[PATCH 0/4] ASoC updates

[PATCH 0/4] ASoC updates

[Mark Brown]

This series fixes a couple of bugs (neither of which affect 2.6.27) and
introduces a new API call for marking disconnected pins, mostly as a
placeholder for future development so machine drivers can start using it
now and then benefit when a more substantial implementation follows.

The following changes since commit 3090258fed179996ac4daecb482bad28d84cb054:
  Jean Delvare (1):
        ALSA: ASoC: Convert tlv320aic3x to a new-style i2c driver (v2)

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Mark Brown (4):
      ASoC: Fix inverted input PGA mute bits in WM8903
      ASoC: Fix build of GTA01 audio driver
      ASoC: Allow machine drivers to mark pins as not connected
      ASoC: Use snd_soc_dapm_nc_pin() in GTA01 audio driver

 include/sound/soc-dapm.h           |    1 +
 sound/soc/codecs/wm8903.c          |    4 ++--
 sound/soc/s3c24xx/neo1973_wm8753.c |   17 ++++++++---------
 sound/soc/soc-dapm.c               |   20 ++++++++++++++++++++
 4 files changed, 31 insertions(+), 11 deletions(-)

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Wed, 24 Sep 2008 12:59:07 +0100,
Mark Brown wrote:
> 
> This series fixes a couple of bugs (neither of which affect 2.6.27) and
> introduces a new API call for marking disconnected pins, mostly as a
> placeholder for future development so machine drivers can start using it
> now and then benefit when a more substantial implementation follows.
> 
> The following changes since commit 3090258fed179996ac4daecb482bad28d84cb054:
>   Jean Delvare (1):
>         ALSA: ASoC: Convert tlv320aic3x to a new-style i2c driver (v2)
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Applied.  Thanks.


Takashi

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc:
  Takashi Iwai (1):
        ALSA: ASoC - Remove unnecessary inclusion of linux/version.h

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

David Anders (1):
      ASoC: Add new parameter to s3c24xx_pcm_enqueue

Marek Vasut (1):
      ASoC: Add Palm/PXA27x unified ASoC audio driver

Mark Brown (1):
      ASoC: Remove core version number

Troy Kisky (1):
      ASoC: TLV320AIC23B Support more sample rates

 arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
 include/sound/soc.h                       |    2 -
 sound/soc/codecs/tlv320aic23.c            |  227 +++++++++++++++++++-----
 sound/soc/pxa/Kconfig                     |    9 +
 sound/soc/pxa/Makefile                    |    2 +
 sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c24xx-pcm.c           |   12 +-
 sound/soc/soc-core.c                      |    1 -
 8 files changed, 483 insertions(+), 52 deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc:
  Takashi Iwai (1):
        ALSA: ASoC - Remove unnecessary inclusion of linux/version.h

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

This series drops the TLV320AIC23 patch for now and makes the Palm
driver non-modular for the time being due to the platform data dodge
being used.

David Anders (1):
      ASoC: Add new parameter to s3c24xx_pcm_enqueue

Grazvydas Ignotas (1):
      ALSA: ASoC: TWL4030 codec - fix 256*Fs clock

Marek Vasut (1):
      ASoC: Add Palm/PXA27x unified ASoC audio driver

Mark Brown (1):
      ASoC: Remove core version number

 arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
 include/sound/soc.h                       |    2 -
 sound/soc/codecs/twl4030.c                |    4 +-
 sound/soc/pxa/Kconfig                     |    9 +
 sound/soc/pxa/Makefile                    |    2 +
 sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
 sound/soc/s3c24xx/s3c24xx-pcm.c           |   12 +-
 sound/soc/soc-core.c                      |    1 -
 8 files changed, 303 insertions(+), 9 deletions(-)
 create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
 create mode 100644 sound/soc/pxa/palm27x.c

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Thu, 6 Nov 2008 11:38:32 +0000,
Mark Brown wrote:
> 
> The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc:
>   Takashi Iwai (1):
>         ALSA: ASoC - Remove unnecessary inclusion of linux/version.h
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, pulled now.


Takashi

> 
> This series drops the TLV320AIC23 patch for now and makes the Palm
> driver non-modular for the time being due to the platform data dodge
> being used.
> 
> David Anders (1):
>       ASoC: Add new parameter to s3c24xx_pcm_enqueue
> 
> Grazvydas Ignotas (1):
>       ALSA: ASoC: TWL4030 codec - fix 256*Fs clock
> 
> Marek Vasut (1):
>       ASoC: Add Palm/PXA27x unified ASoC audio driver
> 
> Mark Brown (1):
>       ASoC: Remove core version number
> 
>  arch/arm/mach-pxa/include/mach/palmasoc.h |   13 ++
>  include/sound/soc.h                       |    2 -
>  sound/soc/codecs/twl4030.c                |    4 +-
>  sound/soc/pxa/Kconfig                     |    9 +
>  sound/soc/pxa/Makefile                    |    2 +
>  sound/soc/pxa/palm27x.c                   |  269 +++++++++++++++++++++++++++++
>  sound/soc/s3c24xx/s3c24xx-pcm.c           |   12 +-
>  sound/soc/soc-core.c                      |    1 -
>  8 files changed, 303 insertions(+), 9 deletions(-)
>  create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h
>  create mode 100644 sound/soc/pxa/palm27x.c
> 

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit e18c94d20224f3df584531a48d944d8cccfda46d:
  Grazvydas Ignotas (1):
        ALSA: ASoC: TWL4030 codec - fix 256*Fs clock

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Christian Pellegrin (1):
      ASoC: s3c24xx 8 bit sound fix

Hugo Villeneuve (1):
      ASoC: Add Right-Justified mode and Codec clock master to davinci-i2s

Naresh Medisetty (1):
      ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback

Troy Kisky (1):
      ASoC: TLV320AIC23B Support more sample rates

 sound/soc/codecs/tlv320aic23.c  |  222 +++++++++++++++++++++++++++++++--------
 sound/soc/davinci/davinci-i2s.c |   89 ++++++++++++++--
 sound/soc/s3c24xx/s3c24xx-i2s.c |    7 ++
 3 files changed, 263 insertions(+), 55 deletions(-)

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Wed, 12 Nov 2008 11:55:07 +0000,
Mark Brown wrote:
> 
> The following changes since commit e18c94d20224f3df584531a48d944d8cccfda46d:
>   Grazvydas Ignotas (1):
>         ALSA: ASoC: TWL4030 codec - fix 256*Fs clock
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, pulled in, and pushed out.


Takashi

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit fb0ef645f2c546f8297b2fbf9b2b8fff4a7455e8:
  Naresh Medisetty (1):
        ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Jarkko Nikula (2):
      ASoC: Fix supported sample rates of TWL4030 audio codec
      ASoC: OMAP: Add more supported sample rates into McBSP DAI driver

Mark Brown (2):
      ASoC: Revert "ASoC: Add new parameter to s3c24xx_pcm_enqueue"
      ASoC: Add WM8728 codec driver

 sound/soc/codecs/Kconfig        |    4 +
 sound/soc/codecs/Makefile       |    2 +
 sound/soc/codecs/twl4030.c      |    2 +-
 sound/soc/codecs/wm8728.c       |  574 +++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8728.h       |   30 ++
 sound/soc/omap/omap-mcbsp.c     |    4 +-
 sound/soc/s3c24xx/s3c24xx-pcm.c |   12 +-
 7 files changed, 616 insertions(+), 12 deletions(-)
 create mode 100644 sound/soc/codecs/wm8728.c
 create mode 100644 sound/soc/codecs/wm8728.h

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Fri, 14 Nov 2008 14:57:06 +0000,
Mark Brown wrote:
> 
> The following changes since commit fb0ef645f2c546f8297b2fbf9b2b8fff4a7455e8:
>   Naresh Medisetty (1):
>         ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Thanks, applied now.


Takashi

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit 6f2a974bfc8d3be7a30674c71e2fef003b39a8d2:
  Daniel Mack (1):
        ASoC: tlv320aic3x: headset/button press support

are available in the git repository at:

  git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Mark Brown (4):
      ASoC: Push debugfs files out of the snd_soc_device structure
      ASoC: Remove device from platform suspend and resume operations
      ASoC: Remove platform device from DAI suspend and resume operations
      ASoC: Remove obsolete declaration of struct snd_soc_clock_info

 include/sound/soc-dai.h         |    6 +--
 include/sound/soc.h             |   15 +++----
 sound/soc/atmel/atmel-pcm.c     |    6 +--
 sound/soc/atmel/atmel_ssc_dai.c |    6 +--
 sound/soc/au1x/psc-ac97.c       |    6 +--
 sound/soc/au1x/psc-i2s.c        |    6 +--
 sound/soc/blackfin/bf5xx-ac97.c |    6 +--
 sound/soc/blackfin/bf5xx-i2s.c  |    6 +--
 sound/soc/pxa/pxa-ssp.c         |    6 +--
 sound/soc/pxa/pxa2xx-ac97.c     |    6 +--
 sound/soc/pxa/pxa2xx-i2s.c      |    6 +--
 sound/soc/s3c24xx/s3c2412-i2s.c |   16 +++----
 sound/soc/s3c24xx/s3c24xx-i2s.c |    6 +--
 sound/soc/soc-core.c            |   85 +++++++++++++++++++++-----------------
 14 files changed, 83 insertions(+), 99 deletions(-)

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Thu, 4 Dec 2008 11:25:43 +0000,
Mark Brown wrote:
> 
> The following changes since commit 6f2a974bfc8d3be7a30674c71e2fef003b39a8d2:
>   Daniel Mack (1):
>         ASoC: tlv320aic3x: headset/button press support
> 
> are available in the git repository at:
> 
>   git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai

Pulled now.  Thanks.


Takashi

[PATCH 0/4] ASoC updates

[Mark Brown]

The following changes since commit 49d92c7d5bbd158734bc34ed578a68b214a48583:
  Stanley.Miao (1):
        ASoC: TWL4030: hands-free start-up sequence.

are available in the git repository at:

  git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai

Alexander Beregalov (1):
      ASoC: switch davinci DPRINTK to pr_debug()

Mark Brown (3):
      ASoC: Ease merge difficulties from new architectures
      ASoC: Complain if we fail to create DAPM controls
      ASoC: Add WM8350 AudioPlus codec driver

 include/linux/mfd/wm8350/audio.h |   38 +-
 sound/soc/Kconfig                |   10 +-
 sound/soc/Makefile               |   12 +-
 sound/soc/codecs/Kconfig         |    4 +
 sound/soc/codecs/Makefile        |    2 +
 sound/soc/codecs/wm8350.c        | 1583 ++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8350.h        |   20 +
 sound/soc/davinci/davinci-pcm.c  |   18 +-
 sound/soc/soc-dapm.c             |    6 +-
 9 files changed, 1669 insertions(+), 24 deletions(-)
 create mode 100644 sound/soc/codecs/wm8350.c
 create mode 100644 sound/soc/codecs/wm8350.h

[PATCH 1/4] ASoC: switch davinci DPRINTK to pr_debug()

[Mark Brown]

From: Alexander Beregalov <a.beregalov@gmail.com>

Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/davinci/davinci-pcm.c |   18 ++++++------------
 1 files changed, 6 insertions(+), 12 deletions(-)

diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index bc83e1c..74abc9b 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -14,6 +14,7 @@
 #include <linux/platform_device.h>
 #include <linux/slab.h>
 #include <linux/dma-mapping.h>
+#include <linux/kernel.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -24,13 +25,6 @@
 
 #include "davinci-pcm.h"
 
-#define DAVINCI_PCM_DEBUG 0
-#if DAVINCI_PCM_DEBUG
-#define DPRINTK(x...) printk(KERN_DEBUG x)
-#else
-#define DPRINTK(x...)
-#endif
-
 static struct snd_pcm_hardware davinci_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
@@ -78,8 +72,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	dma_offset = prtd->period * period_size;
 	dma_pos = runtime->dma_addr + dma_offset;
 
-	DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x "
-		"period_size=%x\n", lch, dma_pos, period_size);
+	pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
+		"dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size);
 
 	data_type = prtd->params->data_type;
 	count = period_size / data_type;
@@ -112,7 +106,7 @@ static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
 	struct snd_pcm_substream *substream = data;
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
 
-	DPRINTK("lch=%d, status=0x%x\n", lch, ch_status);
+	pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status);
 
 	if (unlikely(ch_status != DMA_COMPLETE))
 		return;
@@ -316,8 +310,8 @@ static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
 					   &buf->addr, GFP_KERNEL);
 
-	DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
-		(void *) buf->area, (void *) buf->addr, size);
+	pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, "
+		"size=%d\n", (void *) buf->area, (void *) buf->addr, size);
 
 	if (!buf->area)
 		return -ENOMEM;
-- 
1.5.6.5

[PATCH 2/4] ASoC: Ease merge difficulties from new architectures

[Mark Brown]

Rather than listing lots of architectures per line in Kconfig and
Makefile, causing merge conflicts all the time, have one per line
in alphabetical order.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/Kconfig  |   10 +++++-----
 sound/soc/Makefile |   12 ++++++++++--
 2 files changed, 15 insertions(+), 7 deletions(-)

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 615ebf0..ef025c6 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -22,16 +22,16 @@ if SND_SOC
 config SND_SOC_AC97_BUS
 	bool
 
-# All the supported Soc's
+# All the supported SoCs
 source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
+source "sound/soc/blackfin/Kconfig"
+source "sound/soc/davinci/Kconfig"
+source "sound/soc/fsl/Kconfig"
+source "sound/soc/omap/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
 source "sound/soc/sh/Kconfig"
-source "sound/soc/fsl/Kconfig"
-source "sound/soc/davinci/Kconfig"
-source "sound/soc/omap/Kconfig"
-source "sound/soc/blackfin/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 4d475c3..86a9b1f 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,13 @@
 snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
-obj-$(CONFIG_SND_SOC)	+= codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/
-obj-$(CONFIG_SND_SOC)	+= omap/ au1x/ blackfin/
+obj-$(CONFIG_SND_SOC)	+= codecs/
+obj-$(CONFIG_SND_SOC)	+= atmel/
+obj-$(CONFIG_SND_SOC)	+= au1x/
+obj-$(CONFIG_SND_SOC)	+= blackfin/
+obj-$(CONFIG_SND_SOC)	+= davinci/
+obj-$(CONFIG_SND_SOC)	+= fsl/
+obj-$(CONFIG_SND_SOC)	+= omap/
+obj-$(CONFIG_SND_SOC)	+= pxa/
+obj-$(CONFIG_SND_SOC)	+= s3c24xx/
+obj-$(CONFIG_SND_SOC)	+= sh/
-- 
1.5.6.5

[PATCH 3/4] ASoC: Complain if we fail to create DAPM controls

[Mark Brown]

This should never happen and it's helpful to identify the specific control
that failed when it does happen.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 sound/soc/soc-dapm.c |    6 +++++-
 1 files changed, 5 insertions(+), 1 deletions(-)

diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 61d7d85..8863edd 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1320,8 +1320,12 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
 
 	for (i = 0; i < num; i++) {
 		ret = snd_soc_dapm_new_control(codec, widget);
-		if (ret < 0)
+		if (ret < 0) {
+			printk(KERN_ERR
+			       "ASoC: Failed to create DAPM control %s: %d\n",
+			       widget->name, ret);
 			return ret;
+		}
 		widget++;
 	}
 	return 0;
-- 
1.5.6.5

[PATCH 4/4] ASoC: Add WM8350 AudioPlus codec driver

[Mark Brown]

The WM8350 is an integrated audio and power management subsystem which
provides a single-chip solution for portable audio and multimedia systems.

The integrated audio CODEC provides all the necessary functions for
high-quality stereo recording and playback. Programmable on-chip
amplifiers allow for the direct connection of headphones and microphones
with a minimum of external components. A programmable low-noise bias
voltage is available to feed one or more electret microphones.
Additional audio features include programmable high-pass filter in the
ADC input path.

This driver was originally written by Liam Girdwood with further updates
from me.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/linux/mfd/wm8350/audio.h |   38 +-
 sound/soc/codecs/Kconfig         |    4 +
 sound/soc/codecs/Makefile        |    2 +
 sound/soc/codecs/wm8350.c        | 1583 ++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/wm8350.h        |   20 +
 5 files changed, 1643 insertions(+), 4 deletions(-)
 create mode 100644 sound/soc/codecs/wm8350.c
 create mode 100644 sound/soc/codecs/wm8350.h

diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h
index 217bb22..af95a1d 100644
--- a/include/linux/mfd/wm8350/audio.h
+++ b/include/linux/mfd/wm8350/audio.h
@@ -1,7 +1,7 @@
 /*
  * audio.h  --  Audio Driver for Wolfson WM8350 PMIC
  *
- * Copyright 2007 Wolfson Microelectronics PLC
+ * Copyright 2007, 2008 Wolfson Microelectronics PLC
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -70,9 +70,9 @@
 #define WM8350_CODEC_ISEL_0_5                   3	/* x0.5 */
 
 #define WM8350_VMID_OFF                         0
-#define WM8350_VMID_500K                        1
-#define WM8350_VMID_100K                        2
-#define WM8350_VMID_10K                         3
+#define WM8350_VMID_300K                        1
+#define WM8350_VMID_50K                         2
+#define WM8350_VMID_5K                          3
 
 /*
  * R40 (0x28) - Clock Control 1
@@ -591,8 +591,38 @@
 #define WM8350_IRQ_CODEC_MICSCD			41
 #define WM8350_IRQ_CODEC_MICD			42
 
+/*
+ * WM8350 Platform data.
+ *
+ * This must be initialised per platform for best audio performance.
+ * Please see WM8350 datasheet for information.
+ */
+struct wm8350_audio_platform_data {
+	int vmid_discharge_msecs;	/* VMID --> OFF discharge time */
+	int drain_msecs;	/* OFF drain time */
+	int cap_discharge_msecs;	/* Cap ON (from OFF) discharge time */
+	int vmid_charge_msecs;	/* vmid power up time */
+	u32 vmid_s_curve:2;	/* vmid enable s curve speed */
+	u32 dis_out4:2;		/* out4 discharge speed */
+	u32 dis_out3:2;		/* out3 discharge speed */
+	u32 dis_out2:2;		/* out2 discharge speed */
+	u32 dis_out1:2;		/* out1 discharge speed */
+	u32 vroi_out4:1;	/* out4 tie off */
+	u32 vroi_out3:1;	/* out3 tie off */
+	u32 vroi_out2:1;	/* out2 tie off */
+	u32 vroi_out1:1;	/* out1 tie off */
+	u32 vroi_enable:1;	/* enable tie off */
+	u32 codec_current_on:2;	/* current level ON */
+	u32 codec_current_standby:2;	/* current level STANDBY */
+	u32 codec_current_charge:2;	/* codec current @ vmid charge */
+};
+
+struct snd_soc_codec;
+
 struct wm8350_codec {
 	struct platform_device *pdev;
+	struct snd_soc_codec *codec;
+	struct wm8350_audio_platform_data *platform_data;
 };
 
 #endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index bf68052..c41289b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -13,6 +13,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TWL4030 if TWL4030_CORE
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
+	select SND_SOC_WM8350 if MFD_WM8350
 	select SND_SOC_WM8510 if (I2C || SPI_MASTER)
 	select SND_SOC_WM8580 if I2C
 	select SND_SOC_WM8728 if (I2C || SPI_MASTER)
@@ -100,6 +101,9 @@ config SND_SOC_UDA134X
 config SND_SOC_UDA1380
         tristate
 
+config SND_SOC_WM8350
+	tristate
+
 config SND_SOC_WM8510
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 9a20fdd..c4ddc9a 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -12,6 +12,7 @@ snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-twl4030-objs := twl4030.o
 snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
+snd-soc-wm8350-objs := wm8350.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8580-objs := wm8580.o
 snd-soc-wm8728-objs := wm8728.o
@@ -39,6 +40,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o
 obj-$(CONFIG_SND_SOC_WM8728)	+= snd-soc-wm8728.o
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
new file mode 100644
index 0000000..4bbfb5a
--- /dev/null
+++ b/sound/soc/codecs/wm8350.c
@@ -0,0 +1,1583 @@
+/*
+ * wm8350.c -- WM8350 ALSA SoC audio driver
+ *
+ * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/wm8350/audio.h>
+#include <linux/mfd/wm8350/core.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8350.h"
+
+#define WM8350_OUTn_0dB 0x39
+
+#define WM8350_RAMP_NONE	0
+#define WM8350_RAMP_UP		1
+#define WM8350_RAMP_DOWN	2
+
+/* We only include the analogue supplies here; the digital supplies
+ * need to be available well before this driver can be probed.
+ */
+static const char *supply_names[] = {
+	"AVDD",
+	"HPVDD",
+};
+
+struct wm8350_output {
+	u16 active;
+	u16 left_vol;
+	u16 right_vol;
+	u16 ramp;
+	u16 mute;
+};
+
+struct wm8350_data {
+	struct snd_soc_codec codec;
+	struct wm8350_output out1;
+	struct wm8350_output out2;
+	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
+};
+
+static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec,
+					    unsigned int reg)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350->reg_cache[reg];
+}
+
+static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
+				      unsigned int reg)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350_reg_read(wm8350, reg);
+}
+
+static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+			      unsigned int value)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	return wm8350_reg_write(wm8350, reg, value);
+}
+
+/*
+ * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec)
+{
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_data->out1;
+	struct wm8350 *wm8350 = codec->control_data;
+	int left_complete = 0, right_complete = 0;
+	u16 reg, val;
+
+	/* left channel */
+	reg = wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME);
+	val = (reg & WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+
+	if (out1->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out1->left_vol) {
+			val++;
+			reg &= ~WM8350_OUT1L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else if (out1->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT1L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else
+		return 1;
+
+	/* right channel */
+	reg = wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME);
+	val = (reg & WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	if (out1->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out1->right_vol) {
+			val++;
+			reg &= ~WM8350_OUT1R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	} else if (out1->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT1R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	}
+
+	/* only hit the update bit if either volume has changed this step */
+	if (!left_complete || !right_complete)
+		wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, WM8350_OUT1_VU);
+
+	return left_complete & right_complete;
+}
+
+/*
+ * Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown.
+ */
+static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec)
+{
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out2 = &wm8350_data->out2;
+	struct wm8350 *wm8350 = codec->control_data;
+	int left_complete = 0, right_complete = 0;
+	u16 reg, val;
+
+	/* left channel */
+	reg = wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME);
+	val = (reg & WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	if (out2->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out2->left_vol) {
+			val++;
+			reg &= ~WM8350_OUT2L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else if (out2->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT2L_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
+					 reg | (val << WM8350_OUT1L_VOL_SHIFT));
+		} else
+			left_complete = 1;
+	} else
+		return 1;
+
+	/* right channel */
+	reg = wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME);
+	val = (reg & WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	if (out2->ramp == WM8350_RAMP_UP) {
+		/* ramp step up */
+		if (val < out2->right_vol) {
+			val++;
+			reg &= ~WM8350_OUT2R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	} else if (out2->ramp == WM8350_RAMP_DOWN) {
+		/* ramp step down */
+		if (val > 0) {
+			val--;
+			reg &= ~WM8350_OUT2R_VOL_MASK;
+			wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
+					 reg | (val << WM8350_OUT1R_VOL_SHIFT));
+		} else
+			right_complete = 1;
+	}
+
+	/* only hit the update bit if either volume has changed this step */
+	if (!left_complete || !right_complete)
+		wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, WM8350_OUT2_VU);
+
+	return left_complete & right_complete;
+}
+
+/*
+ * This work ramps both output PGAs at stream start/stop time to
+ * minimise pop associated with DAPM power switching.
+ * It's best to enable Zero Cross when ramping occurs to minimise any
+ * zipper noises.
+ */
+static void wm8350_pga_work(struct work_struct *work)
+{
+	struct snd_soc_codec *codec =
+	    container_of(work, struct snd_soc_codec, delayed_work.work);
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_data->out1,
+	    *out2 = &wm8350_data->out2;
+	int i, out1_complete, out2_complete;
+
+	/* do we need to ramp at all ? */
+	if (out1->ramp == WM8350_RAMP_NONE && out2->ramp == WM8350_RAMP_NONE)
+		return;
+
+	/* PGA volumes have 6 bits of resolution to ramp */
+	for (i = 0; i <= 63; i++) {
+		out1_complete = 1, out2_complete = 1;
+		if (out1->ramp != WM8350_RAMP_NONE)
+			out1_complete = wm8350_out1_ramp_step(codec);
+		if (out2->ramp != WM8350_RAMP_NONE)
+			out2_complete = wm8350_out2_ramp_step(codec);
+
+		/* ramp finished ? */
+		if (out1_complete && out2_complete)
+			break;
+
+		/* we need to delay longer on the up ramp */
+		if (out1->ramp == WM8350_RAMP_UP ||
+		    out2->ramp == WM8350_RAMP_UP) {
+			/* delay is longer over 0dB as increases are larger */
+			if (i >= WM8350_OUTn_0dB)
+				schedule_timeout_interruptible(msecs_to_jiffies
+							       (2));
+			else
+				schedule_timeout_interruptible(msecs_to_jiffies
+							       (1));
+		} else
+			udelay(50);	/* doesn't matter if we delay longer */
+	}
+
+	out1->ramp = WM8350_RAMP_NONE;
+	out2->ramp = WM8350_RAMP_NONE;
+}
+
+/*
+ * WM8350 Controls
+ */
+
+static int pga_event(struct snd_soc_dapm_widget *w,
+		     struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8350_data *wm8350_data = codec->private_data;
+	struct wm8350_output *out;
+
+	switch (w->shift) {
+	case 0:
+	case 1:
+		out = &wm8350_data->out1;
+		break;
+	case 2:
+	case 3:
+		out = &wm8350_data->out2;
+		break;
+
+	default:
+		BUG();
+		return -1;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		out->ramp = WM8350_RAMP_UP;
+		out->active = 1;
+
+		if (!delayed_work_pending(&codec->delayed_work))
+			schedule_delayed_work(&codec->delayed_work,
+					      msecs_to_jiffies(1));
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		out->ramp = WM8350_RAMP_DOWN;
+		out->active = 0;
+
+		if (!delayed_work_pending(&codec->delayed_work))
+			schedule_delayed_work(&codec->delayed_work,
+					      msecs_to_jiffies(1));
+		break;
+	}
+
+	return 0;
+}
+
+static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8350_data *wm8350_priv = codec->private_data;
+	struct wm8350_output *out = NULL;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int ret;
+	unsigned int reg = mc->reg;
+	u16 val;
+
+	/* For OUT1 and OUT2 we shadow the values and only actually write
+	 * them out when active in order to ensure the amplifier comes on
+	 * as quietly as possible. */
+	switch (reg) {
+	case WM8350_LOUT1_VOLUME:
+		out = &wm8350_priv->out1;
+		break;
+	case WM8350_LOUT2_VOLUME:
+		out = &wm8350_priv->out2;
+		break;
+	default:
+		break;
+	}
+
+	if (out) {
+		out->left_vol = ucontrol->value.integer.value[0];
+		out->right_vol = ucontrol->value.integer.value[1];
+		if (!out->active)
+			return 1;
+	}
+
+	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+	if (ret < 0)
+		return ret;
+
+	/* now hit the volume update bits (always bit 8) */
+	val = wm8350_codec_read(codec, reg);
+	wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU);
+	return 1;
+}
+
+static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8350_data *wm8350_priv = codec->private_data;
+	struct wm8350_output *out1 = &wm8350_priv->out1;
+	struct wm8350_output *out2 = &wm8350_priv->out2;
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	unsigned int reg = mc->reg;
+
+	/* If these are cached registers use the cache */
+	switch (reg) {
+	case WM8350_LOUT1_VOLUME:
+		ucontrol->value.integer.value[0] = out1->left_vol;
+		ucontrol->value.integer.value[1] = out1->right_vol;
+		return 0;
+
+	case WM8350_LOUT2_VOLUME:
+		ucontrol->value.integer.value[0] = out2->left_vol;
+		ucontrol->value.integer.value[1] = out2->right_vol;
+		return 0;
+
+	default:
+		break;
+	}
+
+	return snd_soc_get_volsw_2r(kcontrol, ucontrol);
+}
+
+/* double control with volume update */
+#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+				xinvert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+		SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+		SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw_2r, \
+	.get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+		 .rshift = xshift, .max = xmax, .invert = xinvert}, }
+
+static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
+static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
+static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
+static const char *wm8350_dacmutes[] = { "Fast", "Slow" };
+static const char *wm8350_dacfilter[] = { "Normal", "Sloping" };
+static const char *wm8350_adcfilter[] = { "None", "High Pass" };
+static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" };
+static const char *wm8350_lr[] = { "Left", "Right" };
+
+static const struct soc_enum wm8350_enum[] = {
+	SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 4, 4, wm8350_deemp),
+	SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes),
+	SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp),
+	SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol),
+	SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr),
+};
+
+static DECLARE_TLV_DB_LINEAR(pre_amp_tlv, -1200, 3525);
+static DECLARE_TLV_DB_LINEAR(out_pga_tlv, -5700, 600);
+static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1);
+static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1);
+static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1);
+
+static const unsigned int capture_sd_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 12, TLV_DB_SCALE_ITEM(-3600, 300, 1),
+	13, 15, TLV_DB_SCALE_ITEM(0, 0, 0),
+};
+
+static const struct snd_kcontrol_new wm8350_snd_controls[] = {
+	SOC_ENUM("Playback Deemphasis", wm8350_enum[0]),
+	SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]),
+	SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume",
+				WM8350_DAC_DIGITAL_VOLUME_L,
+				WM8350_DAC_DIGITAL_VOLUME_R,
+				0, 255, 0, dac_pcm_tlv),
+	SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
+	SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
+	SOC_ENUM("Playback PCM Filter", wm8350_enum[4]),
+	SOC_ENUM("Capture PCM Filter", wm8350_enum[5]),
+	SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]),
+	SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]),
+	SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume",
+				WM8350_ADC_DIGITAL_VOLUME_L,
+				WM8350_ADC_DIGITAL_VOLUME_R,
+				0, 255, 0, adc_pcm_tlv),
+	SOC_DOUBLE_TLV("Capture Sidetone Volume",
+		       WM8350_ADC_DIVIDER,
+		       8, 4, 15, 1, capture_sd_tlv),
+	SOC_WM8350_DOUBLE_R_TLV("Capture Volume",
+				WM8350_LEFT_INPUT_VOLUME,
+				WM8350_RIGHT_INPUT_VOLUME,
+				2, 63, 0, pre_amp_tlv),
+	SOC_DOUBLE_R("Capture ZC Switch",
+		     WM8350_LEFT_INPUT_VOLUME,
+		     WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0),
+	SOC_SINGLE_TLV("Left Input Left Sidetone Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Left Input Right Sidetone Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Left Input Bypass Volume",
+		       WM8350_OUTPUT_LEFT_MIXER_VOLUME,
+		       9, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Left Sidetone Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       1, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Right Sidetone Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+	SOC_SINGLE_TLV("Right Input Bypass Volume",
+		       WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
+		       13, 7, 0, out_mix_tlv),
+	SOC_SINGLE("Left Input Mixer +20dB Switch",
+		   WM8350_INPUT_MIXER_VOLUME_L, 0, 1, 0),
+	SOC_SINGLE("Right Input Mixer +20dB Switch",
+		   WM8350_INPUT_MIXER_VOLUME_R, 0, 1, 0),
+	SOC_SINGLE_TLV("Out4 Capture Volume",
+		       WM8350_INPUT_MIXER_VOLUME,
+		       1, 7, 0, out_mix_tlv),
+	SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume",
+				WM8350_LOUT1_VOLUME,
+				WM8350_ROUT1_VOLUME,
+				2, 63, 0, out_pga_tlv),
+	SOC_DOUBLE_R("Out1 Playback ZC Switch",
+		     WM8350_LOUT1_VOLUME,
+		     WM8350_ROUT1_VOLUME, 13, 1, 0),
+	SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume",
+				WM8350_LOUT2_VOLUME,
+				WM8350_ROUT2_VOLUME,
+				2, 63, 0, out_pga_tlv),
+	SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME,
+		     WM8350_ROUT2_VOLUME, 13, 1, 0),
+	SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0),
+	SOC_SINGLE_TLV("Out2 Beep Volume", WM8350_BEEP_VOLUME,
+		       5, 7, 0, out_mix_tlv),
+
+	SOC_DOUBLE_R("Out1 Playback Switch",
+		     WM8350_LOUT1_VOLUME,
+		     WM8350_ROUT1_VOLUME,
+		     14, 1, 1),
+	SOC_DOUBLE_R("Out2 Playback Switch",
+		     WM8350_LOUT2_VOLUME,
+		     WM8350_ROUT2_VOLUME,
+		     14, 1, 1),
+};
+
+/*
+ * DAPM Controls
+ */
+
+/* Left Playback Mixer */
+static const struct snd_kcontrol_new wm8350_left_play_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch",
+			WM8350_LEFT_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch",
+			WM8350_LEFT_MIXER_CONTROL, 2, 1, 0),
+	SOC_DAPM_SINGLE("Right Playback Switch",
+			WM8350_LEFT_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Left Sidetone Switch",
+			WM8350_LEFT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("Right Sidetone Switch",
+			WM8350_LEFT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Right Playback Mixer */
+static const struct snd_kcontrol_new wm8350_right_play_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 3, 1, 0),
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Sidetone Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("Right Sidetone Switch",
+			WM8350_RIGHT_MIXER_CONTROL, 1, 1, 0),
+};
+
+/* Out4 Mixer */
+static const struct snd_kcontrol_new wm8350_out4_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Right Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 12, 1, 0),
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Right Capture Switch",
+			WM8350_OUT4_MIXER_CONTROL, 9, 1, 0),
+	SOC_DAPM_SINGLE("Out3 Playback Switch",
+			WM8350_OUT4_MIXER_CONTROL, 2, 1, 0),
+	SOC_DAPM_SINGLE("Right Mixer Switch",
+			WM8350_OUT4_MIXER_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("Left Mixer Switch",
+			WM8350_OUT4_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Out3 Mixer */
+static const struct snd_kcontrol_new wm8350_out3_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left Playback Switch",
+			WM8350_OUT3_MIXER_CONTROL, 11, 1, 0),
+	SOC_DAPM_SINGLE("Left Capture Switch",
+			WM8350_OUT3_MIXER_CONTROL, 8, 1, 0),
+	SOC_DAPM_SINGLE("Out4 Playback Switch",
+			WM8350_OUT3_MIXER_CONTROL, 3, 1, 0),
+	SOC_DAPM_SINGLE("Left Mixer Switch",
+			WM8350_OUT3_MIXER_CONTROL, 0, 1, 0),
+};
+
+/* Left Input Mixer */
+static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
+	SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_L, 1, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE("PGA Capture Switch",
+			WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Right Input Mixer */
+static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
+	SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_R, 5, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
+			    WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
+	SOC_DAPM_SINGLE("PGA Capture Switch",
+			WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+};
+
+/* Left Mic Mixer */
+static const struct snd_kcontrol_new wm8350_left_mic_mixer_controls[] = {
+	SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 1, 1, 0),
+	SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 0, 1, 0),
+	SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 2, 1, 0),
+};
+
+/* Right Mic Mixer */
+static const struct snd_kcontrol_new wm8350_right_mic_mixer_controls[] = {
+	SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 9, 1, 0),
+	SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 8, 1, 0),
+	SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 10, 1, 0),
+};
+
+/* Beep Switch */
+static const struct snd_kcontrol_new wm8350_beep_switch_controls =
+SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1);
+
+/* Out4 Capture Mux */
+static const struct snd_kcontrol_new wm8350_out4_capture_controls =
+SOC_DAPM_ENUM("Route", wm8350_enum[8]);
+
+static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
+
+	SND_SOC_DAPM_PGA("IN3R PGA", WM8350_POWER_MGMT_2, 11, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IN3L PGA", WM8350_POWER_MGMT_2, 10, 0, NULL, 0),
+	SND_SOC_DAPM_PGA_E("Right Out2 PGA", WM8350_POWER_MGMT_3, 3, 0, NULL,
+			   0, pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Left Out2 PGA", WM8350_POWER_MGMT_3, 2, 0, NULL, 0,
+			   pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Right Out1 PGA", WM8350_POWER_MGMT_3, 1, 0, NULL,
+			   0, pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Left Out1 PGA", WM8350_POWER_MGMT_3, 0, 0, NULL, 0,
+			   pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	SND_SOC_DAPM_MIXER("Right Capture Mixer", WM8350_POWER_MGMT_2,
+			   7, 0, &wm8350_right_capt_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_capt_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Capture Mixer", WM8350_POWER_MGMT_2,
+			   6, 0, &wm8350_left_capt_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_capt_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Out4 Mixer", WM8350_POWER_MGMT_2, 5, 0,
+			   &wm8350_out4_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_out4_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Out3 Mixer", WM8350_POWER_MGMT_2, 4, 0,
+			   &wm8350_out3_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_out3_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Right Playback Mixer", WM8350_POWER_MGMT_2, 1, 0,
+			   &wm8350_right_play_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_play_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Playback Mixer", WM8350_POWER_MGMT_2, 0, 0,
+			   &wm8350_left_play_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_play_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Left Mic Mixer", WM8350_POWER_MGMT_2, 8, 0,
+			   &wm8350_left_mic_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_left_mic_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("Right Mic Mixer", WM8350_POWER_MGMT_2, 9, 0,
+			   &wm8350_right_mic_mixer_controls[0],
+			   ARRAY_SIZE(wm8350_right_mic_mixer_controls)),
+
+	/* virtual mixer for Beep and Out2R */
+	SND_SOC_DAPM_MIXER("Out2 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("Beep", WM8350_POWER_MGMT_3, 7, 0,
+			    &wm8350_beep_switch_controls),
+
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
+			 WM8350_POWER_MGMT_4, 3, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture",
+			 WM8350_POWER_MGMT_4, 2, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback",
+			 WM8350_POWER_MGMT_4, 5, 0),
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback",
+			 WM8350_POWER_MGMT_4, 4, 0),
+
+	SND_SOC_DAPM_MICBIAS("Mic Bias", WM8350_POWER_MGMT_1, 4, 0),
+
+	SND_SOC_DAPM_MUX("Out4 Capture Channel", SND_SOC_NOPM, 0, 0,
+			 &wm8350_out4_capture_controls),
+
+	SND_SOC_DAPM_OUTPUT("OUT1R"),
+	SND_SOC_DAPM_OUTPUT("OUT1L"),
+	SND_SOC_DAPM_OUTPUT("OUT2R"),
+	SND_SOC_DAPM_OUTPUT("OUT2L"),
+	SND_SOC_DAPM_OUTPUT("OUT3"),
+	SND_SOC_DAPM_OUTPUT("OUT4"),
+
+	SND_SOC_DAPM_INPUT("IN1RN"),
+	SND_SOC_DAPM_INPUT("IN1RP"),
+	SND_SOC_DAPM_INPUT("IN2R"),
+	SND_SOC_DAPM_INPUT("IN1LP"),
+	SND_SOC_DAPM_INPUT("IN1LN"),
+	SND_SOC_DAPM_INPUT("IN2L"),
+	SND_SOC_DAPM_INPUT("IN3R"),
+	SND_SOC_DAPM_INPUT("IN3L"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* left playback mixer */
+	{"Left Playback Mixer", "Playback Switch", "Left DAC"},
+	{"Left Playback Mixer", "Left Bypass Switch", "IN3L PGA"},
+	{"Left Playback Mixer", "Right Playback Switch", "Right DAC"},
+	{"Left Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+	{"Left Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+	/* right playback mixer */
+	{"Right Playback Mixer", "Playback Switch", "Right DAC"},
+	{"Right Playback Mixer", "Right Bypass Switch", "IN3R PGA"},
+	{"Right Playback Mixer", "Left Playback Switch", "Left DAC"},
+	{"Right Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
+	{"Right Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
+
+	/* out4 playback mixer */
+	{"Out4 Mixer", "Right Playback Switch", "Right DAC"},
+	{"Out4 Mixer", "Left Playback Switch", "Left DAC"},
+	{"Out4 Mixer", "Right Capture Switch", "Right Capture Mixer"},
+	{"Out4 Mixer", "Out3 Playback Switch", "Out3 Mixer"},
+	{"Out4 Mixer", "Right Mixer Switch", "Right Playback Mixer"},
+	{"Out4 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+	{"OUT4", NULL, "Out4 Mixer"},
+
+	/* out3 playback mixer */
+	{"Out3 Mixer", "Left Playback Switch", "Left DAC"},
+	{"Out3 Mixer", "Left Capture Switch", "Left Capture Mixer"},
+	{"Out3 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
+	{"Out3 Mixer", "Out4 Playback Switch", "Out4 Mixer"},
+	{"OUT3", NULL, "Out3 Mixer"},
+
+	/* out2 */
+	{"Right Out2 PGA", NULL, "Right Playback Mixer"},
+	{"Left Out2 PGA", NULL, "Left Playback Mixer"},
+	{"OUT2L", NULL, "Left Out2 PGA"},
+	{"OUT2R", NULL, "Right Out2 PGA"},
+
+	/* out1 */
+	{"Right Out1 PGA", NULL, "Right Playback Mixer"},
+	{"Left Out1 PGA", NULL, "Left Playback Mixer"},
+	{"OUT1L", NULL, "Left Out1 PGA"},
+	{"OUT1R", NULL, "Right Out1 PGA"},
+
+	/* ADCs */
+	{"Left ADC", NULL, "Left Capture Mixer"},
+	{"Right ADC", NULL, "Right Capture Mixer"},
+
+	/* Left capture mixer */
+	{"Left Capture Mixer", "L2 Capture Volume", "IN2L"},
+	{"Left Capture Mixer", "L3 Capture Volume", "IN3L PGA"},
+	{"Left Capture Mixer", "PGA Capture Switch", "Left Mic Mixer"},
+	{"Left Capture Mixer", NULL, "Out4 Capture Channel"},
+
+	/* Right capture mixer */
+	{"Right Capture Mixer", "L2 Capture Volume", "IN2R"},
+	{"Right Capture Mixer", "L3 Capture Volume", "IN3R PGA"},
+	{"Right Capture Mixer", "PGA Capture Switch", "Right Mic Mixer"},
+	{"Right Capture Mixer", NULL, "Out4 Capture Channel"},
+
+	/* L3 Inputs */
+	{"IN3L PGA", NULL, "IN3L"},
+	{"IN3R PGA", NULL, "IN3R"},
+
+	/* Left Mic mixer */
+	{"Left Mic Mixer", "INN Capture Switch", "IN1LN"},
+	{"Left Mic Mixer", "INP Capture Switch", "IN1LP"},
+	{"Left Mic Mixer", "IN2 Capture Switch", "IN2L"},
+
+	/* Right Mic mixer */
+	{"Right Mic Mixer", "INN Capture Switch", "IN1RN"},
+	{"Right Mic Mixer", "INP Capture Switch", "IN1RP"},
+	{"Right Mic Mixer", "IN2 Capture Switch", "IN2R"},
+
+	/* out 4 capture */
+	{"Out4 Capture Channel", NULL, "Out4 Mixer"},
+
+	/* Beep */
+	{"Beep", NULL, "IN3R PGA"},
+};
+
+static int wm8350_add_controls(struct snd_soc_codec *codec)
+{
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&wm8350_snd_controls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static int wm8350_add_widgets(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(codec,
+					wm8350_dapm_widgets,
+					ARRAY_SIZE(wm8350_dapm_widgets));
+	if (ret != 0) {
+		dev_err(codec->dev, "dapm control register failed\n");
+		return ret;
+	}
+
+	/* set up audio paths */
+	ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	if (ret != 0) {
+		dev_err(codec->dev, "DAPM route register failed\n");
+		return ret;
+	}
+
+	return snd_soc_dapm_new_widgets(codec);
+}
+
+static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				 int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	u16 fll_4;
+
+	switch (clk_id) {
+	case WM8350_MCLK_SEL_MCLK:
+		wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+				  WM8350_MCLK_SEL);
+		break;
+	case WM8350_MCLK_SEL_PLL_MCLK:
+	case WM8350_MCLK_SEL_PLL_DAC:
+	case WM8350_MCLK_SEL_PLL_ADC:
+	case WM8350_MCLK_SEL_PLL_32K:
+		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1,
+				WM8350_MCLK_SEL);
+		fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+		    ~WM8350_FLL_CLK_SRC_MASK;
+		wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
+		break;
+	}
+
+	/* MCLK direction */
+	if (dir == WM8350_MCLK_DIR_OUT)
+		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+				WM8350_MCLK_DIR);
+	else
+		wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_2,
+				  WM8350_MCLK_DIR);
+
+	return 0;
+}
+
+static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 val;
+
+	switch (div_id) {
+	case WM8350_ADC_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) &
+		    ~WM8350_ADC_CLKDIV_MASK;
+		wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div);
+		break;
+	case WM8350_DAC_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) &
+		    ~WM8350_DAC_CLKDIV_MASK;
+		wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div);
+		break;
+	case WM8350_BCLK_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_BCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_OPCLK_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_OPCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_SYS_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) &
+		    ~WM8350_MCLK_DIV_MASK;
+		wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div);
+		break;
+	case WM8350_DACLR_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+		    ~WM8350_DACLRC_RATE_MASK;
+		wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div);
+		break;
+	case WM8350_ADCLR_CLKDIV:
+		val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+		    ~WM8350_ADCLRC_RATE_MASK;
+		wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+	    ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK);
+	u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) &
+	    ~WM8350_BCLK_MSTR;
+	u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) &
+	    ~WM8350_DACLRC_ENA;
+	u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) &
+	    ~WM8350_ADCLRC_ENA;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		master |= WM8350_BCLK_MSTR;
+		dac_lrc |= WM8350_DACLRC_ENA;
+		adc_lrc |= WM8350_ADCLRC_ENA;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= 0x2 << 8;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface |= 0x1 << 8;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface |= 0x3 << 8;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface |= 0x3 << 8;	/* lg not sure which mode */
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		iface |= WM8350_AIF_LRCLK_INV | WM8350_AIF_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		iface |= WM8350_AIF_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		iface |= WM8350_AIF_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+	wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master);
+	wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc);
+	wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc);
+	return 0;
+}
+
+static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
+			      int cmd, struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) &
+	    WM8350_BCLK_MSTR;
+	int enabled = 0;
+
+	/* Check that the DACs or ADCs are enabled since they are
+	 * required for LRC in master mode. The DACs or ADCs need a
+	 * valid audio path i.e. pin -> ADC or DAC -> pin before
+	 * the LRC will be enabled in master mode. */
+	if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+		return 0;
+
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+		    (WM8350_ADCR_ENA | WM8350_ADCL_ENA);
+	} else {
+		enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) &
+		    (WM8350_DACR_ENA | WM8350_DACL_ENA);
+	}
+
+	if (!enabled) {
+		dev_err(codec->dev,
+		       "%s: invalid audio path - no clocks available\n",
+		       __func__);
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *codec_dai)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) &
+	    ~WM8350_AIF_WL_MASK;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface |= 0x1 << 10;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface |= 0x2 << 10;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface |= 0x3 << 10;
+		break;
+	}
+
+	wm8350_codec_write(codec, WM8350_AI_FORMATING, iface);
+	return 0;
+}
+
+static int wm8350_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+
+	if (mute)
+		wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+	else
+		wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+	return 0;
+}
+
+/* FLL divisors */
+struct _fll_div {
+	int div;		/* FLL_OUTDIV */
+	int n;
+	int k;
+	int ratio;		/* FLL_FRATIO */
+};
+
+/* The size in bits of the fll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static inline int fll_factors(struct _fll_div *fll_div, unsigned int input,
+			      unsigned int output)
+{
+	u64 Kpart;
+	unsigned int t1, t2, K, Nmod;
+
+	if (output >= 2815250 && output <= 3125000)
+		fll_div->div = 0x4;
+	else if (output >= 5625000 && output <= 6250000)
+		fll_div->div = 0x3;
+	else if (output >= 11250000 && output <= 12500000)
+		fll_div->div = 0x2;
+	else if (output >= 22500000 && output <= 25000000)
+		fll_div->div = 0x1;
+	else {
+		printk(KERN_ERR "wm8350: fll freq %d out of range\n", output);
+		return -EINVAL;
+	}
+
+	if (input > 48000)
+		fll_div->ratio = 1;
+	else
+		fll_div->ratio = 8;
+
+	t1 = output * (1 << (fll_div->div + 1));
+	t2 = input * fll_div->ratio;
+
+	fll_div->n = t1 / t2;
+	Nmod = t1 % t2;
+
+	if (Nmod) {
+		Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+		do_div(Kpart, t2);
+		K = Kpart & 0xFFFFFFFF;
+
+		/* Check if we need to round */
+		if ((K % 10) >= 5)
+			K += 5;
+
+		/* Move down to proper range now rounding is done */
+		K /= 10;
+		fll_div->k = K;
+	} else
+		fll_div->k = 0;
+
+	return 0;
+}
+
+static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
+			  int pll_id, unsigned int freq_in,
+			  unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	struct _fll_div fll_div;
+	int ret = 0;
+	u16 fll_1, fll_4;
+
+	/* power down FLL - we need to do this for reconfiguration */
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+			  WM8350_FLL_ENA | WM8350_FLL_OSC_ENA);
+
+	if (freq_out == 0 || freq_in == 0)
+		return ret;
+
+	ret = fll_factors(&fll_div, freq_in, freq_out);
+	if (ret < 0)
+		return ret;
+	dev_dbg(wm8350->dev,
+		"FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+		freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
+		fll_div.ratio);
+
+	/* set up N.K & dividers */
+	fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) &
+	    ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_1,
+			   fll_1 | (fll_div.div << 8) | 0x50);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_2,
+			   (fll_div.ratio << 11) | (fll_div.
+						    n & WM8350_FLL_N_MASK));
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k);
+	fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) &
+	    ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF);
+	wm8350_codec_write(codec, WM8350_FLL_CONTROL_4,
+			   fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) |
+			   (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0));
+
+	/* power FLL on */
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA);
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA);
+
+	return 0;
+}
+
+static int wm8350_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350_data *priv = codec->private_data;
+	struct wm8350_audio_platform_data *platform =
+		wm8350->codec.platform_data;
+	u16 pm1;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_VMID_50K |
+				 platform->codec_current_on << 14);
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1);
+		pm1 &= ~WM8350_VMID_MASK;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_VMID_50K);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
+						    priv->supplies);
+			if (ret != 0)
+				return ret;
+
+			/* Enable the system clock */
+			wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4,
+					WM8350_SYSCLK_ENA);
+
+			/* mute DAC & outputs */
+			wm8350_set_bits(wm8350, WM8350_DAC_MUTE,
+					WM8350_DAC_MUTE_ENA);
+
+			/* discharge cap memory */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+					 platform->dis_out1 |
+					 (platform->dis_out2 << 2) |
+					 (platform->dis_out3 << 4) |
+					 (platform->dis_out4 << 6));
+
+			/* wait for discharge */
+			schedule_timeout_interruptible(msecs_to_jiffies
+						       (platform->
+							cap_discharge_msecs));
+
+			/* enable antipop */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+					 (platform->vmid_s_curve << 8));
+
+			/* ramp up vmid */
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 (platform->
+					  codec_current_charge << 14) |
+					 WM8350_VMID_5K | WM8350_VMIDEN |
+					 WM8350_VBUFEN);
+
+			/* wait for vmid */
+			schedule_timeout_interruptible(msecs_to_jiffies
+						       (platform->
+							vmid_charge_msecs));
+
+			/* turn on vmid 300k  */
+			pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+			    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+			pm1 |= WM8350_VMID_300K |
+				(platform->codec_current_standby << 14);
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 pm1);
+
+
+			/* enable analogue bias */
+			pm1 |= WM8350_BIASEN;
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+			/* disable antipop */
+			wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+		} else {
+			/* turn on vmid 300k and reduce current */
+			pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+			    ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
+			wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+					 pm1 | WM8350_VMID_300K |
+					 (platform->
+					  codec_current_standby << 14));
+
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+
+		/* mute DAC & enable outputs */
+		wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
+
+		wm8350_set_bits(wm8350, WM8350_POWER_MGMT_3,
+				WM8350_OUT1L_ENA | WM8350_OUT1R_ENA |
+				WM8350_OUT2L_ENA | WM8350_OUT2R_ENA);
+
+		/* enable anti pop S curve */
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+				 (platform->vmid_s_curve << 8));
+
+		/* turn off vmid  */
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~WM8350_VMIDEN;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+		/* wait */
+		schedule_timeout_interruptible(msecs_to_jiffies
+					       (platform->
+						vmid_discharge_msecs));
+
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
+				 (platform->vmid_s_curve << 8) |
+				 platform->dis_out1 |
+				 (platform->dis_out2 << 2) |
+				 (platform->dis_out3 << 4) |
+				 (platform->dis_out4 << 6));
+
+		/* turn off VBuf and drain */
+		pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
+		    ~(WM8350_VBUFEN | WM8350_VMID_MASK);
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
+				 pm1 | WM8350_OUTPUT_DRAIN_EN);
+
+		/* wait */
+		schedule_timeout_interruptible(msecs_to_jiffies
+					       (platform->drain_msecs));
+
+		pm1 &= ~WM8350_BIASEN;
+		wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
+
+		/* disable anti-pop */
+		wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
+
+		wm8350_clear_bits(wm8350, WM8350_LOUT1_VOLUME,
+				  WM8350_OUT1L_ENA);
+		wm8350_clear_bits(wm8350, WM8350_ROUT1_VOLUME,
+				  WM8350_OUT1R_ENA);
+		wm8350_clear_bits(wm8350, WM8350_LOUT2_VOLUME,
+				  WM8350_OUT2L_ENA);
+		wm8350_clear_bits(wm8350, WM8350_ROUT2_VOLUME,
+				  WM8350_OUT2R_ENA);
+
+		/* disable clock gen */
+		wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
+				  WM8350_SYSCLK_ENA);
+
+		regulator_bulk_disable(ARRAY_SIZE(priv->supplies),
+				       priv->supplies);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8350_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+		wm8350_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+	return 0;
+}
+
+static struct snd_soc_codec *wm8350_codec;
+
+static int wm8350_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct wm8350 *wm8350;
+	struct wm8350_data *priv;
+	int ret;
+	struct wm8350_output *out1;
+	struct wm8350_output *out2;
+
+	BUG_ON(!wm8350_codec);
+
+	socdev->codec = wm8350_codec;
+	codec = socdev->codec;
+	wm8350 = codec->control_data;
+	priv = codec->private_data;
+
+	/* Enable the codec */
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	/* Enable robust clocking mode in ADC */
+	wm8350_codec_write(codec, WM8350_SECURITY, 0xa7);
+	wm8350_codec_write(codec, 0xde, 0x13);
+	wm8350_codec_write(codec, WM8350_SECURITY, 0);
+
+	/* read OUT1 & OUT2 volumes */
+	out1 = &priv->out1;
+	out2 = &priv->out2;
+	out1->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME) &
+			  WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	out1->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME) &
+			   WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	out2->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME) &
+			  WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
+	out2->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME) &
+			   WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
+	wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, 0);
+	wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, 0);
+
+	/* Latch VU bits & mute */
+	wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME,
+			WM8350_OUT1_VU | WM8350_OUT1L_MUTE);
+	wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME,
+			WM8350_OUT2_VU | WM8350_OUT2L_MUTE);
+	wm8350_set_bits(wm8350, WM8350_ROUT1_VOLUME,
+			WM8350_OUT1_VU | WM8350_OUT1R_MUTE);
+	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
+			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
+
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to create pcms\n");
+		return ret;
+	}
+
+	wm8350_add_controls(codec);
+	wm8350_add_widgets(codec);
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to register card\n");
+		goto card_err;
+	}
+
+	return 0;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+	return ret;
+}
+
+static int wm8350_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	struct wm8350 *wm8350 = codec->control_data;
+	int ret;
+
+	/* cancel any work waiting to be queued. */
+	ret = cancel_delayed_work(&codec->delayed_work);
+
+	/* if there was any work waiting then we run it now and
+	 * wait for its completion */
+	if (ret) {
+		schedule_delayed_work(&codec->delayed_work, 0);
+		flush_scheduled_work();
+	}
+
+	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	return 0;
+}
+
+#define WM8350_RATES (SNDRV_PCM_RATE_8000_96000)
+
+#define WM8350_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+			SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S24_LE)
+
+struct snd_soc_dai wm8350_dai = {
+	.name = "WM8350",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8350_RATES,
+		.formats = WM8350_FORMATS,
+	},
+	.capture = {
+		 .stream_name = "Capture",
+		 .channels_min = 1,
+		 .channels_max = 2,
+		 .rates = WM8350_RATES,
+		 .formats = WM8350_FORMATS,
+	 },
+	.ops = {
+		 .hw_params = wm8350_pcm_hw_params,
+		 .digital_mute = wm8350_mute,
+		 .trigger = wm8350_pcm_trigger,
+		 .set_fmt = wm8350_set_dai_fmt,
+		 .set_sysclk = wm8350_set_dai_sysclk,
+		 .set_pll = wm8350_set_fll,
+		 .set_clkdiv = wm8350_set_clkdiv,
+	 },
+};
+EXPORT_SYMBOL_GPL(wm8350_dai);
+
+struct snd_soc_codec_device soc_codec_dev_wm8350 = {
+	.probe = 	wm8350_probe,
+	.remove = 	wm8350_remove,
+	.suspend = 	wm8350_suspend,
+	.resume =	wm8350_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350);
+
+static int wm8350_codec_probe(struct platform_device *pdev)
+{
+	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+	struct wm8350_data *priv;
+	struct snd_soc_codec *codec;
+	int ret, i;
+
+	if (wm8350->codec.platform_data == NULL) {
+		dev_err(&pdev->dev, "No audio platform data supplied\n");
+		return -EINVAL;
+	}
+
+	priv = kzalloc(sizeof(struct wm8350_data), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+		priv->supplies[i].supply = supply_names[i];
+
+	ret = regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies),
+				 priv->supplies);
+	if (ret != 0)
+		goto err_priv;
+
+	codec = &priv->codec;
+	wm8350->codec.codec = codec;
+
+	wm8350_dai.dev = &pdev->dev;
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	codec->dev = &pdev->dev;
+	codec->name = "WM8350";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8350_codec_read;
+	codec->write = wm8350_codec_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8350_set_bias_level;
+	codec->dai = &wm8350_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8350_MAX_REGISTER;
+	codec->private_data = priv;
+	codec->control_data = wm8350;
+
+	/* Put the codec into reset if it wasn't already */
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
+
+	INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work);
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0)
+		goto err_supply;
+
+	wm8350_codec = codec;
+
+	ret = snd_soc_register_dai(&wm8350_dai);
+	if (ret != 0)
+		goto err_codec;
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err_supply:
+	regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+err_priv:
+	kfree(priv);
+	wm8350_codec = NULL;
+	return ret;
+}
+
+static int wm8350_codec_remove(struct platform_device *pdev)
+{
+	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = wm8350->codec.codec;
+	struct wm8350_data *priv = codec->private_data;
+
+	snd_soc_unregister_dai(&wm8350_dai);
+	snd_soc_unregister_codec(codec);
+	regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies);
+	kfree(priv);
+	wm8350_codec = NULL;
+	return 0;
+}
+
+static struct platform_driver wm8350_codec_driver = {
+	.driver = {
+		   .name = "wm8350-codec",
+		   .owner = THIS_MODULE,
+		   },
+	.probe = wm8350_codec_probe,
+	.remove = __devexit_p(wm8350_codec_remove),
+};
+
+static __init int wm8350_init(void)
+{
+	return platform_driver_register(&wm8350_codec_driver);
+}
+module_init(wm8350_init);
+
+static __exit void wm8350_exit(void)
+{
+	platform_driver_unregister(&wm8350_codec_driver);
+}
+module_exit(wm8350_exit);
+
+MODULE_DESCRIPTION("ASoC WM8350 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8350-codec");
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
new file mode 100644
index 0000000..cc2887a
--- /dev/null
+++ b/sound/soc/codecs/wm8350.h
@@ -0,0 +1,20 @@
+/*
+ * wm8350.h - WM8903 audio codec interface
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#ifndef _WM8350_H
+#define _WM8350_H
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm8350_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8350;
+
+#endif
-- 
1.5.6.5

Re: [PATCH 4/4] ASoC: Add WM8350 AudioPlus codec driver

[Liam Girdwood]

On Thu, 2008-12-18 at 17:30 +0000, Mark Brown wrote:
> The WM8350 is an integrated audio and power management subsystem which
> provides a single-chip solution for portable audio and multimedia systems.
> 
> The integrated audio CODEC provides all the necessary functions for
> high-quality stereo recording and playback. Programmable on-chip
> amplifiers allow for the direct connection of headphones and microphones
> with a minimum of external components. A programmable low-noise bias
> voltage is available to feed one or more electret microphones.
> Additional audio features include programmable high-pass filter in the
> ADC input path.
> 
> This driver was originally written by Liam Girdwood with further updates
> from me.
> 
> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>

Liam

Re: [PATCH 0/4] ASoC updates

[Takashi Iwai]

At Thu, 18 Dec 2008 17:30:21 +0000,
Mark Brown wrote:
> 
> The following changes since commit 49d92c7d5bbd158734bc34ed578a68b214a48583:
>   Stanley.Miao (1):
>         ASoC: TWL4030: hands-free start-up sequence.
> 
> are available in the git repository at:
> 
>   git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai

Applied now.  Thanks.


Takashi

> 
> Alexander Beregalov (1):
>       ASoC: switch davinci DPRINTK to pr_debug()
> 
> Mark Brown (3):
>       ASoC: Ease merge difficulties from new architectures
>       ASoC: Complain if we fail to create DAPM controls
>       ASoC: Add WM8350 AudioPlus codec driver
> 
>  include/linux/mfd/wm8350/audio.h |   38 +-
>  sound/soc/Kconfig                |   10 +-
>  sound/soc/Makefile               |   12 +-
>  sound/soc/codecs/Kconfig         |    4 +
>  sound/soc/codecs/Makefile        |    2 +
>  sound/soc/codecs/wm8350.c        | 1583 ++++++++++++++++++++++++++++++++++++++
>  sound/soc/codecs/wm8350.h        |   20 +
>  sound/soc/davinci/davinci-pcm.c  |   18 +-
>  sound/soc/soc-dapm.c             |    6 +-
>  9 files changed, 1669 insertions(+), 24 deletions(-)
>  create mode 100644 sound/soc/codecs/wm8350.c
>  create mode 100644 sound/soc/codecs/wm8350.h
>