#define PSC_SEL(x)	((unsigned long)((x)->mmio) + PSC_SEL_OFFSET)
+#define I2S_STAT(x)	((unsigned long)((x)->mmio) + PSC_I2SSTAT_OFFSET)
+#define I2S_CFG(x)	((unsigned long)((x)->mmio) + PSC_I2SCFG_OFFSET)
+#define I2S_PCR(x)	((unsigned long)((x)->mmio) + PSC_I2SPCR_OFFSET)
+#define AC97_CFG(x)	((unsigned long)((x)->mmio) + PSC_AC97CFG_OFFSET)
+#define AC97_CDC(x)	((unsigned long)((x)->mmio) + PSC_AC97CDC_OFFSET)
+#define AC97_EVNT(x)	((unsigned long)((x)->mmio) + PSC_AC97EVNT_OFFSET)
+#define AC97_PCR(x)	((unsigned long)((x)->mmio) + PSC_AC97PCR_OFFSET)
+#define AC97_RST(x)	((unsigned long)((x)->mmio) + PSC_AC97RST_OFFSET)
+#define AC97_STAT(x)	((unsigned long)((x)->mmio) + PSC_AC97STAT_OFFSET)
+
+#endif
diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c
new file mode 100644
index 000000000000..f75ae7f62c3d
--- /dev/null
+++ b/sound/soc/au1x/sample-ac97.c
@@ -0,0 +1,144 @@
+/*
+ * Sample Au12x0/Au1550 PSC AC97 sound machine.
+ *
+ * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms outlined in the file COPYING at the root of this
+ *  source archive.
+ *
+ * This is a very generic AC97 sound machine driver for boards which
+ * have (AC97) audio at PSC1 (e.g. DB1200 demoboards).
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.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-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1xxx_psc.h>
+#include <asm/mach-au1x00/au1xxx_dbdma.h>
+
+#include "../codecs/ac97.h"
+#include "psc.h"
+
+static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_sync(codec);
+	return 0;
+}
+
+static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
+	.name		= "AC97",
+	.stream_name	= "AC97 HiFi",
+	.cpu_dai	= &au1xpsc_ac97_dai,	/* see psc-ac97.c */
+	.codec_dai	= &ac97_dai,		/* see codecs/ac97.c */
+	.init		= au1xpsc_sample_ac97_init,
+	.ops		= NULL,
+};
+
+static struct snd_soc_machine 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,
+	.platform	= &au1xpsc_soc_platform, /* see dbdma2.c */
+	.codec_dev	= &soc_codec_dev_ac97,
+};
+
+static struct resource au1xpsc_psc1_res[] = {
+	[0] = {
+		.start	= CPHYSADDR(PSC1_BASE_ADDR),
+		.end	= CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff,
+		.flags	= IORESOURCE_MEM,
+	},
+	[1] = {
+#ifdef CONFIG_SOC_AU1200
+		.start	= AU1200_PSC1_INT,
+		.end	= AU1200_PSC1_INT,
+#elif defined(CONFIG_SOC_AU1550)
+		.start	= AU1550_PSC1_INT,
+		.end	= AU1550_PSC1_INT,
+#endif
+		.flags	= IORESOURCE_IRQ,
+	},
+	[2] = {
+		.start	= DSCR_CMD0_PSC1_TX,
+		.end	= DSCR_CMD0_PSC1_TX,
+		.flags	= IORESOURCE_DMA,
+	},
+	[3] = {
+		.start	= DSCR_CMD0_PSC1_RX,
+		.end	= DSCR_CMD0_PSC1_RX,
+		.flags	= IORESOURCE_DMA,
+	},
+};
+
+static struct platform_device *au1xpsc_sample_ac97_dev;
+
+static int __init au1xpsc_sample_ac97_load(void)
+{
+	int ret;
+
+#ifdef CONFIG_SOC_AU1200
+	unsigned long io;
+
+	/* modify sys_pinfunc for AC97 on PSC1 */
+	io = au_readl(SYS_PINFUNC);
+	io |= SYS_PINFUNC_P1C;
+	io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B);
+	au_writel(io, SYS_PINFUNC);
+	au_sync();
+#endif
+
+	ret = -ENOMEM;
+
+	/* setup PSC clock source for AC97 part: external clock provided
+	 * by codec.  The psc-ac97.c driver depends on this setting!
+	 */
+	au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET);
+	au_sync();
+
+	au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1);
+	if (!au1xpsc_sample_ac97_dev)
+		goto out;
+
+	au1xpsc_sample_ac97_dev->resource =
+		kmemdup(au1xpsc_psc1_res, sizeof(struct resource) *
+			ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL);
+	au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res);
+	au1xpsc_sample_ac97_dev->id = 1;
+
+	platform_set_drvdata(au1xpsc_sample_ac97_dev,
+			     &au1xpsc_sample_ac97_devdata);
+	au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev;
+	ret = platform_device_add(au1xpsc_sample_ac97_dev);
+
+	if (ret) {
+		platform_device_put(au1xpsc_sample_ac97_dev);
+		au1xpsc_sample_ac97_dev = NULL;
+	}
+
+out:
+	return ret;
+}
+
+static void __exit au1xpsc_sample_ac97_exit(void)
+{
+	platform_device_unregister(au1xpsc_sample_ac97_dev);
+}
+
+module_init(au1xpsc_sample_ac97_load);
+module_exit(au1xpsc_sample_ac97_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine");
+MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 3903ab7dfa4a..1db04a28a53d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -1,31 +1,37 @@
 config SND_SOC_AC97_CODEC
 	tristate
-	depends on SND_SOC
+	select SND_AC97_CODEC
+
+config SND_SOC_AK4535
+	tristate
+
+config SND_SOC_UDA1380
+        tristate
+
+config SND_SOC_WM8510
+	tristate
 
 config SND_SOC_WM8731
 	tristate
-	depends on SND_SOC
 
 config SND_SOC_WM8750
 	tristate
-	depends on SND_SOC
 
 config SND_SOC_WM8753
 	tristate
-	depends on SND_SOC
+
+config SND_SOC_WM8990
+	tristate
 
 config SND_SOC_WM9712
 	tristate
-	depends on SND_SOC
 
 config SND_SOC_WM9713
 	tristate
-	depends on SND_SOC
 
 # Cirrus Logic CS4270 Codec
 config SND_SOC_CS4270
 	tristate
-	depends on SND_SOC
 
 # Cirrus Logic CS4270 Codec Hardware Mute Support
 # Select if you have external muting circuitry attached to your CS4270.
@@ -43,4 +49,4 @@ config SND_SOC_CS4270_VD33_ERRATA
 
 config SND_SOC_TLV320AIC3X
 	tristate
-	depends on SND_SOC && I2C
+	depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4e1314c9d3ec..d7b97abcf729 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,16 +1,24 @@
 snd-soc-ac97-objs := ac97.o
+snd-soc-ak4535-objs := ak4535.o
+snd-soc-uda1380-objs := uda1380.o
+snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8731-objs := wm8731.o
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
+snd-soc-wm8990-objs := wm8990.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
+obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
+obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 2a1ffe396908..61fd96ca7bc7 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -10,9 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    17th Oct 2005   Initial version.
- *
  * Generic AC97 support.
  */
 
@@ -24,6 +21,7 @@
 #include <sound/ac97_codec.h>
 #include <sound/initval.h>
 #include <sound/soc.h>
+#include "ac97.h"
 
 #define AC97_VERSION "0.6"
 
@@ -43,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream)
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
-struct snd_soc_codec_dai ac97_dai = {
+struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
 	.type = SND_SOC_DAI_AC97,
 	.playback = {
@@ -146,9 +144,34 @@ static int ac97_soc_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_ac97_suspend(socdev->codec->ac97);
+
+	return 0;
+}
+
+static int ac97_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_ac97_resume(socdev->codec->ac97);
+
+	return 0;
+}
+#else
+#define ac97_soc_suspend NULL
+#define ac97_soc_resume NULL
+#endif
+
 struct snd_soc_codec_device soc_codec_dev_ac97 = {
 	.probe = 	ac97_soc_probe,
 	.remove = 	ac97_soc_remove,
+	.suspend =	ac97_soc_suspend,
+	.resume =	ac97_soc_resume,
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ac97);
 
diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h
index 2bf6d69fd069..281aa42e2bbb 100644
--- a/sound/soc/codecs/ac97.h
+++ b/sound/soc/codecs/ac97.h
@@ -14,6 +14,6 @@
 #define __LINUX_SND_SOC_AC97_H
 
 extern struct snd_soc_codec_device soc_codec_dev_ac97;
-extern struct snd_soc_codec_dai ac97_dai;
+extern struct snd_soc_dai ac97_dai;
 
 #endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
new file mode 100644
index 000000000000..b26003c4f3e8
--- /dev/null
+++ b/sound/soc/codecs/ak4535.c
@@ -0,0 +1,696 @@
+/*
+ * ak4535.c  --  AK4535 ALSA Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/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 "ak4535.h"
+
+#define AUDIO_NAME "ak4535"
+#define AK4535_VERSION "0.3"
+
+struct snd_soc_codec_device soc_codec_dev_ak4535;
+
+/* codec private data */
+struct ak4535_priv {
+	unsigned int sysclk;
+};
+
+/*
+ * ak4535 register cache
+ */
+static const u16 ak4535_reg[AK4535_CACHEREGNUM] = {
+    0x0000, 0x0080, 0x0000, 0x0003,
+    0x0002, 0x0000, 0x0011, 0x0001,
+    0x0000, 0x0040, 0x0036, 0x0010,
+    0x0000, 0x0000, 0x0057, 0x0000,
+};
+
+/*
+ * read ak4535 register cache
+ */
+static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= AK4535_CACHEREGNUM)
+		return -1;
+	return cache[reg];
+}
+
+static inline unsigned int ak4535_read(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u8 data;
+	data = reg;
+
+	if (codec->hw_write(codec->control_data, &data, 1) != 1)
+		return -EIO;
+
+	if (codec->hw_read(codec->control_data, &data, 1) != 1)
+		return -EIO;
+
+	return data;
+};
+
+/*
+ * write ak4535 register cache
+ */
+static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec,
+	u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= AK4535_CACHEREGNUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the AK4535 register space
+ */
+static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[2];
+
+	/* data is
+	 *   D15..D8 AK4535 register offset
+	 *   D7...D0 register data
+	 */
+	data[0] = reg & 0xff;
+	data[1] = value & 0xff;
+
+	ak4535_write_reg_cache(codec, reg, value);
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+static int ak4535_sync(struct snd_soc_codec *codec)
+{
+	u16 *cache = codec->reg_cache;
+	int i, r = 0;
+
+	for (i = 0; i < AK4535_CACHEREGNUM; i++)
+		r |= ak4535_write(codec, i, cache[i]);
+
+	return r;
+};
+
+static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"};
+static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"};
+static const char *ak4535_hp_out[] = {"Stereo", "Mono"};
+static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"};
+static const char *ak4535_mic_select[] = {"Internal", "External"};
+
+static const struct soc_enum ak4535_enum[] = {
+	SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain),
+	SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out),
+	SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out),
+	SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp),
+	SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select),
+};
+
+static const struct snd_kcontrol_new ak4535_snd_controls[] = {
+	SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0),
+	SOC_ENUM("Mono 1 Output", ak4535_enum[1]),
+	SOC_ENUM("Mono 1 Gain", ak4535_enum[0]),
+	SOC_ENUM("Headphone Output", ak4535_enum[2]),
+	SOC_ENUM("Playback Deemphasis", ak4535_enum[3]),
+	SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0),
+	SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0),
+	SOC_ENUM("Mic Select", ak4535_enum[4]),
+	SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0),
+	SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0),
+	SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0),
+	SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0),
+	SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0),
+	SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0),
+	SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0),
+	SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1),
+	SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1),
+	SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0),
+	SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
+};
+
+/* add non dapm controls */
+static int ak4535_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+			snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/* Mono 1 Mixer */
+static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
+	SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0),
+};
+
+/* Stereo Mixer */
+static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0),
+	SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0),
+};
+
+/* Input Mixer */
+static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0),
+	SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new ak4535_input_mux_control =
+	SOC_DAPM_ENUM("Input Select", ak4535_enum[4]);
+
+/* HP L switch */
+static const struct snd_kcontrol_new ak4535_hpl_control =
+	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1);
+
+/* HP R switch */
+static const struct snd_kcontrol_new ak4535_hpr_control =
+	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1);
+
+/* mono 2 switch */
+static const struct snd_kcontrol_new ak4535_mono2_control =
+	SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0);
+
+/* Line out switch */
+static const struct snd_kcontrol_new ak4535_line_control =
+	SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0);
+
+/* ak4535 dapm widgets */
+static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = {
+	SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
+		&ak4535_stereo_mixer_controls[0],
+		ARRAY_SIZE(ak4535_stereo_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
+		&ak4535_mono1_mixer_controls[0],
+		ARRAY_SIZE(ak4535_mono1_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
+		&ak4535_input_mixer_controls[0],
+		ARRAY_SIZE(ak4535_input_mixer_controls)),
+	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+		&ak4535_input_mux_control),
+	SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0),
+	SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
+		&ak4535_mono2_control),
+	/* speaker powersave bit */
+	SND_SOC_DAPM_PGA("Speaker Enable", AK4535_MODE2, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
+		&ak4535_line_control),
+	SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0,
+		&ak4535_hpl_control),
+	SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0,
+		&ak4535_hpr_control),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+	SND_SOC_DAPM_OUTPUT("SPP"),
+	SND_SOC_DAPM_OUTPUT("SPN"),
+	SND_SOC_DAPM_OUTPUT("MOUT1"),
+	SND_SOC_DAPM_OUTPUT("MOUT2"),
+	SND_SOC_DAPM_OUTPUT("MICOUT"),
+	SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 0),
+	SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0),
+
+	SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0),
+	SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0),
+	SND_SOC_DAPM_INPUT("MICIN"),
+	SND_SOC_DAPM_INPUT("MICEXT"),
+	SND_SOC_DAPM_INPUT("AUX"),
+	SND_SOC_DAPM_INPUT("MIN"),
+	SND_SOC_DAPM_INPUT("AIN"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/*stereo mixer */
+	{"Stereo Mixer", "Playback Switch", "DAC"},
+	{"Stereo Mixer", "Mic Sidetone Switch", "Mic"},
+	{"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
+
+	/* mono1 mixer */
+	{"Mono1 Mixer", "Mic Sidetone Switch", "Mic"},
+	{"Mono1 Mixer", "Mono Playback Switch", "DAC"},
+
+	/* Mic */
+	{"Mic", NULL, "AIN"},
+	{"Input Mux", "Internal", "Mic Int Bias"},
+	{"Input Mux", "External", "Mic Ext Bias"},
+	{"Mic Int Bias", NULL, "MICIN"},
+	{"Mic Ext Bias", NULL, "MICEXT"},
+	{"MICOUT", NULL, "Input Mux"},
+
+	/* line out */
+	{"LOUT", NULL, "Line Out Enable"},
+	{"ROUT", NULL, "Line Out Enable"},
+	{"Line Out Enable", "Switch", "Line Out"},
+	{"Line Out", NULL, "Stereo Mixer"},
+
+	/* mono1 out */
+	{"MOUT1", NULL, "Mono Out"},
+	{"Mono Out", NULL, "Mono1 Mixer"},
+
+	/* left HP */
+	{"HPL", NULL, "Left HP Enable"},
+	{"Left HP Enable", "Switch", "HP L Amp"},
+	{"HP L Amp", NULL, "Stereo Mixer"},
+
+	/* right HP */
+	{"HPR", NULL, "Right HP Enable"},
+	{"Right HP Enable", "Switch", "HP R Amp"},
+	{"HP R Amp", NULL, "Stereo Mixer"},
+
+	/* speaker */
+	{"SPP", NULL, "Speaker Enable"},
+	{"SPN", NULL, "Speaker Enable"},
+	{"Speaker Enable", "Switch", "Spk Amp"},
+	{"Spk Amp", NULL, "MIN"},
+
+	/* mono 2 */
+	{"MOUT2", NULL, "Mono 2 Enable"},
+	{"Mono 2 Enable", "Switch", "Stereo Mixer"},
+
+	/* Aux In */
+	{"Aux In", NULL, "AUX"},
+
+	/* ADC */
+	{"ADC", NULL, "Input Mixer"},
+	{"Input Mixer", "Mic Capture Switch", "Mic"},
+	{"Input Mixer", "Aux Capture Switch", "Aux In"},
+};
+
+static int ak4535_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets,
+				  ARRAY_SIZE(ak4535_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static int ak4535_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 ak4535_priv *ak4535 = codec->private_data;
+
+	ak4535->sysclk = freq;
+	return 0;
+}
+
+static int ak4535_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct ak4535_priv *ak4535 = codec->private_data;
+	u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
+	int rate = params_rate(params), fs = 256;
+
+	if (rate)
+		fs = ak4535->sysclk / rate;
+
+	/* set fs */
+	switch (fs) {
+	case 1024:
+		mode2 |= (0x2 << 5);
+		break;
+	case 512:
+		mode2 |= (0x1 << 5);
+		break;
+	case 256:
+		break;
+	}
+
+	/* set rate */
+	ak4535_write(codec, AK4535_MODE2, mode2);
+	return 0;
+}
+
+static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 mode1 = 0;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		mode1 = 0x0002;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode1 = 0x0001;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* use 32 fs for BCLK to save power */
+	mode1 |= 0x4;
+
+	ak4535_write(codec, AK4535_MODE1, mode1);
+	return 0;
+}
+
+static int ak4535_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf;
+	if (!mute)
+		ak4535_write(codec, AK4535_DAC, mute_reg);
+	else
+		ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
+	return 0;
+}
+
+static int ak4535_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	u16 i;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		ak4535_mute(codec->dai, 0);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		ak4535_mute(codec->dai, 1);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		i = ak4535_read_reg_cache(codec, AK4535_PM1);
+		ak4535_write(codec, AK4535_PM1, i | 0x80);
+		i = ak4535_read_reg_cache(codec, AK4535_PM2);
+		ak4535_write(codec, AK4535_PM2, i & (~0x80));
+		break;
+	case SND_SOC_BIAS_OFF:
+		i = ak4535_read_reg_cache(codec, AK4535_PM1);
+		ak4535_write(codec, AK4535_PM1, i & (~0x80));
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define AK4535_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)
+
+struct snd_soc_dai ak4535_dai = {
+	.name = "AK4535",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = AK4535_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = AK4535_RATES,
+		.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,
+	},
+};
+EXPORT_SYMBOL_GPL(ak4535_dai);
+
+static int ak4535_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;
+
+	ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int ak4535_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	ak4535_sync(codec);
+	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	ak4535_set_bias_level(codec, codec->suspend_bias_level);
+	return 0;
+}
+
+/*
+ * initialise the AK4535 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int ak4535_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	codec->name = "AK4535";
+	codec->owner = THIS_MODULE;
+	codec->read = ak4535_read_reg_cache;
+	codec->write = ak4535_write;
+	codec->set_bias_level = ak4535_set_bias_level;
+	codec->dai = &ak4535_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(ak4535_reg);
+	codec->reg_cache = kmemdup(ak4535_reg, sizeof(ak4535_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 "ak4535: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ak4535_add_controls(codec);
+	ak4535_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "ak4535: 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 *ak4535_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+#define I2C_DRIVERID_AK4535 0xfefe /* liam -  need a proper id */
+
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver ak4535_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 ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+	struct snd_soc_device *socdev = ak4535_socdev;
+	struct ak4535_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) {
+		printk(KERN_ERR "failed to attach codec at addr %x\n", addr);
+		goto err;
+	}
+
+	ret = ak4535_init(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "failed to initialise AK4535\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+
+static int ak4535_i2c_detach(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	i2c_detach_client(client);
+	kfree(codec->reg_cache);
+	kfree(client);
+	return 0;
+}
+
+static int ak4535_i2c_attach(struct i2c_adapter *adap)
+{
+	return i2c_probe(adap, &addr_data, ak4535_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver ak4535_i2c_driver = {
+	.driver = {
+		.name = "AK4535 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.id =             I2C_DRIVERID_AK4535,
+	.attach_adapter = ak4535_i2c_attach,
+	.detach_client =  ak4535_i2c_detach,
+	.command =        NULL,
+};
+
+static struct i2c_client client_template = {
+	.name =   "AK4535",
+	.driver = &ak4535_i2c_driver,
+};
+#endif
+
+static int ak4535_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct ak4535_setup_data *setup;
+	struct snd_soc_codec *codec;
+	struct ak4535_priv *ak4535;
+	int ret = 0;
+
+	printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION);
+
+	setup = socdev->codec_data;
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL);
+	if (ak4535 == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+
+	codec->private_data = ak4535;
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	ak4535_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;
+		codec->hw_read = (hw_read_t)i2c_master_recv;
+		ret = i2c_add_driver(&ak4535_i2c_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add i2c driver");
+	}
+#else
+	/* Add other interfaces here */
+#endif
+	return ret;
+}
+
+/* power down chip */
+static int ak4535_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)
+		ak4535_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(&ak4535_i2c_driver);
+#endif
+	kfree(codec->private_data);
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ak4535 = {
+	.probe = 	ak4535_probe,
+	.remove = 	ak4535_remove,
+	.suspend = 	ak4535_suspend,
+	.resume =	ak4535_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535);
+
+MODULE_DESCRIPTION("Soc AK4535 driver");
+MODULE_AUTHOR("Richard Purdie");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h
new file mode 100644
index 000000000000..e9fe30e2c056
--- /dev/null
+++ b/sound/soc/codecs/ak4535.h
@@ -0,0 +1,46 @@
+/*
+ * ak4535.h  --  AK4535 Soc Audio driver
+ *
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on wm8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AK4535_H
+#define _AK4535_H
+
+/* AK4535 register space */
+
+#define AK4535_PM1		0x0
+#define AK4535_PM2		0x1
+#define AK4535_SIG1		0x2
+#define AK4535_SIG2		0x3
+#define AK4535_MODE1		0x4
+#define AK4535_MODE2		0x5
+#define AK4535_DAC		0x6
+#define AK4535_MIC		0x7
+#define AK4535_TIMER		0x8
+#define AK4535_ALC1		0x9
+#define AK4535_ALC2		0xa
+#define AK4535_PGA		0xb
+#define AK4535_LATT		0xc
+#define AK4535_RATT		0xd
+#define AK4535_VOL		0xe
+#define AK4535_STATUS		0xf
+
+#define AK4535_CACHEREGNUM 	0x10
+
+struct ak4535_setup_data {
+	unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai ak4535_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ak4535;
+
+#endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index e73fcfd9f5cd..9deb8c74fdfd 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -201,7 +201,7 @@ static struct {
  * driver what the input settings can be.  This would need to be implemented
  * for stand-alone mode to work.
  */
-static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int cs4270_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;
@@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
  * data for playback only, but ASoC currently does not support different
  * formats for playback vs. record.
  */
-static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			      unsigned int format)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
  * board does not have the MUTEA or MUTEB pins connected to such circuitry,
  * then this function will do nothing.
  */
-static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute)
+static int cs4270_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	int reg6;
@@ -667,7 +667,7 @@ error:
 
 #endif /* USE_I2C*/
 
-struct snd_soc_codec_dai cs4270_dai = {
+struct snd_soc_dai cs4270_dai = {
 	.name = "CS4270",
 	.playback = {
 		.stream_name = "Playback",
diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h
index 0ced49b7804d..adc6cd9667d4 100644
--- a/sound/soc/codecs/cs4270.h
+++ b/sound/soc/codecs/cs4270.h
@@ -16,7 +16,7 @@
  * The ASoC codec DAI structure for the CS4270.  Assign this structure to
  * the .codec_dai field of your machine driver's snd_soc_dai_link structure.
  */
-extern struct snd_soc_codec_dai cs4270_dai;
+extern struct snd_soc_dai cs4270_dai;
 
 /*
  * The ASoC codec device structure for the CS4270.  Assign this structure
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 09b1661b8a3a..b1dce5f459db 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -29,7 +29,7 @@
  *  ---------------------------------------
  *
  *  Hence the machine layer should disable unsupported inputs/outputs by
- *  snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc.
+ *  snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc.
  */
 
 #include <linux/module.h>
@@ -49,7 +49,7 @@
 #include "tlv320aic3x.h"
 
 #define AUDIO_NAME "aic3x"
-#define AIC3X_VERSION "0.1"
+#define AIC3X_VERSION "0.2"
 
 /* codec private data */
 struct aic3x_priv {
@@ -138,6 +138,20 @@ static int aic3x_write(struct snd_soc_codec *codec, unsigned int reg,
 		return -EIO;
 }
 
+/*
+ * read from the aic3x register space
+ */
+static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
+		      u8 *value)
+{
+	*value = reg & 0xff;
+	if (codec->hw_read(codec->control_data, value, 1) != 1)
+		return -EIO;
+
+	aic3x_write_reg_cache(codec, reg, *value);
+	return 0;
+}
+
 #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, \
@@ -192,7 +206,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 		}
 
 		if (found)
-			snd_soc_dapm_sync_endpoints(widget->codec);
+			snd_soc_dapm_sync(widget->codec);
 	}
 
 	ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
@@ -209,6 +223,8 @@ static const char *aic3x_right_hpcom_mux[] =
     { "differential of HPROUT", "constant VCM", "single-ended",
       "differential of HPLCOM", "external feedback" };
 static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" };
+static const char *aic3x_adc_hpf[] =
+    { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
 
 #define LDAC_ENUM	0
 #define RDAC_ENUM	1
@@ -218,6 +234,7 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" };
 #define LINE1R_ENUM	5
 #define LINE2L_ENUM	6
 #define LINE2R_ENUM	7
+#define ADC_HPF_ENUM	8
 
 static const struct soc_enum aic3x_enum[] = {
 	SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux),
@@ -228,6 +245,7 @@ static const struct soc_enum aic3x_enum[] = {
 	SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
 	SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
 	SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
+	SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
 };
 
 static const struct snd_kcontrol_new aic3x_snd_controls[] = {
@@ -278,6 +296,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	/* Input */
 	SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0),
 	SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
+
+	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
 };
 
 /* add non dapm controls */
@@ -441,11 +461,34 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
 			 &aic3x_right_line2_mux_controls),
 
+	/*
+	 * Not a real mic bias widget but similar function. This is for dynamic
+	 * control of GPIO1 digital mic modulator clock output function when
+	 * using digital mic.
+	 */
+	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk",
+			 AIC3X_GPIO1_REG, 4, 0xf,
+			 AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK,
+			 AIC3X_GPIO1_FUNC_DISABLED),
+
+	/*
+	 * Also similar function like mic bias. Selects digital mic with
+	 * configurable oversampling rate instead of ADC converter.
+	 */
+	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128",
+			 AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0),
+	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64",
+			 AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0),
+	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32",
+			 AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0),
+
 	/* Mic Bias */
-	SND_SOC_DAPM_MICBIAS("Mic Bias 2V", MICBIAS_CTRL, 6, 0),
-	SND_SOC_DAPM_MICBIAS("Mic Bias 2.5V", MICBIAS_CTRL, 7, 0),
-	SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0),
-	SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0),
+	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V",
+			 MICBIAS_CTRL, 6, 3, 1, 0),
+	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V",
+			 MICBIAS_CTRL, 6, 3, 2, 0),
+	SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD",
+			 MICBIAS_CTRL, 6, 3, 3, 0),
 
 	/* Left PGA to Left Output bypass */
 	SND_SOC_DAPM_MIXER("Left PGA Bypass Mixer", SND_SOC_NOPM, 0, 0,
@@ -483,7 +526,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("LINE2R"),
 };
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 	/* Left Output */
 	{"Left DAC Mux", "DAC_L1", "Left DAC"},
 	{"Left DAC Mux", "DAC_L2", "Left DAC"},
@@ -554,6 +597,7 @@ static const char *intercon[][3] = {
 	{"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
 
 	{"Left ADC", NULL, "Left PGA Mixer"},
+	{"Left ADC", NULL, "GPIO1 dmic modclk"},
 
 	/* Right Input */
 	{"Right Line1R Mux", "single-ended", "LINE1R"},
@@ -567,6 +611,7 @@ static const char *intercon[][3] = {
 	{"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"},
@@ -628,101 +673,27 @@ static const char *intercon[][3] = {
 	{"Mono Out", NULL, "Right Line2 Bypass Mixer"},
 	{"Right HP Out", NULL, "Right Line2 Bypass Mixer"},
 
-	/* terminator */
-	{NULL, NULL, NULL},
+	/*
+	 * Logical path between digital mic enable and GPIO1 modulator clock
+	 * output function
+	 */
+	{"GPIO1 dmic modclk", NULL, "DMic Rate 128"},
+	{"GPIO1 dmic modclk", NULL, "DMic Rate 64"},
+	{"GPIO1 dmic modclk", NULL, "DMic Rate 32"},
 };
 
 static int aic3x_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* set up audio path interconnects */
-	for (i = 0; intercon[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, intercon[i][0],
-					   intercon[i][1], intercon[i][2]);
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
-struct aic3x_rate_divs {
-	u32 mclk;
-	u32 rate;
-	u32 fsref_reg;
-	u8 sr_reg:4;
-	u8 pllj_reg;
-	u16 plld_reg;
-};
-
-/* AIC3X codec mclk clock divider coefficients */
-static const struct aic3x_rate_divs aic3x_divs[] = {
-	/* 8k */
-	{12000000, 8000, 48000, 0xa, 16, 3840},
-	{19200000, 8000, 48000, 0xa, 10, 2400},
-	{22579200, 8000, 48000, 0xa, 8, 7075},
-	{33868800, 8000, 48000, 0xa, 5, 8049},
-	/* 11.025k */
-	{12000000, 11025, 44100, 0x6, 15, 528},
-	{19200000, 11025, 44100, 0x6, 9, 4080},
-	{22579200, 11025, 44100, 0x6, 8, 0},
-	{33868800, 11025, 44100, 0x6, 5, 3333},
-	/* 16k */
-	{12000000, 16000, 48000, 0x4, 16, 3840},
-	{19200000, 16000, 48000, 0x4, 10, 2400},
-	{22579200, 16000, 48000, 0x4, 8, 7075},
-	{33868800, 16000, 48000, 0x4, 5, 8049},
-	/* 22.05k */
-	{12000000, 22050, 44100, 0x2, 15, 528},
-	{19200000, 22050, 44100, 0x2, 9, 4080},
-	{22579200, 22050, 44100, 0x2, 8, 0},
-	{33868800, 22050, 44100, 0x2, 5, 3333},
-	/* 32k */
-	{12000000, 32000, 48000, 0x1, 16, 3840},
-	{19200000, 32000, 48000, 0x1, 10, 2400},
-	{22579200, 32000, 48000, 0x1, 8, 7075},
-	{33868800, 32000, 48000, 0x1, 5, 8049},
-	/* 44.1k */
-	{12000000, 44100, 44100, 0x0, 15, 528},
-	{19200000, 44100, 44100, 0x0, 9, 4080},
-	{22579200, 44100, 44100, 0x0, 8, 0},
-	{33868800, 44100, 44100, 0x0, 5, 3333},
-	/* 48k */
-	{12000000, 48000, 48000, 0x0, 16, 3840},
-	{19200000, 48000, 48000, 0x0, 10, 2400},
-	{22579200, 48000, 48000, 0x0, 8, 7075},
-	{33868800, 48000, 48000, 0x0, 5, 8049},
-	/* 64k */
-	{12000000, 64000, 96000, 0x1, 16, 3840},
-	{19200000, 64000, 96000, 0x1, 10, 2400},
-	{22579200, 64000, 96000, 0x1, 8, 7075},
-	{33868800, 64000, 96000, 0x1, 5, 8049},
-	/* 88.2k */
-	{12000000, 88200, 88200, 0x0, 15, 528},
-	{19200000, 88200, 88200, 0x0, 9, 4080},
-	{22579200, 88200, 88200, 0x0, 8, 0},
-	{33868800, 88200, 88200, 0x0, 5, 3333},
-	/* 96k */
-	{12000000, 96000, 96000, 0x0, 16, 3840},
-	{19200000, 96000, 96000, 0x0, 10, 2400},
-	{22579200, 96000, 96000, 0x0, 8, 7075},
-	{33868800, 96000, 96000, 0x0, 5, 8049},
-};
-
-static inline int aic3x_get_divs(int mclk, int rate)
-{
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(aic3x_divs); i++) {
-		if (aic3x_divs[i].rate == rate && aic3x_divs[i].mclk == mclk)
-			return i;
-	}
-
-	return 0;
-}
-
 static int aic3x_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params)
 {
@@ -730,49 +701,107 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
-	int i;
-	u8 data, pll_p, pll_r, pll_j;
-	u16 pll_d;
-
-	i = aic3x_get_divs(aic3x->sysclk, params_rate(params));
+	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
+	u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
+	u16 pll_d = 1;
 
-	/* Route Left DAC to left channel input and
-	 * right DAC to right channel input */
-	data = (LDAC2LCH | RDAC2RCH);
-	switch (aic3x_divs[i].fsref_reg) {
-	case 44100:
-		data |= FSREF_44100;
+	/* select data word length */
+	data =
+	    aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
-	case 48000:
-		data |= FSREF_48000;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		data |= (0x01 << 4);
 		break;
-	case 88200:
-		data |= FSREF_44100 | DUAL_RATE_MODE;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		data |= (0x02 << 4);
 		break;
-	case 96000:
-		data |= FSREF_48000 | DUAL_RATE_MODE;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		data |= (0x03 << 4);
 		break;
 	}
+	aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data);
+
+	/* Fsref can be 44100 or 48000 */
+	fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000;
+
+	/* Try to find a value for Q which allows us to bypass the PLL and
+	 * generate CODEC_CLK directly. */
+	for (pll_q = 2; pll_q < 18; pll_q++)
+		if (aic3x->sysclk / (128 * pll_q) == fsref) {
+			bypass_pll = 1;
+			break;
+		}
+
+	if (bypass_pll) {
+		pll_q &= 0xf;
+		aic3x_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
+		aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
+	} else
+		aic3x_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
+
+	/* Route Left DAC to left channel input and
+	 * right DAC to right channel input */
+	data = (LDAC2LCH | RDAC2RCH);
+	data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000;
+	if (params_rate(params) >= 64000)
+		data |= DUAL_RATE_MODE;
 	aic3x_write(codec, AIC3X_CODEC_DATAPATH_REG, data);
 
 	/* codec sample rate select */
-	data = aic3x_divs[i].sr_reg;
+	data = (fsref * 20) / params_rate(params);
+	if (params_rate(params) < 64000)
+		data /= 2;
+	data /= 5;
+	data -= 2;
 	data |= (data << 4);
 	aic3x_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data);
 
-	/* Use PLL for generation Fsref by equation:
-	 * Fsref = (MCLK * K * R)/(2048 * P);
-	 * Fix P = 2 and R = 1 and calculate K, if
-	 * K = J.D, i.e. J - an interger portion of K and D is the fractional
-	 * one with 4 digits of precision;
-	 * Example:
-	 * For MCLK = 22.5792 MHz and Fsref = 48kHz:
-	 * Select P = 2, R= 1, K = 8.7074, which results in J = 8, D = 7074
+	if (bypass_pll)
+		return 0;
+
+	/* Use PLL
+	 * find an apropriate setup for j, d, r and p by iterating over
+	 * p and r - j and d are calculated for each fraction.
+	 * Up to 128 values are probed, the closest one wins the game.
+	 * The sysclk is divided by 1000 to prevent integer overflows.
 	 */
-	pll_p = 2;
-	pll_r = 1;
-	pll_j = aic3x_divs[i].pllj_reg;
-	pll_d = aic3x_divs[i].plld_reg;
+	codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000);
+
+	for (r = 1; r <= 16; r++)
+		for (p = 1; p <= 8; p++) {
+			int clk, tmp = (codec_clk * pll_r * 10) / pll_p;
+			u8 j = tmp / 10000;
+			u16 d = tmp % 10000;
+
+			if (j > 63)
+				continue;
+
+			if (d != 0 && aic3x->sysclk < 10000000)
+				continue;
+
+			/* This is actually 1000 * ((j + (d/10000)) * r) / p
+			 * The term had to be converted to get rid of the
+			 * division by 10000 */
+			clk = ((10000 * j * r) + (d * r)) / (10 * p);
+
+			/* check whether this values get closer than the best
+			 * ones we had before */
+			if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) {
+				pll_j = j; pll_d = d; pll_r = r; pll_p = p;
+				last_clk = clk;
+			}
+
+			/* Early exit for exact matches */
+			if (clk == codec_clk)
+				break;
+		}
+
+	if (last_clk == 0) {
+		printk(KERN_ERR "%s(): unable to setup PLL\n", __func__);
+		return -EINVAL;
+	}
 
 	data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
 	aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT));
@@ -782,28 +811,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	aic3x_write(codec, AIC3X_PLL_PROGD_REG,
 		    (pll_d & 0x3F) << PLLD_LSB_SHIFT);
 
-	/* select data word length */
-	data =
-	    aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
-		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
-		data |= (0x01 << 4);
-		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-		data |= (0x02 << 4);
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-		data |= (0x03 << 4);
-		break;
-	}
-	aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, data);
-
 	return 0;
 }
 
-static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute)
+static int aic3x_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON;
@@ -820,31 +831,25 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int aic3x_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 aic3x_priv *aic3x = codec->private_data;
 
-	switch (freq) {
-	case 12000000:
-	case 19200000:
-	case 22579200:
-	case 33868800:
-		aic3x->sysclk = freq;
-		return 0;
-	}
-
-	return -EINVAL;
+	aic3x->sysclk = freq;
+	return 0;
 }
 
-static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			     unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
-	u8 iface_areg = 0;
-	u8 iface_breg = 0;
+	u8 iface_areg, iface_breg;
+
+	iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
+	iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
 
 	/* set master/slave audio interface */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -883,13 +888,14 @@ static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
+static int aic3x_set_bias_level(struct snd_soc_codec *codec,
+				enum snd_soc_bias_level level)
 {
 	struct aic3x_priv *aic3x = codec->private_data;
 	u8 reg;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0:
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* all power is driven by DAPM system */
 		if (aic3x->master) {
 			/* enable pll */
@@ -898,10 +904,9 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
 				    reg | PLL_ENABLE);
 		}
 		break;
-	case SNDRV_CTL_POWER_D1:
-	case SNDRV_CTL_POWER_D2:
+	case SND_SOC_BIAS_PREPARE:
 		break;
-	case SNDRV_CTL_POWER_D3hot:
+	case SND_SOC_BIAS_STANDBY:
 		/*
 		 * all power is driven by DAPM system,
 		 * so output power is safe if bypass was set
@@ -913,7 +918,7 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
 				    reg & ~PLL_ENABLE);
 		}
 		break;
-	case SNDRV_CTL_POWER_D3cold:
+	case SND_SOC_BIAS_OFF:
 		/* force all power off */
 		reg = aic3x_read_reg_cache(codec, LINE1L_2_LADC_CTRL);
 		aic3x_write(codec, LINE1L_2_LADC_CTRL, reg & ~LADC_PWR_ON);
@@ -949,16 +954,43 @@ static int aic3x_dapm_event(struct snd_soc_codec *codec, int event)
 		}
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 
 	return 0;
 }
 
+void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state)
+{
+	u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
+	u8 bit = gpio ? 3: 0;
+	u8 val = aic3x_read_reg_cache(codec, reg) & ~(1 << bit);
+	aic3x_write(codec, reg, val | (!!state << bit));
+}
+EXPORT_SYMBOL_GPL(aic3x_set_gpio);
+
+int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
+{
+	u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
+	u8 val, bit = gpio ? 2: 1;
+
+	aic3x_read(codec, reg, &val);
+	return (val >> bit) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_get_gpio);
+
+int aic3x_headset_detected(struct snd_soc_codec *codec)
+{
+	u8 val;
+	aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val);
+	return (val >> 2) & 1;
+}
+EXPORT_SYMBOL_GPL(aic3x_headset_detected);
+
 #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)
 
-struct snd_soc_codec_dai aic3x_dai = {
+struct snd_soc_dai aic3x_dai = {
 	.name = "aic3x",
 	.playback = {
 		.stream_name = "Playback",
@@ -988,7 +1020,7 @@ static int aic3x_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;
 
-	aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }
@@ -1008,7 +1040,7 @@ static int aic3x_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	aic3x_dapm_event(codec, codec->suspend_dapm_state);
+	aic3x_set_bias_level(codec, codec->suspend_bias_level);
 
 	return 0;
 }
@@ -1020,16 +1052,17 @@ static int aic3x_resume(struct platform_device *pdev)
 static int aic3x_init(struct snd_soc_device *socdev)
 {
 	struct snd_soc_codec *codec = socdev->codec;
+	struct aic3x_setup_data *setup = socdev->codec_data;
 	int reg, ret = 0;
 
 	codec->name = "aic3x";
 	codec->owner = THIS_MODULE;
 	codec->read = aic3x_read_reg_cache;
 	codec->write = aic3x_write;
-	codec->dapm_event = aic3x_dapm_event;
+	codec->set_bias_level = aic3x_set_bias_level;
 	codec->dai = &aic3x_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = sizeof(aic3x_reg);
+	codec->reg_cache_size = ARRAY_SIZE(aic3x_reg);
 	codec->reg_cache = kmemdup(aic3x_reg, sizeof(aic3x_reg), GFP_KERNEL);
 	if (codec->reg_cache == NULL)
 		return -ENOMEM;
@@ -1108,7 +1141,11 @@ static int aic3x_init(struct snd_soc_device *socdev)
 	aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
 
 	/* off, with power on */
-	aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* setup GPIO functions */
+	aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
+	aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
 
 	aic3x_add_controls(codec);
 	aic3x_add_widgets(codec);
@@ -1217,6 +1254,12 @@ static struct i2c_client client_template = {
 	.name = "AIC3X",
 	.driver = &aic3x_i2c_driver,
 };
+
+static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len)
+{
+	value[0] = i2c_smbus_read_byte_data(client, value[0]);
+	return (len == 1);
+}
 #endif
 
 static int aic3x_probe(struct platform_device *pdev)
@@ -1251,6 +1294,7 @@ static int aic3x_probe(struct platform_device *pdev)
 	if (setup->i2c_address) {
 		normal_i2c[0] = setup->i2c_address;
 		codec->hw_write = (hw_write_t) i2c_master_send;
+		codec->hw_read = (hw_read_t) aic3x_i2c_read;
 		ret = i2c_add_driver(&aic3x_i2c_driver);
 		if (ret != 0)
 			printk(KERN_ERR "can't add i2c driver");
@@ -1268,7 +1312,7 @@ static int aic3x_remove(struct platform_device *pdev)
 
 	/* power down chip */
 	if (codec->control_data)
-		aic3x_dapm_event(codec, SNDRV_CTL_POWER_D3);
+		aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index d0cdeeb629de..d76c079b86e7 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -37,6 +37,8 @@
 #define AIC3X_ASD_INTF_CTRLB		9
 /* 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
 
 /* ADC PGA Gain control registers */
 #define LADC_VOL			15
@@ -108,6 +110,13 @@
 #define DACR1_2_RLOPM_VOL		92
 #define LLOPM_CTRL			86
 #define RLOPM_CTRL			93
+/* GPIO/IRQ registers */
+#define AIC3X_STICKY_IRQ_FLAGS_REG	96
+#define AIC3X_RT_IRQ_FLAGS_REG		97
+#define AIC3X_GPIO1_REG			98
+#define AIC3X_GPIO2_REG			99
+#define AIC3X_GPIOA_REG			100
+#define AIC3X_GPIOB_REG			101
 /* Clock generation control register */
 #define AIC3X_CLKGEN_CTRL_REG		102
 
@@ -128,12 +137,15 @@
 
 /* PLL registers bitfields */
 #define PLLP_SHIFT		0
+#define PLLQ_SHIFT		3
 #define PLLR_SHIFT		0
 #define PLLJ_SHIFT		2
 #define PLLD_MSB_SHIFT		0
 #define PLLD_LSB_SHIFT		2
 
 /* Clock generation register bits */
+#define CODEC_CLKIN_PLLDIV	0
+#define CODEC_CLKIN_CLKDIV	1
 #define PLL_CLKIN_SHIFT		4
 #define MCLK_SOURCE		0x0
 #define PLL_CLKDIV_SHIFT	0
@@ -171,11 +183,52 @@
 /* Default input volume */
 #define DEFAULT_GAIN    0x20
 
+/* GPIO API */
+enum {
+	AIC3X_GPIO1_FUNC_DISABLED		= 0,
+	AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC	= 1,
+	AIC3X_GPIO1_FUNC_CLOCK_MUX		= 2,
+	AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2		= 3,
+	AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4		= 4,
+	AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8		= 5,
+	AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ	= 6,
+	AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ		= 7,
+	AIC3X_GPIO1_FUNC_INPUT			= 8,
+	AIC3X_GPIO1_FUNC_OUTPUT			= 9,
+	AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK	= 10,
+	AIC3X_GPIO1_FUNC_AUDIO_WORDCLK		= 11,
+	AIC3X_GPIO1_FUNC_BUTTON_IRQ		= 12,
+	AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ	= 13,
+	AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ	= 14,
+	AIC3X_GPIO1_FUNC_ALL_IRQ		= 16
+};
+
+enum {
+	AIC3X_GPIO2_FUNC_DISABLED		= 0,
+	AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ	= 2,
+	AIC3X_GPIO2_FUNC_INPUT			= 3,
+	AIC3X_GPIO2_FUNC_OUTPUT			= 4,
+	AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT	= 5,
+	AIC3X_GPIO2_FUNC_AUDIO_BITCLK		= 8,
+	AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9,
+	AIC3X_GPIO2_FUNC_ALL_IRQ		= 10,
+	AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11,
+	AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12,
+	AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ	= 13,
+	AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ		= 14,
+	AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ	= 15
+};
+
+void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
+int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
+int aic3x_headset_detected(struct snd_soc_codec *codec);
+
 struct aic3x_setup_data {
 	unsigned short i2c_address;
+	unsigned int gpio_func[2];
 };
 
-extern struct snd_soc_codec_dai aic3x_dai;
+extern struct snd_soc_dai aic3x_dai;
 extern struct snd_soc_codec_device soc_codec_dev_aic3x;
 
 #endif /* _AIC3X_H */
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
new file mode 100644
index 000000000000..a52d6d9e007a
--- /dev/null
+++ b/sound/soc/codecs/uda1380.c
@@ -0,0 +1,852 @@
+/*
+ * uda1380.c - Philips UDA1380 ALSA SoC audio driver
+ *
+ * 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.
+ *
+ * Copyright (c) 2007 Philipp Zabel <philipp.zabel@gmail.com>
+ * Improved support for DAPM and audio routing/mixing capabilities,
+ * added TLV support.
+ *
+ * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC
+ * codec model.
+ *
+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
+ * Copyright 2005 Openedhand Ltd.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/types.h>
+#include <linux/string.h>
+#include <linux/slab.h>
+#include <linux/errno.h>
+#include <linux/ioctl.h>
+#include <linux/delay.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "uda1380.h"
+
+#define UDA1380_VERSION "0.6"
+#define AUDIO_NAME "uda1380"
+
+/*
+ * uda1380 register cache
+ */
+static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
+	0x0502, 0x0000, 0x0000, 0x3f3f,
+	0x0202, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0000, 0xff00, 0x0000, 0x4800,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x8000, 0x0002, 0x0000,
+};
+
+/*
+ * read uda1380 register cache
+ */
+static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg == UDA1380_RESET)
+		return 0;
+	if (reg >= UDA1380_CACHEREGNUM)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write uda1380 register cache
+ */
+static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
+	u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= UDA1380_CACHEREGNUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the UDA1380 register space
+ */
+static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[3];
+
+	/* data is
+	 *   data[0] is register offset
+	 *   data[1] is MS byte
+	 *   data[2] is LS byte
+	 */
+	data[0] = reg;
+	data[1] = (value & 0xff00) >> 8;
+	data[2] = value & 0x00ff;
+
+	uda1380_write_reg_cache(codec, reg, value);
+
+	/* the interpolator & decimator regs must only be written when the
+	 * codec DAI is active.
+	 */
+	if (!codec->active && (reg >= UDA1380_MVOL))
+		return 0;
+	pr_debug("uda1380: hw write %x val %x\n", reg, value);
+	if (codec->hw_write(codec->control_data, data, 3) == 3) {
+		unsigned int val;
+		i2c_master_send(codec->control_data, data, 1);
+		i2c_master_recv(codec->control_data, data, 2);
+		val = (data[0]<<8) | data[1];
+		if (val != value) {
+			pr_debug("uda1380: READ BACK VAL %x\n",
+					(data[0]<<8) | data[1]);
+			return -EIO;
+		}
+		return 0;
+	} else
+		return -EIO;
+}
+
+#define uda1380_reset(c)	uda1380_write(c, UDA1380_RESET, 0)
+
+/* declarations of ALSA reg_elem_REAL controls */
+static const char *uda1380_deemp[] = {
+	"None",
+	"32kHz",
+	"44.1kHz",
+	"48kHz",
+	"96kHz",
+};
+static const char *uda1380_input_sel[] = {
+	"Line",
+	"Mic + Line R",
+	"Line L",
+	"Mic",
+};
+static const char *uda1380_output_sel[] = {
+	"DAC",
+	"Analog Mixer",
+};
+static const char *uda1380_spf_mode[] = {
+	"Flat",
+	"Minimum1",
+	"Minimum2",
+	"Maximum"
+};
+static const char *uda1380_capture_sel[] = {
+	"ADC",
+	"Digital Mixer"
+};
+static const char *uda1380_sel_ns[] = {
+	"3rd-order",
+	"5th-order"
+};
+static const char *uda1380_mix_control[] = {
+	"off",
+	"PCM only",
+	"before sound processing",
+	"after sound processing"
+};
+static const char *uda1380_sdet_setting[] = {
+	"3200",
+	"4800",
+	"9600",
+	"19200"
+};
+static const char *uda1380_os_setting[] = {
+	"single-speed",
+	"double-speed (no mixing)",
+	"quad-speed (no mixing)"
+};
+
+static const struct soc_enum uda1380_deemp_enum[] = {
+	SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp),
+	SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp),
+};
+static const struct soc_enum uda1380_input_sel_enum =
+	SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel);		/* SEL_MIC, SEL_LNA */
+static const struct soc_enum uda1380_output_sel_enum =
+	SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel);		/* R02_EN_AVC */
+static const struct soc_enum uda1380_spf_enum =
+	SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode);		/* M */
+static const struct soc_enum uda1380_capture_sel_enum =
+	SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel);	/* SEL_SOURCE */
+static const struct soc_enum uda1380_sel_ns_enum =
+	SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns);		/* SEL_NS */
+static const struct soc_enum uda1380_mix_enum =
+	SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control);	/* MIX, MIX_POS */
+static const struct soc_enum uda1380_sdet_enum =
+	SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting);	/* SD_VALUE */
+static const struct soc_enum uda1380_os_enum =
+	SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting);	/* OS */
+
+/*
+ * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB)
+ */
+static DECLARE_TLV_DB_SCALE(amix_tlv, -4950, 150, 1);
+
+/*
+ * from -78 dB in 1 dB steps (3 dB steps, really. LSB are ignored),
+ * from -66 dB in 0.5 dB steps (2 dB steps, really) and
+ * from -52 dB in 0.25 dB steps
+ */
+static const unsigned int mvol_tlv[] = {
+	TLV_DB_RANGE_HEAD(3),
+	0, 15, TLV_DB_SCALE_ITEM(-8200, 100, 1),
+	16, 43, TLV_DB_SCALE_ITEM(-6600, 50, 0),
+	44, 252, TLV_DB_SCALE_ITEM(-5200, 25, 0),
+};
+
+/*
+ * from -72 dB in 1.5 dB steps (6 dB steps really),
+ * from -66 dB in 0.75 dB steps (3 dB steps really),
+ * from -60 dB in 0.5 dB steps (2 dB steps really) and
+ * from -46 dB in 0.25 dB steps
+ */
+static const unsigned int vc_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0, 7, TLV_DB_SCALE_ITEM(-7800, 150, 1),
+	8, 15, TLV_DB_SCALE_ITEM(-6600, 75, 0),
+	16, 43, TLV_DB_SCALE_ITEM(-6000, 50, 0),
+	44, 228, TLV_DB_SCALE_ITEM(-4600, 25, 0),
+};
+
+/* from 0 to 6 dB in 2 dB steps if SPF mode != flat */
+static DECLARE_TLV_DB_SCALE(tr_tlv, 0, 200, 0);
+
+/* from 0 to 24 dB in 2 dB steps, if SPF mode == maximum, otherwise cuts
+ * off at 18 dB max) */
+static DECLARE_TLV_DB_SCALE(bb_tlv, 0, 200, 0);
+
+/* from -63 to 24 dB in 0.5 dB steps (-128...48) */
+static DECLARE_TLV_DB_SCALE(dec_tlv, -6400, 50, 1);
+
+/* from 0 to 24 dB in 3 dB steps */
+static DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
+
+/* from 0 to 30 dB in 2 dB steps */
+static DECLARE_TLV_DB_SCALE(vga_tlv, 0, 200, 0);
+
+static const struct snd_kcontrol_new uda1380_snd_controls[] = {
+	SOC_DOUBLE_TLV("Analog Mixer Volume", UDA1380_AMIX, 0, 8, 44, 1, amix_tlv),	/* AVCR, AVCL */
+	SOC_DOUBLE_TLV("Master Playback Volume", UDA1380_MVOL, 0, 8, 252, 1, mvol_tlv),	/* MVCL, MVCR */
+	SOC_SINGLE_TLV("ADC Playback Volume", UDA1380_MIXVOL, 8, 228, 1, vc_tlv),	/* VC2 */
+	SOC_SINGLE_TLV("PCM Playback Volume", UDA1380_MIXVOL, 0, 228, 1, vc_tlv),	/* VC1 */
+	SOC_ENUM("Sound Processing Filter", uda1380_spf_enum),				/* M */
+	SOC_DOUBLE_TLV("Tone Control - Treble", UDA1380_MODE, 4, 12, 3, 0, tr_tlv), 	/* TRL, TRR */
+	SOC_DOUBLE_TLV("Tone Control - Bass", UDA1380_MODE, 0, 8, 15, 0, bb_tlv),	/* BBL, BBR */
+/**/	SOC_SINGLE("Master Playback Switch", UDA1380_DEEMP, 14, 1, 1),		/* MTM */
+	SOC_SINGLE("ADC Playback Switch", UDA1380_DEEMP, 11, 1, 1),		/* MT2 from decimation filter */
+	SOC_ENUM("ADC Playback De-emphasis", uda1380_deemp_enum[0]),		/* DE2 */
+	SOC_SINGLE("PCM Playback Switch", UDA1380_DEEMP, 3, 1, 1),		/* MT1, from digital data input */
+	SOC_ENUM("PCM Playback De-emphasis", uda1380_deemp_enum[1]),		/* DE1 */
+	SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0),	/* DA_POL_INV */
+	SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum),				/* SEL_NS */
+	SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum),		/* MIX_POS, MIX */
+	SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0),			/* SILENCE, force DAC output to silence */
+	SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0),		/* SDET_ON */
+	SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum),		/* SD_VALUE */
+	SOC_ENUM("Oversampling Input", uda1380_os_enum),			/* OS */
+	SOC_DOUBLE_S8_TLV("ADC Capture Volume", UDA1380_DEC, -128, 48, dec_tlv),	/* ML_DEC, MR_DEC */
+/**/	SOC_SINGLE("ADC Capture Switch", UDA1380_PGA, 15, 1, 1),		/* MT_ADC */
+	SOC_DOUBLE_TLV("Line Capture Volume", UDA1380_PGA, 0, 8, 8, 0, pga_tlv), /* PGA_GAINCTRLL, PGA_GAINCTRLR */
+	SOC_SINGLE("ADC Polarity inverting Switch", UDA1380_ADC, 12, 1, 0),	/* ADCPOL_INV */
+	SOC_SINGLE_TLV("Mic Capture Volume", UDA1380_ADC, 8, 15, 0, vga_tlv),	/* VGA_CTRL */
+	SOC_SINGLE("DC Filter Bypass Switch", UDA1380_ADC, 1, 1, 0),		/* SKIP_DCFIL (before decimator) */
+	SOC_SINGLE("DC Filter Enable Switch", UDA1380_ADC, 0, 1, 0),		/* EN_DCFIL (at output of decimator) */
+	SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0),			/* TODO: enum, see table 62 */
+	SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1),			/* AGC_LEVEL */
+	/* -5.5, -8, -11.5, -14 dBFS */
+	SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
+};
+
+/* add non dapm controls */
+static int uda1380_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+			snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/* Input mux */
+static const struct snd_kcontrol_new uda1380_input_mux_control =
+	SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
+
+/* Output mux */
+static const struct snd_kcontrol_new uda1380_output_mux_control =
+	SOC_DAPM_ENUM("Route", uda1380_output_sel_enum);
+
+/* Capture mux */
+static const struct snd_kcontrol_new uda1380_capture_mux_control =
+	SOC_DAPM_ENUM("Route", uda1380_capture_sel_enum);
+
+
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+		&uda1380_input_mux_control),
+	SND_SOC_DAPM_MUX("Output Mux", SND_SOC_NOPM, 0, 0,
+		&uda1380_output_mux_control),
+	SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0,
+		&uda1380_capture_mux_control),
+	SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0),
+	SND_SOC_DAPM_INPUT("VINM"),
+	SND_SOC_DAPM_INPUT("VINL"),
+	SND_SOC_DAPM_INPUT("VINR"),
+	SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("VOUTLHP"),
+	SND_SOC_DAPM_OUTPUT("VOUTRHP"),
+	SND_SOC_DAPM_OUTPUT("VOUTL"),
+	SND_SOC_DAPM_OUTPUT("VOUTR"),
+	SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0),
+	SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* output mux */
+	{"HeadPhone Driver", NULL, "Output Mux"},
+	{"VOUTR", NULL, "Output Mux"},
+	{"VOUTL", NULL, "Output Mux"},
+
+	{"Analog Mixer", NULL, "VINR"},
+	{"Analog Mixer", NULL, "VINL"},
+	{"Analog Mixer", NULL, "DAC"},
+
+	{"Output Mux", "DAC", "DAC"},
+	{"Output Mux", "Analog Mixer", "Analog Mixer"},
+
+	/* {"DAC", "Digital Mixer", "I2S" } */
+
+	/* headphone driver */
+	{"VOUTLHP", NULL, "HeadPhone Driver"},
+	{"VOUTRHP", NULL, "HeadPhone Driver"},
+
+	/* input mux */
+	{"Left ADC", NULL, "Input Mux"},
+	{"Input Mux", "Mic", "Mic LNA"},
+	{"Input Mux", "Mic + Line R", "Mic LNA"},
+	{"Input Mux", "Line L", "Left PGA"},
+	{"Input Mux", "Line", "Left PGA"},
+
+	/* right input */
+	{"Right ADC", "Mic + Line R", "Right PGA"},
+	{"Right ADC", "Line", "Right PGA"},
+
+	/* inputs */
+	{"Mic LNA", NULL, "VINM"},
+	{"Left PGA", NULL, "VINL"},
+	{"Right PGA", NULL, "VINR"},
+};
+
+static int uda1380_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+				  ARRAY_SIZE(uda1380_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int iface;
+
+	/* set up DAI based upon fmt */
+	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+	iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
+
+	/* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= R01_SFORI_I2S | R01_SFORO_I2S;
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		iface |= R01_SFORI_MSB | R01_SFORO_I2S;
+	}
+
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
+		iface |= R01_SIM;
+
+	uda1380_write(codec, UDA1380_IFACE, iface);
+
+	return 0;
+}
+
+/*
+ * Flush reg cache
+ * We can only write the interpolator and decimator registers
+ * 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)
+{
+	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, reg_start, reg_end, clk;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		reg_start = UDA1380_MVOL;
+		reg_end = UDA1380_MIXER;
+	} else {
+		reg_start = UDA1380_DEC;
+		reg_end = UDA1380_AGC;
+	}
+
+	/* FIXME disable DAC_CLK */
+	clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+	uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
+
+	for (reg = reg_start; reg <= reg_end; reg++) {
+		pr_debug("uda1380: flush reg %x val %x:", reg,
+				uda1380_read_reg_cache(codec, reg));
+		uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
+	}
+
+	/* FIXME enable DAC_CLK */
+	uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
+
+	return 0;
+}
+
+static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+
+	/* set WSPLL power and divider if running from this clock */
+	if (clk & R00_DAC_CLK) {
+		int rate = params_rate(params);
+		u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+		clk &= ~0x3; /* clear SEL_LOOP_DIV */
+		switch (rate) {
+		case 6250 ... 12500:
+			clk |= 0x0;
+			break;
+		case 12501 ... 25000:
+			clk |= 0x1;
+			break;
+		case 25001 ... 50000:
+			clk |= 0x2;
+			break;
+		case 50001 ... 100000:
+			clk |= 0x3;
+			break;
+		}
+		uda1380_write(codec, UDA1380_PM, R02_PON_PLL | pm);
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		clk |= R00_EN_DAC | R00_EN_INT;
+	else
+		clk |= R00_EN_ADC | R00_EN_DEC;
+
+	uda1380_write(codec, UDA1380_CLK, clk);
+	return 0;
+}
+
+static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+
+	/* shut down WSPLL power if running from this clock */
+	if (clk & R00_DAC_CLK) {
+		u16 pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+		uda1380_write(codec, UDA1380_PM, ~R02_PON_PLL & pm);
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		clk &= ~(R00_EN_DAC | R00_EN_INT);
+	else
+		clk &= ~(R00_EN_ADC | R00_EN_DEC);
+
+	uda1380_write(codec, UDA1380_CLK, clk);
+}
+
+static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
+
+	/* FIXME: mute(codec,0) is called when the magician clock is already
+	 * set to WSPLL, but for some unknown reason writing to interpolator
+	 * registers works only when clocked by SYSCLK */
+	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+	uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
+	if (mute)
+		uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
+	else
+		uda1380_write(codec, UDA1380_DEEMP, mute_reg);
+	uda1380_write(codec, UDA1380_CLK, clk);
+	return 0;
+}
+
+static int uda1380_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
+		break;
+	case SND_SOC_BIAS_OFF:
+		uda1380_write(codec, UDA1380_PM, 0x0);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define UDA1380_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)
+
+struct snd_soc_dai uda1380_dai[] = {
+{
+	.name = "UDA1380",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA1380_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA1380_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = {
+		.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,
+	},
+},
+{ /* playback only - dual interface */
+	.name = "UDA1380",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA1380_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = {
+		.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,
+	},
+},
+{ /* capture only - dual interface*/
+	.name = "UDA1380",
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = UDA1380_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = {
+		.hw_params = uda1380_pcm_hw_params,
+		.shutdown = uda1380_pcm_shutdown,
+		.prepare = uda1380_pcm_prepare,
+	},
+	.dai_ops = {
+		.set_fmt = uda1380_set_dai_fmt,
+	},
+},
+};
+EXPORT_SYMBOL_GPL(uda1380_dai);
+
+static int uda1380_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;
+
+	uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int uda1380_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+	u8 data[2];
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) {
+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+		data[1] = cache[i] & 0x00ff;
+		codec->hw_write(codec->control_data, data, 2);
+	}
+	uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	uda1380_set_bias_level(codec, codec->suspend_bias_level);
+	return 0;
+}
+
+/*
+ * initialise the UDA1380 driver
+ * register mixer and dsp interfaces with the kernel
+ */
+static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	codec->name = "UDA1380";
+	codec->owner = THIS_MODULE;
+	codec->read = uda1380_read_reg_cache;
+	codec->write = uda1380_write;
+	codec->set_bias_level = uda1380_set_bias_level;
+	codec->dai = uda1380_dai;
+	codec->num_dai = ARRAY_SIZE(uda1380_dai);
+	codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg),
+				   GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+	codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
+	codec->reg_cache_step = 1;
+	uda1380_reset(codec);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		pr_err("uda1380: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	/* set clock input */
+	switch (dac_clk) {
+	case UDA1380_DAC_CLK_SYSCLK:
+		uda1380_write(codec, UDA1380_CLK, 0);
+		break;
+	case UDA1380_DAC_CLK_WSPLL:
+		uda1380_write(codec, UDA1380_CLK, R00_DAC_CLK);
+		break;
+	}
+
+	/* uda1380 init */
+	uda1380_add_controls(codec);
+	uda1380_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		pr_err("uda1380: 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 *uda1380_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+#define I2C_DRIVERID_UDA1380 0xfefe /* liam -  need a proper id */
+
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver uda1380_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 uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+	struct snd_soc_device *socdev = uda1380_socdev;
+	struct uda1380_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) {
+		pr_err("uda1380: failed to attach codec at addr %x\n", addr);
+		goto err;
+	}
+
+	ret = uda1380_init(socdev, setup->dac_clk);
+	if (ret < 0) {
+		pr_err("uda1380: failed to initialise UDA1380\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+
+static int uda1380_i2c_detach(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	i2c_detach_client(client);
+	kfree(codec->reg_cache);
+	kfree(client);
+	return 0;
+}
+
+static int uda1380_i2c_attach(struct i2c_adapter *adap)
+{
+	return i2c_probe(adap, &addr_data, uda1380_codec_probe);
+}
+
+static struct i2c_driver uda1380_i2c_driver = {
+	.driver = {
+		.name =  "UDA1380 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.id =             I2C_DRIVERID_UDA1380,
+	.attach_adapter = uda1380_i2c_attach,
+	.detach_client =  uda1380_i2c_detach,
+	.command =        NULL,
+};
+
+static struct i2c_client client_template = {
+	.name =   "UDA1380",
+	.driver = &uda1380_i2c_driver,
+};
+#endif
+
+static int uda1380_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct uda1380_setup_data *setup;
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
+
+	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);
+
+	uda1380_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(&uda1380_i2c_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add i2c driver");
+	}
+#else
+	/* Add other interfaces here */
+#endif
+	return ret;
+}
+
+/* power down chip */
+static int uda1380_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)
+		uda1380_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(&uda1380_i2c_driver);
+#endif
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_uda1380 = {
+	.probe = 	uda1380_probe,
+	.remove = 	uda1380_remove,
+	.suspend = 	uda1380_suspend,
+	.resume =	uda1380_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380);
+
+MODULE_AUTHOR("Giorgio Padrin");
+MODULE_DESCRIPTION("Audio support for codec Philips UDA1380");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h
new file mode 100644
index 000000000000..50c603e2c9f2
--- /dev/null
+++ b/sound/soc/codecs/uda1380.h
@@ -0,0 +1,89 @@
+/*
+ * Audio support for Philips UDA1380
+ *
+ * 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.
+ *
+ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org>
+ */
+
+#ifndef _UDA1380_H
+#define _UDA1380_H
+
+#define UDA1380_CLK	0x00
+#define UDA1380_IFACE	0x01
+#define UDA1380_PM	0x02
+#define UDA1380_AMIX	0x03
+#define UDA1380_HP	0x04
+#define UDA1380_MVOL	0x10
+#define UDA1380_MIXVOL	0x11
+#define UDA1380_MODE	0x12
+#define UDA1380_DEEMP	0x13
+#define UDA1380_MIXER	0x14
+#define UDA1380_INTSTAT	0x18
+#define UDA1380_DEC	0x20
+#define UDA1380_PGA	0x21
+#define UDA1380_ADC	0x22
+#define UDA1380_AGC	0x23
+#define UDA1380_DECSTAT	0x28
+#define UDA1380_RESET	0x7f
+
+#define UDA1380_CACHEREGNUM 0x24
+
+/* Register flags */
+#define R00_EN_ADC	0x0800
+#define R00_EN_DEC	0x0400
+#define R00_EN_DAC	0x0200
+#define R00_EN_INT	0x0100
+#define R00_DAC_CLK	0x0010
+#define R01_SFORI_I2S   0x0000
+#define R01_SFORI_LSB16 0x0100
+#define R01_SFORI_LSB18 0x0200
+#define R01_SFORI_LSB20 0x0300
+#define R01_SFORI_MSB   0x0500
+#define R01_SFORI_MASK  0x0700
+#define R01_SFORO_I2S   0x0000
+#define R01_SFORO_LSB16 0x0001
+#define R01_SFORO_LSB18 0x0002
+#define R01_SFORO_LSB20 0x0003
+#define R01_SFORO_LSB24 0x0004
+#define R01_SFORO_MSB   0x0005
+#define R01_SFORO_MASK  0x0007
+#define R01_SEL_SOURCE  0x0040
+#define R01_SIM		0x0010
+#define R02_PON_PLL	0x8000
+#define R02_PON_HP	0x2000
+#define R02_PON_DAC	0x0400
+#define R02_PON_BIAS	0x0100
+#define R02_EN_AVC	0x0080
+#define R02_PON_AVC	0x0040
+#define R02_PON_LNA	0x0010
+#define R02_PON_PGAL	0x0008
+#define R02_PON_ADCL	0x0004
+#define R02_PON_PGAR	0x0002
+#define R02_PON_ADCR	0x0001
+#define R13_MTM		0x4000
+#define R14_SILENCE	0x0080
+#define R14_SDET_ON	0x0040
+#define R21_MT_ADC	0x8000
+#define R22_SEL_LNA	0x0008
+#define R22_SEL_MIC	0x0004
+#define R22_SKIP_DCFIL	0x0002
+#define R23_AGC_EN	0x0001
+
+struct uda1380_setup_data {
+	unsigned short i2c_address;
+	int            dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL  1
+};
+
+#define UDA1380_DAI_DUPLEX	0 /* playback and capture on single DAI */
+#define UDA1380_DAI_PLAYBACK	1 /* playback DAI */
+#define UDA1380_DAI_CAPTURE	2 /* capture DAI */
+
+extern struct snd_soc_dai uda1380_dai[3];
+extern struct snd_soc_codec_device soc_codec_dev_uda1380;
+
+#endif /* _UDA1380_H */
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
new file mode 100644
index 000000000000..67325fd95447
--- /dev/null
+++ b/sound/soc/codecs/wm8510.c
@@ -0,0 +1,817 @@
+/*
+ * wm8510.c  --  WM8510 ALSA Soc Audio driver
+ *
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ *
+ * Author: 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 version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.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 "wm8510.h"
+
+#define AUDIO_NAME "wm8510"
+#define WM8510_VERSION "0.6"
+
+struct snd_soc_codec_device soc_codec_dev_wm8510;
+
+/*
+ * wm8510 register cache
+ * We can't read the WM8510 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0050, 0x0000, 0x0140, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x00ff,
+	0x0000, 0x0000, 0x0100, 0x00ff,
+	0x0000, 0x0000, 0x012c, 0x002c,
+	0x002c, 0x002c, 0x002c, 0x0000,
+	0x0032, 0x0000, 0x0000, 0x0000,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0038, 0x000b, 0x0032, 0x0000,
+	0x0008, 0x000c, 0x0093, 0x00e9,
+	0x0000, 0x0000, 0x0000, 0x0000,
+	0x0003, 0x0010, 0x0000, 0x0000,
+	0x0000, 0x0002, 0x0001, 0x0000,
+	0x0000, 0x0000, 0x0039, 0x0000,
+	0x0001,
+};
+
+/*
+ * read wm8510 register cache
+ */
+static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg == WM8510_RESET)
+		return 0;
+	if (reg >= WM8510_CACHEREGNUM)
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write wm8510 register cache
+ */
+static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec,
+	u16 reg, unsigned int value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= WM8510_CACHEREGNUM)
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the WM8510 register space
+ */
+static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[2];
+
+	/* data is
+	 *   D15..D9 WM8510 register offset
+	 *   D8...D0 register data
+	 */
+	data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+	data[1] = value & 0x00ff;
+
+	wm8510_write_reg_cache(codec, reg, value);
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+#define wm8510_reset(c)	wm8510_write(c, WM8510_RESET, 0)
+
+static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" };
+static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8510_alc[] = { "ALC", "Limiter" };
+
+static const struct soc_enum wm8510_enum[] = {
+	SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */
+	SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */
+	SOC_ENUM_SINGLE(WM8510_DAC,  4, 4, wm8510_deemp),
+	SOC_ENUM_SINGLE(WM8510_ALC3,  8, 2, wm8510_alc),
+};
+
+static const struct snd_kcontrol_new wm8510_snd_controls[] = {
+
+SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0),
+
+SOC_ENUM("DAC Companding", wm8510_enum[1]),
+SOC_ENUM("ADC Companding", wm8510_enum[0]),
+
+SOC_ENUM("Playback De-emphasis", wm8510_enum[2]),
+SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0),
+
+SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0),
+
+SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0),
+SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0),
+SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0),
+
+SOC_SINGLE("Capture Volume", WM8510_ADCVOL,  0, 127, 0),
+
+SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1,  8, 1, 0),
+SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1,  4, 15, 0),
+SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1,  0, 15, 0),
+
+SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2,  4, 7, 0),
+SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2,  0, 15, 0),
+
+SOC_SINGLE("ALC Enable Switch", WM8510_ALC1,  8, 1, 0),
+SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1,  3, 7, 0),
+SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1,  0, 7, 0),
+
+SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2,  8, 1, 0),
+SOC_SINGLE("ALC Capture Hold", WM8510_ALC2,  4, 7, 0),
+SOC_SINGLE("ALC Capture Target", WM8510_ALC2,  0, 15, 0),
+
+SOC_ENUM("ALC Capture Mode", wm8510_enum[3]),
+SOC_SINGLE("ALC Capture Decay", WM8510_ALC3,  4, 15, 0),
+SOC_SINGLE("ALC Capture Attack", WM8510_ALC3,  0, 15, 0),
+
+SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE,  3, 1, 0),
+SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE,  0, 7, 0),
+
+SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA,  7, 1, 0),
+SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA,  0, 63, 0),
+
+SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL,  7, 1, 0),
+SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL,  6, 1, 1),
+SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL,  0, 63, 0),
+SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0),
+
+SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST,  8, 1, 0),
+SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1),
+};
+
+/* add non dapm controls */
+static int wm8510_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&wm8510_snd_controls[i], codec,
+					NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+/* Speaker Output Mixer */
+static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0),
+};
+
+/* Mono Output Mixer */
+static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0),
+SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0),
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8510_boost_controls[] = {
+SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA,  6, 1, 0),
+SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0),
+SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0),
+};
+
+static const struct snd_kcontrol_new wm8510_micpga_controls[] = {
+SOC_DAPM_SINGLE("MICP Switch", WM8510_INPUT, 0, 1, 0),
+SOC_DAPM_SINGLE("MICN Switch", WM8510_INPUT, 1, 1, 0),
+SOC_DAPM_SINGLE("AUX Switch", WM8510_INPUT, 2, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0,
+	&wm8510_speaker_mixer_controls[0],
+	ARRAY_SIZE(wm8510_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0,
+	&wm8510_mono_mixer_controls[0],
+	ARRAY_SIZE(wm8510_mono_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0),
+SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0,
+		 &wm8510_micpga_controls[0],
+		 ARRAY_SIZE(wm8510_micpga_controls)),
+SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0,
+	&wm8510_boost_controls[0],
+	ARRAY_SIZE(wm8510_boost_controls)),
+
+SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0),
+
+SND_SOC_DAPM_INPUT("MICN"),
+SND_SOC_DAPM_INPUT("MICP"),
+SND_SOC_DAPM_INPUT("AUX"),
+SND_SOC_DAPM_OUTPUT("MONOOUT"),
+SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Mono output mixer */
+	{"Mono Mixer", "PCM Playback Switch", "DAC"},
+	{"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+	{"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+	/* Speaker output mixer */
+	{"Speaker Mixer", "PCM Playback Switch", "DAC"},
+	{"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+	{"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+	/* Outputs */
+	{"Mono Out", NULL, "Mono Mixer"},
+	{"MONOOUT", NULL, "Mono Out"},
+	{"SpkN Out", NULL, "Speaker Mixer"},
+	{"SpkP Out", NULL, "Speaker Mixer"},
+	{"SPKOUTN", NULL, "SpkN Out"},
+	{"SPKOUTP", NULL, "SpkP Out"},
+
+	/* Microphone PGA */
+	{"Mic PGA", "MICN Switch", "MICN"},
+	{"Mic PGA", "MICP Switch", "MICP"},
+	{ "Mic PGA", "AUX Switch", "Aux Input" },
+
+	/* Boost Mixer */
+	{"Boost Mixer", "Mic PGA Switch", "Mic PGA"},
+	{"Boost Mixer", "Mic Volume", "MICP"},
+	{"Boost Mixer", "Aux Volume", "Aux Input"},
+
+	{"ADC", NULL, "Boost Mixer"},
+};
+
+static int wm8510_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8510_dapm_widgets,
+				  ARRAY_SIZE(wm8510_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+struct pll_ {
+	unsigned int pre_div:4; /* prescale - 1 */
+	unsigned int n:4;
+	unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+
+static void pll_factors(unsigned int target, unsigned int source)
+{
+	unsigned long long Kpart;
+	unsigned int K, Ndiv, Nmod;
+
+	Ndiv = target / source;
+	if (Ndiv < 6) {
+		source >>= 1;
+		pll_div.pre_div = 1;
+		Ndiv = target / source;
+	} else
+		pll_div.pre_div = 0;
+
+	if ((Ndiv < 6) || (Ndiv > 12))
+		printk(KERN_WARNING
+			"WM8510 N value %d outwith recommended range!d\n",
+			Ndiv);
+
+	pll_div.n = Ndiv;
+	Nmod = target % source;
+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, source);
+
+	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;
+
+	pll_div.k = K;
+}
+
+static int wm8510_set_dai_pll(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;
+	u16 reg;
+
+	if (freq_in == 0 || freq_out == 0) {
+		/* Clock CODEC directly from MCLK */
+		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK);
+		wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff);
+
+		/* Turn off PLL */
+		reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
+		wm8510_write(codec, WM8510_POWER1, reg & 0x1df);
+		return 0;
+	}
+
+	pll_factors(freq_out*8, freq_in);
+
+	wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
+	wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
+	wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff);
+	wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff);
+	reg = wm8510_read_reg_cache(codec, WM8510_POWER1);
+	wm8510_write(codec, WM8510_POWER1, reg | 0x020);
+
+	/* Run CODEC from PLL instead of MCLK */
+	reg = wm8510_read_reg_cache(codec, WM8510_CLOCK);
+	wm8510_write(codec, WM8510_CLOCK, reg | 0x100);
+
+	return 0;
+}
+
+/*
+ * Configure WM8510 clock dividers.
+ */
+static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+		int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	switch (div_id) {
+	case WM8510_OPCLKDIV:
+		reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf;
+		wm8510_write(codec, WM8510_GPIO, reg | div);
+		break;
+	case WM8510_MCLKDIV:
+		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f;
+		wm8510_write(codec, WM8510_CLOCK, reg | div);
+		break;
+	case WM8510_ADCCLK:
+		reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7;
+		wm8510_write(codec, WM8510_ADC, reg | div);
+		break;
+	case WM8510_DACCLK:
+		reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7;
+		wm8510_write(codec, WM8510_DAC, reg | div);
+		break;
+	case WM8510_BCLKDIV:
+		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3;
+		wm8510_write(codec, WM8510_CLOCK, reg | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = 0;
+	u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		clk |= 0x0001;
+		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 |= 0x0010;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x0008;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x00018;
+		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 |= 0x0180;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= 0x0100;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= 0x0080;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8510_write(codec, WM8510_IFACE, iface);
+	wm8510_write(codec, WM8510_CLOCK, clk);
+	return 0;
+}
+
+static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f;
+	u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x0020;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x0040;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= 0x0060;
+		break;
+	}
+
+	/* filter coefficient */
+	switch (params_rate(params)) {
+	case SNDRV_PCM_RATE_8000:
+		adn |= 0x5 << 1;
+		break;
+	case SNDRV_PCM_RATE_11025:
+		adn |= 0x4 << 1;
+		break;
+	case SNDRV_PCM_RATE_16000:
+		adn |= 0x3 << 1;
+		break;
+	case SNDRV_PCM_RATE_22050:
+		adn |= 0x2 << 1;
+		break;
+	case SNDRV_PCM_RATE_32000:
+		adn |= 0x1 << 1;
+		break;
+	case SNDRV_PCM_RATE_44100:
+	case SNDRV_PCM_RATE_48000:
+		break;
+	}
+
+	wm8510_write(codec, WM8510_IFACE, iface);
+	wm8510_write(codec, WM8510_ADD, adn);
+	return 0;
+}
+
+static int wm8510_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf;
+
+	if (mute)
+		wm8510_write(codec, WM8510_DAC, mute_reg | 0x40);
+	else
+		wm8510_write(codec, WM8510_DAC, mute_reg);
+	return 0;
+}
+
+/* liam need to make this lower power with dapm */
+static int wm8510_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		wm8510_write(codec, WM8510_POWER1, 0x1ff);
+		wm8510_write(codec, WM8510_POWER2, 0x1ff);
+		wm8510_write(codec, WM8510_POWER3, 0x1ff);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* everything off, dac mute, inactive */
+		wm8510_write(codec, WM8510_POWER1, 0x0);
+		wm8510_write(codec, WM8510_POWER2, 0x0);
+		wm8510_write(codec, WM8510_POWER3, 0x0);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8510_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)
+
+#define WM8510_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 wm8510_dai = {
+	.name = "WM8510 HiFi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = WM8510_RATES,
+		.formats = WM8510_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = WM8510_RATES,
+		.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,
+		.set_pll = wm8510_set_dai_pll,
+	},
+};
+EXPORT_SYMBOL_GPL(wm8510_dai);
+
+static int wm8510_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;
+
+	wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8510_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+	u8 data[2];
+	u16 *cache = codec->reg_cache;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) {
+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+		data[1] = cache[i] & 0x00ff;
+		codec->hw_write(codec->control_data, data, 2);
+	}
+	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	wm8510_set_bias_level(codec, codec->suspend_bias_level);
+	return 0;
+}
+
+/*
+ * initialise the WM8510 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8510_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+
+	codec->name = "WM8510";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8510_read_reg_cache;
+	codec->write = wm8510_write;
+	codec->set_bias_level = wm8510_set_bias_level;
+	codec->dai = &wm8510_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(wm8510_reg);
+	codec->reg_cache = kmemdup(wm8510_reg, sizeof(wm8510_reg), GFP_KERNEL);
+
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	wm8510_reset(codec);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8510: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	wm8510_add_controls(codec);
+	wm8510_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8510: 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 *wm8510_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM8510 2 wire address is 0x1a
+ */
+#define I2C_DRIVERID_WM8510 0xfefe /* liam -  need a proper id */
+
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8510_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 wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+	struct snd_soc_device *socdev = wm8510_socdev;
+	struct wm8510_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) {
+		pr_err("failed to attach codec at addr %x\n", addr);
+		goto err;
+	}
+
+	ret = wm8510_init(socdev);
+	if (ret < 0) {
+		pr_err("failed to initialise WM8510\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+
+static int wm8510_i2c_detach(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	i2c_detach_client(client);
+	kfree(codec->reg_cache);
+	kfree(client);
+	return 0;
+}
+
+static int wm8510_i2c_attach(struct i2c_adapter *adap)
+{
+	return i2c_probe(adap, &addr_data, wm8510_codec_probe);
+}
+
+/* corgi i2c codec control layer */
+static struct i2c_driver wm8510_i2c_driver = {
+	.driver = {
+		.name = "WM8510 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.id =             I2C_DRIVERID_WM8510,
+	.attach_adapter = wm8510_i2c_attach,
+	.detach_client =  wm8510_i2c_detach,
+	.command =        NULL,
+};
+
+static struct i2c_client client_template = {
+	.name =   "WM8510",
+	.driver = &wm8510_i2c_driver,
+};
+#endif
+
+static int wm8510_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct wm8510_setup_data *setup;
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	pr_info("WM8510 Audio Codec %s", WM8510_VERSION);
+
+	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);
+
+	wm8510_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(&wm8510_i2c_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add i2c driver");
+	}
+#else
+	/* Add other interfaces here */
+#endif
+	return ret;
+}
+
+/* power down chip */
+static int wm8510_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)
+		wm8510_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(&wm8510_i2c_driver);
+#endif
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8510 = {
+	.probe = 	wm8510_probe,
+	.remove = 	wm8510_remove,
+	.suspend = 	wm8510_suspend,
+	.resume =	wm8510_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510);
+
+MODULE_DESCRIPTION("ASoC WM8510 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h
new file mode 100644
index 000000000000..f5d2e42eb3f4
--- /dev/null
+++ b/sound/soc/codecs/wm8510.h
@@ -0,0 +1,103 @@
+/*
+ * wm8510.h  --  WM8510 Soc Audio driver
+ *
+ * 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 _WM8510_H
+#define _WM8510_H
+
+/* WM8510 register space */
+
+#define WM8510_RESET		0x0
+#define WM8510_POWER1		0x1
+#define WM8510_POWER2		0x2
+#define WM8510_POWER3		0x3
+#define WM8510_IFACE		0x4
+#define WM8510_COMP			0x5
+#define WM8510_CLOCK		0x6
+#define WM8510_ADD			0x7
+#define WM8510_GPIO			0x8
+#define WM8510_DAC			0xa
+#define WM8510_DACVOL		0xb
+#define WM8510_ADC			0xe
+#define WM8510_ADCVOL		0xf
+#define WM8510_EQ1			0x12
+#define WM8510_EQ2			0x13
+#define WM8510_EQ3			0x14
+#define WM8510_EQ4			0x15
+#define WM8510_EQ5			0x16
+#define WM8510_DACLIM1		0x18
+#define WM8510_DACLIM2		0x19
+#define WM8510_NOTCH1		0x1b
+#define WM8510_NOTCH2		0x1c
+#define WM8510_NOTCH3		0x1d
+#define WM8510_NOTCH4		0x1e
+#define WM8510_ALC1			0x20
+#define WM8510_ALC2			0x21
+#define WM8510_ALC3			0x22
+#define WM8510_NGATE		0x23
+#define WM8510_PLLN			0x24
+#define WM8510_PLLK1		0x25
+#define WM8510_PLLK2		0x26
+#define WM8510_PLLK3		0x27
+#define WM8510_ATTEN		0x28
+#define WM8510_INPUT		0x2c
+#define WM8510_INPPGA		0x2d
+#define WM8510_ADCBOOST		0x2f
+#define WM8510_OUTPUT		0x31
+#define WM8510_SPKMIX		0x32
+#define WM8510_SPKVOL		0x36
+#define WM8510_MONOMIX		0x38
+
+#define WM8510_CACHEREGNUM 	57
+
+/* Clock divider Id's */
+#define WM8510_OPCLKDIV		0
+#define WM8510_MCLKDIV		1
+#define WM8510_ADCCLK		2
+#define WM8510_DACCLK		3
+#define WM8510_BCLKDIV		4
+
+/* DAC clock dividers */
+#define WM8510_DACCLK_F2	(1 << 3)
+#define WM8510_DACCLK_F4	(0 << 3)
+
+/* ADC clock dividers */
+#define WM8510_ADCCLK_F2	(1 << 3)
+#define WM8510_ADCCLK_F4	(0 << 3)
+
+/* PLL Out dividers */
+#define WM8510_OPCLKDIV_1	(0 << 4)
+#define WM8510_OPCLKDIV_2	(1 << 4)
+#define WM8510_OPCLKDIV_3	(2 << 4)
+#define WM8510_OPCLKDIV_4	(3 << 4)
+
+/* BCLK clock dividers */
+#define WM8510_BCLKDIV_1	(0 << 2)
+#define WM8510_BCLKDIV_2	(1 << 2)
+#define WM8510_BCLKDIV_4	(2 << 2)
+#define WM8510_BCLKDIV_8	(3 << 2)
+#define WM8510_BCLKDIV_16	(4 << 2)
+#define WM8510_BCLKDIV_32	(5 << 2)
+
+/* MCLK clock dividers */
+#define WM8510_MCLKDIV_1	(0 << 5)
+#define WM8510_MCLKDIV_1_5	(1 << 5)
+#define WM8510_MCLKDIV_2	(2 << 5)
+#define WM8510_MCLKDIV_3	(3 << 5)
+#define WM8510_MCLKDIV_4	(4 << 5)
+#define WM8510_MCLKDIV_6	(5 << 5)
+#define WM8510_MCLKDIV_8	(6 << 5)
+#define WM8510_MCLKDIV_12	(7 << 5)
+
+struct wm8510_setup_data {
+	unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8510_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8510;
+
+#endif
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 0cf9265fca8f..369d39c3f745 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -31,25 +31,6 @@
 #define AUDIO_NAME "wm8731"
 #define WM8731_VERSION "0.13"
 
-/*
- * Debug
- */
-
-#define WM8731_DEBUG 0
-
-#ifdef WM8731_DEBUG
-#define dbg(format, arg...) \
-	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
-	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
-	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
-	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
 struct snd_soc_codec_device soc_codec_dev_wm8731;
 
 /* codec private data */
@@ -193,7 +174,7 @@ SND_SOC_DAPM_INPUT("RLINEIN"),
 SND_SOC_DAPM_INPUT("LLINEIN"),
 };
 
-static const char *intercon[][3] = {
+static const struct snd_soc_dapm_route intercon[] = {
 	/* output mixer */
 	{"Output Mixer", "Line Bypass Switch", "Line Input"},
 	{"Output Mixer", "HiFi Playback Switch", "DAC"},
@@ -214,22 +195,14 @@ static const char *intercon[][3] = {
 	{"Line Input", NULL, "LLINEIN"},
 	{"Line Input", NULL, "RLINEIN"},
 	{"Mic Bias", NULL, "MICIN"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8731_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+				  ARRAY_SIZE(wm8731_dapm_widgets));
 
-	/* set up audio path interconnects */
-	for (i = 0; intercon[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, intercon[i][0],
-			intercon[i][1], intercon[i][2]);
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -345,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream)
 	}
 }
 
-static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8731_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7;
@@ -357,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8731_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;
@@ -376,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 }
 
 
-static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -435,29 +408,29 @@ static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int wm8731_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8731_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* vref/mid, osc on, dac unmute */
 		wm8731_write(codec, WM8731_PWR, reg);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* everything off except vref/vmid, */
 		wm8731_write(codec, WM8731_PWR, reg | 0x0040);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		/* everything off, dac mute, inactive */
 		wm8731_write(codec, WM8731_ACTIVE, 0x0);
 		wm8731_write(codec, WM8731_PWR, 0xffff);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -470,7 +443,7 @@ static int wm8731_dapm_event(struct snd_soc_codec *codec, int event)
 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-struct snd_soc_codec_dai wm8731_dai = {
+struct snd_soc_dai wm8731_dai = {
 	.name = "WM8731",
 	.playback = {
 		.stream_name = "Playback",
@@ -503,7 +476,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	wm8731_write(codec, WM8731_ACTIVE, 0x0);
-	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -521,8 +494,8 @@ static int wm8731_resume(struct platform_device *pdev)
 		data[1] = cache[i] & 0x00ff;
 		codec->hw_write(codec->control_data, data, 2);
 	}
-	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
-	wm8731_dapm_event(codec, codec->suspend_dapm_state);
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	wm8731_set_bias_level(codec, codec->suspend_bias_level);
 	return 0;
 }
 
@@ -539,10 +512,10 @@ static int wm8731_init(struct snd_soc_device *socdev)
 	codec->owner = THIS_MODULE;
 	codec->read = wm8731_read_reg_cache;
 	codec->write = wm8731_write;
-	codec->dapm_event = wm8731_dapm_event;
+	codec->set_bias_level = wm8731_set_bias_level;
 	codec->dai = &wm8731_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = sizeof(wm8731_reg);
+	codec->reg_cache_size = ARRAY_SIZE(wm8731_reg);
 	codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL);
 	if (codec->reg_cache == NULL)
 		return -ENOMEM;
@@ -557,7 +530,7 @@ static int wm8731_init(struct snd_soc_device *socdev)
 	}
 
 	/* power on device */
-	wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* set the update bits */
 	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
@@ -632,13 +605,13 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 
 	ret = i2c_attach_client(i2c);
 	if (ret < 0) {
-		err("failed to attach codec at addr %x\n", addr);
+		pr_err("failed to attach codec at addr %x\n", addr);
 		goto err;
 	}
 
 	ret = wm8731_init(socdev);
 	if (ret < 0) {
-		err("failed to initialise WM8731\n");
+		pr_err("failed to initialise WM8731\n");
 		goto err;
 	}
 	return ret;
@@ -689,7 +662,7 @@ static int wm8731_probe(struct platform_device *pdev)
 	struct wm8731_priv *wm8731;
 	int ret = 0;
 
-	info("WM8731 Audio Codec %s", WM8731_VERSION);
+	pr_info("WM8731 Audio Codec %s", WM8731_VERSION);
 
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
@@ -730,7 +703,7 @@ static int wm8731_remove(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	if (codec->control_data)
-		wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+		wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
index 5bcab6a7afb4..99f2e3c60e33 100644
--- a/sound/soc/codecs/wm8731.h
+++ b/sound/soc/codecs/wm8731.h
@@ -38,7 +38,7 @@ struct wm8731_setup_data {
 	unsigned short i2c_address;
 };
 
-extern struct snd_soc_codec_dai wm8731_dai;
+extern struct snd_soc_dai wm8731_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8731;
 
 #endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 16cd5d4d5ad9..e23cb09f0d14 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -31,25 +31,6 @@
 #define AUDIO_NAME "WM8750"
 #define WM8750_VERSION "0.12"
 
-/*
- * Debug
- */
-
-#define WM8750_DEBUG 0
-
-#ifdef WM8750_DEBUG
-#define dbg(format, arg...) \
-	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
-	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
-	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
-	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
 /* codec private data */
 struct wm8750_priv {
 	unsigned int sysclk;
@@ -378,7 +359,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("RINPUT3"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left mixer */
 	{"Left Mixer", "Playback Switch", "Left DAC"},
 	{"Left Mixer", "Left Bypass Switch", "Left Line Mux"},
@@ -470,22 +451,14 @@ static const char *audio_map[][3] = {
 	/* ADC */
 	{"Left ADC", NULL, "Left ADC Mux"},
 	{"Right ADC", NULL, "Right ADC Mux"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8750_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+				  ARRAY_SIZE(wm8750_dapm_widgets));
 
-	/* set up audio path audio_mapnects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -563,7 +536,7 @@ static inline int get_coeff(int mclk, int rate)
 	return -EINVAL;
 }
 
-static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8750_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;
@@ -581,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 	return -EINVAL;
 }
 
-static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -674,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8750_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7;
@@ -686,29 +659,29 @@ static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8750_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* set vmid to 50k and unmute dac */
 		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		/* set vmid to 5k for quick power up */
 		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* mute dac and set vmid to 500k, enable VREF */
 		wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		wm8750_write(codec, WM8750_PWR1, 0x0001);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -719,7 +692,7 @@ static int wm8750_dapm_event(struct snd_soc_codec *codec, int event)
 #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
-struct snd_soc_codec_dai wm8750_dai = {
+struct snd_soc_dai wm8750_dai = {
 	.name = "WM8750",
 	.playback = {
 		.stream_name = "Playback",
@@ -748,7 +721,7 @@ static void wm8750_work(struct work_struct *work)
 {
 	struct snd_soc_codec *codec =
 		container_of(work, struct snd_soc_codec, delayed_work.work);
-	wm8750_dapm_event(codec, codec->dapm_state);
+	wm8750_set_bias_level(codec, codec->bias_level);
 }
 
 static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
@@ -756,7 +729,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->codec;
 
-	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -777,12 +750,12 @@ static int wm8750_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge wm8750 caps */
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
-		wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
-		codec->dapm_state = SNDRV_CTL_POWER_D0;
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+		wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+		codec->bias_level = SND_SOC_BIAS_ON;
 		schedule_delayed_work(&codec->delayed_work,
 					msecs_to_jiffies(1000));
 	}
@@ -803,10 +776,10 @@ static int wm8750_init(struct snd_soc_device *socdev)
 	codec->owner = THIS_MODULE;
 	codec->read = wm8750_read_reg_cache;
 	codec->write = wm8750_write;
-	codec->dapm_event = wm8750_dapm_event;
+	codec->set_bias_level = wm8750_set_bias_level;
 	codec->dai = &wm8750_dai;
 	codec->num_dai = 1;
-	codec->reg_cache_size = sizeof(wm8750_reg);
+	codec->reg_cache_size = ARRAY_SIZE(wm8750_reg);
 	codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL);
 	if (codec->reg_cache == NULL)
 		return -ENOMEM;
@@ -821,8 +794,8 @@ static int wm8750_init(struct snd_soc_device *socdev)
 	}
 
 	/* charge output caps */
-	wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
-	codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+	wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
 
 	/* set the update bits */
@@ -904,13 +877,13 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 
 	ret = i2c_attach_client(i2c);
 	if (ret < 0) {
-		err("failed to attach codec at addr %x\n", addr);
+		pr_err("failed to attach codec at addr %x\n", addr);
 		goto err;
 	}
 
 	ret = wm8750_init(socdev);
 	if (ret < 0) {
-	err("failed to initialise WM8750\n");
+		pr_err("failed to initialise WM8750\n");
 		goto err;
 	}
 	return ret;
@@ -961,7 +934,7 @@ static int wm8750_probe(struct platform_device *pdev)
 	struct wm8750_priv *wm8750;
 	int ret = 0;
 
-	info("WM8750 Audio Codec %s", WM8750_VERSION);
+	pr_info("WM8750 Audio Codec %s", WM8750_VERSION);
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
@@ -1021,7 +994,7 @@ static int wm8750_remove(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	if (codec->control_data)
-		wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+		wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	run_delayed_work(&codec->delayed_work);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h
index a97a54a6348e..8ef30e628b21 100644
--- a/sound/soc/codecs/wm8750.h
+++ b/sound/soc/codecs/wm8750.h
@@ -61,7 +61,7 @@ struct wm8750_setup_data {
 	unsigned short i2c_address;
 };
 
-extern struct snd_soc_codec_dai wm8750_dai;
+extern struct snd_soc_dai wm8750_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8750;
 
 #endif
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index fb41826c4c4c..8604809f0c36 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -55,25 +55,6 @@
 #define AUDIO_NAME "wm8753"
 #define WM8753_VERSION "0.16"
 
-/*
- * Debug
- */
-
-#define WM8753_DEBUG 0
-
-#ifdef WM8753_DEBUG
-#define dbg(format, arg...) \
-	printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
-#else
-#define dbg(format, arg...) do {} while (0)
-#endif
-#define err(format, arg...) \
-	printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
-#define info(format, arg...) \
-	printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
-#define warn(format, arg...) \
-	printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
-
 static int caps_charge = 2000;
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
@@ -260,28 +241,50 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const unsigned int out_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	/* 0000000 - 0101111 = "Analogue mute" */
+	0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0),
+	48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0),
+};
+static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0);
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
 
 static const struct snd_kcontrol_new wm8753_snd_controls[] = {
-SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0),
-
-SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0),
-
-SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0),
-SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0),
-
-SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0),
-
-SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1),
-SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1),
-SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1),
-
-SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0),
-SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0),
-
-SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1),
-SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1),
-SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1),
+SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv),
+
+SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0,
+		 adc_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V,
+		 0, 127, 0, out_tlv),
+SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0,
+		 127, 0, out_tlv),
+
+SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv),
+
+SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7,
+		 1, mix_tlv),
+SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4,
+		 7, 1, mix_tlv),
+SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7,
+		 1, voice_mix_tlv),
+
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7,
+	     1, 0),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7,
+	     1, 0),
+
+SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv),
+SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1,
+	       mix_tlv),
+SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1,
+	       voice_mix_tlv),
 SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
 
 SOC_ENUM("Bass Boost", wm8753_enum[0]),
@@ -291,10 +294,13 @@ SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1),
 SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1),
 SOC_ENUM("Treble Cut-off", wm8753_enum[2]),
 
-SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1, rec_mix_tlv),
-SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1, rec_mix_tlv),
+SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1,
+	       rec_mix_tlv),
+SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1,
+	       rec_mix_tlv),
 
-SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0),
+SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0,
+		 pga_tlv),
 SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0),
 SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1),
 
@@ -326,8 +332,8 @@ SOC_ENUM("De-emphasis", wm8753_enum[8]),
 SOC_ENUM("Playback Mono Mix", wm8753_enum[9]),
 SOC_ENUM("Playback Phase", wm8753_enum[10]),
 
-SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0),
-SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0),
+SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv),
+SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv),
 
 SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
 
@@ -523,7 +529,7 @@ SND_SOC_DAPM_INPUT("MIC2"),
 SND_SOC_DAPM_VMID("VREF"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left mixer */
 	{"Left Mixer", "Left Playback Switch", "Left DAC"},
 	{"Left Mixer", "Voice Playback Switch", "Voice DAC"},
@@ -674,23 +680,14 @@ static const char *audio_map[][3] = {
 
 	/* ACOP */
 	{"ACOP", NULL, "ALC Mixer"},
-
-	/* terminator */
-	{NULL, NULL, NULL},
 };
 
 static int wm8753_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
+	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
 
-	for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
-
-	/* set up the WM8753 audio map */
-	for (i = 0; audio_map[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -743,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 	pll_div->k = K;
 }
 
-static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai,
 		int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
 	u16 reg, enable;
@@ -866,7 +863,7 @@ static int get_coeff(int mclk, int rate)
 /*
  * Clock after PLL and dividers
  */
-static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_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;
@@ -893,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai,
 /*
  * Set's ADC and Voice DAC format.
  */
-static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -963,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
 /*
  * Set's PCM dai fmt and BCLK.
  */
-static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1029,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1057,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
 /*
  * Set's HiFi DAC format.
  */
-static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1090,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 /*
  * Set's I2S DAI format.
  */
-static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1198,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1213,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
 }
 
-static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
@@ -1221,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
 }
 
-static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1236,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
 }
 
-static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1253,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
 }
 
-static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
+static int wm8753_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
 	u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7;
@@ -1274,29 +1271,29 @@ static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute)
 	return 0;
 }
 
-static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm8753_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* set vmid to 50k and unmute dac */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		/* set vmid to 5k for quick power up */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1);
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* mute dac and set vmid to 500k, enable VREF */
 		wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		wm8753_write(codec, WM8753_PWR1, 0x0001);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -1319,7 +1316,7 @@ static int wm8753_dapm_event(struct snd_soc_codec *codec, int event)
  * 3. Voice disabled - HIFI over HIFI
  * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
  */
-static const struct snd_soc_codec_dai wm8753_all_dai[] = {
+static const struct snd_soc_dai wm8753_all_dai[] = {
 /* DAI HiFi mode 1 */
 {	.name = "WM8753 HiFi",
 	.id = 1,
@@ -1459,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = {
 },
 };
 
-struct snd_soc_codec_dai wm8753_dai[2];
+struct snd_soc_dai wm8753_dai[2];
 EXPORT_SYMBOL_GPL(wm8753_dai);
 
 static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
@@ -1500,7 +1497,7 @@ static void wm8753_work(struct work_struct *work)
 {
 	struct snd_soc_codec *codec =
 		container_of(work, struct snd_soc_codec, delayed_work.work);
-	wm8753_dapm_event(codec, codec->dapm_state);
+	wm8753_set_bias_level(codec, codec->bias_level);
 }
 
 static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
@@ -1512,7 +1509,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 	if (!codec->card)
 		return 0;
 
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -1537,12 +1534,12 @@ static int wm8753_resume(struct platform_device *pdev)
 		codec->hw_write(codec->control_data, data, 2);
 	}
 
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge wm8753 caps */
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
-		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
-		codec->dapm_state = SNDRV_CTL_POWER_D0;
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
+		wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+		codec->bias_level = SND_SOC_BIAS_ON;
 		schedule_delayed_work(&codec->delayed_work,
 			msecs_to_jiffies(caps_charge));
 	}
@@ -1563,10 +1560,10 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	codec->owner = THIS_MODULE;
 	codec->read = wm8753_read_reg_cache;
 	codec->write = wm8753_write;
-	codec->dapm_event = wm8753_dapm_event;
+	codec->set_bias_level = wm8753_set_bias_level;
 	codec->dai = wm8753_dai;
 	codec->num_dai = 2;
-	codec->reg_cache_size = sizeof(wm8753_reg);
+	codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
 	codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL);
 
 	if (codec->reg_cache == NULL)
@@ -1584,8 +1581,8 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	}
 
 	/* charge output caps */
-	wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
-	codec->dapm_state = SNDRV_CTL_POWER_D3hot;
+	wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	schedule_delayed_work(&codec->delayed_work,
 		msecs_to_jiffies(caps_charge));
 
@@ -1673,13 +1670,13 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind)
 
 	ret = i2c_attach_client(i2c);
 	if (ret < 0) {
-		err("failed to attach codec at addr %x\n", addr);
+		pr_err("failed to attach codec at addr %x\n", addr);
 		goto err;
 	}
 
 	ret = wm8753_init(socdev);
 	if (ret < 0) {
-		err("failed to initialise WM8753\n");
+		pr_err("failed to initialise WM8753\n");
 		goto err;
 	}
 
@@ -1731,7 +1728,7 @@ static int wm8753_probe(struct platform_device *pdev)
 	struct wm8753_priv *wm8753;
 	int ret = 0;
 
-	info("WM8753 Audio Codec %s", WM8753_VERSION);
+	pr_info("WM8753 Audio Codec %s", WM8753_VERSION);
 
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
@@ -1792,7 +1789,7 @@ static int wm8753_remove(struct platform_device *pdev)
 	struct snd_soc_codec *codec = socdev->codec;
 
 	if (codec->control_data)
-		wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+		wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	run_delayed_work(&codec->delayed_work);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index 95e2a1f53169..44f5f1ff0cc7 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -120,7 +120,7 @@ struct wm8753_setup_data {
 #define WM8753_DAI_HIFI		0
 #define WM8753_DAI_VOICE		1
 
-extern struct snd_soc_codec_dai wm8753_dai[2];
+extern struct snd_soc_dai wm8753_dai[2];
 extern struct snd_soc_codec_device soc_codec_dev_wm8753;
 
 #endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
new file mode 100644
index 000000000000..3ecce5168e94
--- /dev/null
+++ b/sound/soc/codecs/wm8990.c
@@ -0,0 +1,1626 @@
+/*
+ * wm8990.c  --  WM8990 ALSA Soc Audio driver
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood
+ *         lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.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 <asm/div64.h>
+
+#include "wm8990.h"
+
+#define AUDIO_NAME "wm8990"
+#define WM8990_VERSION "0.2"
+
+/* codec private data */
+struct wm8990_priv {
+	unsigned int sysclk;
+	unsigned int pcmclk;
+};
+
+/*
+ * wm8990 register cache.  Note that register 0 is not included in the
+ * cache.
+ */
+static const u16 wm8990_reg[] = {
+	0x8990,     /* R0  - Reset */
+	0x0000,     /* R1  - Power Management (1) */
+	0x6000,     /* R2  - Power Management (2) */
+	0x0000,     /* R3  - Power Management (3) */
+	0x4050,     /* R4  - Audio Interface (1) */
+	0x4000,     /* R5  - Audio Interface (2) */
+	0x01C8,     /* R6  - Clocking (1) */
+	0x0000,     /* R7  - Clocking (2) */
+	0x0040,     /* R8  - Audio Interface (3) */
+	0x0040,     /* R9  - Audio Interface (4) */
+	0x0004,     /* R10 - DAC CTRL */
+	0x00C0,     /* R11 - Left DAC Digital Volume */
+	0x00C0,     /* R12 - Right DAC Digital Volume */
+	0x0000,     /* R13 - Digital Side Tone */
+	0x0100,     /* R14 - ADC CTRL */
+	0x00C0,     /* R15 - Left ADC Digital Volume */
+	0x00C0,     /* R16 - Right ADC Digital Volume */
+	0x0000,     /* R17 */
+	0x0000,     /* R18 - GPIO CTRL 1 */
+	0x1000,     /* R19 - GPIO1 & GPIO2 */
+	0x1010,     /* R20 - GPIO3 & GPIO4 */
+	0x1010,     /* R21 - GPIO5 & GPIO6 */
+	0x8000,     /* R22 - GPIOCTRL 2 */
+	0x0800,     /* R23 - GPIO_POL */
+	0x008B,     /* R24 - Left Line Input 1&2 Volume */
+	0x008B,     /* R25 - Left Line Input 3&4 Volume */
+	0x008B,     /* R26 - Right Line Input 1&2 Volume */
+	0x008B,     /* R27 - Right Line Input 3&4 Volume */
+	0x0000,     /* R28 - Left Output Volume */
+	0x0000,     /* R29 - Right Output Volume */
+	0x0066,     /* R30 - Line Outputs Volume */
+	0x0022,     /* R31 - Out3/4 Volume */
+	0x0079,     /* R32 - Left OPGA Volume */
+	0x0079,     /* R33 - Right OPGA Volume */
+	0x0003,     /* R34 - Speaker Volume */
+	0x0003,     /* R35 - ClassD1 */
+	0x0000,     /* R36 */
+	0x0100,     /* R37 - ClassD3 */
+	0x0000,     /* R38 */
+	0x0000,     /* R39 - Input Mixer1 */
+	0x0000,     /* R40 - Input Mixer2 */
+	0x0000,     /* R41 - Input Mixer3 */
+	0x0000,     /* R42 - Input Mixer4 */
+	0x0000,     /* R43 - Input Mixer5 */
+	0x0000,     /* R44 - Input Mixer6 */
+	0x0000,     /* R45 - Output Mixer1 */
+	0x0000,     /* R46 - Output Mixer2 */
+	0x0000,     /* R47 - Output Mixer3 */
+	0x0000,     /* R48 - Output Mixer4 */
+	0x0000,     /* R49 - Output Mixer5 */
+	0x0000,     /* R50 - Output Mixer6 */
+	0x0180,     /* R51 - Out3/4 Mixer */
+	0x0000,     /* R52 - Line Mixer1 */
+	0x0000,     /* R53 - Line Mixer2 */
+	0x0000,     /* R54 - Speaker Mixer */
+	0x0000,     /* R55 - Additional Control */
+	0x0000,     /* R56 - AntiPOP1 */
+	0x0000,     /* R57 - AntiPOP2 */
+	0x0000,     /* R58 - MICBIAS */
+	0x0000,     /* R59 */
+	0x0008,     /* R60 - PLL1 */
+	0x0031,     /* R61 - PLL2 */
+	0x0026,     /* R62 - PLL3 */
+};
+
+/*
+ * read wm8990 register cache
+ */
+static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec,
+	unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+	return cache[reg];
+}
+
+/*
+ * write wm8990 register cache
+ */
+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)
+		return;
+
+	cache[reg] = value;
+}
+
+/*
+ * write to the wm8990 register space
+ */
+static int wm8990_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	u8 data[3];
+
+	data[0] = reg & 0xFF;
+	data[1] = (value >> 8) & 0xFF;
+	data[2] = value & 0xFF;
+
+	wm8990_write_reg_cache(codec, reg, value);
+
+	if (codec->hw_write(codec->control_data, data, 3) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+#define wm8990_reset(c) wm8990_write(c, WM8990_RESET, 0)
+
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, 0, -2100);
+
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+
+static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int ret;
+	u16 val;
+
+	ret = snd_soc_put_volsw(kcontrol, ucontrol);
+	if (ret < 0)
+		return ret;
+
+	/* now hit the volume update bits (always bit 8) */
+	val = wm8990_read_reg_cache(codec, reg);
+	return wm8990_write(codec, reg, val | 0x0100);
+}
+
+#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\
+	 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, .put = wm899x_outpga_put_volsw_vu, \
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+
+static const char *wm8990_digital_sidetone[] =
+	{"None", "Left ADC", "Right ADC", "Reserved"};
+
+static const struct soc_enum wm8990_left_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
+	WM8990_ADC_TO_DACL_SHIFT,
+	WM8990_ADC_TO_DACL_MASK,
+	wm8990_digital_sidetone);
+
+static const struct soc_enum wm8990_right_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE,
+	WM8990_ADC_TO_DACR_SHIFT,
+	WM8990_ADC_TO_DACR_MASK,
+	wm8990_digital_sidetone);
+
+static const char *wm8990_adcmode[] =
+	{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
+
+static const struct soc_enum wm8990_right_adcmode_enum =
+SOC_ENUM_SINGLE(WM8990_ADC_CTRL,
+	WM8990_ADC_HPF_CUT_SHIFT,
+	WM8990_ADC_HPF_CUT_MASK,
+	wm8990_adcmode);
+
+static const struct snd_kcontrol_new wm8990_snd_controls[] = {
+/* INMIXL */
+SOC_SINGLE("LIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L12MNBST_BIT, 1, 0),
+SOC_SINGLE("LIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_L34MNBST_BIT, 1, 0),
+/* INMIXR */
+SOC_SINGLE("RIN12 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R12MNBST_BIT, 1, 0),
+SOC_SINGLE("RIN34 PGA Boost", WM8990_INPUT_MIXER3, WM8990_R34MNBST_BIT, 1, 0),
+
+/* LOMIX */
+SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER3,
+	WM8990_LLI3LOVOL_SHIFT, WM8990_LLI3LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3,
+	WM8990_LR12LOVOL_SHIFT, WM8990_LR12LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER3,
+	WM8990_LL12LOVOL_SHIFT, WM8990_LL12LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER5,
+	WM8990_LRI3LOVOL_SHIFT, WM8990_LRI3LOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER5,
+	WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER5,
+	WM8990_LRBLOVOL_SHIFT, WM8990_LRBLOVOL_MASK, 1, out_mix_tlv),
+
+/* ROMIX */
+SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8990_OUTPUT_MIXER4,
+	WM8990_RRI3ROVOL_SHIFT, WM8990_RRI3ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4,
+	WM8990_RL12ROVOL_SHIFT, WM8990_RL12ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8990_OUTPUT_MIXER4,
+	WM8990_RR12ROVOL_SHIFT, WM8990_RR12ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8990_OUTPUT_MIXER6,
+	WM8990_RLI3ROVOL_SHIFT, WM8990_RLI3ROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8990_OUTPUT_MIXER6,
+	WM8990_RLBROVOL_SHIFT, WM8990_RLBROVOL_MASK, 1, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8990_OUTPUT_MIXER6,
+	WM8990_RRBROVOL_SHIFT, WM8990_RRBROVOL_MASK, 1, out_mix_tlv),
+
+/* LOUT */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8990_LEFT_OUTPUT_VOLUME,
+	WM8990_LOUTVOL_SHIFT, WM8990_LOUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOUT ZC", WM8990_LEFT_OUTPUT_VOLUME, WM8990_LOZC_BIT, 1, 0),
+
+/* ROUT */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8990_RIGHT_OUTPUT_VOLUME,
+	WM8990_ROUTVOL_SHIFT, WM8990_ROUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROUT ZC", WM8990_RIGHT_OUTPUT_VOLUME, WM8990_ROZC_BIT, 1, 0),
+
+/* LOPGA */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8990_LEFT_OPGA_VOLUME,
+	WM8990_LOPGAVOL_SHIFT, WM8990_LOPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOPGA ZC Switch", WM8990_LEFT_OPGA_VOLUME,
+	WM8990_LOPGAZC_BIT, 1, 0),
+
+/* ROPGA */
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8990_RIGHT_OPGA_VOLUME,
+	WM8990_ROPGAVOL_SHIFT, WM8990_ROPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROPGA ZC Switch", WM8990_RIGHT_OPGA_VOLUME,
+	WM8990_ROPGAZC_BIT, 1, 0),
+
+SOC_SINGLE("LON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+	WM8990_LONMUTE_BIT, 1, 0),
+SOC_SINGLE("LOP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+	WM8990_LOPMUTE_BIT, 1, 0),
+SOC_SINGLE("LOP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME,
+	WM8990_LOATTN_BIT, 1, 0),
+SOC_SINGLE("RON Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+	WM8990_RONMUTE_BIT, 1, 0),
+SOC_SINGLE("ROP Mute Switch", WM8990_LINE_OUTPUTS_VOLUME,
+	WM8990_ROPMUTE_BIT, 1, 0),
+SOC_SINGLE("ROP Attenuation Switch", WM8990_LINE_OUTPUTS_VOLUME,
+	WM8990_ROATTN_BIT, 1, 0),
+
+SOC_SINGLE("OUT3 Mute Switch", WM8990_OUT3_4_VOLUME,
+	WM8990_OUT3MUTE_BIT, 1, 0),
+SOC_SINGLE("OUT3 Attenuation Switch", WM8990_OUT3_4_VOLUME,
+	WM8990_OUT3ATTN_BIT, 1, 0),
+
+SOC_SINGLE("OUT4 Mute Switch", WM8990_OUT3_4_VOLUME,
+	WM8990_OUT4MUTE_BIT, 1, 0),
+SOC_SINGLE("OUT4 Attenuation Switch", WM8990_OUT3_4_VOLUME,
+	WM8990_OUT4ATTN_BIT, 1, 0),
+
+SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1,
+	WM8990_CDMODE_BIT, 1, 0),
+
+SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME,
+	WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0),
+SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3,
+	WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0),
+SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3,
+	WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
+	WM8990_LEFT_DAC_DIGITAL_VOLUME,
+	WM8990_DACL_VOL_SHIFT,
+	WM8990_DACL_VOL_MASK,
+	0,
+	out_dac_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume",
+	WM8990_RIGHT_DAC_DIGITAL_VOLUME,
+	WM8990_DACR_VOL_SHIFT,
+	WM8990_DACR_VOL_MASK,
+	0,
+	out_dac_tlv),
+
+SOC_ENUM("Left Digital Sidetone", wm8990_left_digital_sidetone_enum),
+SOC_ENUM("Right Digital Sidetone", wm8990_right_digital_sidetone_enum),
+
+SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE,
+	WM8990_ADCL_DAC_SVOL_SHIFT, WM8990_ADCL_DAC_SVOL_MASK, 0,
+	out_sidetone_tlv),
+SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8990_DIGITAL_SIDE_TONE,
+	WM8990_ADCR_DAC_SVOL_SHIFT, WM8990_ADCR_DAC_SVOL_MASK, 0,
+	out_sidetone_tlv),
+
+SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8990_ADC_CTRL,
+	WM8990_ADC_HPF_ENA_BIT, 1, 0),
+
+SOC_ENUM("ADC HPF Mode", wm8990_right_adcmode_enum),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume",
+	WM8990_LEFT_ADC_DIGITAL_VOLUME,
+	WM8990_ADCL_VOL_SHIFT,
+	WM8990_ADCL_VOL_MASK,
+	0,
+	in_adc_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume",
+	WM8990_RIGHT_ADC_DIGITAL_VOLUME,
+	WM8990_ADCR_VOL_SHIFT,
+	WM8990_ADCR_VOL_MASK,
+	0,
+	in_adc_tlv),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume",
+	WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8990_LIN12VOL_SHIFT,
+	WM8990_LIN12VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("LIN12 ZC Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8990_LI12ZC_BIT, 1, 0),
+
+SOC_SINGLE("LIN12 Mute Switch", WM8990_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8990_LI12MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume",
+	WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8990_LIN34VOL_SHIFT,
+	WM8990_LIN34VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("LIN34 ZC Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8990_LI34ZC_BIT, 1, 0),
+
+SOC_SINGLE("LIN34 Mute Switch", WM8990_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8990_LI34MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume",
+	WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8990_RIN12VOL_SHIFT,
+	WM8990_RIN12VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("RIN12 ZC Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8990_RI12ZC_BIT, 1, 0),
+
+SOC_SINGLE("RIN12 Mute Switch", WM8990_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8990_RI12MUTE_BIT, 1, 0),
+
+SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume",
+	WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8990_RIN34VOL_SHIFT,
+	WM8990_RIN34VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("RIN34 ZC Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8990_RI34ZC_BIT, 1, 0),
+
+SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8990_RI34MUTE_BIT, 1, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8990_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				snd_soc_cnew(&wm8990_snd_controls[i], codec,
+					NULL));
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+/*
+ * _DAPM_ Controls
+ */
+
+static int inmixer_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	u16 reg, fakepower;
+
+	reg = wm8990_read_reg_cache(w->codec, WM8990_POWER_MANAGEMENT_2);
+	fakepower = wm8990_read_reg_cache(w->codec, WM8990_INTDRIVBITS);
+
+	if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) |
+		(1 << WM8990_AINLMUX_PWR_BIT))) {
+		reg |= WM8990_AINL_ENA;
+	} else {
+		reg &= ~WM8990_AINL_ENA;
+	}
+
+	if (fakepower & ((1 << WM8990_INMIXR_PWR_BIT) |
+		(1 << WM8990_AINRMUX_PWR_BIT))) {
+		reg |= WM8990_AINR_ENA;
+	} else {
+		reg &= ~WM8990_AINL_ENA;
+	}
+	wm8990_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg);
+
+	return 0;
+}
+
+static int outmixer_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	u32 reg_shift = kcontrol->private_value & 0xfff;
+	int ret = 0;
+	u16 reg;
+
+	switch (reg_shift) {
+	case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) :
+		reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER1);
+		if (reg & WM8990_LDLO) {
+			printk(KERN_WARNING
+			"Cannot set as Output Mixer 1 LDLO Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8):
+		reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER2);
+		if (reg & WM8990_RDRO) {
+			printk(KERN_WARNING
+			"Cannot set as Output Mixer 2 RDRO Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8):
+		reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER);
+		if (reg & WM8990_LDSPK) {
+			printk(KERN_WARNING
+			"Cannot set as Speaker Mixer LDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8):
+		reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER);
+		if (reg & WM8990_RDSPK) {
+			printk(KERN_WARNING
+			"Cannot set as Speaker Mixer RDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	}
+
+	return ret;
+}
+
+/* INMIX dB values */
+static const unsigned int in_mix_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0, 7, TLV_DB_LINEAR_ITEM(-1200, 600),
+};
+
+/* Left In PGA Connections */
+static const struct snd_kcontrol_new wm8990_dapm_lin12_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN1 Switch", WM8990_INPUT_MIXER2, WM8990_LMN1_BIT, 1, 0),
+SOC_DAPM_SINGLE("LIN2 Switch", WM8990_INPUT_MIXER2, WM8990_LMP2_BIT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8990_dapm_lin34_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN3 Switch", WM8990_INPUT_MIXER2, WM8990_LMN3_BIT, 1, 0),
+SOC_DAPM_SINGLE("LIN4 Switch", WM8990_INPUT_MIXER2, WM8990_LMP4_BIT, 1, 0),
+};
+
+/* Right In PGA Connections */
+static const struct snd_kcontrol_new wm8990_dapm_rin12_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN1 Switch", WM8990_INPUT_MIXER2, WM8990_RMN1_BIT, 1, 0),
+SOC_DAPM_SINGLE("RIN2 Switch", WM8990_INPUT_MIXER2, WM8990_RMP2_BIT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8990_dapm_rin34_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN3 Switch", WM8990_INPUT_MIXER2, WM8990_RMN3_BIT, 1, 0),
+SOC_DAPM_SINGLE("RIN4 Switch", WM8990_INPUT_MIXER2, WM8990_RMP4_BIT, 1, 0),
+};
+
+/* INMIXL */
+static const struct snd_kcontrol_new wm8990_dapm_inmixl_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8990_INPUT_MIXER3,
+	WM8990_LDBVOL_SHIFT, WM8990_LDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8990_INPUT_MIXER5, WM8990_LI2BVOL_SHIFT,
+	7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("LINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT,
+	1, 0),
+SOC_DAPM_SINGLE("LINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT,
+	1, 0),
+};
+
+/* INMIXR */
+static const struct snd_kcontrol_new wm8990_dapm_inmixr_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8990_INPUT_MIXER4,
+	WM8990_RDBVOL_SHIFT, WM8990_RDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8990_INPUT_MIXER6, WM8990_RI2BVOL_SHIFT,
+	7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("RINPGA12 Switch", WM8990_INPUT_MIXER3, WM8990_L12MNB_BIT,
+	1, 0),
+SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT,
+	1, 0),
+};
+
+/* AINLMUX */
+static const char *wm8990_ainlmux[] =
+	{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
+
+static const struct soc_enum wm8990_ainlmux_enum =
+SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT,
+	ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux);
+
+static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls =
+SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum);
+
+/* DIFFINL */
+
+/* AINRMUX */
+static const char *wm8990_ainrmux[] =
+	{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
+
+static const struct soc_enum wm8990_ainrmux_enum =
+SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT,
+	ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux);
+
+static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls =
+SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum);
+
+/* RXVOICE */
+static const struct snd_kcontrol_new wm8990_dapm_rxvoice_controls[] = {
+SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8990_INPUT_MIXER5, WM8990_LR4BVOL_SHIFT,
+			WM8990_LR4BVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8990_INPUT_MIXER6, WM8990_RL4BVOL_SHIFT,
+			WM8990_RL4BVOL_MASK, 0, in_mix_tlv),
+};
+
+/* LOMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lomix_controls[] = {
+SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER1,
+	WM8990_LRBLO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER1,
+	WM8990_LLBLO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER1,
+	WM8990_LRI3LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER1,
+	WM8990_LLI3LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1,
+	WM8990_LR12LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER1,
+	WM8990_LL12LO_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8990_OUTPUT_MIXER1,
+	WM8990_LDLO_BIT, 1, 0),
+};
+
+/* ROMIX */
+static const struct snd_kcontrol_new wm8990_dapm_romix_controls[] = {
+SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8990_OUTPUT_MIXER2,
+	WM8990_RLBRO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8990_OUTPUT_MIXER2,
+	WM8990_RRBRO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8990_OUTPUT_MIXER2,
+	WM8990_RLI3RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8990_OUTPUT_MIXER2,
+	WM8990_RRI3RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2,
+	WM8990_RL12RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8990_OUTPUT_MIXER2,
+	WM8990_RR12RO_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8990_OUTPUT_MIXER2,
+	WM8990_RDRO_BIT, 1, 0),
+};
+
+/* LONMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lonmix_controls[] = {
+SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1,
+	WM8990_LLOPGALON_BIT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER1,
+	WM8990_LROPGALON_BIT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8990_LINE_MIXER1,
+	WM8990_LOPLON_BIT, 1, 0),
+};
+
+/* LOPMIX */
+static const struct snd_kcontrol_new wm8990_dapm_lopmix_controls[] = {
+SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER1,
+	WM8990_LR12LOP_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER1,
+	WM8990_LL12LOP_BIT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8990_LINE_MIXER1,
+	WM8990_LLOPGALOP_BIT, 1, 0),
+};
+
+/* RONMIX */
+static const struct snd_kcontrol_new wm8990_dapm_ronmix_controls[] = {
+SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2,
+	WM8990_RROPGARON_BIT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8990_LINE_MIXER2,
+	WM8990_RLOPGARON_BIT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8990_LINE_MIXER2,
+	WM8990_ROPRON_BIT, 1, 0),
+};
+
+/* ROPMIX */
+static const struct snd_kcontrol_new wm8990_dapm_ropmix_controls[] = {
+SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8990_LINE_MIXER2,
+	WM8990_RL12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8990_LINE_MIXER2,
+	WM8990_RR12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8990_LINE_MIXER2,
+	WM8990_RROPGAROP_BIT, 1, 0),
+};
+
+/* OUT3MIX */
+static const struct snd_kcontrol_new wm8990_dapm_out3mix_controls[] = {
+SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER,
+	WM8990_LI4O3_BIT, 1, 0),
+SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8990_OUT3_4_MIXER,
+	WM8990_LPGAO3_BIT, 1, 0),
+};
+
+/* OUT4MIX */
+static const struct snd_kcontrol_new wm8990_dapm_out4mix_controls[] = {
+SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8990_OUT3_4_MIXER,
+	WM8990_RPGAO4_BIT, 1, 0),
+SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8990_OUT3_4_MIXER,
+	WM8990_RI4O4_BIT, 1, 0),
+};
+
+/* SPKMIX */
+static const struct snd_kcontrol_new wm8990_dapm_spkmix_controls[] = {
+SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8990_SPEAKER_MIXER,
+	WM8990_LI2SPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8990_SPEAKER_MIXER,
+	WM8990_LB2SPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8990_SPEAKER_MIXER,
+	WM8990_LOPGASPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8990_SPEAKER_MIXER,
+	WM8990_LDSPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8990_SPEAKER_MIXER,
+	WM8990_RDSPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8990_SPEAKER_MIXER,
+	WM8990_ROPGASPK_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8990_SPEAKER_MIXER,
+	WM8990_RL12ROP_BIT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8990_SPEAKER_MIXER,
+	WM8990_RI2SPK_BIT, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8990_dapm_widgets[] = {
+/* Input Side */
+/* Input Lines */
+SND_SOC_DAPM_INPUT("LIN1"),
+SND_SOC_DAPM_INPUT("LIN2"),
+SND_SOC_DAPM_INPUT("LIN3"),
+SND_SOC_DAPM_INPUT("LIN4/RXN"),
+SND_SOC_DAPM_INPUT("RIN3"),
+SND_SOC_DAPM_INPUT("RIN4/RXP"),
+SND_SOC_DAPM_INPUT("RIN1"),
+SND_SOC_DAPM_INPUT("RIN2"),
+SND_SOC_DAPM_INPUT("Internal ADC Source"),
+
+/* DACs */
+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8990_POWER_MANAGEMENT_2,
+	WM8990_ADCL_ENA_BIT, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8990_POWER_MANAGEMENT_2,
+	WM8990_ADCR_ENA_BIT, 0),
+
+/* Input PGAs */
+SND_SOC_DAPM_MIXER("LIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN12_ENA_BIT,
+	0, &wm8990_dapm_lin12_pga_controls[0],
+	ARRAY_SIZE(wm8990_dapm_lin12_pga_controls)),
+SND_SOC_DAPM_MIXER("LIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_LIN34_ENA_BIT,
+	0, &wm8990_dapm_lin34_pga_controls[0],
+	ARRAY_SIZE(wm8990_dapm_lin34_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN12 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN12_ENA_BIT,
+	0, &wm8990_dapm_rin12_pga_controls[0],
+	ARRAY_SIZE(wm8990_dapm_rin12_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN34 PGA", WM8990_POWER_MANAGEMENT_2, WM8990_RIN34_ENA_BIT,
+	0, &wm8990_dapm_rin34_pga_controls[0],
+	ARRAY_SIZE(wm8990_dapm_rin34_pga_controls)),
+
+/* INMIXL */
+SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
+	&wm8990_dapm_inmixl_controls[0],
+	ARRAY_SIZE(wm8990_dapm_inmixl_controls),
+	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINLMUX */
+SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+	&wm8990_dapm_ainlmux_controls, inmixer_event,
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* INMIXR */
+SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
+	&wm8990_dapm_inmixr_controls[0],
+	ARRAY_SIZE(wm8990_dapm_inmixr_controls),
+	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINRMUX */
+SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+	&wm8990_dapm_ainrmux_controls, inmixer_event,
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* Output Side */
+/* DACs */
+SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8990_POWER_MANAGEMENT_3,
+	WM8990_DACL_ENA_BIT, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8990_POWER_MANAGEMENT_3,
+	WM8990_DACR_ENA_BIT, 0),
+
+/* LOMIX */
+SND_SOC_DAPM_MIXER_E("LOMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOMIX_ENA_BIT,
+	0, &wm8990_dapm_lomix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_lomix_controls),
+	outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LONMIX */
+SND_SOC_DAPM_MIXER("LONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LON_ENA_BIT, 0,
+	&wm8990_dapm_lonmix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_lonmix_controls)),
+
+/* LOPMIX */
+SND_SOC_DAPM_MIXER("LOPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_LOP_ENA_BIT, 0,
+	&wm8990_dapm_lopmix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_lopmix_controls)),
+
+/* OUT3MIX */
+SND_SOC_DAPM_MIXER("OUT3MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT3_ENA_BIT, 0,
+	&wm8990_dapm_out3mix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_out3mix_controls)),
+
+/* SPKMIX */
+SND_SOC_DAPM_MIXER_E("SPKMIX", WM8990_POWER_MANAGEMENT_1, WM8990_SPK_ENA_BIT, 0,
+	&wm8990_dapm_spkmix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_spkmix_controls), outmixer_event,
+	SND_SOC_DAPM_PRE_REG),
+
+/* OUT4MIX */
+SND_SOC_DAPM_MIXER("OUT4MIX", WM8990_POWER_MANAGEMENT_1, WM8990_OUT4_ENA_BIT, 0,
+	&wm8990_dapm_out4mix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_out4mix_controls)),
+
+/* ROPMIX */
+SND_SOC_DAPM_MIXER("ROPMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROP_ENA_BIT, 0,
+	&wm8990_dapm_ropmix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_ropmix_controls)),
+
+/* RONMIX */
+SND_SOC_DAPM_MIXER("RONMIX", WM8990_POWER_MANAGEMENT_3, WM8990_RON_ENA_BIT, 0,
+	&wm8990_dapm_ronmix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_ronmix_controls)),
+
+/* ROMIX */
+SND_SOC_DAPM_MIXER_E("ROMIX", WM8990_POWER_MANAGEMENT_3, WM8990_ROMIX_ENA_BIT,
+	0, &wm8990_dapm_romix_controls[0],
+	ARRAY_SIZE(wm8990_dapm_romix_controls),
+	outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LOUT PGA */
+SND_SOC_DAPM_PGA("LOUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_LOUT_ENA_BIT, 0,
+	NULL, 0),
+
+/* ROUT PGA */
+SND_SOC_DAPM_PGA("ROUT PGA", WM8990_POWER_MANAGEMENT_1, WM8990_ROUT_ENA_BIT, 0,
+	NULL, 0),
+
+/* LOPGA */
+SND_SOC_DAPM_PGA("LOPGA", WM8990_POWER_MANAGEMENT_3, WM8990_LOPGA_ENA_BIT, 0,
+	NULL, 0),
+
+/* ROPGA */
+SND_SOC_DAPM_PGA("ROPGA", WM8990_POWER_MANAGEMENT_3, WM8990_ROPGA_ENA_BIT, 0,
+	NULL, 0),
+
+/* MICBIAS */
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8990_POWER_MANAGEMENT_1,
+	WM8990_MICBIAS_ENA_BIT, 0),
+
+SND_SOC_DAPM_OUTPUT("LON"),
+SND_SOC_DAPM_OUTPUT("LOP"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_OUTPUT("ROP"),
+SND_SOC_DAPM_OUTPUT("RON"),
+
+SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Make DACs turn on when playing even if not mixed into any outputs */
+	{"Internal DAC Sink", NULL, "Left DAC"},
+	{"Internal DAC Sink", NULL, "Right DAC"},
+
+	/* Make ADCs turn on when recording even if not mixed from any inputs */
+	{"Left ADC", NULL, "Internal ADC Source"},
+	{"Right ADC", NULL, "Internal ADC Source"},
+
+	/* Input Side */
+	/* LIN12 PGA */
+	{"LIN12 PGA", "LIN1 Switch", "LIN1"},
+	{"LIN12 PGA", "LIN2 Switch", "LIN2"},
+	/* LIN34 PGA */
+	{"LIN34 PGA", "LIN3 Switch", "LIN3"},
+	{"LIN34 PGA", "LIN4 Switch", "LIN4"},
+	/* INMIXL */
+	{"INMIXL", "Record Left Volume", "LOMIX"},
+	{"INMIXL", "LIN2 Volume", "LIN2"},
+	{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
+	{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
+	/* AILNMUX */
+	{"AILNMUX", "INMIXL Mix", "INMIXL"},
+	{"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
+	{"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
+	{"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* ADC */
+	{"Left ADC", NULL, "AILNMUX"},
+
+	/* RIN12 PGA */
+	{"RIN12 PGA", "RIN1 Switch", "RIN1"},
+	{"RIN12 PGA", "RIN2 Switch", "RIN2"},
+	/* RIN34 PGA */
+	{"RIN34 PGA", "RIN3 Switch", "RIN3"},
+	{"RIN34 PGA", "RIN4 Switch", "RIN4"},
+	/* INMIXL */
+	{"INMIXR", "Record Right Volume", "ROMIX"},
+	{"INMIXR", "RIN2 Volume", "RIN2"},
+	{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
+	{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
+	/* AIRNMUX */
+	{"AIRNMUX", "INMIXR Mix", "INMIXR"},
+	{"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
+	{"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
+	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
+	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* ADC */
+	{"Right ADC", NULL, "AIRNMUX"},
+
+	/* LOMIX */
+	{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
+	{"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"},
+	{"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"},
+	{"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"},
+	{"LOMIX", "LOMIX Left DAC Switch", "Left DAC"},
+
+	/* ROMIX */
+	{"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"},
+	{"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"},
+	{"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"},
+	{"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"},
+	{"ROMIX", "ROMIX Right DAC Switch", "Right DAC"},
+
+	/* SPKMIX */
+	{"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"},
+	{"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"},
+	{"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"},
+	{"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"},
+	{"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
+	{"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
+	{"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
+	{"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+
+	/* LONMIX */
+	{"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"},
+
+	/* LOPMIX */
+	{"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
+
+	/* OUT3MIX */
+	{"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+	{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
+
+	/* OUT4MIX */
+	{"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"},
+	{"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"},
+
+	/* RONMIX */
+	{"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"},
+
+	/* ROPMIX */
+	{"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"},
+
+	/* Out Mixer PGAs */
+	{"LOPGA", NULL, "LOMIX"},
+	{"ROPGA", NULL, "ROMIX"},
+
+	{"LOUT PGA", NULL, "LOMIX"},
+	{"ROUT PGA", NULL, "ROMIX"},
+
+	/* Output Pins */
+	{"LON", NULL, "LONMIX"},
+	{"LOP", NULL, "LOPMIX"},
+	{"OUT", NULL, "OUT3MIX"},
+	{"LOUT", NULL, "LOUT PGA"},
+	{"SPKN", NULL, "SPKMIX"},
+	{"ROUT", NULL, "ROUT PGA"},
+	{"OUT4", NULL, "OUT4MIX"},
+	{"ROP", NULL, "ROPMIX"},
+	{"RON", NULL, "RONMIX"},
+};
+
+static int wm8990_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8990_dapm_widgets,
+				  ARRAY_SIZE(wm8990_dapm_widgets));
+
+	/* set up the WM8990 audio map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+	u32 div2;
+	u32 n;
+	u32 k;
+};
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 16) * 10)
+
+static void pll_factors(struct _pll_div *pll_div, unsigned int target,
+	unsigned int source)
+{
+	u64 Kpart;
+	unsigned int K, Ndiv, Nmod;
+
+
+	Ndiv = target / source;
+	if (Ndiv < 6) {
+		source >>= 1;
+		pll_div->div2 = 1;
+		Ndiv = target / source;
+	} else
+		pll_div->div2 = 0;
+
+	if ((Ndiv < 6) || (Ndiv > 12))
+		printk(KERN_WARNING
+		"WM8990 N value outwith recommended range! N = %d\n", Ndiv);
+
+	pll_div->n = Ndiv;
+	Nmod = target % source;
+	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, source);
+
+	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;
+
+	pll_div->k = K;
+}
+
+static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
+		int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+	u16 reg;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct _pll_div pll_div;
+
+	if (freq_in && freq_out) {
+		pll_factors(&pll_div, freq_out * 4, freq_in);
+
+		/* Turn on PLL */
+		reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+		reg |= WM8990_PLL_ENA;
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
+
+		/* sysclk comes from PLL */
+		reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2);
+		wm8990_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC);
+
+		/* set up N , fractional mode and pre-divisor if neccessary */
+		wm8990_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM |
+			(pll_div.div2?WM8990_PRESCALE:0));
+		wm8990_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8));
+		wm8990_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF));
+	} else {
+		/* Turn on PLL */
+		reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+		reg &= ~WM8990_PLL_ENA;
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
+	}
+	return 0;
+}
+
+/*
+ * Clock after PLL and dividers
+ */
+static int wm8990_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 wm8990_priv *wm8990 = codec->private_data;
+
+	wm8990->sysclk = freq;
+	return 0;
+}
+
+/*
+ * Set's ADC and Voice DAC format.
+ */
+static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 audio1, audio3;
+
+	audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
+	audio3 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_3);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		audio3 &= ~WM8990_AIF_MSTR1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		audio3 |= WM8990_AIF_MSTR1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	audio1 &= ~WM8990_AIF_FMT_MASK;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		audio1 |= WM8990_AIF_TMF_I2S;
+		audio1 &= ~WM8990_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		audio1 |= WM8990_AIF_TMF_RIGHTJ;
+		audio1 &= ~WM8990_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		audio1 |= WM8990_AIF_TMF_LEFTJ;
+		audio1 &= ~WM8990_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		audio1 |= WM8990_AIF_TMF_DSP;
+		audio1 &= ~WM8990_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		audio1 |= WM8990_AIF_TMF_DSP | WM8990_AIF_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
+	wm8990_write(codec, WM8990_AUDIO_INTERFACE_3, audio3);
+	return 0;
+}
+
+static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+		int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	switch (div_id) {
+	case WM8990_MCLK_DIV:
+		reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+			~WM8990_MCLK_DIV_MASK;
+		wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+		break;
+	case WM8990_DACCLK_DIV:
+		reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+			~WM8990_DAC_CLKDIV_MASK;
+		wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+		break;
+	case WM8990_ADCCLK_DIV:
+		reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) &
+			~WM8990_ADC_CLKDIV_MASK;
+		wm8990_write(codec, WM8990_CLOCKING_2, reg | div);
+		break;
+	case WM8990_BCLK_DIV:
+		reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_1) &
+			~WM8990_BCLK_DIV_MASK;
+		wm8990_write(codec, WM8990_CLOCKING_1, reg | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * 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_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
+
+	audio1 &= ~WM8990_AIF_WL_MASK;
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		audio1 |= WM8990_AIF_WL_20BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		audio1 |= WM8990_AIF_WL_24BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		audio1 |= WM8990_AIF_WL_32BITS;
+		break;
+	}
+
+	wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1);
+	return 0;
+}
+
+static int wm8990_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 val;
+
+	val  = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE;
+
+	if (mute)
+		wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+	else
+		wm8990_write(codec, WM8990_DAC_CTRL, val);
+
+	return 0;
+}
+
+static int wm8990_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	u16 val;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Enable all output discharge bits */
+			wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
+				WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
+				WM8990_DIS_OUT4 | WM8990_DIS_LOUT |
+				WM8990_DIS_ROUT);
+
+			/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
+			wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+				     WM8990_BUFDCOPEN | WM8990_POBCTRL |
+				     WM8990_VMIDTOG);
+
+			/* Delay to allow output caps to discharge */
+			msleep(msecs_to_jiffies(300));
+
+			/* Disable VMIDTOG */
+			wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+				     WM8990_BUFDCOPEN | WM8990_POBCTRL);
+
+			/* disable all output discharge bits */
+			wm8990_write(codec, WM8990_ANTIPOP1, 0);
+
+			/* Enable outputs */
+			wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00);
+
+			msleep(msecs_to_jiffies(50));
+
+			/* Enable VMID at 2x50k */
+			wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02);
+
+			msleep(msecs_to_jiffies(100));
+
+			/* Enable VREF */
+			wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
+
+			msleep(msecs_to_jiffies(600));
+
+			/* Enable BUFIOEN */
+			wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+				     WM8990_BUFDCOPEN | WM8990_POBCTRL |
+				     WM8990_BUFIOEN);
+
+			/* Disable outputs */
+			wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3);
+
+			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+			wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN);
+		} else {
+			/* ON -> standby */
+
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* Enable POBCTRL and SOFT_ST */
+		wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+			WM8990_POBCTRL | WM8990_BUFIOEN);
+
+		/* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
+		wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST |
+			WM8990_BUFDCOPEN | WM8990_POBCTRL |
+			WM8990_BUFIOEN);
+
+		/* mute DAC */
+		val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL);
+		wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+
+		/* Enable any disabled outputs */
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
+
+		/* Disable VMID */
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01);
+
+		msleep(msecs_to_jiffies(300));
+
+		/* Enable all output discharge bits */
+		wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
+			WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
+			WM8990_DIS_OUT4 | WM8990_DIS_LOUT |
+			WM8990_DIS_ROUT);
+
+		/* Disable VREF */
+		wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0);
+
+		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+		wm8990_write(codec, WM8990_ANTIPOP2, 0x0);
+		break;
+	}
+
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8990_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)
+
+#define WM8990_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+/*
+ * The WM8990 supports 2 different and mutually exclusive DAI
+ * configurations.
+ *
+ * 1. ADC/DAC on Primary Interface
+ * 2. ADC on Primary Interface/DAC on secondary
+ */
+struct snd_soc_dai wm8990_dai = {
+/* ADC/DAC on primary */
+	.name = "WM8990 ADC/DAC Primary",
+	.id = 1,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8990_RATES,
+		.formats = WM8990_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8990_RATES,
+		.formats = WM8990_FORMATS,},
+	.ops = {
+		.hw_params = wm8990_hw_params,},
+	.dai_ops = {
+		.digital_mute = wm8990_mute,
+		.set_fmt = wm8990_set_dai_fmt,
+		.set_clkdiv = wm8990_set_dai_clkdiv,
+		.set_pll = wm8990_set_dai_pll,
+		.set_sysclk = wm8990_set_dai_sysclk,
+	},
+};
+EXPORT_SYMBOL_GPL(wm8990_dai);
+
+static int wm8990_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;
+
+	/* we only need to suspend if we are a valid card */
+	if (!codec->card)
+		return 0;
+
+	wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8990_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+	u8 data[2];
+	u16 *cache = codec->reg_cache;
+
+	/* we only need to resume if we are a valid card */
+	if (!codec->card)
+		return 0;
+
+	/* Sync reg_cache with the hardware */
+	for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) {
+		if (i + 1 == WM8990_RESET)
+			continue;
+		data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
+		data[1] = cache[i] & 0x00ff;
+		codec->hw_write(codec->control_data, data, 2);
+	}
+
+	wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+/*
+ * initialise the WM8990 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int wm8990_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 reg;
+	int ret = 0;
+
+	codec->name = "WM8990";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8990_read_reg_cache;
+	codec->write = wm8990_write;
+	codec->set_bias_level = wm8990_set_bias_level;
+	codec->dai = &wm8990_dai;
+	codec->num_dai = 2;
+	codec->reg_cache_size = ARRAY_SIZE(wm8990_reg);
+	codec->reg_cache = kmemdup(wm8990_reg, sizeof(wm8990_reg), GFP_KERNEL);
+
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	wm8990_reset(codec);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8990: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* charge output caps */
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	reg = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_4);
+	wm8990_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1);
+
+	reg = wm8990_read_reg_cache(codec, WM8990_GPIO1_GPIO2) &
+		~WM8990_GPIO1_SEL_MASK;
+	wm8990_write(codec, WM8990_GPIO1_GPIO2, reg | 1);
+
+	reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2);
+	wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA);
+
+	wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+	wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+
+	wm8990_add_controls(codec);
+	wm8990_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8990: failed to register card\n");
+		goto card_err;
+	}
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+	return ret;
+}
+
+/* If the i2c layer weren't so broken, we could pass this kind of data
+   around */
+static struct snd_soc_device *wm8990_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+/*
+ * WM891 2 wire address is determined by GPIO5
+ * state during powerup.
+ *    low  = 0x34
+ *    high = 0x36
+ */
+static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
+
+/* Magic definition of all other variables and things */
+I2C_CLIENT_INSMOD;
+
+static struct i2c_driver wm8990_i2c_driver;
+static struct i2c_client client_template;
+
+static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind)
+{
+	struct snd_soc_device *socdev = wm8990_socdev;
+	struct wm8990_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) {
+		pr_err("failed to attach codec at addr %x\n", addr);
+		goto err;
+	}
+
+	ret = wm8990_init(socdev);
+	if (ret < 0) {
+		pr_err("failed to initialise WM8990\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+
+static int wm8990_i2c_detach(struct i2c_client *client)
+{
+	struct snd_soc_codec *codec = i2c_get_clientdata(client);
+	i2c_detach_client(client);
+	kfree(codec->reg_cache);
+	kfree(client);
+	return 0;
+}
+
+static int wm8990_i2c_attach(struct i2c_adapter *adap)
+{
+	return i2c_probe(adap, &addr_data, wm8990_codec_probe);
+}
+
+static struct i2c_driver wm8990_i2c_driver = {
+	.driver = {
+		.name = "WM8990 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.attach_adapter = wm8990_i2c_attach,
+	.detach_client =  wm8990_i2c_detach,
+	.command =        NULL,
+};
+
+static struct i2c_client client_template = {
+	.name =   "WM8990",
+	.driver = &wm8990_i2c_driver,
+};
+#endif
+
+static int wm8990_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct wm8990_setup_data *setup;
+	struct snd_soc_codec *codec;
+	struct wm8990_priv *wm8990;
+	int ret = 0;
+
+	pr_info("WM8990 Audio Codec %s\n", WM8990_VERSION);
+
+	setup = socdev->codec_data;
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	wm8990 = kzalloc(sizeof(struct wm8990_priv), GFP_KERNEL);
+	if (wm8990 == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+
+	codec->private_data = wm8990;
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	wm8990_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(&wm8990_i2c_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add i2c driver");
+	}
+#else
+		/* Add other interfaces here */
+#endif
+	return ret;
+}
+
+/* power down chip */
+static int wm8990_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)
+		wm8990_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(&wm8990_i2c_driver);
+#endif
+	kfree(codec->private_data);
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8990 = {
+	.probe =	wm8990_probe,
+	.remove =	wm8990_remove,
+	.suspend =	wm8990_suspend,
+	.resume =	wm8990_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990);
+
+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
new file mode 100644
index 000000000000..6bea57485283
--- /dev/null
+++ b/sound/soc/codecs/wm8990.h
@@ -0,0 +1,832 @@
+/*
+ * wm8990.h  --  audio driver for WM8990
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef __WM8990REGISTERDEFS_H__
+#define __WM8990REGISTERDEFS_H__
+
+/*
+ * Register values.
+ */
+#define WM8990_RESET                            0x00
+#define WM8990_POWER_MANAGEMENT_1               0x01
+#define WM8990_POWER_MANAGEMENT_2               0x02
+#define WM8990_POWER_MANAGEMENT_3               0x03
+#define WM8990_AUDIO_INTERFACE_1                0x04
+#define WM8990_AUDIO_INTERFACE_2                0x05
+#define WM8990_CLOCKING_1                       0x06
+#define WM8990_CLOCKING_2                       0x07
+#define WM8990_AUDIO_INTERFACE_3                0x08
+#define WM8990_AUDIO_INTERFACE_4                0x09
+#define WM8990_DAC_CTRL                         0x0A
+#define WM8990_LEFT_DAC_DIGITAL_VOLUME          0x0B
+#define WM8990_RIGHT_DAC_DIGITAL_VOLUME         0x0C
+#define WM8990_DIGITAL_SIDE_TONE                0x0D
+#define WM8990_ADC_CTRL                         0x0E
+#define WM8990_LEFT_ADC_DIGITAL_VOLUME          0x0F
+#define WM8990_RIGHT_ADC_DIGITAL_VOLUME         0x10
+#define WM8990_GPIO_CTRL_1                      0x12
+#define WM8990_GPIO1_GPIO2                      0x13
+#define WM8990_GPIO3_GPIO4                      0x14
+#define WM8990_GPIO5_GPIO6                      0x15
+#define WM8990_GPIOCTRL_2                       0x16
+#define WM8990_GPIO_POL                         0x17
+#define WM8990_LEFT_LINE_INPUT_1_2_VOLUME       0x18
+#define WM8990_LEFT_LINE_INPUT_3_4_VOLUME       0x19
+#define WM8990_RIGHT_LINE_INPUT_1_2_VOLUME      0x1A
+#define WM8990_RIGHT_LINE_INPUT_3_4_VOLUME      0x1B
+#define WM8990_LEFT_OUTPUT_VOLUME               0x1C
+#define WM8990_RIGHT_OUTPUT_VOLUME              0x1D
+#define WM8990_LINE_OUTPUTS_VOLUME              0x1E
+#define WM8990_OUT3_4_VOLUME                    0x1F
+#define WM8990_LEFT_OPGA_VOLUME                 0x20
+#define WM8990_RIGHT_OPGA_VOLUME                0x21
+#define WM8990_SPEAKER_VOLUME                   0x22
+#define WM8990_CLASSD1                          0x23
+#define WM8990_CLASSD3                          0x25
+#define WM8990_INPUT_MIXER1                     0x27
+#define WM8990_INPUT_MIXER2                     0x28
+#define WM8990_INPUT_MIXER3                     0x29
+#define WM8990_INPUT_MIXER4                     0x2A
+#define WM8990_INPUT_MIXER5                     0x2B
+#define WM8990_INPUT_MIXER6                     0x2C
+#define WM8990_OUTPUT_MIXER1                    0x2D
+#define WM8990_OUTPUT_MIXER2                    0x2E
+#define WM8990_OUTPUT_MIXER3                    0x2F
+#define WM8990_OUTPUT_MIXER4                    0x30
+#define WM8990_OUTPUT_MIXER5                    0x31
+#define WM8990_OUTPUT_MIXER6                    0x32
+#define WM8990_OUT3_4_MIXER                     0x33
+#define WM8990_LINE_MIXER1                      0x34
+#define WM8990_LINE_MIXER2                      0x35
+#define WM8990_SPEAKER_MIXER                    0x36
+#define WM8990_ADDITIONAL_CONTROL               0x37
+#define WM8990_ANTIPOP1                         0x38
+#define WM8990_ANTIPOP2                         0x39
+#define WM8990_MICBIAS                          0x3A
+#define WM8990_PLL1                             0x3C
+#define WM8990_PLL2                             0x3D
+#define WM8990_PLL3                             0x3E
+#define WM8990_INTDRIVBITS			0x3F
+
+#define WM8990_REGISTER_COUNT                   60
+#define WM8990_MAX_REGISTER                     0x3F
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Reset
+ */
+#define WM8990_SW_RESET_CHIP_ID_MASK            0xFFFF  /* SW_RESET_CHIP_ID */
+
+/*
+ * R1 (0x01) - Power Management (1)
+ */
+#define WM8990_SPK_ENA                          0x1000  /* SPK_ENA */
+#define WM8990_SPK_ENA_BIT			12
+#define WM8990_OUT3_ENA                         0x0800  /* OUT3_ENA */
+#define WM8990_OUT3_ENA_BIT			11
+#define WM8990_OUT4_ENA                         0x0400  /* OUT4_ENA */
+#define WM8990_OUT4_ENA_BIT			10
+#define WM8990_LOUT_ENA                         0x0200  /* LOUT_ENA */
+#define WM8990_LOUT_ENA_BIT			9
+#define WM8990_ROUT_ENA                         0x0100  /* ROUT_ENA */
+#define WM8990_ROUT_ENA_BIT			8
+#define WM8990_MICBIAS_ENA                      0x0010  /* MICBIAS_ENA */
+#define WM8990_MICBIAS_ENA_BIT			4
+#define WM8990_VMID_MODE_MASK                   0x0006  /* VMID_MODE - [2:1] */
+#define WM8990_VREF_ENA                         0x0001  /* VREF_ENA */
+#define WM8990_VREF_ENA_BIT			0
+
+/*
+ * R2 (0x02) - Power Management (2)
+ */
+#define WM8990_PLL_ENA                          0x8000  /* PLL_ENA */
+#define WM8990_PLL_ENA_BIT			15
+#define WM8990_TSHUT_ENA                        0x4000  /* TSHUT_ENA */
+#define WM8990_TSHUT_ENA_BIT			14
+#define WM8990_TSHUT_OPDIS                      0x2000  /* TSHUT_OPDIS */
+#define WM8990_TSHUT_OPDIS_BIT			13
+#define WM8990_OPCLK_ENA                        0x0800  /* OPCLK_ENA */
+#define WM8990_OPCLK_ENA_BIT			11
+#define WM8990_AINL_ENA                         0x0200  /* AINL_ENA */
+#define WM8990_AINL_ENA_BIT			9
+#define WM8990_AINR_ENA                         0x0100  /* AINR_ENA */
+#define WM8990_AINR_ENA_BIT			8
+#define WM8990_LIN34_ENA                        0x0080  /* LIN34_ENA */
+#define WM8990_LIN34_ENA_BIT			7
+#define WM8990_LIN12_ENA                        0x0040  /* LIN12_ENA */
+#define WM8990_LIN12_ENA_BIT			6
+#define WM8990_RIN34_ENA                        0x0020  /* RIN34_ENA */
+#define WM8990_RIN34_ENA_BIT			5
+#define WM8990_RIN12_ENA                        0x0010  /* RIN12_ENA */
+#define WM8990_RIN12_ENA_BIT			4
+#define WM8990_ADCL_ENA                         0x0002  /* ADCL_ENA */
+#define WM8990_ADCL_ENA_BIT			1
+#define WM8990_ADCR_ENA                         0x0001  /* ADCR_ENA */
+#define WM8990_ADCR_ENA_BIT			0
+
+/*
+ * R3 (0x03) - Power Management (3)
+ */
+#define WM8990_LON_ENA                          0x2000  /* LON_ENA */
+#define WM8990_LON_ENA_BIT			13
+#define WM8990_LOP_ENA                          0x1000  /* LOP_ENA */
+#define WM8990_LOP_ENA_BIT			12
+#define WM8990_RON_ENA                          0x0800  /* RON_ENA */
+#define WM8990_RON_ENA_BIT			11
+#define WM8990_ROP_ENA                          0x0400  /* ROP_ENA */
+#define WM8990_ROP_ENA_BIT			10
+#define WM8990_LOPGA_ENA                        0x0080  /* LOPGA_ENA */
+#define WM8990_LOPGA_ENA_BIT			7
+#define WM8990_ROPGA_ENA                        0x0040  /* ROPGA_ENA */
+#define WM8990_ROPGA_ENA_BIT			6
+#define WM8990_LOMIX_ENA                        0x0020  /* LOMIX_ENA */
+#define WM8990_LOMIX_ENA_BIT			5
+#define WM8990_ROMIX_ENA                        0x0010  /* ROMIX_ENA */
+#define WM8990_ROMIX_ENA_BIT			4
+#define WM8990_DACL_ENA                         0x0002  /* DACL_ENA */
+#define WM8990_DACL_ENA_BIT			1
+#define WM8990_DACR_ENA                         0x0001  /* DACR_ENA */
+#define WM8990_DACR_ENA_BIT			0
+
+/*
+ * R4 (0x04) - Audio Interface (1)
+ */
+#define WM8990_AIFADCL_SRC                      0x8000  /* AIFADCL_SRC */
+#define WM8990_AIFADCR_SRC                      0x4000  /* AIFADCR_SRC */
+#define WM8990_AIFADC_TDM                       0x2000  /* AIFADC_TDM */
+#define WM8990_AIFADC_TDM_CHAN                  0x1000  /* AIFADC_TDM_CHAN */
+#define WM8990_AIF_BCLK_INV                     0x0100  /* AIF_BCLK_INV */
+#define WM8990_AIF_LRCLK_INV                    0x0080  /* AIF_LRCLK_INV */
+#define WM8990_AIF_WL_MASK                      0x0060  /* AIF_WL - [6:5] */
+#define WM8990_AIF_WL_16BITS			(0 << 5)
+#define WM8990_AIF_WL_20BITS			(1 << 5)
+#define WM8990_AIF_WL_24BITS			(2 << 5)
+#define WM8990_AIF_WL_32BITS			(3 << 5)
+#define WM8990_AIF_FMT_MASK                     0x0018  /* AIF_FMT - [4:3] */
+#define WM8990_AIF_TMF_RIGHTJ			(0 << 3)
+#define WM8990_AIF_TMF_LEFTJ			(1 << 3)
+#define WM8990_AIF_TMF_I2S			(2 << 3)
+#define WM8990_AIF_TMF_DSP			(3 << 3)
+
+/*
+ * R5 (0x05) - Audio Interface (2)
+ */
+#define WM8990_DACL_SRC                         0x8000  /* DACL_SRC */
+#define WM8990_DACR_SRC                         0x4000  /* DACR_SRC */
+#define WM8990_AIFDAC_TDM                       0x2000  /* AIFDAC_TDM */
+#define WM8990_AIFDAC_TDM_CHAN                  0x1000  /* AIFDAC_TDM_CHAN */
+#define WM8990_DAC_BOOST_MASK                   0x0C00  /* DAC_BOOST */
+#define WM8990_DAC_COMP                         0x0010  /* DAC_COMP */
+#define WM8990_DAC_COMPMODE                     0x0008  /* DAC_COMPMODE */
+#define WM8990_ADC_COMP                         0x0004  /* ADC_COMP */
+#define WM8990_ADC_COMPMODE                     0x0002  /* ADC_COMPMODE */
+#define WM8990_LOOPBACK                         0x0001  /* LOOPBACK */
+
+/*
+ * R6 (0x06) - Clocking (1)
+ */
+#define WM8990_TOCLK_RATE                       0x8000  /* TOCLK_RATE */
+#define WM8990_TOCLK_ENA                        0x4000  /* TOCLK_ENA */
+#define WM8990_OPCLKDIV_MASK                    0x1E00  /* OPCLKDIV - [12:9] */
+#define WM8990_DCLKDIV_MASK                     0x01C0  /* DCLKDIV - [8:6] */
+#define WM8990_BCLK_DIV_MASK                    0x001E  /* BCLK_DIV - [4:1] */
+#define WM8990_BCLK_DIV_1			(0x0 << 1)
+#define WM8990_BCLK_DIV_1_5			(0x1 << 1)
+#define WM8990_BCLK_DIV_2			(0x2 << 1)
+#define WM8990_BCLK_DIV_3			(0x3 << 1)
+#define WM8990_BCLK_DIV_4			(0x4 << 1)
+#define WM8990_BCLK_DIV_5_5			(0x5 << 1)
+#define WM8990_BCLK_DIV_6			(0x6 << 1)
+#define WM8990_BCLK_DIV_8			(0x7 << 1)
+#define WM8990_BCLK_DIV_11			(0x8 << 1)
+#define WM8990_BCLK_DIV_12			(0x9 << 1)
+#define WM8990_BCLK_DIV_16			(0xA << 1)
+#define WM8990_BCLK_DIV_22			(0xB << 1)
+#define WM8990_BCLK_DIV_24			(0xC << 1)
+#define WM8990_BCLK_DIV_32			(0xD << 1)
+#define WM8990_BCLK_DIV_44			(0xE << 1)
+#define WM8990_BCLK_DIV_48			(0xF << 1)
+
+/*
+ * R7 (0x07) - Clocking (2)
+ */
+#define WM8990_MCLK_SRC                         0x8000  /* MCLK_SRC */
+#define WM8990_SYSCLK_SRC                       0x4000  /* SYSCLK_SRC */
+#define WM8990_CLK_FORCE                        0x2000  /* CLK_FORCE */
+#define WM8990_MCLK_DIV_MASK                    0x1800  /* MCLK_DIV - [12:11] */
+#define WM8990_MCLK_DIV_1			(0 << 11)
+#define WM8990_MCLK_DIV_2			(2 << 11)
+#define WM8990_MCLK_INV                         0x0400  /* MCLK_INV */
+#define WM8990_ADC_CLKDIV_MASK                  0x00E0  /* ADC_CLKDIV */
+#define WM8990_ADC_CLKDIV_1			(0 << 5)
+#define WM8990_ADC_CLKDIV_1_5			(1 << 5)
+#define WM8990_ADC_CLKDIV_2			(2 << 5)
+#define WM8990_ADC_CLKDIV_3			(3 << 5)
+#define WM8990_ADC_CLKDIV_4			(4 << 5)
+#define WM8990_ADC_CLKDIV_5_5			(5 << 5)
+#define WM8990_ADC_CLKDIV_6			(6 << 5)
+#define WM8990_DAC_CLKDIV_MASK                  0x001C  /* DAC_CLKDIV - [4:2] */
+#define WM8990_DAC_CLKDIV_1			(0 << 2)
+#define WM8990_DAC_CLKDIV_1_5			(1 << 2)
+#define WM8990_DAC_CLKDIV_2			(2 << 2)
+#define WM8990_DAC_CLKDIV_3			(3 << 2)
+#define WM8990_DAC_CLKDIV_4			(4 << 2)
+#define WM8990_DAC_CLKDIV_5_5			(5 << 2)
+#define WM8990_DAC_CLKDIV_6			(6 << 2)
+
+/*
+ * R8 (0x08) - Audio Interface (3)
+ */
+#define WM8990_AIF_MSTR1                        0x8000  /* AIF_MSTR1 */
+#define WM8990_AIF_MSTR2                        0x4000  /* AIF_MSTR2 */
+#define WM8990_AIF_SEL                          0x2000  /* AIF_SEL */
+#define WM8990_ADCLRC_DIR                       0x0800  /* ADCLRC_DIR */
+#define WM8990_ADCLRC_RATE_MASK                 0x07FF  /* ADCLRC_RATE */
+
+/*
+ * R9 (0x09) - Audio Interface (4)
+ */
+#define WM8990_ALRCGPIO1                        0x8000  /* ALRCGPIO1 */
+#define WM8990_ALRCBGPIO6                       0x4000  /* ALRCBGPIO6 */
+#define WM8990_AIF_TRIS                         0x2000  /* AIF_TRIS */
+#define WM8990_DACLRC_DIR                       0x0800  /* DACLRC_DIR */
+#define WM8990_DACLRC_RATE_MASK                 0x07FF  /* DACLRC_RATE */
+
+/*
+ * R10 (0x0A) - DAC CTRL
+ */
+#define WM8990_AIF_LRCLKRATE                    0x0400  /* AIF_LRCLKRATE */
+#define WM8990_DAC_MONO                         0x0200  /* DAC_MONO */
+#define WM8990_DAC_SB_FILT                      0x0100  /* DAC_SB_FILT */
+#define WM8990_DAC_MUTERATE                     0x0080  /* DAC_MUTERATE */
+#define WM8990_DAC_MUTEMODE                     0x0040  /* DAC_MUTEMODE */
+#define WM8990_DEEMP_MASK                       0x0030  /* DEEMP - [5:4] */
+#define WM8990_DAC_MUTE                         0x0004  /* DAC_MUTE */
+#define WM8990_DACL_DATINV                      0x0002  /* DACL_DATINV */
+#define WM8990_DACR_DATINV                      0x0001  /* DACR_DATINV */
+
+/*
+ * R11 (0x0B) - Left DAC Digital Volume
+ */
+#define WM8990_DAC_VU                           0x0100  /* DAC_VU */
+#define WM8990_DACL_VOL_MASK                    0x00FF  /* DACL_VOL - [7:0] */
+#define WM8990_DACL_VOL_SHIFT			0
+/*
+ * R12 (0x0C) - Right DAC Digital Volume
+ */
+#define WM8990_DAC_VU                           0x0100  /* DAC_VU */
+#define WM8990_DACR_VOL_MASK                    0x00FF  /* DACR_VOL - [7:0] */
+#define WM8990_DACR_VOL_SHIFT			0
+/*
+ * R13 (0x0D) - Digital Side Tone
+ */
+#define WM8990_ADCL_DAC_SVOL_MASK               0x0F  /* ADCL_DAC_SVOL */
+#define WM8990_ADCL_DAC_SVOL_SHIFT		9
+#define WM8990_ADCR_DAC_SVOL_MASK               0x0F  /* ADCR_DAC_SVOL */
+#define WM8990_ADCR_DAC_SVOL_SHIFT		5
+#define WM8990_ADC_TO_DACL_MASK                 0x03  /* ADC_TO_DACL - [3:2] */
+#define WM8990_ADC_TO_DACL_SHIFT		2
+#define WM8990_ADC_TO_DACR_MASK                 0x03  /* ADC_TO_DACR - [1:0] */
+#define WM8990_ADC_TO_DACR_SHIFT		0
+
+/*
+ * R14 (0x0E) - ADC CTRL
+ */
+#define WM8990_ADC_HPF_ENA                      0x0100  /* ADC_HPF_ENA */
+#define WM8990_ADC_HPF_ENA_BIT			8
+#define WM8990_ADC_HPF_CUT_MASK                 0x03  /* ADC_HPF_CUT - [6:5] */
+#define WM8990_ADC_HPF_CUT_SHIFT		5
+#define WM8990_ADCL_DATINV                      0x0002  /* ADCL_DATINV */
+#define WM8990_ADCL_DATINV_BIT			1
+#define WM8990_ADCR_DATINV                      0x0001  /* ADCR_DATINV */
+#define WM8990_ADCR_DATINV_BIT			0
+
+/*
+ * R15 (0x0F) - Left ADC Digital Volume
+ */
+#define WM8990_ADC_VU                           0x0100  /* ADC_VU */
+#define WM8990_ADCL_VOL_MASK                    0x00FF  /* ADCL_VOL - [7:0] */
+#define WM8990_ADCL_VOL_SHIFT			0
+
+/*
+ * R16 (0x10) - Right ADC Digital Volume
+ */
+#define WM8990_ADC_VU                           0x0100  /* ADC_VU */
+#define WM8990_ADCR_VOL_MASK                    0x00FF  /* ADCR_VOL - [7:0] */
+#define WM8990_ADCR_VOL_SHIFT			0
+
+/*
+ * R18 (0x12) - GPIO CTRL 1
+ */
+#define WM8990_IRQ                              0x1000  /* IRQ */
+#define WM8990_TEMPOK                           0x0800  /* TEMPOK */
+#define WM8990_MICSHRT                          0x0400  /* MICSHRT */
+#define WM8990_MICDET                           0x0200  /* MICDET */
+#define WM8990_PLL_LCK                          0x0100  /* PLL_LCK */
+#define WM8990_GPI8_STATUS                      0x0080  /* GPI8_STATUS */
+#define WM8990_GPI7_STATUS                      0x0040  /* GPI7_STATUS */
+#define WM8990_GPIO6_STATUS                     0x0020  /* GPIO6_STATUS */
+#define WM8990_GPIO5_STATUS                     0x0010  /* GPIO5_STATUS */
+#define WM8990_GPIO4_STATUS                     0x0008  /* GPIO4_STATUS */
+#define WM8990_GPIO3_STATUS                     0x0004  /* GPIO3_STATUS */
+#define WM8990_GPIO2_STATUS                     0x0002  /* GPIO2_STATUS */
+#define WM8990_GPIO1_STATUS                     0x0001  /* GPIO1_STATUS */
+
+/*
+ * R19 (0x13) - GPIO1 & GPIO2
+ */
+#define WM8990_GPIO2_DEB_ENA                    0x8000  /* GPIO2_DEB_ENA */
+#define WM8990_GPIO2_IRQ_ENA                    0x4000  /* GPIO2_IRQ_ENA */
+#define WM8990_GPIO2_PU                         0x2000  /* GPIO2_PU */
+#define WM8990_GPIO2_PD                         0x1000  /* GPIO2_PD */
+#define WM8990_GPIO2_SEL_MASK                   0x0F00  /* GPIO2_SEL - [11:8] */
+#define WM8990_GPIO1_DEB_ENA                    0x0080  /* GPIO1_DEB_ENA */
+#define WM8990_GPIO1_IRQ_ENA                    0x0040  /* GPIO1_IRQ_ENA */
+#define WM8990_GPIO1_PU                         0x0020  /* GPIO1_PU */
+#define WM8990_GPIO1_PD                         0x0010  /* GPIO1_PD */
+#define WM8990_GPIO1_SEL_MASK                   0x000F  /* GPIO1_SEL - [3:0] */
+
+/*
+ * R20 (0x14) - GPIO3 & GPIO4
+ */
+#define WM8990_GPIO4_DEB_ENA                    0x8000  /* GPIO4_DEB_ENA */
+#define WM8990_GPIO4_IRQ_ENA                    0x4000  /* GPIO4_IRQ_ENA */
+#define WM8990_GPIO4_PU                         0x2000  /* GPIO4_PU */
+#define WM8990_GPIO4_PD                         0x1000  /* GPIO4_PD */
+#define WM8990_GPIO4_SEL_MASK                   0x0F00  /* GPIO4_SEL - [11:8] */
+#define WM8990_GPIO3_DEB_ENA                    0x0080  /* GPIO3_DEB_ENA */
+#define WM8990_GPIO3_IRQ_ENA                    0x0040  /* GPIO3_IRQ_ENA */
+#define WM8990_GPIO3_PU                         0x0020  /* GPIO3_PU */
+#define WM8990_GPIO3_PD                         0x0010  /* GPIO3_PD */
+#define WM8990_GPIO3_SEL_MASK                   0x000F  /* GPIO3_SEL - [3:0] */
+
+/*
+ * R21 (0x15) - GPIO5 & GPIO6
+ */
+#define WM8990_GPIO6_DEB_ENA                    0x8000  /* GPIO6_DEB_ENA */
+#define WM8990_GPIO6_IRQ_ENA                    0x4000  /* GPIO6_IRQ_ENA */
+#define WM8990_GPIO6_PU                         0x2000  /* GPIO6_PU */
+#define WM8990_GPIO6_PD                         0x1000  /* GPIO6_PD */
+#define WM8990_GPIO6_SEL_MASK                   0x0F00  /* GPIO6_SEL - [11:8] */
+#define WM8990_GPIO5_DEB_ENA                    0x0080  /* GPIO5_DEB_ENA */
+#define WM8990_GPIO5_IRQ_ENA                    0x0040  /* GPIO5_IRQ_ENA */
+#define WM8990_GPIO5_PU                         0x0020  /* GPIO5_PU */
+#define WM8990_GPIO5_PD                         0x0010  /* GPIO5_PD */
+#define WM8990_GPIO5_SEL_MASK                   0x000F  /* GPIO5_SEL - [3:0] */
+
+/*
+ * R22 (0x16) - GPIOCTRL 2
+ */
+#define WM8990_RD_3W_ENA                        0x8000  /* RD_3W_ENA */
+#define WM8990_MODE_3W4W                        0x4000  /* MODE_3W4W */
+#define WM8990_TEMPOK_IRQ_ENA                   0x0800  /* TEMPOK_IRQ_ENA */
+#define WM8990_MICSHRT_IRQ_ENA                  0x0400  /* MICSHRT_IRQ_ENA */
+#define WM8990_MICDET_IRQ_ENA                   0x0200  /* MICDET_IRQ_ENA */
+#define WM8990_PLL_LCK_IRQ_ENA                  0x0100  /* PLL_LCK_IRQ_ENA */
+#define WM8990_GPI8_DEB_ENA                     0x0080  /* GPI8_DEB_ENA */
+#define WM8990_GPI8_IRQ_ENA                     0x0040  /* GPI8_IRQ_ENA */
+#define WM8990_GPI8_ENA                         0x0010  /* GPI8_ENA */
+#define WM8990_GPI7_DEB_ENA                     0x0008  /* GPI7_DEB_ENA */
+#define WM8990_GPI7_IRQ_ENA                     0x0004  /* GPI7_IRQ_ENA */
+#define WM8990_GPI7_ENA                         0x0001  /* GPI7_ENA */
+
+/*
+ * R23 (0x17) - GPIO_POL
+ */
+#define WM8990_IRQ_INV                          0x1000  /* IRQ_INV */
+#define WM8990_TEMPOK_POL                       0x0800  /* TEMPOK_POL */
+#define WM8990_MICSHRT_POL                      0x0400  /* MICSHRT_POL */
+#define WM8990_MICDET_POL                       0x0200  /* MICDET_POL */
+#define WM8990_PLL_LCK_POL                      0x0100  /* PLL_LCK_POL */
+#define WM8990_GPI8_POL                         0x0080  /* GPI8_POL */
+#define WM8990_GPI7_POL                         0x0040  /* GPI7_POL */
+#define WM8990_GPIO6_POL                        0x0020  /* GPIO6_POL */
+#define WM8990_GPIO5_POL                        0x0010  /* GPIO5_POL */
+#define WM8990_GPIO4_POL                        0x0008  /* GPIO4_POL */
+#define WM8990_GPIO3_POL                        0x0004  /* GPIO3_POL */
+#define WM8990_GPIO2_POL                        0x0002  /* GPIO2_POL */
+#define WM8990_GPIO1_POL                        0x0001  /* GPIO1_POL */
+
+/*
+ * R24 (0x18) - Left Line Input 1&2 Volume
+ */
+#define WM8990_IPVU                             0x0100  /* IPVU */
+#define WM8990_LI12MUTE                         0x0080  /* LI12MUTE */
+#define WM8990_LI12MUTE_BIT			7
+#define WM8990_LI12ZC                           0x0040  /* LI12ZC */
+#define WM8990_LI12ZC_BIT			6
+#define WM8990_LIN12VOL_MASK                    0x001F  /* LIN12VOL - [4:0] */
+#define WM8990_LIN12VOL_SHIFT			0
+/*
+ * R25 (0x19) - Left Line Input 3&4 Volume
+ */
+#define WM8990_IPVU                             0x0100  /* IPVU */
+#define WM8990_LI34MUTE                         0x0080  /* LI34MUTE */
+#define WM8990_LI34MUTE_BIT			7
+#define WM8990_LI34ZC                           0x0040  /* LI34ZC */
+#define WM8990_LI34ZC_BIT			6
+#define WM8990_LIN34VOL_MASK                    0x001F  /* LIN34VOL - [4:0] */
+#define WM8990_LIN34VOL_SHIFT			0
+
+/*
+ * R26 (0x1A) - Right Line Input 1&2 Volume
+ */
+#define WM8990_IPVU                             0x0100  /* IPVU */
+#define WM8990_RI12MUTE                         0x0080  /* RI12MUTE */
+#define WM8990_RI12MUTE_BIT			7
+#define WM8990_RI12ZC                           0x0040  /* RI12ZC */
+#define WM8990_RI12ZC_BIT			6
+#define WM8990_RIN12VOL_MASK                    0x001F  /* RIN12VOL - [4:0] */
+#define WM8990_RIN12VOL_SHIFT			0
+
+/*
+ * R27 (0x1B) - Right Line Input 3&4 Volume
+ */
+#define WM8990_IPVU                             0x0100  /* IPVU */
+#define WM8990_RI34MUTE                         0x0080  /* RI34MUTE */
+#define WM8990_RI34MUTE_BIT			7
+#define WM8990_RI34ZC                           0x0040  /* RI34ZC */
+#define WM8990_RI34ZC_BIT			6
+#define WM8990_RIN34VOL_MASK                    0x001F  /* RIN34VOL - [4:0] */
+#define WM8990_RIN34VOL_SHIFT			0
+
+/*
+ * R28 (0x1C) - Left Output Volume
+ */
+#define WM8990_OPVU                             0x0100  /* OPVU */
+#define WM8990_LOZC                             0x0080  /* LOZC */
+#define WM8990_LOZC_BIT				7
+#define WM8990_LOUTVOL_MASK                     0x007F  /* LOUTVOL - [6:0] */
+#define WM8990_LOUTVOL_SHIFT			0
+/*
+ * R29 (0x1D) - Right Output Volume
+ */
+#define WM8990_OPVU                             0x0100  /* OPVU */
+#define WM8990_ROZC                             0x0080  /* ROZC */
+#define WM8990_ROZC_BIT				7
+#define WM8990_ROUTVOL_MASK                     0x007F  /* ROUTVOL - [6:0] */
+#define WM8990_ROUTVOL_SHIFT			0
+/*
+ * R30 (0x1E) - Line Outputs Volume
+ */
+#define WM8990_LONMUTE                          0x0040  /* LONMUTE */
+#define WM8990_LONMUTE_BIT			6
+#define WM8990_LOPMUTE                          0x0020  /* LOPMUTE */
+#define WM8990_LOPMUTE_BIT			5
+#define WM8990_LOATTN                           0x0010  /* LOATTN */
+#define WM8990_LOATTN_BIT			4
+#define WM8990_RONMUTE                          0x0004  /* RONMUTE */
+#define WM8990_RONMUTE_BIT			2
+#define WM8990_ROPMUTE                          0x0002  /* ROPMUTE */
+#define WM8990_ROPMUTE_BIT			1
+#define WM8990_ROATTN                           0x0001  /* ROATTN */
+#define WM8990_ROATTN_BIT			0
+
+/*
+ * R31 (0x1F) - Out3/4 Volume
+ */
+#define WM8990_OUT3MUTE                         0x0020  /* OUT3MUTE */
+#define WM8990_OUT3MUTE_BIT			5
+#define WM8990_OUT3ATTN                         0x0010  /* OUT3ATTN */
+#define WM8990_OUT3ATTN_BIT			4
+#define WM8990_OUT4MUTE                         0x0002  /* OUT4MUTE */
+#define WM8990_OUT4MUTE_BIT			1
+#define WM8990_OUT4ATTN                         0x0001  /* OUT4ATTN */
+#define WM8990_OUT4ATTN_BIT			0
+
+/*
+ * R32 (0x20) - Left OPGA Volume
+ */
+#define WM8990_OPVU                             0x0100  /* OPVU */
+#define WM8990_LOPGAZC                          0x0080  /* LOPGAZC */
+#define WM8990_LOPGAZC_BIT			7
+#define WM8990_LOPGAVOL_MASK                    0x007F  /* LOPGAVOL - [6:0] */
+#define WM8990_LOPGAVOL_SHIFT			0
+
+/*
+ * R33 (0x21) - Right OPGA Volume
+ */
+#define WM8990_OPVU                             0x0100  /* OPVU */
+#define WM8990_ROPGAZC                          0x0080  /* ROPGAZC */
+#define WM8990_ROPGAZC_BIT			7
+#define WM8990_ROPGAVOL_MASK                    0x007F  /* ROPGAVOL - [6:0] */
+#define WM8990_ROPGAVOL_SHIFT			0
+/*
+ * R34 (0x22) - Speaker Volume
+ */
+#define WM8990_SPKVOL_MASK                      0x0003  /* SPKVOL - [1:0] */
+#define WM8990_SPKVOL_SHIFT			0
+
+/*
+ * R35 (0x23) - ClassD1
+ */
+#define WM8990_CDMODE                           0x0100  /* CDMODE */
+#define WM8990_CDMODE_BIT			8
+
+/*
+ * R37 (0x25) - ClassD3
+ */
+#define WM8990_DCGAIN_MASK                      0x0007  /* DCGAIN - [5:3] */
+#define WM8990_DCGAIN_SHIFT			3
+#define WM8990_ACGAIN_MASK                      0x0007  /* ACGAIN - [2:0] */
+#define WM8990_ACGAIN_SHIFT			0
+/*
+ * R39 (0x27) - Input Mixer1
+ */
+#define WM8990_AINLMODE_MASK                    0x000C  /* AINLMODE - [3:2] */
+#define WM8990_AINLMODE_SHIFT			2
+#define WM8990_AINRMODE_MASK                    0x0003  /* AINRMODE - [1:0] */
+#define WM8990_AINRMODE_SHIFT			0
+
+/*
+ * R40 (0x28) - Input Mixer2
+ */
+#define WM8990_LMP4				0x0080	/* LMP4 */
+#define WM8990_LMP4_BIT                         7	/* LMP4 */
+#define WM8990_LMN3                             0x0040  /* LMN3 */
+#define WM8990_LMN3_BIT                         6       /* LMN3 */
+#define WM8990_LMP2                             0x0020  /* LMP2 */
+#define WM8990_LMP2_BIT                         5       /* LMP2 */
+#define WM8990_LMN1                             0x0010  /* LMN1 */
+#define WM8990_LMN1_BIT                         4       /* LMN1 */
+#define WM8990_RMP4                             0x0008  /* RMP4 */
+#define WM8990_RMP4_BIT                         3       /* RMP4 */
+#define WM8990_RMN3                             0x0004  /* RMN3 */
+#define WM8990_RMN3_BIT                         2       /* RMN3 */
+#define WM8990_RMP2                             0x0002  /* RMP2 */
+#define WM8990_RMP2_BIT                         1       /* RMP2 */
+#define WM8990_RMN1                             0x0001  /* RMN1 */
+#define WM8990_RMN1_BIT                         0       /* RMN1 */
+
+/*
+ * R41 (0x29) - Input Mixer3
+ */
+#define WM8990_L34MNB                           0x0100  /* L34MNB */
+#define WM8990_L34MNB_BIT			8
+#define WM8990_L34MNBST                         0x0080  /* L34MNBST */
+#define WM8990_L34MNBST_BIT			7
+#define WM8990_L12MNB                           0x0020  /* L12MNB */
+#define WM8990_L12MNB_BIT			5
+#define WM8990_L12MNBST                         0x0010  /* L12MNBST */
+#define WM8990_L12MNBST_BIT			4
+#define WM8990_LDBVOL_MASK                      0x0007  /* LDBVOL - [2:0] */
+#define WM8990_LDBVOL_SHIFT			0
+
+/*
+ * R42 (0x2A) - Input Mixer4
+ */
+#define WM8990_R34MNB                           0x0100  /* R34MNB */
+#define WM8990_R34MNB_BIT			8
+#define WM8990_R34MNBST                         0x0080  /* R34MNBST */
+#define WM8990_R34MNBST_BIT			7
+#define WM8990_R12MNB                           0x0020  /* R12MNB */
+#define WM8990_R12MNB_BIT			5
+#define WM8990_R12MNBST                         0x0010  /* R12MNBST */
+#define WM8990_R12MNBST_BIT			4
+#define WM8990_RDBVOL_MASK                      0x0007  /* RDBVOL - [2:0] */
+#define WM8990_RDBVOL_SHIFT			0
+
+/*
+ * R43 (0x2B) - Input Mixer5
+ */
+#define WM8990_LI2BVOL_MASK                     0x07  /* LI2BVOL - [8:6] */
+#define WM8990_LI2BVOL_SHIFT			6
+#define WM8990_LR4BVOL_MASK                     0x07  /* LR4BVOL - [5:3] */
+#define WM8990_LR4BVOL_SHIFT			3
+#define WM8990_LL4BVOL_MASK                     0x07  /* LL4BVOL - [2:0] */
+#define WM8990_LL4BVOL_SHIFT			0
+
+/*
+ * R44 (0x2C) - Input Mixer6
+ */
+#define WM8990_RI2BVOL_MASK                     0x07  /* RI2BVOL - [8:6] */
+#define WM8990_RI2BVOL_SHIFT			6
+#define WM8990_RL4BVOL_MASK                     0x07  /* RL4BVOL - [5:3] */
+#define WM8990_RL4BVOL_SHIFT			3
+#define WM8990_RR4BVOL_MASK                     0x07  /* RR4BVOL - [2:0] */
+#define WM8990_RR4BVOL_SHIFT			0
+
+/*
+ * R45 (0x2D) - Output Mixer1
+ */
+#define WM8990_LRBLO                            0x0080  /* LRBLO */
+#define WM8990_LRBLO_BIT			7
+#define WM8990_LLBLO                            0x0040  /* LLBLO */
+#define WM8990_LLBLO_BIT			6
+#define WM8990_LRI3LO                           0x0020  /* LRI3LO */
+#define WM8990_LRI3LO_BIT			5
+#define WM8990_LLI3LO                           0x0010  /* LLI3LO */
+#define WM8990_LLI3LO_BIT			4
+#define WM8990_LR12LO                           0x0008  /* LR12LO */
+#define WM8990_LR12LO_BIT			3
+#define WM8990_LL12LO                           0x0004  /* LL12LO */
+#define WM8990_LL12LO_BIT			2
+#define WM8990_LDLO                             0x0001  /* LDLO */
+#define WM8990_LDLO_BIT				0
+
+/*
+ * R46 (0x2E) - Output Mixer2
+ */
+#define WM8990_RLBRO                            0x0080  /* RLBRO */
+#define WM8990_RLBRO_BIT			7
+#define WM8990_RRBRO                            0x0040  /* RRBRO */
+#define WM8990_RRBRO_BIT			6
+#define WM8990_RLI3RO                           0x0020  /* RLI3RO */
+#define WM8990_RLI3RO_BIT			5
+#define WM8990_RRI3RO                           0x0010  /* RRI3RO */
+#define WM8990_RRI3RO_BIT			4
+#define WM8990_RL12RO                           0x0008  /* RL12RO */
+#define WM8990_RL12RO_BIT			3
+#define WM8990_RR12RO                           0x0004  /* RR12RO */
+#define WM8990_RR12RO_BIT			2
+#define WM8990_RDRO                             0x0001  /* RDRO */
+#define WM8990_RDRO_BIT				0
+
+/*
+ * R47 (0x2F) - Output Mixer3
+ */
+#define WM8990_LLI3LOVOL_MASK                   0x07  /* LLI3LOVOL - [8:6] */
+#define WM8990_LLI3LOVOL_SHIFT			6
+#define WM8990_LR12LOVOL_MASK                   0x07  /* LR12LOVOL - [5:3] */
+#define WM8990_LR12LOVOL_SHIFT			3
+#define WM8990_LL12LOVOL_MASK                   0x07  /* LL12LOVOL - [2:0] */
+#define WM8990_LL12LOVOL_SHIFT			0
+
+/*
+ * R48 (0x30) - Output Mixer4
+ */
+#define WM8990_RRI3ROVOL_MASK                   0x07  /* RRI3ROVOL - [8:6] */
+#define WM8990_RRI3ROVOL_SHIFT			6
+#define WM8990_RL12ROVOL_MASK                   0x07  /* RL12ROVOL - [5:3] */
+#define WM8990_RL12ROVOL_SHIFT			3
+#define WM8990_RR12ROVOL_MASK                   0x07  /* RR12ROVOL - [2:0] */
+#define WM8990_RR12ROVOL_SHIFT			0
+
+/*
+ * R49 (0x31) - Output Mixer5
+ */
+#define WM8990_LRI3LOVOL_MASK                   0x07  /* LRI3LOVOL - [8:6] */
+#define WM8990_LRI3LOVOL_SHIFT			6
+#define WM8990_LRBLOVOL_MASK                    0x07  /* LRBLOVOL - [5:3] */
+#define WM8990_LRBLOVOL_SHIFT			3
+#define WM8990_LLBLOVOL_MASK                    0x07  /* LLBLOVOL - [2:0] */
+#define WM8990_LLBLOVOL_SHIFT			0
+
+/*
+ * R50 (0x32) - Output Mixer6
+ */
+#define WM8990_RLI3ROVOL_MASK                   0x07  /* RLI3ROVOL - [8:6] */
+#define WM8990_RLI3ROVOL_SHIFT			6
+#define WM8990_RLBROVOL_MASK                    0x07  /* RLBROVOL - [5:3] */
+#define WM8990_RLBROVOL_SHIFT			3
+#define WM8990_RRBROVOL_MASK                    0x07  /* RRBROVOL - [2:0] */
+#define WM8990_RRBROVOL_SHIFT			0
+
+/*
+ * R51 (0x33) - Out3/4 Mixer
+ */
+#define WM8990_VSEL_MASK                        0x0180  /* VSEL - [8:7] */
+#define WM8990_LI4O3                            0x0020  /* LI4O3 */
+#define WM8990_LI4O3_BIT			5
+#define WM8990_LPGAO3                           0x0010  /* LPGAO3 */
+#define WM8990_LPGAO3_BIT			4
+#define WM8990_RI4O4                            0x0002  /* RI4O4 */
+#define WM8990_RI4O4_BIT			1
+#define WM8990_RPGAO4                           0x0001  /* RPGAO4 */
+#define WM8990_RPGAO4_BIT			0
+/*
+ * R52 (0x34) - Line Mixer1
+ */
+#define WM8990_LLOPGALON                        0x0040  /* LLOPGALON */
+#define WM8990_LLOPGALON_BIT			6
+#define WM8990_LROPGALON                        0x0020  /* LROPGALON */
+#define WM8990_LROPGALON_BIT			5
+#define WM8990_LOPLON                           0x0010  /* LOPLON */
+#define WM8990_LOPLON_BIT			4
+#define WM8990_LR12LOP                          0x0004  /* LR12LOP */
+#define WM8990_LR12LOP_BIT			2
+#define WM8990_LL12LOP                          0x0002  /* LL12LOP */
+#define WM8990_LL12LOP_BIT			1
+#define WM8990_LLOPGALOP                        0x0001  /* LLOPGALOP */
+#define WM8990_LLOPGALOP_BIT			0
+/*
+ * R53 (0x35) - Line Mixer2
+ */
+#define WM8990_RROPGARON                        0x0040  /* RROPGARON */
+#define WM8990_RROPGARON_BIT			6
+#define WM8990_RLOPGARON                        0x0020  /* RLOPGARON */
+#define WM8990_RLOPGARON_BIT			5
+#define WM8990_ROPRON                           0x0010  /* ROPRON */
+#define WM8990_ROPRON_BIT			4
+#define WM8990_RL12ROP                          0x0004  /* RL12ROP */
+#define WM8990_RL12ROP_BIT			2
+#define WM8990_RR12ROP                          0x0002  /* RR12ROP */
+#define WM8990_RR12ROP_BIT			1
+#define WM8990_RROPGAROP                        0x0001  /* RROPGAROP */
+#define WM8990_RROPGAROP_BIT			0
+
+/*
+ * R54 (0x36) - Speaker Mixer
+ */
+#define WM8990_LB2SPK                           0x0080  /* LB2SPK */
+#define WM8990_LB2SPK_BIT			7
+#define WM8990_RB2SPK                           0x0040  /* RB2SPK */
+#define WM8990_RB2SPK_BIT			6
+#define WM8990_LI2SPK                           0x0020  /* LI2SPK */
+#define WM8990_LI2SPK_BIT			5
+#define WM8990_RI2SPK                           0x0010  /* RI2SPK */
+#define WM8990_RI2SPK_BIT			4
+#define WM8990_LOPGASPK                         0x0008  /* LOPGASPK */
+#define WM8990_LOPGASPK_BIT			3
+#define WM8990_ROPGASPK                         0x0004  /* ROPGASPK */
+#define WM8990_ROPGASPK_BIT			2
+#define WM8990_LDSPK                            0x0002  /* LDSPK */
+#define WM8990_LDSPK_BIT			1
+#define WM8990_RDSPK                            0x0001  /* RDSPK */
+#define WM8990_RDSPK_BIT			0
+
+/*
+ * R55 (0x37) - Additional Control
+ */
+#define WM8990_VROI                             0x0001  /* VROI */
+
+/*
+ * R56 (0x38) - AntiPOP1
+ */
+#define WM8990_DIS_LLINE                        0x0020  /* DIS_LLINE */
+#define WM8990_DIS_RLINE                        0x0010  /* DIS_RLINE */
+#define WM8990_DIS_OUT3                         0x0008  /* DIS_OUT3 */
+#define WM8990_DIS_OUT4                         0x0004  /* DIS_OUT4 */
+#define WM8990_DIS_LOUT                         0x0002  /* DIS_LOUT */
+#define WM8990_DIS_ROUT                         0x0001  /* DIS_ROUT */
+
+/*
+ * R57 (0x39) - AntiPOP2
+ */
+#define WM8990_SOFTST                           0x0040  /* SOFTST */
+#define WM8990_BUFIOEN                          0x0008  /* BUFIOEN */
+#define WM8990_BUFDCOPEN                        0x0004  /* BUFDCOPEN */
+#define WM8990_POBCTRL                          0x0002  /* POBCTRL */
+#define WM8990_VMIDTOG                          0x0001  /* VMIDTOG */
+
+/*
+ * R58 (0x3A) - MICBIAS
+ */
+#define WM8990_MCDSCTH_MASK                     0x00C0  /* MCDSCTH - [7:6] */
+#define WM8990_MCDTHR_MASK                      0x0038  /* MCDTHR - [5:3] */
+#define WM8990_MCD                              0x0004  /* MCD */
+#define WM8990_MBSEL                            0x0001  /* MBSEL */
+
+/*
+ * R60 (0x3C) - PLL1
+ */
+#define WM8990_SDM                              0x0080  /* SDM */
+#define WM8990_PRESCALE                         0x0040  /* PRESCALE */
+#define WM8990_PLLN_MASK                        0x000F  /* PLLN - [3:0] */
+
+/*
+ * R61 (0x3D) - PLL2
+ */
+#define WM8990_PLLK1_MASK                       0x00FF  /* PLLK1 - [7:0] */
+
+/*
+ * R62 (0x3E) - PLL3
+ */
+#define WM8990_PLLK2_MASK                       0x00FF  /* PLLK2 - [7:0] */
+
+/*
+ * R63 (0x3F) - Internal Driver Bits
+ */
+#define WM8990_INMIXL_PWR_BIT			0
+#define WM8990_AINLMUX_PWR_BIT			1
+#define WM8990_INMIXR_PWR_BIT			2
+#define WM8990_AINRMUX_PWR_BIT			3
+
+struct wm8990_setup_data {
+	unsigned short i2c_address;
+};
+
+#define WM8990_MCLK_DIV 0
+#define WM8990_DACCLK_DIV 1
+#define WM8990_ADCCLK_DIV 2
+#define WM8990_BCLK_DIV 3
+
+extern struct snd_soc_dai wm8990_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8990;
+
+#endif	/* __WM8990REGISTERDEFS_H__ */
+/*------------------------------ END OF FILE ---------------------------------*/
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 76c1e2d33e7d..9fc8edd82225 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -9,9 +9,6 @@
  *  under  the terms of  the GNU General  Public License as published by the
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
- *
- *  Revision history
- *    4th Feb 2006   Initial version.
  */
 
 #include <linux/init.h>
@@ -25,6 +22,7 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
+#include "wm9712.h"
 
 #define WM9712_VERSION "0.4"
 
@@ -351,7 +349,7 @@ SND_SOC_DAPM_INPUT("MIC1"),
 SND_SOC_DAPM_INPUT("MIC2"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* virtual mixer - mixes left & right channels for spk and mono */
 	{"AC97 Mixer", NULL, "Left DAC"},
 	{"AC97 Mixer", NULL, "Right DAC"},
@@ -446,21 +444,14 @@ static const char *audio_map[][3] = {
 	{"Speaker PGA", NULL, "Speaker Mux"},
 	{"LOUT2", NULL, "Speaker PGA"},
 	{"ROUT2", NULL, "Speaker PGA"},
-
-	{NULL, NULL, NULL},
 };
 
 static int wm9712_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
+				  ARRAY_SIZE(wm9712_dapm_widgets));
 
-	/* set up audio path connects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-					   audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -541,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
-struct snd_soc_codec_dai wm9712_dai[] = {
+struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
 	.type = SND_SOC_DAI_AC97_BUS,
@@ -574,23 +565,23 @@ struct snd_soc_codec_dai wm9712_dai[] = {
 };
 EXPORT_SYMBOL_GPL(wm9712_dai);
 
-static int wm9712_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm9712_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		ac97_write(codec, AC97_POWERDOWN, 0x0000);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		/* disable everything including AC link */
 		ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
 		ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -598,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
 {
 	if (try_warm && soc_ac97_ops.warm_reset) {
 		soc_ac97_ops.warm_reset(codec->ac97);
-		if (!(ac97_read(codec, 0) & 0x8000))
+		if (ac97_read(codec, 0) == wm9712_reg[0])
 			return 1;
 	}
 
 	soc_ac97_ops.reset(codec->ac97);
-	if (ac97_read(codec, 0) & 0x8000)
+	if (ac97_read(codec, 0) != wm9712_reg[0])
 		goto err;
 	return 0;
 
@@ -618,7 +609,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->codec;
 
-	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
+	wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
 
@@ -635,7 +626,7 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 		return ret;
 	}
 
-	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	if (ret == 0) {
 		/* Sync reg_cache with the hardware after cold reset */
@@ -647,8 +638,8 @@ static int wm9712_soc_resume(struct platform_device *pdev)
 		}
 	}
 
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
-		wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0);
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		wm9712_set_bias_level(codec, SND_SOC_BIAS_ON);
 
 	return ret;
 }
@@ -682,7 +673,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	codec->num_dai = ARRAY_SIZE(wm9712_dai);
 	codec->write = ac97_write;
 	codec->read = ac97_read;
-	codec->dapm_event = wm9712_dapm_event;
+	codec->set_bias_level = wm9712_set_bias_level;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -706,7 +697,7 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	/* set alc mux to none */
 	ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
 
-	wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm9712_add_controls(codec);
 	wm9712_add_widgets(codec);
 	ret = snd_soc_register_card(socdev);
diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h
index 719105d61e65..d29e8a18ca6d 100644
--- a/sound/soc/codecs/wm9712.h
+++ b/sound/soc/codecs/wm9712.h
@@ -8,7 +8,7 @@
 #define WM9712_DAI_AC97_HIFI	0
 #define WM9712_DAI_AC97_AUX		1
 
-extern struct snd_soc_codec_dai wm9712_dai[2];
+extern struct snd_soc_dai wm9712_dai[2];
 extern struct snd_soc_codec_device soc_codec_dev_wm9712;
 
 #endif
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 1f241161445c..38d1fe0971fc 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -10,9 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    4th Feb 2006   Initial version.
- *
  *  Features:-
  *
  *   o Support for AC97 Codec, Voice DAC and Aux DAC
@@ -456,7 +453,7 @@ SND_SOC_DAPM_INPUT("MIC2B"),
 SND_SOC_DAPM_VMID("VMID"),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* left HP mixer */
 	{"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
 	{"Left HP Mixer", "Voice Playback Switch",   "Voice DAC"},
@@ -607,21 +604,14 @@ static const char *audio_map[][3] = {
 	{"Capture Mono Mux", "Stereo", "Capture Mixer"},
 	{"Capture Mono Mux", "Left", "Left Capture Source"},
 	{"Capture Mono Mux", "Right", "Right Capture Source"},
-
-	{NULL, NULL, NULL},
 };
 
 static int wm9713_add_widgets(struct snd_soc_codec *codec)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
+				  ARRAY_SIZE(wm9713_dapm_widgets));
 
-	/* set up audio path audio_mapnects */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	snd_soc_dapm_new_widgets(codec);
 	return 0;
@@ -799,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_pll(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;
@@ -810,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
  * Tristate the PCM DAI lines, tristate can be disabled by calling
  * wm9713_set_dai_fmt()
  */
-static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
 	int tristate)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -826,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai,
  * Configure WM9713 clock dividers.
  * Voice DAC needs 256 FS
  */
-static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -868,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai,
 	return 0;
 }
 
-static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
+static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -886,7 +876,7 @@ static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
 		gpio |= 0x0018;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
-		reg |= 0x0200;
+		reg |= 0x2000;
 		gpio |= 0x001a;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFM:
@@ -1011,15 +1001,24 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream)
 	return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
 }
 
-#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
-		SNDRV_PCM_RATE_48000)
+#define WM9713_RATES (SNDRV_PCM_RATE_8000  |	\
+		      SNDRV_PCM_RATE_11025 |	\
+		      SNDRV_PCM_RATE_22050 |	\
+		      SNDRV_PCM_RATE_44100 |	\
+		      SNDRV_PCM_RATE_48000)
+
+#define WM9713_PCM_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)
 
 #define WM9713_PCM_FORMATS \
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
-struct snd_soc_codec_dai wm9713_dai[] = {
+struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
 	.type = SND_SOC_DAI_AC97_BUS,
@@ -1061,13 +1060,13 @@ struct snd_soc_codec_dai wm9713_dai[] = {
 		.stream_name = "Voice Playback",
 		.channels_min = 1,
 		.channels_max = 1,
-		.rates = WM9713_RATES,
+		.rates = WM9713_PCM_RATES,
 		.formats = WM9713_PCM_FORMATS,},
 	.capture = {
 		.stream_name = "Voice Capture",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = WM9713_RATES,
+		.rates = WM9713_PCM_RATES,
 		.formats = WM9713_PCM_FORMATS,},
 	.ops = {
 		.hw_params = wm9713_pcm_hw_params,
@@ -1086,44 +1085,44 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
 {
 	if (try_warm && soc_ac97_ops.warm_reset) {
 		soc_ac97_ops.warm_reset(codec->ac97);
-		if (!(ac97_read(codec, 0) & 0x8000))
+		if (ac97_read(codec, 0) == wm9713_reg[0])
 			return 1;
 	}
 
 	soc_ac97_ops.reset(codec->ac97);
-	if (ac97_read(codec, 0) & 0x8000)
+	if (ac97_read(codec, 0) != wm9713_reg[0])
 		return -EIO;
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm9713_reset);
 
-static int wm9713_dapm_event(struct snd_soc_codec *codec, int event)
+static int wm9713_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
 {
 	u16 reg;
 
-	switch (event) {
-	case SNDRV_CTL_POWER_D0: /* full On */
+	switch (level) {
+	case SND_SOC_BIAS_ON:
 		/* enable thermal shutdown */
 		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
 		ac97_write(codec, AC97_EXTENDED_MID, reg);
 		break;
-	case SNDRV_CTL_POWER_D1: /* partial On */
-	case SNDRV_CTL_POWER_D2: /* partial On */
+	case SND_SOC_BIAS_PREPARE:
 		break;
-	case SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	case SND_SOC_BIAS_STANDBY:
 		/* enable master bias and vmid */
 		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
 		ac97_write(codec, AC97_EXTENDED_MID, reg);
 		ac97_write(codec, AC97_POWERDOWN, 0x0000);
 		break;
-	case SNDRV_CTL_POWER_D3cold: /* Off, without power */
+	case SND_SOC_BIAS_OFF:
 		/* disable everything including AC link */
 		ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
 		ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
 		ac97_write(codec, AC97_POWERDOWN, 0xffff);
 		break;
 	}
-	codec->dapm_state = event;
+	codec->bias_level = level;
 	return 0;
 }
 
@@ -1160,7 +1159,7 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 		return ret;
 	}
 
-	wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* do we need to re-start the PLL ? */
 	if (wm9713->pll_out)
@@ -1176,8 +1175,8 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 		}
 	}
 
-	if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0)
-		wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0);
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		wm9713_set_bias_level(codec, SND_SOC_BIAS_ON);
 
 	return ret;
 }
@@ -1216,7 +1215,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	codec->num_dai = ARRAY_SIZE(wm9713_dai);
 	codec->write = ac97_write;
 	codec->read = ac97_read;
-	codec->dapm_event = wm9713_dapm_event;
+	codec->set_bias_level = wm9713_set_bias_level;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -1238,7 +1237,7 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 		goto reset_err;
 	}
 
-	wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* unmute the adc - move to kcontrol */
 	reg = ac97_read(codec, AC97_CD) & 0x7fff;
diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h
index d357b6c8134b..63b8d81756e3 100644
--- a/sound/soc/codecs/wm9713.h
+++ b/sound/soc/codecs/wm9713.h
@@ -46,7 +46,7 @@
 #define WM9713_DAI_PCM_VOICE	2
 
 extern struct snd_soc_codec_device soc_codec_dev_wm9713;
-extern struct snd_soc_codec_dai wm9713_dai[3];
+extern struct snd_soc_dai wm9713_dai[3];
 
 int wm9713_reset(struct snd_soc_codec *codec,  int try_warm);
 
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 20680c551aab..8f7e33834902 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -1,6 +1,6 @@
 config SND_DAVINCI_SOC
 	tristate "SoC Audio for the TI DAVINCI chip"
-	depends on ARCH_DAVINCI && SND_SOC
+	depends on ARCH_DAVINCI
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the DAVINCI AC97 or I2S interface. You will also need
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index fcd165240333..5e2c306399ed 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -33,24 +33,24 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 			 struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	int ret = 0;
 
 	/* 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_CBM_CFM);
 	if (ret < 0)
 		return ret;
 
 	/* set cpu DAI configuration */
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
 				       SND_SOC_DAIFMT_IB_NF);
 	if (ret < 0)
 		return ret;
 
 	/* set the codec system clock */
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK,
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK,
 					    SND_SOC_CLOCK_OUT);
 	if (ret < 0)
 		return ret;
@@ -71,7 +71,7 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 };
 
 /* davinci-evm machine audio_mapnections to the codec pins */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	/* Headphone connected to HPLOUT, HPROUT */
 	{"Headphone Jack", NULL, "HPLOUT"},
 	{"Headphone Jack", NULL, "HPROUT"},
@@ -90,36 +90,30 @@ static const char *audio_map[][3] = {
 	{"LINE2L", NULL, "Line In"},
 	{"LINE1R", NULL, "Line In"},
 	{"LINE2R", NULL, "Line In"},
-
-	{NULL, NULL, NULL},
 };
 
 /* Logic for a aic3x as connected on a davinci-evm */
 static int evm_aic3x_init(struct snd_soc_codec *codec)
 {
-	int i;
-
 	/* Add davinci-evm specific widgets */
-	for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* Set up davinci-evm specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-					   audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
 	/* not connected */
-	snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
-	snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0);
-	snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0);
+	snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
+	snd_soc_dapm_disable_pin(codec, "HPLCOM");
+	snd_soc_dapm_disable_pin(codec, "HPRCOM");
 
 	/* always connected */
-	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
-	snd_soc_dapm_set_endpoint(codec, "Line Out", 1);
-	snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
-	snd_soc_dapm_set_endpoint(codec, "Line In", 1);
+	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Line Out");
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+	snd_soc_dapm_enable_pin(codec, "Line In");
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 
 	return 0;
 }
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index c421774b33ee..5ebf1ff71c4c 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -147,7 +147,7 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream)
 static int davinci_i2s_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data;
 
 	cpu_dai->dma_data = dev->dma_params[substream->stream];
@@ -155,7 +155,7 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+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;
@@ -295,11 +295,12 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
 	return ret;
 }
 
-static int davinci_i2s_probe(struct platform_device *pdev)
+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_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
 	struct davinci_mcbsp_dev *dev;
 	struct resource *mem, *ioarea;
 	struct evm_snd_platform_data *pdata;
@@ -356,11 +357,12 @@ err_release_region:
 	return ret;
 }
 
-static void davinci_i2s_remove(struct platform_device *pdev)
+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_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
+	struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai;
 	struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
 	struct resource *mem;
 
@@ -376,7 +378,7 @@ static void davinci_i2s_remove(struct platform_device *pdev)
 
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
-struct snd_soc_cpu_dai davinci_i2s_dai = {
+struct snd_soc_dai davinci_i2s_dai = {
 	.name = "davinci-i2s",
 	.id = 0,
 	.type = SND_SOC_DAI_I2S,
diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h
index 9592d17db320..c5b091807eec 100644
--- a/sound/soc/davinci/davinci-i2s.h
+++ b/sound/soc/davinci/davinci-i2s.h
@@ -12,6 +12,6 @@
 #ifndef _DAVINCI_I2S_H
 #define _DAVINCI_I2S_H
 
-extern struct snd_soc_cpu_dai davinci_i2s_dai;
+extern struct snd_soc_dai davinci_i2s_dai;
 
 #endif
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 6a76927c9971..6a5e56a782bb 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -350,7 +350,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm)
 static u64 davinci_pcm_dmamask = 0xffffffff;
 
 static int davinci_pcm_new(struct snd_card *card,
-			   struct snd_soc_codec_dai *dai, struct snd_pcm *pcm)
+			   struct snd_soc_dai *dai, struct snd_pcm *pcm)
 {
 	int ret;
 
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 257101f44e9e..3368ace60977 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,8 +1,6 @@
-menu "ALSA SoC audio for Freescale SOCs"
-
 config SND_SOC_MPC8610
 	bool "ALSA SoC support for the MPC8610 SOC"
-	depends on SND_SOC && MPC8610_HPCD
+	depends on MPC8610_HPCD
 	default y if MPC8610
 	help
 	  Say Y if you want to add support for codecs attached to the SSI
@@ -16,5 +14,3 @@ config SND_SOC_MPC8610_HPCD
 	default y if MPC8610_HPCD
 	help
 	  Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
-
-endmenu
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 78de7168d2ba..da2bc5902864 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -282,7 +282,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
  * once for each .dai_link in the machine driver's snd_soc_machine
  * structure.
  */
-static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
 	struct snd_pcm *pcm)
 {
 	static u64 fsl_dma_dmamask = DMA_BIT_MASK(32);
diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h
index 430a6ce8b0d0..385d4a42603c 100644
--- a/sound/soc/fsl/fsl_dma.h
+++ b/sound/soc/fsl/fsl_dma.h
@@ -126,7 +126,7 @@ struct fsl_dma_link_descriptor {
 	u8 res[4];      /* Reserved */
 } __attribute__ ((aligned(32), packed));
 
-/* DMA information needed to create a snd_soc_cpu_dai object
+/* DMA information needed to create a snd_soc_dai object
  *
  * ssi_stx_phys: bus address of SSI STX register to use
  * ssi_srx_phys: bus address of SSI SRX register to use
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index f588545698f3..71bff33f5528 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -82,7 +82,7 @@ struct fsl_ssi_private {
 	struct device *dev;
 	unsigned int playback;
 	unsigned int capture;
-	struct snd_soc_cpu_dai cpu_dai;
+	struct snd_soc_dai cpu_dai;
 	struct device_attribute dev_attr;
 
 	struct {
@@ -479,7 +479,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream)
  * @freq: the frequency of the given clock ID, currently ignored
  * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master)
  */
-static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai,
 			      int clk_id, unsigned int freq, int dir)
 {
 
@@ -497,7 +497,7 @@ static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
  *
  * @format: one of SND_SOC_DAIFMT_xxx
  */
-static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
+static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 {
 	return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
 }
@@ -505,7 +505,7 @@ static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format)
 /**
  * fsl_ssi_dai_template: template CPU DAI for the SSI
  */
-static struct snd_soc_cpu_dai fsl_ssi_dai_template = {
+static struct snd_soc_dai fsl_ssi_dai_template = {
 	.playback = {
 		/* The SSI does not support monaural audio. */
 		.channels_min = 2,
@@ -569,15 +569,15 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev,
 }
 
 /**
- * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure
+ * fsl_ssi_create_dai: create a snd_soc_dai structure
  *
- * This function is called by the machine driver to create a snd_soc_cpu_dai
+ * This function is called by the machine driver to create a snd_soc_dai
  * structure.  The function creates an ssi_private object, which contains
- * the snd_soc_cpu_dai.  It also creates the sysfs statistics device.
+ * the snd_soc_dai.  It also creates the sysfs statistics device.
  */
-struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
+struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 {
-	struct snd_soc_cpu_dai *fsl_ssi_dai;
+	struct snd_soc_dai *fsl_ssi_dai;
 	struct fsl_ssi_private *ssi_private;
 	int ret = 0;
 	struct device_attribute *dev_attr;
@@ -588,7 +588,7 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 		return NULL;
 	}
 	memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template,
-	       sizeof(struct snd_soc_cpu_dai));
+	       sizeof(struct snd_soc_dai));
 
 	fsl_ssi_dai = &ssi_private->cpu_dai;
 	dev_attr = &ssi_private->dev_attr;
@@ -623,11 +623,11 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 EXPORT_SYMBOL_GPL(fsl_ssi_create_dai);
 
 /**
- * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object
+ * fsl_ssi_destroy_dai: destroy the snd_soc_dai object
  *
  * This function undoes the operations of fsl_ssi_create_dai()
  */
-void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai)
+void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
 {
 	struct fsl_ssi_private *ssi_private =
 	container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai);
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index c5ce88e15651..83b44d700e33 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -217,8 +217,8 @@ struct fsl_ssi_info {
 	struct device *dev;
 };
 
-struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
-void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai);
+struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
+void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai);
 
 #endif
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index a00aac7a71f1..4bdc9d8fc90e 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -58,9 +58,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device)
 		sound_device->dev.platform_data;
 
 	/* Program the signal routing between the SSI and the DMA */
-	guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1,
+	guts_set_dmacr(machine_data->guts, machine_data->dma_id,
 		machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI);
-	guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1,
+	guts_set_dmacr(machine_data->guts, machine_data->dma_id,
 		machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI);
 
 	guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id,
@@ -96,62 +96,52 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device)
 static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct mpc8610_hpcd_data *machine_data =
 		rtd->socdev->dev->platform_data;
 	int ret = 0;
 
 	/* Tell the CPU driver what the serial protocol is. */
-	if (cpu_dai->dai_ops.set_fmt) {
-		ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
-			machine_data->dai_format);
-		if (ret < 0) {
-			dev_err(substream->pcm->card->dev,
-				"could not set CPU driver audio format\n");
-			return ret;
-		}
+	ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format);
+	if (ret < 0) {
+		dev_err(substream->pcm->card->dev,
+			"could not set CPU driver audio format\n");
+		return ret;
 	}
 
 	/* Tell the codec driver what the serial protocol is. */
-	if (codec_dai->dai_ops.set_fmt) {
-		ret = codec_dai->dai_ops.set_fmt(codec_dai,
-			machine_data->dai_format);
-		if (ret < 0) {
-			dev_err(substream->pcm->card->dev,
-				"could not set codec driver audio format\n");
-			return ret;
-		}
+	ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format);
+	if (ret < 0) {
+		dev_err(substream->pcm->card->dev,
+			"could not set codec driver audio format\n");
+		return ret;
 	}
 
 	/*
 	 * Tell the CPU driver what the clock frequency is, and whether it's a
 	 * slave or master.
 	 */
-	if (cpu_dai->dai_ops.set_sysclk) {
-		ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0,
-			machine_data->clk_frequency,
-			machine_data->cpu_clk_direction);
-		if (ret < 0) {
-			dev_err(substream->pcm->card->dev,
-				"could not set CPU driver clock parameters\n");
-			return ret;
-		}
+	ret = snd_soc_dai_set_sysclk(cpu_dai, 0,
+					machine_data->clk_frequency,
+					machine_data->cpu_clk_direction);
+	if (ret < 0) {
+		dev_err(substream->pcm->card->dev,
+			"could not set CPU driver clock parameters\n");
+		return ret;
 	}
 
 	/*
 	 * Tell the codec driver what the MCLK frequency is, and whether it's
 	 * a slave or master.
 	 */
-	if (codec_dai->dai_ops.set_sysclk) {
-		ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0,
-			machine_data->clk_frequency,
-			machine_data->codec_clk_direction);
-		if (ret < 0) {
-			dev_err(substream->pcm->card->dev,
-				"could not set codec driver clock params\n");
-			return ret;
-		}
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+					machine_data->clk_frequency,
+					machine_data->codec_clk_direction);
+	if (ret < 0) {
+		dev_err(substream->pcm->card->dev,
+			"could not set codec driver clock params\n");
+		return ret;
 	}
 
 	return 0;
@@ -170,9 +160,9 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device)
 
 	/* Restore the signal routing */
 
-	guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1,
+	guts_set_dmacr(machine_data->guts, machine_data->dma_id,
 		machine_data->dma_channel_id[0], 0);
-	guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1,
+	guts_set_dmacr(machine_data->guts, machine_data->dma_id,
 		machine_data->dma_channel_id[1], 0);
 
 	switch (machine_data->ssi_id) {
@@ -182,7 +172,7 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device)
 		break;
 	case 1:
 		clrsetbits_be32(&machine_data->guts->pmuxcr,
-			CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA);
+			CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA);
 		break;
 	}
 
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 0230d83e8e5e..aea27e70043c 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -1,5 +1,3 @@
-menu "SoC Audio for the Texas Instruments OMAP"
-
 config SND_OMAP_SOC
 	tristate "SoC Audio for the Texas Instruments OMAP chips"
 	depends on ARCH_OMAP && SND_SOC
@@ -15,5 +13,3 @@ config SND_OMAP_SOC_N810
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
-
-endmenu
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 6533563a6011..02cec96859b8 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -30,15 +30,15 @@
 
 #include <asm/mach-types.h>
 #include <asm/arch/hardware.h>
-#include <asm/arch/gpio.h>
+#include <linux/gpio.h>
 #include <asm/arch/mcbsp.h>
 
 #include "omap-mcbsp.h"
 #include "omap-pcm.h"
 #include "../codecs/tlv320aic3x.h"
 
-#define RX44_HEADSET_AMP_GPIO	10
-#define RX44_SPEAKER_AMP_GPIO	101
+#define N810_HEADSET_AMP_GPIO	10
+#define N810_SPEAKER_AMP_GPIO	101
 
 static struct clk *sys_clkout2;
 static struct clk *sys_clkout2_src;
@@ -46,13 +46,26 @@ static struct clk *func96m_clk;
 
 static int n810_spk_func;
 static int n810_jack_func;
+static int n810_dmic_func;
 
 static void n810_ext_control(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func);
-	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func);
+	if (n810_spk_func)
+		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(codec, "Ext Spk");
+
+	if (n810_jack_func)
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
 
-	snd_soc_dapm_sync_endpoints(codec);
+	if (n810_dmic_func)
+		snd_soc_dapm_enable_pin(codec, "DMic");
+	else
+		snd_soc_dapm_disable_pin(codec, "DMic");
+
+	snd_soc_dapm_sync(codec);
 }
 
 static int n810_startup(struct snd_pcm_substream *substream)
@@ -73,12 +86,12 @@ static int n810_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	int err;
 
 	/* Set codec DAI configuration */
-	err = codec_dai->dai_ops.set_fmt(codec_dai,
+	err = snd_soc_dai_set_fmt(codec_dai,
 					 SND_SOC_DAIFMT_I2S |
 					 SND_SOC_DAIFMT_NB_NF |
 					 SND_SOC_DAIFMT_CBM_CFM);
@@ -86,7 +99,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream,
 		return err;
 
 	/* Set cpu DAI configuration */
-	err = cpu_dai->dai_ops.set_fmt(cpu_dai,
+	err = snd_soc_dai_set_fmt(cpu_dai,
 				       SND_SOC_DAIFMT_I2S |
 				       SND_SOC_DAIFMT_NB_NF |
 				       SND_SOC_DAIFMT_CBM_CFM);
@@ -94,7 +107,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream,
 		return err;
 
 	/* Set the codec system clock for DAC and ADC */
-	err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000,
+	err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000,
 					    SND_SOC_CLOCK_IN);
 
 	return err;
@@ -150,13 +163,35 @@ static int n810_set_jack(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+static int n810_get_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = n810_dmic_func;
+
+	return 0;
+}
+
+static int n810_set_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+
+	if (n810_dmic_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	n810_dmic_func = ucontrol->value.integer.value[0];
+	n810_ext_control(codec);
+
+	return 1;
+}
+
 static int n810_spk_event(struct snd_soc_dapm_widget *w,
 			  struct snd_kcontrol *k, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
-		omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 1);
+		gpio_set_value(N810_SPEAKER_AMP_GPIO, 1);
 	else
-		omap_set_gpio_dataout(RX44_SPEAKER_AMP_GPIO, 0);
+		gpio_set_value(N810_SPEAKER_AMP_GPIO, 0);
 
 	return 0;
 }
@@ -165,9 +200,9 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w,
 			   struct snd_kcontrol *k, int event)
 {
 	if (SND_SOC_DAPM_EVENT_ON(event))
-		omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 1);
+		gpio_set_value(N810_HEADSET_AMP_GPIO, 1);
 	else
-		omap_set_gpio_dataout(RX44_HEADSET_AMP_GPIO, 0);
+		gpio_set_value(N810_HEADSET_AMP_GPIO, 0);
 
 	return 0;
 }
@@ -175,21 +210,27 @@ static int n810_jack_event(struct snd_soc_dapm_widget *w,
 static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
 	SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
 	SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
+	SND_SOC_DAPM_MIC("DMic", NULL),
 };
 
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 	{"Headphone Jack", NULL, "HPLOUT"},
 	{"Headphone Jack", NULL, "HPROUT"},
 
 	{"Ext Spk", NULL, "LLOUT"},
 	{"Ext Spk", NULL, "RLOUT"},
+
+	{"DMic Rate 64", NULL, "Mic Bias 2V"},
+	{"Mic Bias 2V", NULL, "DMic"},
 };
 
 static const char *spk_function[] = {"Off", "On"};
 static const char *jack_function[] = {"Off", "Headphone"};
+static const char *input_function[] = {"ADC", "Digital Mic"};
 static const struct soc_enum n810_enum[] = {
 	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
 	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
 };
 
 static const struct snd_kcontrol_new aic33_n810_controls[] = {
@@ -197,6 +238,8 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = {
 		     n810_get_spk, n810_set_spk),
 	SOC_ENUM_EXT("Jack Function", n810_enum[1],
 		     n810_get_jack, n810_set_jack),
+	SOC_ENUM_EXT("Input Select",  n810_enum[2],
+		     n810_get_input, n810_set_input),
 };
 
 static int n810_aic33_init(struct snd_soc_codec *codec)
@@ -204,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
 	int i, err;
 
 	/* Not connected */
-	snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0);
-	snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0);
-	snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0);
+	snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
+	snd_soc_dapm_disable_pin(codec, "HPLCOM");
+	snd_soc_dapm_disable_pin(codec, "HPRCOM");
 
 	/* Add N810 specific controls */
 	for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
@@ -217,15 +260,13 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
 	}
 
 	/* Add N810 specific widgets */
-	for (i = 0; i < ARRAY_SIZE(aic33_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &aic33_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
+				  ARRAY_SIZE(aic33_dapm_widgets));
 
 	/* Set up N810 specific audio path audio_map */
-	for (i = 0; i < ARRAY_SIZE(audio_map); i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 
 	return 0;
 }
@@ -250,6 +291,8 @@ static struct snd_soc_machine snd_soc_machine_n810 = {
 /* Audio private data */
 static struct aic3x_setup_data n810_aic33_setup = {
 	.i2c_address = 0x18,
+	.gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
+	.gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
 };
 
 /* Audio subsystem */
@@ -267,7 +310,7 @@ static int __init n810_soc_init(void)
 	int err;
 	struct device *dev;
 
-	if (!machine_is_nokia_n810())
+	if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
 		return -ENODEV;
 
 	n810_snd_device = platform_device_alloc("soc-audio", -1);
@@ -305,12 +348,12 @@ static int __init n810_soc_init(void)
 	clk_set_parent(sys_clkout2_src, func96m_clk);
 	clk_set_rate(sys_clkout2, 12000000);
 
-	if (omap_request_gpio(RX44_HEADSET_AMP_GPIO) < 0)
+	if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0)
 		BUG();
-	if (omap_request_gpio(RX44_SPEAKER_AMP_GPIO) < 0)
+	if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)
 		BUG();
-	omap_set_gpio_direction(RX44_HEADSET_AMP_GPIO, 0);
-	omap_set_gpio_direction(RX44_SPEAKER_AMP_GPIO, 0);
+	gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
+	gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
 
 	return 0;
 err2:
@@ -325,6 +368,9 @@ err1:
 
 static void __exit n810_soc_exit(void)
 {
+	gpio_free(N810_SPEAKER_AMP_GPIO);
+	gpio_free(N810_HEADSET_AMP_GPIO);
+
 	platform_device_unregister(n810_snd_device);
 }
 
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 40d87e6d0de8..00b0c9d73cd4 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -103,7 +103,7 @@ static const unsigned long omap2420_mcbsp_port[][2] = {};
 static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	int err = 0;
 
@@ -116,7 +116,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
 static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 
 	if (!cpu_dai->active) {
@@ -128,7 +128,7 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream)
 static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	int err = 0;
 
@@ -157,7 +157,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 				    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	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;
@@ -223,7 +223,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
  * This must be called before _set_clkdiv and _set_sysclk since McBSP register
  * cache is initialized here
  */
-static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 				      unsigned int fmt)
 {
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
@@ -292,7 +292,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
 	return 0;
 }
 
-static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
 				     int div_id, int div)
 {
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
@@ -347,7 +347,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 	return 0;
 }
 
-static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 					 int clk_id, unsigned int freq,
 					 int dir)
 {
@@ -376,7 +376,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
 	return err;
 }
 
-struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = {
+struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = {
 {
 	.name = "omap-mcbsp-dai",
 	.id = 0,
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index 9965fd4b0427..ed8afb550671 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -44,6 +44,6 @@ enum omap_mcbsp_div {
  */
 #define NUM_LINKS	1
 
-extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS];
+extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
 
 #endif
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 62370202c649..e092f3d836d0 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -316,7 +316,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
 	}
 }
 
-int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 		 struct snd_pcm *pcm)
 {
 	int ret = 0;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 484f883459e0..12f6ac99b04c 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,6 +1,6 @@
 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
-	depends on ARCH_PXA && SND_SOC
+	depends on ARCH_PXA
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the PXA2xx AC97, I2S or SSP interface. You will also need
@@ -62,3 +62,12 @@ config SND_PXA2XX_SOC_E800
 	help
 	  Say Y if you want to add support for SoC audio on the
 	  Toshiba e800 PDA
+
+config SND_PXA2XX_SOC_EM_X270
+	tristate "SoC Audio support for CompuLab EM-x270"
+	depends on SND_PXA2XX_SOC && MACH_EM_X270
+	select SND_PXA2XX_SOC_AC97
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on
+	  CompuLab EM-x270.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 04e5646f75ba..5bc8edf9dca9 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -13,10 +13,11 @@ snd-soc-poodle-objs := poodle.o
 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
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
 obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
 obj-$(CONFIG_SND_PXA2XX_SOC_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
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 7f32a1167572..c0294464a23a 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -11,10 +11,6 @@
  *  under  the terms of  the GNU General  Public License as published by the
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
- *
- *  Revision history
- *    30th Nov 2005   Initial version.
- *
  */
 
 #include <linux/module.h>
@@ -54,47 +50,51 @@ static int corgi_spk_func;
 
 static void corgi_ext_control(struct snd_soc_codec *codec)
 {
-	int spk = 0, mic = 0, line = 0, hp = 0, hs = 0;
-
 	/* set up jack connection */
 	switch (corgi_jack_func) {
 	case CORGI_HP:
-		hp = 1;
 		/* set = unmute headphone */
 		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
 		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		snd_soc_dapm_disable_pin(codec, "Mic Jack");
+		snd_soc_dapm_disable_pin(codec, "Line Jack");
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		break;
 	case CORGI_MIC:
-		mic = 1;
 		/* reset = mute headphone */
 		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
 		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		snd_soc_dapm_enable_pin(codec, "Mic Jack");
+		snd_soc_dapm_disable_pin(codec, "Line Jack");
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		break;
 	case CORGI_LINE:
-		line = 1;
 		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
 		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		snd_soc_dapm_disable_pin(codec, "Mic Jack");
+		snd_soc_dapm_enable_pin(codec, "Line Jack");
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		break;
 	case CORGI_HEADSET:
-		hs = 1;
-		mic = 1;
 		reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L);
 		set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R);
+		snd_soc_dapm_enable_pin(codec, "Mic Jack");
+		snd_soc_dapm_disable_pin(codec, "Line Jack");
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_enable_pin(codec, "Headset Jack");
 		break;
 	}
 
 	if (corgi_spk_func == CORGI_SPK_ON)
-		spk = 1;
-
-	/* set the enpoints to their new connetion states */
-	snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
-	snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic);
-	snd_soc_dapm_set_endpoint(codec, "Line Jack", line);
-	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
-	snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
+		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(codec, "Ext Spk");
 
 	/* signal a DAPM event */
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 }
 
 static int corgi_startup(struct snd_pcm_substream *substream)
@@ -123,8 +123,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	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;
 
@@ -143,25 +143,25 @@ static int corgi_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_CBS_CFS);
 	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_CBS_CFS);
 	if (ret < 0)
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
 
 	/* set the I2S system clock as input (unused) */
-	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -247,7 +247,7 @@ SND_SOC_DAPM_HP("Headset Jack", NULL),
 };
 
 /* Corgi machine audio map (connections to the codec pins) */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* headset Jack  - in = micin, out = LHPOUT*/
 	{"Headset Jack", NULL, "LHPOUT"},
@@ -265,8 +265,6 @@ static const char *audio_map[][3] = {
 
 	/* Same as the above but no mic bias for line signals */
 	{"MICIN", NULL, "Line Jack"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -291,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
-	snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
 
 	/* Add corgi specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
@@ -303,15 +301,13 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
 	}
 
 	/* Add corgi specific widgets */
-	for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+				  ARRAY_SIZE(wm8731_dapm_widgets));
 
 	/* Set up corgi specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
new file mode 100644
index 000000000000..02dcac39cdf6
--- /dev/null
+++ b/sound/soc/pxa/em-x270.c
@@ -0,0 +1,102 @@
+/*
+ * em-x270.c  --  SoC audio for EM-X270
+ *
+ * Copyright 2007 CompuLab, Ltd.
+ *
+ * Author: Mike Rapoport <mike@compulab.co.il>
+ *
+ * Copied from tosa.c:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <asm/arch/pxa-regs.h>
+#include <asm/arch/hardware.h>
+#include <asm/arch/audio.h>
+
+#include "../codecs/wm9712.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static struct snd_soc_dai_link em_x270_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	},
+};
+
+static struct snd_soc_machine em_x270 = {
+	.name = "EM-X270",
+	.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,
+	.codec_dev = &soc_codec_dev_wm9712,
+};
+
+static struct platform_device *em_x270_snd_device;
+
+static int __init em_x270_init(void)
+{
+	int ret;
+
+	if (!machine_is_em_x270())
+		return -ENODEV;
+
+	em_x270_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!em_x270_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata);
+	em_x270_snd_devdata.dev = &em_x270_snd_device->dev;
+	ret = platform_device_add(em_x270_snd_device);
+
+	if (ret)
+		platform_device_put(em_x270_snd_device);
+
+	return ret;
+}
+
+static void __exit em_x270_exit(void)
+{
+	platform_device_unregister(em_x270_snd_device);
+}
+
+module_init(em_x270_init);
+module_exit(em_x270_exit);
+
+/* Module information */
+MODULE_AUTHOR("Mike Rapoport");
+MODULE_DESCRIPTION("ALSA SoC EM-X270");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 7e830b218943..65a4e9a8c39e 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -48,8 +48,6 @@ static int poodle_spk_func;
 
 static void poodle_ext_control(struct snd_soc_codec *codec)
 {
-	int spk = 0;
-
 	/* set up jack connection */
 	if (poodle_jack_func == POODLE_HP) {
 		/* set = unmute headphone */
@@ -57,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec)
 			POODLE_LOCOMO_GPIO_MUTE_L, 1);
 		locomo_gpio_write(&poodle_locomo_device.dev,
 			POODLE_LOCOMO_GPIO_MUTE_R, 1);
-		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
 	} else {
 		locomo_gpio_write(&poodle_locomo_device.dev,
 			POODLE_LOCOMO_GPIO_MUTE_L, 0);
 		locomo_gpio_write(&poodle_locomo_device.dev,
 			POODLE_LOCOMO_GPIO_MUTE_R, 0);
-		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
 	}
 
-	if (poodle_spk_func == POODLE_SPK_ON)
-		spk = 1;
-
 	/* set the enpoints to their new connetion states */
-	snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk);
+	if (poodle_spk_func == POODLE_SPK_ON)
+		snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin(codec, "Ext Spk");
 
 	/* signal a DAPM event */
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 }
 
 static int poodle_startup(struct snd_pcm_substream *substream)
@@ -104,8 +102,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	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;
 
@@ -124,25 +122,25 @@ static int poodle_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_CBS_CFS);
 	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_CBS_CFS);
 	if (ret < 0)
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
 
 	/* set the I2S system clock as input (unused) */
-	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -215,8 +213,8 @@ SND_SOC_DAPM_HP("Headphone Jack", NULL),
 SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
 };
 
-/* Corgi machine audio_mapnections to the codec pins */
-static const char *audio_map[][3] = {
+/* Corgi machine connections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* headphone connected to LHPOUT1, RHPOUT1 */
 	{"Headphone Jack", NULL, "LHPOUT"},
@@ -225,8 +223,6 @@ static const char *audio_map[][3] = {
 	/* speaker connected to LOUT, ROUT */
 	{"Ext Spk", NULL, "ROUT"},
 	{"Ext Spk", NULL, "LOUT"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *jack_function[] = {"Off", "Headphone"};
@@ -250,9 +246,9 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0);
-	snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0);
-	snd_soc_dapm_set_endpoint(codec, "MICIN", 1);
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_enable_pin(codec, "MICIN");
 
 	/* Add poodle specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
@@ -263,15 +259,13 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
 	}
 
 	/* Add poodle specific widgets */
-	for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
+				  ARRAY_SIZE(wm8731_dapm_widgets));
 
 	/* Set up poodle specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 97ec2d90547c..059af815ea0c 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -283,7 +283,7 @@ 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_cpu_dai *dai)
+	struct snd_soc_dai *dai)
 {
 	GCR |= GCR_ACLINK_OFF;
 	clk_disable(ac97_clk);
@@ -291,7 +291,7 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev,
 }
 
 static int pxa2xx_ac97_resume(struct platform_device *pdev,
-	struct snd_soc_cpu_dai *dai)
+	struct snd_soc_dai *dai)
 {
 	pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
 	pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD);
@@ -310,7 +310,8 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev,
 #define pxa2xx_ac97_resume	NULL
 #endif
 
-static int pxa2xx_ac97_probe(struct platform_device *pdev)
+static int pxa2xx_ac97_probe(struct platform_device *pdev,
+			     struct snd_soc_dai *dai)
 {
 	int ret;
 
@@ -355,7 +356,8 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev)
 	return ret;
 }
 
-static void pxa2xx_ac97_remove(struct platform_device *pdev)
+static void pxa2xx_ac97_remove(struct platform_device *pdev,
+			       struct snd_soc_dai *dai)
 {
 	GCR |= GCR_ACLINK_OFF;
 	free_irq(IRQ_AC97, NULL);
@@ -372,7 +374,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out;
@@ -386,7 +388,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
@@ -400,7 +402,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
@@ -418,7 +420,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
  * There is only 1 physical AC97 interface for pxa2xx, but it
  * has extra fifo's that can be used for aux DACs and ADCs.
  */
-struct snd_soc_cpu_dai pxa_ac97_dai[] = {
+struct snd_soc_dai pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97",
 	.id = 0,
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
index b8ccfee095c4..e390de8edcd4 100644
--- a/sound/soc/pxa/pxa2xx-ac97.h
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -14,7 +14,7 @@
 #define PXA2XX_DAI_AC97_AUX		1
 #define PXA2XX_DAI_AC97_MIC		2
 
-extern struct snd_soc_cpu_dai pxa_ac97_dai[3];
+extern struct snd_soc_dai pxa_ac97_dai[3];
 
 /* platform data */
 extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 425071030970..8f96d87f7b4b 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -9,15 +9,13 @@
  *  under  the terms of  the GNU General  Public License as published by the
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
- *
- *  Revision history
- *    12th Aug 2005   Initial version.
  */
 
 #include <linux/init.h>
 #include <linux/module.h>
 #include <linux/device.h>
 #include <linux/delay.h>
+#include <linux/clk.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
@@ -40,6 +38,7 @@ struct pxa_i2s_port {
 	u32 fmt;
 };
 static struct pxa_i2s_port pxa_i2s;
+static struct clk *clk_i2s;
 
 static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
 	.name			= "I2S PCM Stereo out",
@@ -80,7 +79,11 @@ static struct pxa2xx_gpio gpio_bus[] = {
 static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+
+	clk_i2s = clk_get(NULL, "I2SCLK");
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(clk_i2s);
 
 	if (!cpu_dai->active) {
 		SACR0 |= SACR0_RST;
@@ -101,7 +104,7 @@ static int pxa_i2s_wait(void)
 	return 0;
 }
 
-static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		unsigned int fmt)
 {
 	/* interface format */
@@ -127,7 +130,7 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai,
 	return 0;
 }
 
-static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		int clk_id, unsigned int freq, int dir)
 {
 	if (clk_id != PXA2XX_I2S_SYSCLK)
@@ -143,13 +146,13 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
 	pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
 	pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
 	pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
 	pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
-	pxa_set_cken(CKEN_I2S, 1);
+	clk_enable(clk_i2s);
 	pxa_i2s_wait();
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -234,13 +237,15 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream)
 	if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
 		SACR0 &= ~SACR0_ENB;
 		pxa_i2s_wait();
-		pxa_set_cken(CKEN_I2S, 0);
+		clk_disable(clk_i2s);
 	}
+
+	clk_put(clk_i2s);
 }
 
 #ifdef CONFIG_PM
 static int pxa2xx_i2s_suspend(struct platform_device *dev,
-	struct snd_soc_cpu_dai *dai)
+	struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -258,7 +263,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev,
 }
 
 static int pxa2xx_i2s_resume(struct platform_device *pdev,
-	struct snd_soc_cpu_dai *dai)
+	struct snd_soc_dai *dai)
 {
 	if (!dai->active)
 		return 0;
@@ -283,7 +288,7 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
 
-struct snd_soc_cpu_dai pxa_i2s_dai = {
+struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
 	.type = SND_SOC_DAI_I2S,
diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h
index 4435bd9f884f..e2def441153e 100644
--- a/sound/soc/pxa/pxa2xx-i2s.h
+++ b/sound/soc/pxa/pxa2xx-i2s.h
@@ -15,6 +15,6 @@
 /* I2S clock */
 #define PXA2XX_I2S_SYSCLK		0
 
-extern struct snd_soc_cpu_dai pxa_i2s_dai;
+extern struct snd_soc_dai pxa_i2s_dai;
 
 #endif
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index 01ad7bf716b7..2df03ee5819e 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK;
 
-int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	struct snd_pcm *pcm)
 {
 	int ret = 0;
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d8b8372db00e..64385797da5d 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -12,9 +12,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    30th Nov 2005   Initial version.
- *
  */
 
 #include <linux/module.h>
@@ -54,60 +51,60 @@ static int spitz_spk_func;
 static void spitz_ext_control(struct snd_soc_codec *codec)
 {
 	if (spitz_spk_func == SPITZ_SPK_ON)
-		snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1);
+		snd_soc_dapm_enable_pin(codec, "Ext Spk");
 	else
-		snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0);
+		snd_soc_dapm_disable_pin(codec, "Ext Spk");
 
 	/* set up jack connection */
 	switch (spitz_jack_func) {
 	case SPITZ_HP:
 		/* enable and unmute hp jack, disable mic bias */
-		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(codec, "Mic Jack");
+		snd_soc_dapm_disable_pin(codec, "Line Jack");
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
 		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
 		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
 		break;
 	case SPITZ_MIC:
 		/* enable mic jack and bias, mute hp */
-		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(codec, "Line Jack");
+		snd_soc_dapm_enable_pin(codec, "Mic Jack");
 		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
 		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
 		break;
 	case SPITZ_LINE:
 		/* enable line jack, disable mic bias and mute hp */
-		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Line Jack", 1);
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(codec, "Mic Jack");
+		snd_soc_dapm_enable_pin(codec, "Line Jack");
 		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
 		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
 		break;
 	case SPITZ_HEADSET:
 		/* enable and unmute headset jack enable mic bias, mute L hp */
-		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
-		snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1);
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_enable_pin(codec, "Mic Jack");
+		snd_soc_dapm_disable_pin(codec, "Line Jack");
+		snd_soc_dapm_enable_pin(codec, "Headset Jack");
 		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
 		set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
 		break;
 	case SPITZ_HP_OFF:
 
 		/* jack removed, everything off */
-		snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
-		snd_soc_dapm_set_endpoint(codec, "Line Jack", 0);
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
+		snd_soc_dapm_disable_pin(codec, "Mic Jack");
+		snd_soc_dapm_disable_pin(codec, "Line Jack");
 		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L);
 		reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R);
 		break;
 	}
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 }
 
 static int spitz_startup(struct snd_pcm_substream *substream)
@@ -124,8 +121,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	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;
 
@@ -144,25 +141,25 @@ static int spitz_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_CBS_CFS);
 	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_CBS_CFS);
 	if (ret < 0)
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
 
 	/* set the I2S system clock as input (unused) */
-	ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -250,7 +247,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
 };
 
 /* Spitz machine audio_map */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* headphone connected to LOUT1, ROUT1 */
 	{"Headphone Jack", NULL, "LOUT1"},
@@ -269,8 +266,6 @@ static const char *audio_map[][3] = {
 
 	/* line is connected to input 1 - no bias */
 	{"LINPUT1", NULL, "Line Jack"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -296,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	int i, err;
 
 	/* NC codec pins */
-	snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0);
-	snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0);
-	snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0);
-	snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0);
-	snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0);
-	snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
-	snd_soc_dapm_set_endpoint(codec, "MONO", 0);
+	snd_soc_dapm_disable_pin(codec, "RINPUT1");
+	snd_soc_dapm_disable_pin(codec, "LINPUT2");
+	snd_soc_dapm_disable_pin(codec, "RINPUT2");
+	snd_soc_dapm_disable_pin(codec, "LINPUT3");
+	snd_soc_dapm_disable_pin(codec, "RINPUT3");
+	snd_soc_dapm_disable_pin(codec, "OUT3");
+	snd_soc_dapm_disable_pin(codec, "MONO");
 
 	/* Add spitz specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
@@ -313,15 +308,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	}
 
 	/* Add spitz specific widgets */
-	for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+				  ARRAY_SIZE(wm8750_dapm_widgets));
 
-	/* Set up spitz specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++)
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
+	/* Set up spitz specific audio paths */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 7346d7e5d066..b6edb61a3a30 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -12,9 +12,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    30th Nov 2005   Initial version.
- *
  * GPIO's
  *  1 - Jack Insertion
  *  5 - Hookswitch (headset answer/hang up switch)
@@ -55,29 +52,31 @@ static int tosa_spk_func;
 
 static void tosa_ext_control(struct snd_soc_codec *codec)
 {
-	int spk = 0, mic_int = 0, hp = 0, hs = 0;
-
 	/* set up jack connection */
 	switch (tosa_jack_func) {
 	case TOSA_HP:
-		hp = 1;
+		snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
+		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		break;
 	case TOSA_MIC_INT:
-		mic_int = 1;
+		snd_soc_dapm_enable_pin(codec, "Mic (Internal)");
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_disable_pin(codec, "Headset Jack");
 		break;
 	case TOSA_HEADSET:
-		hs = 1;
+		snd_soc_dapm_disable_pin(codec, "Mic (Internal)");
+		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+		snd_soc_dapm_enable_pin(codec, "Headset Jack");
 		break;
 	}
 
 	if (tosa_spk_func == TOSA_SPK_ON)
-		spk = 1;
+		snd_soc_dapm_enable_pin(codec, "Speaker");
+	else
+		snd_soc_dapm_disable_pin(codec, "Speaker");
 
-	snd_soc_dapm_set_endpoint(codec, "Speaker", spk);
-	snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int);
-	snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp);
-	snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs);
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 }
 
 static int tosa_startup(struct snd_pcm_substream *substream)
@@ -154,7 +153,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL),
 };
 
 /* tosa audio map */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route audio_map[] = {
 
 	/* headphone connected to HPOUTL, HPOUTR */
 	{"Headphone Jack", NULL, "HPOUTL"},
@@ -173,8 +172,6 @@ static const char *audio_map[][3] = {
 	{"Headset Jack", NULL, "HPOUTR"},
 	{"LINEINR", NULL, "Mic Bias"},
 	{"Mic Bias", NULL, "Headset Jack"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
@@ -196,8 +193,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_set_endpoint(codec, "OUT3", 0);
-	snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0);
+	snd_soc_dapm_disable_pin(codec, "OUT3");
+	snd_soc_dapm_disable_pin(codec, "MONOOUT");
 
 	/* add tosa specific controls */
 	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
@@ -208,17 +205,13 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
 	}
 
 	/* add tosa specific widgets */
-	for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) {
-		snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]);
-	}
+	snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
+				  ARRAY_SIZE(tosa_dapm_widgets));
 
 	/* set up tosa specific audio path audio_map */
-	for (i = 0; audio_map[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 	return 0;
 }
 
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 1f6dbfc4caa8..b9f2353effeb 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,7 +1,6 @@
 config SND_S3C24XX_SOC
 	tristate "SoC Audio for the Samsung S3C24XX chips"
-	depends on ARCH_S3C2410 && SND_SOC
-	select SND_PCM
+	depends on ARCH_S3C2410
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the S3C24XX AC97, I2S or SSP interface. You will also need
@@ -16,7 +15,6 @@ config SND_S3C2412_SOC_I2S
 config SND_S3C2443_SOC_AC97
 	tristate
 	select AC97_BUS
-	select SND_AC97_CODEC
 	select SND_SOC_AC97_BUS
 	
 config SND_S3C24XX_SOC_NEO1973_WM8753
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 0e9d1c5f2484..4d7a9aa15f1a 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -10,10 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    20th Jan 2007   Initial version.
- *    05th Feb 2007   Rename all to Neo1973
- *
  */
 
 #include <linux/module.h>
@@ -26,6 +22,7 @@
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
+#include <sound/tlv.h>
 
 #include <asm/mach-types.h>
 #include <asm/hardware/scoop.h>
@@ -43,6 +40,14 @@
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-i2s.h"
 
+/* Debugging stuff */
+#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0
+#if S3C24XX_SOC_NEO1973_WM8753_DEBUG
+#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x)
+#else
+#define DBG(x...)
+#endif
+
 /* define the scenarios */
 #define NEO_AUDIO_OFF			0
 #define NEO_GSM_CALL_AUDIO_HANDSET	1
@@ -61,12 +66,14 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	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, bclk = 0;
 	int ret = 0;
 	unsigned long iis_clkrate;
 
+	DBG("Entered %s\n", __func__);
+
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
 	switch (params_rate(params)) {
@@ -101,44 +108,44 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* set codec DAI configuration */
-	ret = codec_dai->dai_ops.set_fmt(codec_dai,
+	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,
+	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;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
 
 	/* set MCLK division for sample rate */
-	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
 		S3C2410_IISMOD_32FS);
 	if (ret < 0)
 		return ret;
 
 	/* set codec BCLK division for sample rate */
-	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
 	if (ret < 0)
 		return ret;
 
 	/* set prescaler division for sample rate */
-	ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
 		S3C24XX_PRESCALE(4, 4));
 	if (ret < 0)
 		return ret;
 
 	/* codec PLL input is PCLK/4 */
-	ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1,
+	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
 		iis_clkrate / 4, pll_out);
 	if (ret < 0)
 		return ret;
@@ -149,10 +156,12 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+	DBG("Entered %s\n", __func__);
 
 	/* disable the PLL */
-	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
+	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
 }
 
 /*
@@ -167,11 +176,13 @@ static int neo1973_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_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 	unsigned int pcmdiv = 0;
 	int ret = 0;
 	unsigned long iis_clkrate;
 
+	DBG("Entered %s\n", __func__);
+
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
 	if (params_rate(params) != 8000)
@@ -183,24 +194,24 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
 
 	/* todo: gg check mode (DSP_B) against CSR datasheet */
 	/* set codec DAI configuration */
-	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
 	if (ret < 0)
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
 
 	/* set codec PCM division for sample rate */
-	ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
 	if (ret < 0)
 		return ret;
 
 	/* configue and enable PLL for 12.288MHz output */
-	ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2,
+	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
 		iis_clkrate / 4, 12288000);
 	if (ret < 0)
 		return ret;
@@ -211,10 +222,12 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
 static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+
+	DBG("Entered %s\n", __func__);
 
 	/* disable the PLL */
-	return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
+	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
 }
 
 static struct snd_soc_ops neo1973_voice_ops = {
@@ -233,79 +246,81 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
 
 static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
 {
+	DBG("Entered %s\n", __func__);
+
 	switch (neo1973_scenario) {
 	case NEO_AUDIO_OFF:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
+		snd_soc_dapm_disable_pin(codec, "Audio Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line In");
+		snd_soc_dapm_disable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(codec, "Call Mic");
 		break;
 	case NEO_GSM_CALL_AUDIO_HANDSET:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    1);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  1);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     1);
+		snd_soc_dapm_enable_pin(codec, "Audio Out");
+		snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_enable_pin(codec, "GSM Line In");
+		snd_soc_dapm_disable_pin(codec, "Headset Mic");
+		snd_soc_dapm_enable_pin(codec, "Call Mic");
 		break;
 	case NEO_GSM_CALL_AUDIO_HEADSET:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    1);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  1);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  1);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
+		snd_soc_dapm_enable_pin(codec, "Audio Out");
+		snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_enable_pin(codec, "GSM Line In");
+		snd_soc_dapm_enable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(codec, "Call Mic");
 		break;
 	case NEO_GSM_CALL_AUDIO_BLUETOOTH:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  1);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
+		snd_soc_dapm_disable_pin(codec, "Audio Out");
+		snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_enable_pin(codec, "GSM Line In");
+		snd_soc_dapm_disable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(codec, "Call Mic");
 		break;
 	case NEO_STEREO_TO_SPEAKERS:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    1);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
+		snd_soc_dapm_enable_pin(codec, "Audio Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line In");
+		snd_soc_dapm_disable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(codec, "Call Mic");
 		break;
 	case NEO_STEREO_TO_HEADPHONES:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    1);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
+		snd_soc_dapm_enable_pin(codec, "Audio Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line In");
+		snd_soc_dapm_disable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(codec, "Call Mic");
 		break;
 	case NEO_CAPTURE_HANDSET:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     1);
+		snd_soc_dapm_disable_pin(codec, "Audio Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line In");
+		snd_soc_dapm_disable_pin(codec, "Headset Mic");
+		snd_soc_dapm_enable_pin(codec, "Call Mic");
 		break;
 	case NEO_CAPTURE_HEADSET:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  1);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
+		snd_soc_dapm_disable_pin(codec, "Audio Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line In");
+		snd_soc_dapm_enable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(codec, "Call Mic");
 		break;
 	case NEO_CAPTURE_BLUETOOTH:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
+		snd_soc_dapm_disable_pin(codec, "Audio Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line In");
+		snd_soc_dapm_disable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(codec, "Call Mic");
 		break;
 	default:
-		snd_soc_dapm_set_endpoint(codec, "Audio Out",    0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
-		snd_soc_dapm_set_endpoint(codec, "GSM Line In",  0);
-		snd_soc_dapm_set_endpoint(codec, "Headset Mic",  0);
-		snd_soc_dapm_set_endpoint(codec, "Call Mic",     0);
+		snd_soc_dapm_disable_pin(codec, "Audio Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line Out");
+		snd_soc_dapm_disable_pin(codec, "GSM Line In");
+		snd_soc_dapm_disable_pin(codec, "Headset Mic");
+		snd_soc_dapm_disable_pin(codec, "Call Mic");
 	}
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 
 	return 0;
 }
@@ -315,6 +330,8 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 
+	DBG("Entered %s\n", __func__);
+
 	if (neo1973_scenario == ucontrol->value.integer.value[0])
 		return 0;
 
@@ -327,6 +344,8 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
 
 static void lm4857_write_regs(void)
 {
+	DBG("Entered %s\n", __func__);
+
 	if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
 		printk(KERN_ERR "lm4857: i2c write failed\n");
 }
@@ -338,6 +357,8 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
 	int shift = (kcontrol->private_value >> 8) & 0x0F;
 	int mask = (kcontrol->private_value >> 16) & 0xFF;
 
+	DBG("Entered %s\n", __func__);
+
 	ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
 	return 0;
 }
@@ -364,6 +385,8 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
 {
 	u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
 
+	DBG("Entered %s\n", __func__);
+
 	if (value)
 		value -= 5;
 
@@ -376,6 +399,8 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
 {
 	u8 value = ucontrol->value.integer.value[0];
 
+	DBG("Entered %s\n", __func__);
+
 	if (value)
 		value += 5;
 
@@ -397,8 +422,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
 };
 
 
-/* example machine audio_mapnections */
-static const char *audio_map[][3] = {
+static const struct snd_soc_dapm_route dapm_routes[] = {
 
 	/* Connections to the lm4857 amp */
 	{"Audio Out", NULL, "LOUT1"},
@@ -421,8 +445,6 @@ static const char *audio_map[][3] = {
 
 	/* Connect the ALC pins */
 	{"ACIN", NULL, "ACOP"},
-
-	{NULL, NULL, NULL},
 };
 
 static const char *lm4857_mode[] = {
@@ -453,13 +475,16 @@ static const struct soc_enum neo_scenario_enum[] = {
 	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios),
 };
 
+static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
+
 static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
-	SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg),
-	SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg),
-	SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg),
+	SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
+		lm4857_get_reg, lm4857_set_reg, stereo_tlv),
+	SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
+		lm4857_get_reg, lm4857_set_reg, stereo_tlv),
+	SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
+		lm4857_get_reg, lm4857_set_reg, mono_tlv),
 	SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
 		lm4857_get_mode, lm4857_set_mode),
 	SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
@@ -483,21 +508,23 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
+	DBG("Entered %s\n", __func__);
+
 	/* set up NC codec pins */
-	snd_soc_dapm_set_endpoint(codec, "LOUT2", 0);
-	snd_soc_dapm_set_endpoint(codec, "ROUT2", 0);
-	snd_soc_dapm_set_endpoint(codec, "OUT3",  0);
-	snd_soc_dapm_set_endpoint(codec, "OUT4",  0);
-	snd_soc_dapm_set_endpoint(codec, "LINE1", 0);
-	snd_soc_dapm_set_endpoint(codec, "LINE2", 0);
+	snd_soc_dapm_disable_pin(codec, "LOUT2");
+	snd_soc_dapm_disable_pin(codec, "ROUT2");
+	snd_soc_dapm_disable_pin(codec, "OUT3");
+	snd_soc_dapm_disable_pin(codec, "OUT4");
+	snd_soc_dapm_disable_pin(codec, "LINE1");
+	snd_soc_dapm_disable_pin(codec, "LINE2");
 
 
 	/* set endpoints to default mode */
 	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
 
 	/* Add neo1973 specific widgets */
-	for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
-		snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
+	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
+				  ARRAY_SIZE(wm8753_dapm_widgets));
 
 	/* add neo1973 specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
@@ -508,20 +535,18 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 			return err;
 	}
 
-	/* set up neo1973 specific audio path audio_mapnects */
-	for (i = 0; audio_map[i][0] != NULL; i++) {
-		snd_soc_dapm_connect_input(codec, audio_map[i][0],
-			audio_map[i][1], audio_map[i][2]);
-	}
+	/* set up neo1973 specific audio routes */
+	err = snd_soc_dapm_add_routes(codec, dapm_routes,
+				      ARRAY_SIZE(dapm_routes));
 
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 	return 0;
 }
 
 /*
  * BT Codec DAI
  */
-static struct snd_soc_cpu_dai bt_dai = {
+static struct snd_soc_dai bt_dai = {
 	.name = "Bluetooth",
 	.id = 0,
 	.type = SND_SOC_DAI_PCM,
@@ -583,6 +608,8 @@ static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind)
 {
 	int ret;
 
+	DBG("Entered %s\n", __func__);
+
 	client_template.adapter = adap;
 	client_template.addr = addr;
 
@@ -606,6 +633,8 @@ exit_err:
 
 static int lm4857_i2c_detach(struct i2c_client *client)
 {
+	DBG("Entered %s\n", __func__);
+
 	i2c_detach_client(client);
 	kfree(client);
 	return 0;
@@ -613,6 +642,8 @@ static int lm4857_i2c_detach(struct i2c_client *client)
 
 static int lm4857_i2c_attach(struct i2c_adapter *adap)
 {
+	DBG("Entered %s\n", __func__);
+
 	return i2c_probe(adap, &addr_data, lm4857_amp_probe);
 }
 
@@ -620,6 +651,8 @@ static u8 lm4857_state;
 
 static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
 {
+	DBG("Entered %s\n", __func__);
+
 	dev_dbg(&dev->dev, "lm4857_suspend\n");
 	lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
 	if (lm4857_state) {
@@ -631,6 +664,8 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
 
 static int lm4857_resume(struct i2c_client *dev)
 {
+	DBG("Entered %s\n", __func__);
+
 	if (lm4857_state) {
 		lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
 		lm4857_write_regs();
@@ -640,6 +675,8 @@ static int lm4857_resume(struct i2c_client *dev)
 
 static void lm4857_shutdown(struct i2c_client *dev)
 {
+	DBG("Entered %s\n", __func__);
+
 	dev_dbg(&dev->dev, "lm4857_shutdown\n");
 	lm4857_regs[LM4857_CTRL] &= 0xf0;
 	lm4857_write_regs();
@@ -671,6 +708,8 @@ static int __init neo1973_init(void)
 {
 	int ret;
 
+	DBG("Entered %s\n", __func__);
+
 	neo1973_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!neo1973_snd_device)
 		return -ENOMEM;
@@ -691,6 +730,8 @@ static int __init neo1973_init(void)
 
 static void __exit neo1973_exit(void)
 {
+	DBG("Entered %s\n", __func__);
+
 	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
 }
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index c4a46dd589b3..ee4676ed1283 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -295,7 +295,7 @@ static inline int s3c2412_snd_is_clkmaster(void)
 /*
  * Set S3C2412 I2S DAI format
  */
-static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 			       unsigned int fmt)
 {
 	u32 iismod;
@@ -500,7 +500,7 @@ EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
 /*
  * Set S3C2412 Clock source
  */
-static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 				  int clk_id, unsigned int freq, int dir)
 {
 	u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
@@ -528,7 +528,7 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
 /*
  * Set S3C2412 Clock dividers
  */
-static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 				  int div_id, int div)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
@@ -601,7 +601,8 @@ struct clk *s3c2412_get_iisclk(void)
 EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
 
 
-static int s3c2412_i2s_probe(struct platform_device *pdev)
+static int s3c2412_i2s_probe(struct platform_device *pdev,
+			     struct snd_soc_dai *dai)
 {
 	DBG("Entered %s\n", __func__);
 
@@ -647,7 +648,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev)
 
 #ifdef CONFIG_PM
 static int s3c2412_i2s_suspend(struct platform_device *dev,
-			      struct snd_soc_cpu_dai *dai)
+			      struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 	u32 iismod;
@@ -675,7 +676,7 @@ static int s3c2412_i2s_suspend(struct platform_device *dev,
 }
 
 static int s3c2412_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_cpu_dai *dai)
+			      struct snd_soc_dai *dai)
 {
 	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
 
@@ -707,7 +708,7 @@ static int s3c2412_i2s_resume(struct platform_device *pdev,
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
-struct snd_soc_cpu_dai s3c2412_i2s_dai = {
+struct snd_soc_dai s3c2412_i2s_dai = {
 	.name	= "s3c2412-i2s",
 	.id	= 0,
 	.type	= SND_SOC_DAI_I2S,
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h
index 27f48e1ffa86..aac08a25e541 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.h
+++ b/sound/soc/s3c24xx/s3c2412-i2s.h
@@ -24,7 +24,7 @@
 
 extern struct clk *s3c2412_get_iisclk(void);
 
-extern struct snd_soc_cpu_dai s3c2412_i2s_dai;
+extern struct snd_soc_dai s3c2412_i2s_dai;
 
 struct s3c2412_rate_calc {
 	unsigned int	clk_div;	/* for prescaler */
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index e81d9a6c83da..783349b7fede 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -10,9 +10,6 @@
  *  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.
- *
- *  Revision history
- *	21st Mar 2007   Initial Version
  */
 
 #include <linux/init.h>
@@ -212,7 +209,8 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = {
 	.dma_size	= 4,
 };
 
-static int s3c2443_ac97_probe(struct platform_device *pdev)
+static int s3c2443_ac97_probe(struct platform_device *pdev,
+			      struct snd_soc_dai *dai)
 {
 	int ret;
 	u32 ac_glbctrl;
@@ -263,7 +261,8 @@ static int s3c2443_ac97_probe(struct platform_device *pdev)
 	return ret;
 }
 
-static void s3c2443_ac97_remove(struct platform_device *pdev)
+static void s3c2443_ac97_remove(struct platform_device *pdev,
+				struct snd_soc_dai *dai)
 {
 	free_irq(IRQ_S3C244x_AC97, NULL);
 	clk_disable(s3c24xx_ac97.ac97_clk);
@@ -275,7 +274,7 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out;
@@ -317,7 +316,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
@@ -353,7 +352,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
 		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
-struct snd_soc_cpu_dai s3c2443_ac97_dai[] = {
+struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "s3c2443-ac97",
 	.id = 0,
diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h
index bf03e8ed16c3..a96dcadf28b4 100644
--- a/sound/soc/s3c24xx/s3c24xx-ac97.h
+++ b/sound/soc/s3c24xx/s3c24xx-ac97.h
@@ -26,6 +26,6 @@
 #define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97
 #endif
 
-extern struct snd_soc_cpu_dai s3c2443_ac97_dai[];
+extern struct snd_soc_dai s3c2443_ac97_dai[];
 
 #endif /*S3C24XXAC97_H_*/
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 1ed6afd45459..397524282b57 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -12,11 +12,6 @@
  *  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
- *    11th Dec 2006   Merged with Simtec driver
- *    10th Nov 2006   Initial version.
  */
 
 #include <linux/init.h>
@@ -180,7 +175,7 @@ static void s3c24xx_snd_rxctrl(int on)
 static int s3c24xx_snd_lrsync(void)
 {
 	u32 iiscon;
-	unsigned long timeout = jiffies + msecs_to_jiffies(5);
+	int timeout = 50; /* 5ms */
 
 	DBG("Entered %s\n", __func__);
 
@@ -189,8 +184,9 @@ static int s3c24xx_snd_lrsync(void)
 		if (iiscon & S3C2410_IISCON_LRINDEX)
 			break;
 
-		if (time_after(jiffies, timeout))
+		if (!timeout--)
 			return -ETIMEDOUT;
+		udelay(100);
 	}
 
 	return 0;
@@ -209,7 +205,7 @@ static inline int s3c24xx_snd_is_clkmaster(void)
 /*
  * Set S3C24xx I2S DAI format
  */
-static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 		unsigned int fmt)
 {
 	u32 iismod;
@@ -317,7 +313,7 @@ exit_err:
 /*
  * Set S3C24xx Clock source
  */
-static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 	int clk_id, unsigned int freq, int dir)
 {
 	u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -343,7 +339,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
 /*
  * Set S3C24xx Clock dividers
  */
-static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 	int div_id, int div)
 {
 	u32 reg;
@@ -381,7 +377,8 @@ u32 s3c24xx_i2s_get_clockrate(void)
 }
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
 
-static int s3c24xx_i2s_probe(struct platform_device *pdev)
+static int s3c24xx_i2s_probe(struct platform_device *pdev,
+			     struct snd_soc_dai *dai)
 {
 	DBG("Entered %s\n", __func__);
 
@@ -414,7 +411,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev)
 
 #ifdef CONFIG_PM
 static int s3c24xx_i2s_suspend(struct platform_device *pdev,
-		struct snd_soc_cpu_dai *cpu_dai)
+		struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 
@@ -429,7 +426,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev,
 }
 
 static int s3c24xx_i2s_resume(struct platform_device *pdev,
-		struct snd_soc_cpu_dai *cpu_dai)
+		struct snd_soc_dai *cpu_dai)
 {
 	DBG("Entered %s\n", __func__);
 	clk_enable(s3c24xx_i2s.iis_clk);
@@ -452,7 +449,7 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev,
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
-struct snd_soc_cpu_dai s3c24xx_i2s_dai = {
+struct snd_soc_dai s3c24xx_i2s_dai = {
 	.name = "s3c24xx-i2s",
 	.id = 0,
 	.type = SND_SOC_DAI_I2S,
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h
index 537b4ecce8a3..726d91cf4e1c 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.h
@@ -32,6 +32,6 @@
 
 u32 s3c24xx_i2s_get_clockrate(void);
 
-extern struct snd_soc_cpu_dai s3c24xx_i2s_dai;
+extern struct snd_soc_dai s3c24xx_i2s_dai;
 
 #endif /*S3C24XXI2S_H_*/
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 7806ae614617..cef79b34dc6f 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -12,10 +12,6 @@
  *  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
- *    11th Dec 2006   Merged with Simtec driver
- *    10th Nov 2006   Initial version.
  */
 
 #include <linux/module.h>
@@ -433,7 +429,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK;
 
 static int s3c24xx_pcm_new(struct snd_card *card,
-	struct snd_soc_codec_dai *dai, struct snd_pcm *pcm)
+	struct snd_soc_dai *dai, struct snd_pcm *pcm)
 {
 	int ret = 0;
 
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
index b4a56302b9ab..8515d6ff03f2 100644
--- a/sound/soc/s3c24xx/smdk2443_wm9710.c
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -10,9 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    8th Mar 2007   Initial version.
- *
  */
 
 #include <linux/module.h>
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 4c1e013381c9..54bd604012af 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -3,7 +3,7 @@ menu "SoC Audio support for SuperH"
 
 config SND_SOC_PCM_SH7760
 	tristate "SoC Audio support for Renesas SH7760"
-	depends on CPU_SUBTYPE_SH7760 && SND_SOC && SH_DMABRG
+	depends on CPU_SUBTYPE_SH7760 && SH_DMABRG
 	help
 	  Enable this option for SH7760 AC97/I2S audio support.
 
@@ -13,10 +13,9 @@ config SND_SOC_PCM_SH7760
 ##
 
 config SND_SOC_SH4_HAC
+	tristate
 	select AC97_BUS
 	select SND_SOC_AC97_BUS
-	select SND_AC97_CODEC
-	tristate
 
 config SND_SOC_SH4_SSI
 	tristate
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 7a3ce80d6727..9faa12622d09 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -326,7 +326,7 @@ static void camelot_pcm_free(struct snd_pcm *pcm)
 }
 
 static int camelot_pcm_new(struct snd_card *card,
-			   struct snd_soc_codec_dai *dai,
+			   struct snd_soc_dai *dai,
 			   struct snd_pcm *pcm)
 {
 	/* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index b7b676b3d671..df7bc345c320 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream,
 #define AC97_FMTS	\
 	SNDRV_PCM_FMTBIT_S16_LE
 
-struct snd_soc_cpu_dai sh4_hac_dai[] = {
+struct snd_soc_dai sh4_hac_dai[] = {
 {
 	.name			= "HAC0",
 	.id			= 0,
diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c
index 2f91de84c5c7..92bfaf4774a7 100644
--- a/sound/soc/sh/sh7760-ac97.c
+++ b/sound/soc/sh/sh7760-ac97.c
@@ -20,12 +20,12 @@
 #define IPSEL 0xFE400034
 
 /* platform specific structs can be declared here */
-extern struct snd_soc_cpu_dai sh4_hac_dai[2];
+extern struct snd_soc_dai sh4_hac_dai[2];
 extern struct snd_soc_platform sh7760_soc_platform;
 
 static int machine_init(struct snd_soc_codec *codec)
 {
-	snd_soc_dapm_sync_endpoints(codec);
+	snd_soc_dapm_sync(codec);
 	return 0;
 }
 
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 3388bc3d62d1..55c3464163ab 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -208,7 +208,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id,
+static int ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
 			  unsigned int freq, int dir)
 {
 	struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id];
@@ -222,7 +222,7 @@ static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id,
  * This divider is used to generate the SSI_SCK (I2S bitclock) from the
  * clock at the HAC_BIT_CLK ("oversampling clock") pin.
  */
-static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div)
+static int ssi_set_clkdiv(struct snd_soc_dai *dai, int did, int div)
 {
 	struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
 	unsigned long ssicr;
@@ -245,7 +245,7 @@ static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div)
 	return 0;
 }
 
-static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt)
+static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct ssi_priv *ssi = &ssi_cpu_data[dai->id];
 	unsigned long ssicr = SSIREG(SSICR);
@@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt)
 	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE |	\
 	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_U32_LE)
 
-struct snd_soc_cpu_dai sh4_ssi_dai[] = {
+struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI0",
 	.id			= 0,
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index e148db940cfc..83f1190293a8 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -14,10 +14,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    12th Aug 2005   Initial version.
- *    25th Oct 2005   Working Codec, Interface and Platform registration.
- *
  *  TODO:
  *   o Add hw rules to enforce rates, etc.
  *   o More testing with other codecs/machines.
@@ -112,9 +108,9 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 }
 #endif
 
-static inline const char* get_dai_name(int type)
+static inline const char *get_dai_name(int type)
 {
-	switch(type) {
+	switch (type) {
 	case SND_SOC_DAI_AC97_BUS:
 	case SND_SOC_DAI_AC97:
 		return "AC97";
@@ -138,8 +134,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_dai_link *machine = rtd->dai;
 	struct snd_soc_platform *platform = socdev->platform;
-	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
-	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret = 0;
 
 	mutex_lock(&pcm_mutex);
@@ -182,9 +178,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	/* Check that the codec and cpu DAI's are compatible */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		runtime->hw.rate_min =
-			max(codec_dai->playback.rate_min, cpu_dai->playback.rate_min);
+			max(codec_dai->playback.rate_min,
+			    cpu_dai->playback.rate_min);
 		runtime->hw.rate_max =
-			min(codec_dai->playback.rate_max, cpu_dai->playback.rate_max);
+			min(codec_dai->playback.rate_max,
+			    cpu_dai->playback.rate_max);
 		runtime->hw.channels_min =
 			max(codec_dai->playback.channels_min,
 				cpu_dai->playback.channels_min);
@@ -197,9 +195,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 			codec_dai->playback.rates & cpu_dai->playback.rates;
 	} else {
 		runtime->hw.rate_min =
-			max(codec_dai->capture.rate_min, cpu_dai->capture.rate_min);
+			max(codec_dai->capture.rate_min,
+			    cpu_dai->capture.rate_min);
 		runtime->hw.rate_max =
-			min(codec_dai->capture.rate_max, cpu_dai->capture.rate_max);
+			min(codec_dai->capture.rate_max,
+			    cpu_dai->capture.rate_max);
 		runtime->hw.channels_min =
 			max(codec_dai->capture.channels_min,
 				cpu_dai->capture.channels_min);
@@ -229,7 +229,7 @@ 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: %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);
@@ -272,11 +272,11 @@ static void close_delayed_work(struct work_struct *work)
 	struct snd_soc_device *socdev =
 		container_of(work, struct snd_soc_device, delayed_work.work);
 	struct snd_soc_codec *codec = socdev->codec;
-	struct snd_soc_codec_dai *codec_dai;
+	struct snd_soc_dai *codec_dai;
 	int i;
 
 	mutex_lock(&pcm_mutex);
-	for(i = 0; i < codec->num_dai; i++) {
+	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
 
 		dbg("pop wq checking: %s status: %s waiting: %s\n",
@@ -287,12 +287,12 @@ static void close_delayed_work(struct work_struct *work)
 		/* are we waiting on this codec DAI stream */
 		if (codec_dai->pop_wait == 1) {
 
-			/* power down the codec to D1 if no longer active */
+			/* Reduce power if no longer active */
 			if (codec->active == 0) {
 				dbg("pop wq D1 %s %s\n", codec->name,
 					codec_dai->playback.stream_name);
-				snd_soc_dapm_device_event(socdev,
-					SNDRV_CTL_POWER_D1);
+				snd_soc_dapm_set_bias_level(socdev,
+					SND_SOC_BIAS_PREPARE);
 			}
 
 			codec_dai->pop_wait = 0;
@@ -300,12 +300,12 @@ static void close_delayed_work(struct work_struct *work)
 				codec_dai->playback.stream_name,
 				SND_SOC_DAPM_STREAM_STOP);
 
-			/* power down the codec power domain if no longer active */
+			/* 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);
-				snd_soc_dapm_device_event(socdev,
-					SNDRV_CTL_POWER_D3hot);
+				snd_soc_dapm_set_bias_level(socdev,
+					SND_SOC_BIAS_STANDBY);
 			}
 		}
 	}
@@ -323,8 +323,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
 	struct snd_soc_platform *platform = socdev->platform;
-	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
-	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
 
 	mutex_lock(&pcm_mutex);
@@ -365,8 +365,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 			SND_SOC_DAPM_STREAM_STOP);
 
 		if (codec->active == 0 && codec_dai->pop_wait == 0)
-			snd_soc_dapm_device_event(socdev,
-						SNDRV_CTL_POWER_D3hot);
+			snd_soc_dapm_set_bias_level(socdev,
+						SND_SOC_BIAS_STANDBY);
 	}
 
 	mutex_unlock(&pcm_mutex);
@@ -384,8 +384,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
 	struct snd_soc_platform *platform = socdev->platform;
-	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
-	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
 	int ret = 0;
 
@@ -434,14 +434,14 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		else {
 			codec_dai->pop_wait = 0;
 			cancel_delayed_work(&socdev->delayed_work);
-			if (codec_dai->dai_ops.digital_mute)
-				codec_dai->dai_ops.digital_mute(codec_dai, 0);
+			snd_soc_dai_digital_mute(codec_dai, 0);
 		}
 	} else {
 		/* no delayed work - do we need to power up codec */
-		if (codec->dapm_state != SNDRV_CTL_POWER_D0) {
+		if (codec->bias_level != SND_SOC_BIAS_ON) {
 
-			snd_soc_dapm_device_event(socdev,  SNDRV_CTL_POWER_D1);
+			snd_soc_dapm_set_bias_level(socdev,
+						    SND_SOC_BIAS_PREPARE);
 
 			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 				snd_soc_dapm_stream_event(codec,
@@ -452,9 +452,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 					codec_dai->capture.stream_name,
 					SND_SOC_DAPM_STREAM_START);
 
-			snd_soc_dapm_device_event(socdev, SNDRV_CTL_POWER_D0);
-			if (codec_dai->dai_ops.digital_mute)
-				codec_dai->dai_ops.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 */
@@ -466,8 +465,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 				snd_soc_dapm_stream_event(codec,
 					codec_dai->capture.stream_name,
 					SND_SOC_DAPM_STREAM_START);
-			if (codec_dai->dai_ops.digital_mute)
-				codec_dai->dai_ops.digital_mute(codec_dai, 0);
+
+			snd_soc_dai_digital_mute(codec_dai, 0);
 		}
 	}
 
@@ -488,8 +487,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
 	struct snd_soc_platform *platform = socdev->platform;
-	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
-	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret = 0;
 
 	mutex_lock(&pcm_mutex);
@@ -514,7 +513,7 @@ 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);
 		if (ret < 0) {
-			printk(KERN_ERR "asoc: can't set interface %s hw params\n",
+			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
 			goto interface_err;
 		}
@@ -523,7 +522,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (platform->pcm_ops->hw_params) {
 		ret = platform->pcm_ops->hw_params(substream, params);
 		if (ret < 0) {
-			printk(KERN_ERR "asoc: can't set platform %s hw params\n",
+			printk(KERN_ERR "asoc: platform %s hw params failed\n",
 				platform->name);
 			goto platform_err;
 		}
@@ -542,7 +541,7 @@ interface_err:
 		codec_dai->ops.hw_free(substream);
 
 codec_err:
-	if(machine->ops && machine->ops->hw_free)
+	if (machine->ops && machine->ops->hw_free)
 		machine->ops->hw_free(substream);
 
 	mutex_unlock(&pcm_mutex);
@@ -558,15 +557,15 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
 	struct snd_soc_platform *platform = socdev->platform;
-	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
-	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	struct snd_soc_codec *codec = socdev->codec;
 
 	mutex_lock(&pcm_mutex);
 
 	/* apply codec digital mute */
-	if (!codec->active && codec_dai->dai_ops.digital_mute)
-		codec_dai->dai_ops.digital_mute(codec_dai, 1);
+	if (!codec->active)
+		snd_soc_dai_digital_mute(codec_dai, 1);
 
 	/* free any machine hw params */
 	if (machine->ops && machine->ops->hw_free)
@@ -593,8 +592,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_dai_link *machine = rtd->dai;
 	struct snd_soc_platform *platform = socdev->platform;
-	struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai;
-	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret;
 
 	if (codec_dai->ops.trigger) {
@@ -631,16 +630,26 @@ static struct snd_pcm_ops soc_pcm_ops = {
 /* powers down audio subsystem for suspend */
 static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 {
- 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- 	struct snd_soc_machine *machine = socdev->machine;
- 	struct snd_soc_platform *platform = socdev->platform;
- 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_machine *machine = socdev->machine;
+	struct snd_soc_platform *platform = socdev->platform;
+	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
 	int i;
 
+	/* Due to the resume being scheduled into a workqueue we could
+	* suspend before that's finished - wait for it to complete.
+	 */
+	snd_power_lock(codec->card);
+	snd_power_wait(codec->card, SNDRV_CTL_POWER_D0);
+	snd_power_unlock(codec->card);
+
+	/* we're going to block userspace touching us until resume completes */
+	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_codec_dai *dai = machine->dai_link[i].codec_dai;
+	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);
 	}
@@ -652,8 +661,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	if (machine->suspend_pre)
 		machine->suspend_pre(pdev, state);
 
-	for(i = 0; i < machine->num_links; i++) {
-		struct snd_soc_cpu_dai  *cpu_dai = machine->dai_link[i].cpu_dai;
+	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);
 		if (platform->suspend)
@@ -662,9 +671,9 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 
 	/* close any waiting streams and save state */
 	run_delayed_work(&socdev->delayed_work);
-	codec->suspend_dapm_state = codec->dapm_state;
+	codec->suspend_bias_level = codec->bias_level;
 
-	for(i = 0; i < codec->num_dai; i++) {
+	for (i = 0; i < codec->num_dai; i++) {
 		char *stream = codec->dai[i].playback.stream_name;
 		if (stream != NULL)
 			snd_soc_dapm_stream_event(codec, stream,
@@ -678,8 +687,8 @@ 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_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	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);
 	}
@@ -690,21 +699,32 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	return 0;
 }
 
-/* powers up audio subsystem after a suspend */
-static int soc_resume(struct platform_device *pdev)
+/* deferred resume work, so resume can complete before we finished
+ * setting our codec back up, which can be very slow on I2C
+ */
+static void soc_resume_deferred(struct work_struct *work)
 {
- 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
- 	struct snd_soc_machine *machine = socdev->machine;
- 	struct snd_soc_platform *platform = socdev->platform;
- 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
+	struct snd_soc_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_codec_device *codec_dev = socdev->codec_dev;
 	struct snd_soc_codec *codec = socdev->codec;
+	struct platform_device *pdev = to_platform_device(socdev->dev);
 	int i;
 
+	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
+	 * so userspace apps are blocked from touching us
+	 */
+
+	dev_info(socdev->dev, "starting resume work\n");
+
 	if (machine->resume_pre)
 		machine->resume_pre(pdev);
 
-	for(i = 0; i < machine->num_links; i++) {
-		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	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);
 	}
@@ -712,8 +732,8 @@ static int soc_resume(struct platform_device *pdev)
 	if (codec_dev->resume)
 		codec_dev->resume(pdev);
 
-	for(i = 0; i < codec->num_dai; i++) {
-		char* stream = codec->dai[i].playback.stream_name;
+	for (i = 0; i < codec->num_dai; i++) {
+		char *stream = codec->dai[i].playback.stream_name;
 		if (stream != NULL)
 			snd_soc_dapm_stream_event(codec, stream,
 				SND_SOC_DAPM_STREAM_RESUME);
@@ -723,15 +743,15 @@ static int soc_resume(struct platform_device *pdev)
 				SND_SOC_DAPM_STREAM_RESUME);
 	}
 
-	/* unmute any active DAC's */
-	for(i = 0; i < machine->num_links; i++) {
-		struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
+	/* 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 < machine->num_links; i++) {
-		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+	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);
 		if (platform->resume)
@@ -741,6 +761,22 @@ static int soc_resume(struct platform_device *pdev)
 	if (machine->resume_post)
 		machine->resume_post(pdev);
 
+	dev_info(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);
+}
+
+/* powers up audio subsystem after a suspend */
+static int soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	dev_info(socdev->dev, "scheduling resume work\n");
+
+	if (!schedule_work(&socdev->deferred_resume_work))
+		dev_err(socdev->dev, "work item may be lost\n");
+
 	return 0;
 }
 
@@ -760,33 +796,38 @@ static int soc_probe(struct platform_device *pdev)
 
 	if (machine->probe) {
 		ret = machine->probe(pdev);
-		if(ret < 0)
+		if (ret < 0)
 			return ret;
 	}
 
 	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
 		if (cpu_dai->probe) {
-			ret = cpu_dai->probe(pdev);
-			if(ret < 0)
+			ret = cpu_dai->probe(pdev, cpu_dai);
+			if (ret < 0)
 				goto cpu_dai_err;
 		}
 	}
 
 	if (codec_dev->probe) {
 		ret = codec_dev->probe(pdev);
-		if(ret < 0)
+		if (ret < 0)
 			goto cpu_dai_err;
 	}
 
 	if (platform->probe) {
 		ret = platform->probe(pdev);
-		if(ret < 0)
+		if (ret < 0)
 			goto platform_err;
 	}
 
 	/* DAPM stream work */
 	INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
+#ifdef CONFIG_PM
+	/* deferred resume work */
+	INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred);
+#endif
+
 	return 0;
 
 platform_err:
@@ -795,9 +836,9 @@ platform_err:
 
 cpu_dai_err:
 	for (i--; i >= 0; i--) {
-		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
 		if (cpu_dai->remove)
-			cpu_dai->remove(pdev);
+			cpu_dai->remove(pdev, cpu_dai);
 	}
 
 	if (machine->remove)
@@ -824,9 +865,9 @@ static int soc_remove(struct platform_device *pdev)
 		codec_dev->remove(pdev);
 
 	for (i = 0; i < machine->num_links; i++) {
-		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
 		if (cpu_dai->remove)
-			cpu_dai->remove(pdev);
+			cpu_dai->remove(pdev, cpu_dai);
 	}
 
 	if (machine->remove)
@@ -852,8 +893,8 @@ 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_codec_dai *codec_dai = dai_link->codec_dai;
-	struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai;
+	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;
 	struct snd_pcm *pcm;
 	char new_name[64];
@@ -868,7 +909,7 @@ 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,
+	sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
 		get_dai_name(cpu_dai->type), num);
 
 	if (codec_dai->playback.channels_min)
@@ -879,7 +920,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 	ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
 		capture, &pcm);
 	if (ret < 0) {
-		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+		printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
+			codec->name);
 		kfree(rtd);
 		return ret;
 	}
@@ -928,8 +970,9 @@ static ssize_t codec_reg_show(struct device *dev,
 		step = codec->reg_cache_step;
 
 	count += sprintf(buf, "%s registers\n", codec->name);
-	for(i = 0; i < codec->reg_cache_size; i += step)
-		count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i));
+	for (i = 0; i < codec->reg_cache_size; i += step)
+		count += sprintf(buf + count, "%2x: %4x\n", i,
+			codec->read(codec, i));
 
 	return count;
 }
@@ -1072,7 +1115,7 @@ 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++) {
+	for (i = 0; i < machine->num_links; i++) {
 		ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't create pcm %s\n",
@@ -1102,7 +1145,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 	struct snd_soc_machine *machine = socdev->machine;
 	int ret = 0, i, ac97 = 0, err = 0;
 
-	for(i = 0; i < machine->num_links; i++) {
+	for (i = 0; i < machine->num_links; i++) {
 		if (socdev->machine->dai_link[i].init) {
 			err = socdev->machine->dai_link[i].init(codec);
 			if (err < 0) {
@@ -1111,7 +1154,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 				continue;
 			}
 		}
-		if (socdev->machine->dai_link[i].codec_dai->type == 
+		if (socdev->machine->dai_link[i].codec_dai->type ==
 			SND_SOC_DAI_AC97_BUS)
 			ac97 = 1;
 	}
@@ -1122,7 +1165,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 
 	ret = snd_card_register(codec->card);
 	if (ret < 0) {
-		printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n",
+		printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
 				codec->name);
 		goto out;
 	}
@@ -1146,7 +1189,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev)
 
 	err = device_create_file(socdev->dev, &dev_attr_codec_reg);
 	if (err < 0)
-		printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n");
+		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
 	mutex_unlock(&codec->mutex);
 
@@ -1166,13 +1209,13 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev)
 {
 	struct snd_soc_codec *codec = socdev->codec;
 #ifdef CONFIG_SND_SOC_AC97_BUS
-	struct snd_soc_codec_dai *codec_dai;
+	struct snd_soc_dai *codec_dai;
 	int i;
 #endif
 
 	mutex_lock(&codec->mutex);
 #ifdef CONFIG_SND_SOC_AC97_BUS
-	for(i = 0; i < codec->num_dai; i++) {
+	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
 		if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
 			soc_ac97_dev_unregister(codec);
@@ -1282,7 +1325,8 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
 	for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
 		;
 	val = snd_soc_read(codec, e->reg);
-	ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
+	ucontrol->value.enumerated.item[0]
+		= (val >> e->shift_l) & (bitmask - 1);
 	if (e->shift_l != e->shift_r)
 		ucontrol->value.enumerated.item[1] =
 			(val >> e->shift_r) & (bitmask - 1);
@@ -1576,7 +1620,8 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
 	val = val << shift;
 	val2 = val2 << shift;
 
-	if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0)
+	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);
@@ -1584,6 +1629,204 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
 
+/**
+ * snd_soc_info_volsw_s8 - signed mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a signed mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	int max = (signed char)((kcontrol->private_value >> 16) & 0xff);
+	int min = (signed char)((kcontrol->private_value >> 24) & 0xff);
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = max-min;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);
+
+/**
+ * snd_soc_get_volsw_s8 - signed mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to get the value of a signed mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int min = (signed char)((kcontrol->private_value >> 24) & 0xff);
+	int val = snd_soc_read(codec, reg);
+
+	ucontrol->value.integer.value[0] =
+		((signed char)(val & 0xff))-min;
+	ucontrol->value.integer.value[1] =
+		((signed char)((val >> 8) & 0xff))-min;
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8);
+
+/**
+ * snd_soc_put_volsw_sgn - signed mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to set the value of a signed mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg = kcontrol->private_value & 0xff;
+	int min = (signed char)((kcontrol->private_value >> 24) & 0xff);
+	unsigned short val;
+
+	val = (ucontrol->value.integer.value[0]+min) & 0xff;
+	val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;
+
+	return snd_soc_update_bits(codec, reg, 0xffff, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
+
+/**
+ * snd_soc_dai_set_sysclk - configure DAI system or master clock.
+ * @dai: DAI
+ * @clk_id: DAI specific clock ID
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the DAI master (MCLK) or system (SYSCLK) clocking.
+ */
+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);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
+
+/**
+ * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
+ * @dai: DAI
+ * @clk_id: DAI specific clock divider ID
+ * @div: new clock divisor.
+ *
+ * Configures the clock dividers. This is used to derive the best DAI bit and
+ * frame clocks from the system or master clock. It's best to set the DAI bit
+ * and frame clocks as low as possible to save system power.
+ */
+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);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
+
+/**
+ * snd_soc_dai_set_pll - configure DAI PLL.
+ * @dai: DAI
+ * @pll_id: DAI specific PLL ID
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+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);
+	else
+		return -EINVAL;
+}
+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);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
+
+/**
+ * snd_soc_dai_set_tdm_slot - configure DAI TDM.
+ * @dai: DAI
+ * @mask: DAI specific mask representing used slots.
+ * @slots: Number of slots in use.
+ *
+ * Configures a DAI for TDM operation. Both mask and slots are codec and DAI
+ * specific.
+ */
+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);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
+
+/**
+ * snd_soc_dai_set_tristate - configure DAI system or master clock.
+ * @dai: DAI
+ * @tristate: tristate enable
+ *
+ * Tristates the DAI so that others can use it.
+ */
+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);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
+
+/**
+ * snd_soc_dai_digital_mute - configure DAI system or master clock.
+ * @dai: DAI
+ * @mute: mute enable
+ *
+ * Mutes the DAI DAC.
+ */
+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);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);
+
 static int __devinit snd_soc_init(void)
 {
 	printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
@@ -1592,7 +1835,7 @@ static int __devinit snd_soc_init(void)
 
 static void snd_soc_exit(void)
 {
- 	platform_driver_unregister(&soc_driver);
+	platform_driver_unregister(&soc_driver);
 }
 
 module_init(snd_soc_init);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index af3326c63504..2c87061c2a6b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -10,11 +10,6 @@
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    12th Aug 2005   Initial version.
- *    25th Oct 2005   Implemented path power domain.
- *    18th Dec 2005   Implemented machine and stream level power domain.
- *
  *  Features:
  *    o Changes power status of internal codec blocks depending on the
  *      dynamic configuration of codec internal audio paths and active
@@ -50,23 +45,10 @@
 #include <sound/initval.h>
 
 /* debug */
-#define DAPM_DEBUG 0
-#if DAPM_DEBUG
+#ifdef DEBUG
 #define dump_dapm(codec, action) dbg_dump_dapm(codec, action)
-#define dbg(format, arg...) printk(format, ## arg)
 #else
 #define dump_dapm(codec, action)
-#define dbg(format, arg...)
-#endif
-
-#define POP_DEBUG 0
-#if POP_DEBUG
-#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */
-#define pop_wait(time) schedule_timeout_uninterruptible(msecs_to_jiffies(time))
-#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME)
-#else
-#define pop_dbg(format, arg...)
-#define pop_wait(time)
 #endif
 
 /* dapm power sequences - make this per codec in the future */
@@ -85,6 +67,28 @@ static int dapm_status = 1;
 module_param(dapm_status, int, 0);
 MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
 
+static unsigned int pop_time;
+
+static void pop_wait(void)
+{
+	if (pop_time)
+		schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time));
+}
+
+static void pop_dbg(const char *fmt, ...)
+{
+	va_list args;
+
+	va_start(args, fmt);
+
+	if (pop_time) {
+		vprintk(fmt, args);
+		pop_wait();
+	}
+
+	va_end(args);
+}
+
 /* create a new dapm widget */
 static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
 	const struct snd_soc_dapm_widget *_widget)
@@ -222,11 +226,12 @@ 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);
+			widget->power ? "on" : "off", pop_time);
 		snd_soc_write(codec, widget->reg, new);
-		pop_wait(POP_TIME);
+		pop_wait();
 	}
-	dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change);
+	pr_debug("reg %x old %x new %x change %d\n", widget->reg,
+		 old, new, change);
 	return change;
 }
 
@@ -448,6 +453,25 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
 }
 
 /*
+ * Handler for generic register modifier widget.
+ */
+int dapm_reg_event(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol, int event)
+{
+	unsigned int val;
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		val = w->on_val;
+	else
+		val = w->off_val;
+
+	snd_soc_update_bits(w->codec, -(w->reg + 1),
+			    w->mask << w->shift, val << w->shift);
+
+	return 0;
+}
+
+/*
  * Scan each dapm widget for complete audio path.
  * A complete path is a route that has valid endpoints i.e.:-
  *
@@ -565,8 +589,8 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 			/* call any power change event handlers */
 			if (power_change) {
 				if (w->event) {
-					dbg("power %s event for %s flags %x\n",
-						w->power ? "on" : "off", w->name, w->event_flags);
+					pr_debug("power %s event for %s flags %x\n",
+						 w->power ? "on" : "off", w->name, w->event_flags);
 					if (power) {
 						/* power up event */
 						if (w->event_flags & SND_SOC_DAPM_PRE_PMU) {
@@ -608,7 +632,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 	return ret;
 }
 
-#if DAPM_DEBUG
+#ifdef DEBUG
 static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
 {
 	struct snd_soc_dapm_widget *w;
@@ -693,8 +717,10 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
 			path->connect = 0; /* old connection must be powered down */
 	}
 
-	if (found)
+	if (found) {
 		dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+		dump_dapm(widget->codec, "mux power update");
+	}
 
 	return 0;
 }
@@ -730,8 +756,10 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
 		break;
 	}
 
-	if (found)
+	if (found) {
 		dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
+		dump_dapm(widget->codec, "mixer power update");
+	}
 
 	return 0;
 }
@@ -768,21 +796,18 @@ static ssize_t dapm_widget_show(struct device *dev,
 		}
 	}
 
-	switch(codec->dapm_state){
-	case SNDRV_CTL_POWER_D0:
-		state = "D0";
+	switch (codec->bias_level) {
+	case SND_SOC_BIAS_ON:
+		state = "On";
 		break;
-	case SNDRV_CTL_POWER_D1:
-		state = "D1";
+	case SND_SOC_BIAS_PREPARE:
+		state = "Prepare";
 		break;
-	case SNDRV_CTL_POWER_D2:
-		state = "D2";
+	case SND_SOC_BIAS_STANDBY:
+		state = "Standby";
 		break;
-	case SNDRV_CTL_POWER_D3hot:
-		state = "D3hot";
-		break;
-	case SNDRV_CTL_POWER_D3cold:
-		state = "D3cold";
+	case SND_SOC_BIAS_OFF:
+		state = "Off";
 		break;
 	}
 	count += sprintf(buf + count, "PM State: %s\n", state);
@@ -792,20 +817,51 @@ static ssize_t dapm_widget_show(struct device *dev,
 
 static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
 
+/* pop/click delay times */
+static ssize_t dapm_pop_time_show(struct device *dev,
+	struct device_attribute *attr, char *buf)
+{
+	return sprintf(buf, "%d\n", pop_time);
+}
+
+static ssize_t dapm_pop_time_store(struct device *dev,
+				   struct device_attribute *attr,
+				   const char *buf, size_t count)
+
+{
+	unsigned long val;
+
+	if (strict_strtoul(buf, 10, &val) >= 0)
+		pop_time = val;
+	else
+		printk(KERN_ERR "Unable to parse pop_time setting\n");
+
+	return count;
+}
+
+static DEVICE_ATTR(dapm_pop_time, 0744, dapm_pop_time_show,
+		   dapm_pop_time_store);
+
 int snd_soc_dapm_sys_add(struct device *dev)
 {
 	int ret = 0;
 
-	if (dapm_status)
+	if (dapm_status) {
 		ret = device_create_file(dev, &dev_attr_dapm_widget);
 
+		if (ret == 0)
+			ret = device_create_file(dev, &dev_attr_dapm_pop_time);
+	}
+
 	return ret;
 }
 
 static void snd_soc_dapm_sys_remove(struct device *dev)
 {
-	if (dapm_status)
+	if (dapm_status) {
+		device_remove_file(dev, &dev_attr_dapm_pop_time);
 		device_remove_file(dev, &dev_attr_dapm_widget);
+	}
 }
 
 /* free all dapm widgets and resources */
@@ -826,8 +882,25 @@ static void dapm_free_widgets(struct snd_soc_codec *codec)
 	}
 }
 
+static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
+	char *pin, int status)
+{
+	struct snd_soc_dapm_widget *w;
+
+	list_for_each_entry(w, &codec->dapm_widgets, list) {
+		if (!strcmp(w->name, pin)) {
+			pr_debug("dapm: %s: pin %s\n", codec->name, pin);
+			w->connected = status;
+			return 0;
+		}
+	}
+
+	pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin);
+	return -EINVAL;
+}
+
 /**
- * snd_soc_dapm_sync_endpoints - scan and power dapm paths
+ * snd_soc_dapm_sync - scan and power dapm paths
  * @codec: audio codec
  *
  * Walks all dapm audio paths and powers widgets according to their
@@ -835,27 +908,16 @@ static void dapm_free_widgets(struct snd_soc_codec *codec)
  *
  * Returns 0 for success.
  */
-int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec)
+int snd_soc_dapm_sync(struct snd_soc_codec *codec)
 {
-	return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+	int ret = dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
+	dump_dapm(codec, "sync");
+	return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints);
+EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
 
-/**
- * snd_soc_dapm_connect_input - connect dapm widgets
- * @codec: audio codec
- * @sink: name of target widget
- * @control: mixer control name
- * @source: name of source name
- *
- * Connects 2 dapm widgets together via a named audio path. The sink is
- * the widget receiving the audio signal, whilst the source is the sender
- * of the audio signal.
- *
- * Returns 0 for success else error.
- */
-int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink,
-	const char * control, const char *source)
+static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
+	const char *sink, const char *control, const char *source)
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
@@ -957,9 +1019,64 @@ err:
 	kfree(path);
 	return ret;
 }
+
+/**
+ * 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
+ * @num: number of routes
+ *
+ * Connects 2 dapm widgets together via a named audio path. The sink is
+ * the widget receiving the audio signal, whilst the source is the sender
+ * of the audio signal.
+ *
+ * Returns 0 for success else error. On error all resources can be freed
+ * with a call to snd_soc_card_free().
+ */
+int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
+			    const struct snd_soc_dapm_route *route, int num)
+{
+	int i, ret;
+
+	for (i = 0; i < num; i++) {
+		ret = snd_soc_dapm_add_route(codec, route->sink,
+					     route->control, route->source);
+		if (ret < 0) {
+			printk(KERN_ERR "Failed to add route %s->%s\n",
+			       route->source,
+			       route->sink);
+			return ret;
+		}
+		route++;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
+
+/**
  * snd_soc_dapm_new_widgets - add new dapm widgets
  * @codec: audio codec
  *
@@ -1234,6 +1351,33 @@ int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
 
 /**
+ * snd_soc_dapm_new_controls - create new dapm controls
+ * @codec: audio codec
+ * @widget: widget array
+ * @num: number of widgets
+ *
+ * Creates new DAPM controls based upon the templates.
+ *
+ * Returns 0 for success else error.
+ */
+int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
+	const struct snd_soc_dapm_widget *widget,
+	int num)
+{
+	int i, ret;
+
+	for (i = 0; i < num; i++) {
+		ret = snd_soc_dapm_new_control(codec, widget);
+		if (ret < 0)
+			return ret;
+		widget++;
+	}
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls);
+
+
+/**
  * snd_soc_dapm_stream_event - send a stream event to the dapm core
  * @codec: audio codec
  * @stream: stream name
@@ -1257,8 +1401,8 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
 	{
 		if (!w->sname)
 			continue;
-		dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname,
-			stream, event);
+		pr_debug("widget %s\n %s stream %s event %d\n",
+			 w->name, w->sname, stream, event);
 		if (strstr(w->sname, stream)) {
 			switch(event) {
 			case SND_SOC_DAPM_STREAM_START:
@@ -1294,53 +1438,81 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 
 /**
- * snd_soc_dapm_device_event - send a device event to the dapm core
+ * snd_soc_dapm_set_bias_level - set the bias level for the system
  * @socdev: audio device
- * @event: device event
+ * @level: level to configure
  *
- * Sends a device event to the dapm core. The core then makes any
- * necessary machine or codec power changes..
+ * Configure the bias (power) levels for the SoC audio device.
  *
  * Returns 0 for success else error.
  */
-int snd_soc_dapm_device_event(struct snd_soc_device *socdev, int event)
+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;
+	int ret = 0;
 
-	if (machine->dapm_event)
-		machine->dapm_event(machine, event);
-	if (codec->dapm_event)
-		codec->dapm_event(codec, event);
-	return 0;
+	if (machine->set_bias_level)
+		ret = machine->set_bias_level(machine, level);
+	if (ret == 0 && codec->set_bias_level)
+		ret = codec->set_bias_level(codec, level);
+
+	return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_dapm_device_event);
 
 /**
- * snd_soc_dapm_set_endpoint - set audio endpoint status
+ * snd_soc_dapm_enable_pin - enable pin.
+ * @snd_soc_codec: SoC codec
+ * @pin: pin name
+ *
+ * Enables input/output pin and it's parents or children widgets iff there is
+ * a valid audio route and active audio stream.
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin)
+{
+	return snd_soc_dapm_set_pin(codec, pin, 1);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
+
+/**
+ * snd_soc_dapm_disable_pin - disable pin.
+ * @codec: SoC codec
+ * @pin: pin name
+ *
+ * Disables input/output pin and it's parents or children widgets.
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
+{
+	return snd_soc_dapm_set_pin(codec, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
+
+/**
+ * snd_soc_dapm_get_pin_status - get audio pin status
  * @codec: audio codec
- * @endpoint: audio signal endpoint (or start point)
- * @status: point status
+ * @pin: audio signal pin endpoint (or start point)
  *
- * Set audio endpoint status - connected or disconnected.
+ * Get audio pin status - connected or disconnected.
  *
- * Returns 0 for success else error.
+ * Returns 1 for connected otherwise 0.
  */
-int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
-	char *endpoint, int status)
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin)
 {
 	struct snd_soc_dapm_widget *w;
 
 	list_for_each_entry(w, &codec->dapm_widgets, list) {
-		if (!strcmp(w->name, endpoint)) {
-			w->connected = status;
-			return 0;
-		}
+		if (!strcmp(w->name, pin))
+			return w->connected;
 	}
 
-	return -ENODEV;
+	return 0;
 }
-EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
 
 /**
  * snd_soc_dapm_free - free dapm resources