Merge branch 'topic/asoc' into for-linus

104 files changed, 8717 insertions, 626 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index b1749bc67979..3e598e756e54 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -28,9 +28,13 @@ source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
 source "sound/soc/blackfin/Kconfig"
 source "sound/soc/davinci/Kconfig"
+source "sound/soc/ep93xx/Kconfig"
 source "sound/soc/fsl/Kconfig"
 source "sound/soc/imx/Kconfig"
+source "sound/soc/jz4740/Kconfig"
+source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"
+source "sound/soc/kirkwood/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
 source "sound/soc/s6000/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 1470141d4167..eb183443eee4 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -6,9 +6,13 @@ obj-$(CONFIG_SND_SOC)	+= atmel/
 obj-$(CONFIG_SND_SOC)   += au1x/
 obj-$(CONFIG_SND_SOC)   += blackfin/
 obj-$(CONFIG_SND_SOC)   += davinci/
+obj-$(CONFIG_SND_SOC)   += ep93xx/
 obj-$(CONFIG_SND_SOC)   += fsl/
 obj-$(CONFIG_SND_SOC)   += imx/
+obj-$(CONFIG_SND_SOC)   += jz4740/
+obj-$(CONFIG_SND_SOC)   += nuc900/
 obj-$(CONFIG_SND_SOC)   += omap/
+obj-$(CONFIG_SND_SOC)   += kirkwood/
 obj-$(CONFIG_SND_SOC)   += pxa/
 obj-$(CONFIG_SND_SOC)   += s3c24xx/
 obj-$(CONFIG_SND_SOC)   += s6000/
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index f6b3cc04b34b..dc5249fba85c 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -77,7 +77,6 @@ struct atmel_runtime_data {
         size_t period_size;
 
         dma_addr_t period_ptr;          /* physical address of next period */
-        int periods;                    /* period index of period_ptr */
 
         /* PDC register save */
         u32 pdc_xpr_save;
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index 0b59806905d1..c85844d4845b 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -549,7 +549,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream,
                 printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n",
                         ssc_p->daifmt);
                 return -EINVAL;
-                break;
         }
         pr_debug("atmel_ssc_hw_params: "
                         "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n",
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index a61ccd2d505f..d14a5a91a465 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -375,12 +375,10 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
         }
 
         ret = -EBUSY;
-        wd->ioarea = request_mem_region(r->start, r->end - r->start + 1,
-                                        "au1xpsc_ac97");
-        if (!wd->ioarea)
+        if (!request_mem_region(r->start, resource_size(r), pdev->name))
                 goto out0;
 
-        wd->mmio = ioremap(r->start, 0xffff);
+        wd->mmio = ioremap(r->start, resource_size(r));
         if (!wd->mmio)
                 goto out1;
 
@@ -410,8 +408,7 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 
         snd_soc_unregister_dai(&au1xpsc_ac97_dai);
 out1:
-        release_resource(wd->ioarea);
-        kfree(wd->ioarea);
+        release_mem_region(r->start, resource_size(r));
 out0:
         kfree(wd);
         return ret;
@@ -420,6 +417,7 @@ out0:
 static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev)
 {
         struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
+        struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 
         if (wd->dmapd)
                 au1xpsc_pcm_destroy(wd->dmapd);
@@ -433,8 +431,7 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev)
         au_sync();
 
         iounmap(wd->mmio);
-        release_resource(wd->ioarea);
-        kfree(wd->ioarea);
+        release_mem_region(r->start, resource_size(r));
         kfree(wd);
 
         au1xpsc_ac97_workdata = NULL;   /* MDEV */
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 24454c98d0ee..6083fe7799fa 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -321,12 +321,10 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
         }
 
         ret = -EBUSY;
-        wd->ioarea = request_mem_region(r->start, r->end - r->start + 1,
-                                        "au1xpsc_i2s");
-        if (!wd->ioarea)
+        if (!request_mem_region(r->start, resource_size(r), pdev->name))
                 goto out0;
 
-        wd->mmio = ioremap(r->start, 0xffff);
+        wd->mmio = ioremap(r->start, resource_size(r));
         if (!wd->mmio)
                 goto out1;
 
@@ -362,8 +360,7 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 
         snd_soc_unregister_dai(&au1xpsc_i2s_dai);
 out1:
-        release_resource(wd->ioarea);
-        kfree(wd->ioarea);
+        release_mem_region(r->start, resource_size(r));
 out0:
         kfree(wd);
         return ret;
@@ -372,6 +369,7 @@ out0:
 static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
 {
         struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
+        struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 
         if (wd->dmapd)
                 au1xpsc_pcm_destroy(wd->dmapd);
@@ -384,8 +382,7 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
         au_sync();
 
         iounmap(wd->mmio);
-        release_resource(wd->ioarea);
-        kfree(wd->ioarea);
+        release_mem_region(r->start, resource_size(r));
         kfree(wd);
 
         au1xpsc_i2s_workdata = NULL;    /* MDEV */
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index 32d3807d3f5a..093775d4dc3e 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -32,7 +32,6 @@ struct au1xpsc_audio_data {
         unsigned long rate;
 
         unsigned long pm[2];
-        struct resource *ioarea;
         struct mutex lock;
         struct platform_device *dmapd;
 };
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 523b7fc33f4e..c0eba5109980 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -255,8 +255,7 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops);
 #ifdef CONFIG_PM
 static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
 {
-        struct sport_device *sport =
-                (struct sport_device *)dai->private_data;
+        struct sport_device *sport = dai->private_data;
 
         pr_debug("%s : sport %d\n", __func__, dai->id);
         if (!dai->active)
@@ -271,8 +270,7 @@ static int bf5xx_ac97_suspend(struct snd_soc_dai *dai)
 static int bf5xx_ac97_resume(struct snd_soc_dai *dai)
 {
         int ret;
-        struct sport_device *sport =
-                (struct sport_device *)dai->private_data;
+        struct sport_device *sport = dai->private_data;
 
         pr_debug("%s : sport %d\n", __func__, dai->id);
         if (!dai->active)
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index 4b360124083e..24c14269f4bc 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -210,8 +210,7 @@ static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai,
 #ifdef CONFIG_PM
 static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
 {
-        struct sport_device *sport =
-                (struct sport_device *)dai->private_data;
+        struct sport_device *sport = dai->private_data;
 
         if (!dai->active)
                 return 0;
@@ -225,8 +224,7 @@ static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
 static int bf5xx_tdm_resume(struct snd_soc_dai *dai)
 {
         int ret;
-        struct sport_device *sport =
-                (struct sport_device *)dai->private_data;
+        struct sport_device *sport = dai->private_data;
 
         if (!dai->active)
                 return 0;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 5da30eb6ad00..83f5c67d3c41 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -22,9 +22,11 @@ config SND_SOC_ALL_CODECS
         select SND_SOC_AK4642 if I2C
         select SND_SOC_AK4671 if I2C
         select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
+        select SND_SOC_CS42L51 if I2C
         select SND_SOC_CS4270 if I2C
-        select SND_SOC_MAX9877 if I2C
         select SND_SOC_DA7210 if I2C
+        select SND_SOC_JZ4740 if SOC_JZ4740
+        select SND_SOC_MAX9877 if I2C
         select SND_SOC_PCM3008
         select SND_SOC_SPDIF
         select SND_SOC_SSM2602 if I2C
@@ -48,6 +50,7 @@ config SND_SOC_ALL_CODECS
         select SND_SOC_WM8727
         select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
         select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
+        select SND_SOC_WM8741 if SND_SOC_I2C_AND_SPI
         select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
         select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
         select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
@@ -120,13 +123,13 @@ config SND_SOC_AK4671
 config SND_SOC_CQ0093VC
         tristate
 
+config SND_SOC_CS42L51
+        tristate
+
 # Cirrus Logic CS4270 Codec
 config SND_SOC_CS4270
         tristate
 
-config SND_SOC_DA7210
-        tristate
-
 # Cirrus Logic CS4270 Codec VD = 3.3V Errata
 # Select if you are affected by the errata where the part will not function
 # if MCLK divide-by-1.5 is selected and VD is set to 3.3V.  The driver will
@@ -138,9 +141,15 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_CX20442
         tristate
 
+config SND_SOC_JZ4740_CODEC
+        tristate
+
 config SND_SOC_L3
        tristate
 
+config SND_SOC_DA7210
+        tristate
+
 config SND_SOC_PCM3008
        tristate
 
@@ -206,6 +215,9 @@ config SND_SOC_WM8728
 config SND_SOC_WM8731
         tristate
 
+config SND_SOC_WM8741
+        tristate
+
 config SND_SOC_WM8750
         tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 91429eab0707..53524095759c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -9,6 +9,7 @@ snd-soc-ak4535-objs := ak4535.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
 snd-soc-cq93vc-objs := cq93vc.o
+snd-soc-cs42l51-objs := cs42l51.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
@@ -34,6 +35,7 @@ snd-soc-wm8711-objs := wm8711.o
 snd-soc-wm8727-objs := wm8727.o
 snd-soc-wm8728-objs := wm8728.o
 snd-soc-wm8731-objs := wm8731.o
+snd-soc-wm8741-objs := wm8741.o
 snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8776-objs := wm8776.o
@@ -56,6 +58,7 @@ snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
 snd-soc-wm-hubs-objs := wm_hubs.o
+snd-soc-jz4740-codec-objs := jz4740.o
 
 # Amp
 snd-soc-max9877-objs := max9877.o
@@ -74,10 +77,12 @@ obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_AK4642)    += snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)    += snd-soc-ak4671.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
+obj-$(CONFIG_SND_SOC_CS42L51)   += snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS4270)    += snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CX20442)   += snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)    += snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_L3)        += snd-soc-l3.o
+obj-$(CONFIG_SND_SOC_JZ4740_CODEC)      += snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)   += snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_SPDIF)     += snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)   += snd-soc-ssm2602.o
@@ -99,6 +104,7 @@ obj-$(CONFIG_SND_SOC_WM8711)	+= snd-soc-wm8711.o
 obj-$(CONFIG_SND_SOC_WM8727)    += snd-soc-wm8727.o
 obj-$(CONFIG_SND_SOC_WM8728)    += snd-soc-wm8728.o
 obj-$(CONFIG_SND_SOC_WM8731)    += snd-soc-wm8731.o
+obj-$(CONFIG_SND_SOC_WM8741)    += snd-soc-wm8741.o
 obj-$(CONFIG_SND_SOC_WM8750)    += snd-soc-wm8750.o
 obj-$(CONFIG_SND_SOC_WM8753)    += snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8776)    += snd-soc-wm8776.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 217538423225..a01006c8c606 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -272,6 +272,7 @@ static int ad1836_register(struct ad1836_priv *ad1836)
 
         if (ad1836_codec) {
                 dev_err(codec->dev, "Another ad1836 is registered\n");
+                kfree(ad1836);
                 return -EINVAL;
         }
 
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index c8ca1142b2f4..1def75e4862f 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -24,6 +24,7 @@
 
 /* codec private data */
 struct ad193x_priv {
+        unsigned int sysclk;
         struct snd_soc_codec codec;
         u8 reg_cache[AD193X_NUM_REGS];
 };
@@ -251,15 +252,32 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
         return 0;
 }
 
+static int ad193x_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 ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
+        switch (freq) {
+        case 12288000:
+        case 18432000:
+        case 24576000:
+        case 36864000:
+                ad193x->sysclk = freq;
+                return 0;
+        }
+        return -EINVAL;
+}
+
 static int ad193x_hw_params(struct snd_pcm_substream *substream,
                 struct snd_pcm_hw_params *params,
                 struct snd_soc_dai *dai)
 {
-        int word_len = 0, reg = 0;
+        int word_len = 0, reg = 0, master_rate = 0;
 
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_device *socdev = rtd->socdev;
         struct snd_soc_codec *codec = socdev->card->codec;
+        struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
 
         /* bit size */
         switch (params_format(params)) {
@@ -275,6 +293,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
                 break;
         }
 
+        switch (ad193x->sysclk) {
+        case 12288000:
+                master_rate = AD193X_PLL_INPUT_256;
+                break;
+        case 18432000:
+                master_rate = AD193X_PLL_INPUT_384;
+                break;
+        case 24576000:
+                master_rate = AD193X_PLL_INPUT_512;
+                break;
+        case 36864000:
+                master_rate = AD193X_PLL_INPUT_768;
+                break;
+        }
+
+        reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0);
+        reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate;
+        snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
+
         reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
         reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len;
         snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
@@ -348,6 +385,7 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type)
         /* pll input: mclki/xi */
         snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
         snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04);
+        ad193x->sysclk = 12288000;
 
         ret = snd_soc_register_codec(codec);
         if (ret != 0) {
@@ -383,6 +421,7 @@ static struct snd_soc_dai_ops ad193x_dai_ops = {
         .hw_params = ad193x_hw_params,
         .digital_mute = ad193x_mute,
         .set_tdm_slot = ad193x_set_tdm_slot,
+        .set_sysclk     = ad193x_set_dai_sysclk,
         .set_fmt = ad193x_set_dai_fmt,
 };
 
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index a03c880d52f9..654ba64ae04c 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -11,6 +11,11 @@
 
 #define AD193X_PLL_CLK_CTRL0    0x800
 #define AD193X_PLL_POWERDOWN           0x01
+#define AD193X_PLL_INPUT_MASK   (~0x6)
+#define AD193X_PLL_INPUT_256    (0 << 1)
+#define AD193X_PLL_INPUT_384    (1 << 1)
+#define AD193X_PLL_INPUT_512    (2 << 1)
+#define AD193X_PLL_INPUT_768    (3 << 1)
 #define AD193X_PLL_CLK_CTRL1    0x801
 #define AD193X_DAC_CTRL0        0x802
 #define AD193X_DAC_POWERDOWN           0x01
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 7528a54102b5..3d7dc55305ec 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -22,20 +22,13 @@
  * AK4643 is tested.
  */
 
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
 #include <linux/delay.h>
-#include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
 #include <linux/slab.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 "ak4642.h"
 
@@ -111,6 +104,23 @@
 
 struct snd_soc_codec_device soc_codec_dev_ak4642;
 
+/*
+ * Playback Volume (table 39)
+ *
+ * max : 0x00 : +12.0 dB
+ *       ( 0.5 dB step )
+ * min : 0xFE : -115.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1);
+
+static const struct snd_kcontrol_new ak4642_snd_controls[] = {
+
+        SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
+                         0, 0xFF, 1, out_tlv),
+};
+
+
 /* codec private data */
 struct ak4642_priv {
         struct snd_soc_codec codec;
@@ -204,7 +214,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
                  *
                  * PLL, Master Mode
                  * Audio I/F Format :MSB justified (ADC & DAC)
-                 * Digital Volume: -8dB
                  * Bass Boost Level : Middle
                  *
                  * This operation came from example code of
@@ -214,8 +223,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
                 ak4642_write(codec, 0x0e, 0x19);
                 ak4642_write(codec, 0x09, 0x91);
                 ak4642_write(codec, 0x0c, 0x91);
-                ak4642_write(codec, 0x0a, 0x28);
-                ak4642_write(codec, 0x0d, 0x28);
                 ak4642_write(codec, 0x00, 0x64);
                 snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP);
                 snd_soc_update_bits(codec, PW_MGMT2, HPMTN,     HPMTN);
@@ -491,8 +498,10 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
         codec->control_data = i2c;
 
         ret = ak4642_init(ak4642);
-        if (ret < 0)
+        if (ret < 0) {
                 printk(KERN_ERR "failed to initialise AK4642\n");
+                kfree(ak4642);
+        }
 
         return ret;
 }
@@ -548,6 +557,9 @@ static int ak4642_probe(struct platform_device *pdev)
                 goto pcm_err;
         }
 
+        snd_soc_add_controls(ak4642_codec, ak4642_snd_controls,
+                             ARRAY_SIZE(ak4642_snd_controls));
+
         dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
         return ret;
 
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
new file mode 100644
index 000000000000..dd9b8550c402
--- /dev/null
+++ b/sound/soc/codecs/cs42l51.c
@@ -0,0 +1,763 @@
+/*
+ * cs42l51.c
+ *
+ * ASoC Driver for Cirrus Logic CS42L51 codecs
+ *
+ * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com>
+ *
+ * Based on cs4270.c - Copyright (c) Freescale Semiconductor
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * For now:
+ *  - Only I2C is support. Not SPI
+ *  - master mode *NOT* supported
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/pcm.h>
+#include <linux/i2c.h>
+
+#include "cs42l51.h"
+
+enum master_slave_mode {
+        MODE_SLAVE,
+        MODE_SLAVE_AUTO,
+        MODE_MASTER,
+};
+
+struct cs42l51_private {
+        unsigned int mclk;
+        unsigned int audio_mode;        /* The mode (I2S or left-justified) */
+        enum master_slave_mode func;
+        struct snd_soc_codec codec;
+        u8 reg_cache[CS42L51_NUMREGS];
+};
+
+static struct snd_soc_codec *cs42l51_codec;
+
+#define CS42L51_FORMATS ( \
+                SNDRV_PCM_FMTBIT_S16_LE  | SNDRV_PCM_FMTBIT_S16_BE  | \
+                SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \
+                SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \
+                SNDRV_PCM_FMTBIT_S24_LE  | SNDRV_PCM_FMTBIT_S24_BE)
+
+static int cs42l51_fill_cache(struct snd_soc_codec *codec)
+{
+        u8 *cache = codec->reg_cache + 1;
+        struct i2c_client *i2c_client = codec->control_data;
+        s32 length;
+
+        length = i2c_smbus_read_i2c_block_data(i2c_client,
+                        CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache);
+        if (length != CS42L51_NUMREGS) {
+                dev_err(&i2c_client->dev,
+                                "I2C read failure, addr=0x%x (ret=%d vs %d)\n",
+                                i2c_client->addr, length, CS42L51_NUMREGS);
+                return -EIO;
+        }
+
+        return 0;
+}
+
+static int cs42l51_i2c_probe(struct i2c_client *i2c_client,
+        const struct i2c_device_id *id)
+{
+        struct snd_soc_codec *codec;
+        struct cs42l51_private *cs42l51;
+        int ret = 0;
+        int reg;
+
+        if (cs42l51_codec)
+                return -EBUSY;
+
+        /* Verify that we have a CS42L51 */
+        ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID);
+        if (ret < 0) {
+                dev_err(&i2c_client->dev, "failed to read I2C\n");
+                goto error;
+        }
+
+        if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) &&
+            (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) {
+                dev_err(&i2c_client->dev, "Invalid chip id\n");
+                ret = -ENODEV;
+                goto error;
+        }
+
+        dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n",
+                                ret & 7);
+
+        cs42l51 = kzalloc(sizeof(struct cs42l51_private), GFP_KERNEL);
+        if (!cs42l51) {
+                dev_err(&i2c_client->dev, "could not allocate codec\n");
+                return -ENOMEM;
+        }
+        codec = &cs42l51->codec;
+
+        mutex_init(&codec->mutex);
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        codec->dev = &i2c_client->dev;
+        codec->name = "CS42L51";
+        codec->owner = THIS_MODULE;
+        codec->dai = &cs42l51_dai;
+        codec->num_dai = 1;
+        snd_soc_codec_set_drvdata(codec, cs42l51);
+
+        codec->control_data = i2c_client;
+        codec->reg_cache = cs42l51->reg_cache;
+        codec->reg_cache_size = CS42L51_NUMREGS;
+        i2c_set_clientdata(i2c_client, codec);
+
+        ret = cs42l51_fill_cache(codec);
+        if (ret < 0) {
+                dev_err(&i2c_client->dev, "failed to fill register cache\n");
+                goto error_alloc;
+        }
+
+        ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+        if (ret < 0) {
+                dev_err(&i2c_client->dev, "Failed to set cache I/O: %d\n", ret);
+                goto error_alloc;
+        }
+
+        /*
+         * DAC configuration
+         * - Use signal processor
+         * - auto mute
+         * - vol changes immediate
+         * - no de-emphasize
+         */
+        reg = CS42L51_DAC_CTL_DATA_SEL(1)
+                | CS42L51_DAC_CTL_AMUTE | CS42L51_DAC_CTL_DACSZ(0);
+        ret = snd_soc_write(codec, CS42L51_DAC_CTL, reg);
+        if (ret < 0)
+                goto error_alloc;
+
+        cs42l51_dai.dev = codec->dev;
+        cs42l51_codec = codec;
+
+        ret = snd_soc_register_codec(codec);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+                goto error_alloc;
+        }
+
+        ret = snd_soc_register_dai(&cs42l51_dai);
+        if (ret < 0) {
+                dev_err(&i2c_client->dev, "failed to register DAIe\n");
+                goto error_reg;
+        }
+
+        return 0;
+
+error_reg:
+        snd_soc_unregister_codec(codec);
+error_alloc:
+        kfree(cs42l51);
+error:
+        return ret;
+}
+
+static int cs42l51_i2c_remove(struct i2c_client *client)
+{
+        struct cs42l51_private *cs42l51 = i2c_get_clientdata(client);
+        snd_soc_unregister_dai(&cs42l51_dai);
+        snd_soc_unregister_codec(cs42l51_codec);
+        cs42l51_codec = NULL;
+        kfree(cs42l51);
+        return 0;
+}
+
+
+static const struct i2c_device_id cs42l51_id[] = {
+        {"cs42l51", 0},
+        {}
+};
+MODULE_DEVICE_TABLE(i2c, cs42l51_id);
+
+static struct i2c_driver cs42l51_i2c_driver = {
+        .driver = {
+                .name = "CS42L51 I2C",
+                .owner = THIS_MODULE,
+        },
+        .id_table = cs42l51_id,
+        .probe = cs42l51_i2c_probe,
+        .remove = cs42l51_i2c_remove,
+};
+
+static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol,
+                        struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        unsigned long value = snd_soc_read(codec, CS42L51_PCM_MIXER)&3;
+
+        switch (value) {
+        default:
+        case 0:
+                ucontrol->value.integer.value[0] = 0;
+                break;
+        /* same value : (L+R)/2 and (R+L)/2 */
+        case 1:
+        case 2:
+                ucontrol->value.integer.value[0] = 1;
+                break;
+        case 3:
+                ucontrol->value.integer.value[0] = 2;
+                break;
+        }
+
+        return 0;
+}
+
+#define CHAN_MIX_NORMAL 0x00
+#define CHAN_MIX_BOTH   0x55
+#define CHAN_MIX_SWAP   0xFF
+
+static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol,
+                        struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        unsigned char val;
+
+        switch (ucontrol->value.integer.value[0]) {
+        default:
+        case 0:
+                val = CHAN_MIX_NORMAL;
+                break;
+        case 1:
+                val = CHAN_MIX_BOTH;
+                break;
+        case 2:
+                val = CHAN_MIX_SWAP;
+                break;
+        }
+
+        snd_soc_write(codec, CS42L51_PCM_MIXER, val);
+
+        return 1;
+}
+
+static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0);
+/* This is a lie. after -102 db, it stays at -102 */
+/* maybe a range would be better */
+static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0);
+static const char *chan_mix[] = {
+        "L R",
+        "L+R",
+        "R L",
+};
+
+static const struct soc_enum cs42l51_chan_mix =
+        SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix);
+
+static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
+        SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
+                        CS42L51_PCMA_VOL, CS42L51_PCMB_VOL,
+                        7, 0xffffff99, 0x18, adc_pcm_tlv),
+        SOC_DOUBLE_R("PCM Playback Switch",
+                        CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1),
+        SOC_DOUBLE_R_SX_TLV("Analog Playback Volume",
+                        CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL,
+                        8, 0xffffff19, 0x18, aout_tlv),
+        SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
+                        CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
+                        7, 0xffffff99, 0x18, adc_pcm_tlv),
+        SOC_DOUBLE_R("ADC Mixer Switch",
+                        CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1),
+        SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
+        SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0),
+        SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0),
+        SOC_SINGLE("Zero Cross Switch", CS42L51_DAC_CTL, 0, 0, 0),
+        SOC_DOUBLE_TLV("Mic Boost Volume",
+                        CS42L51_MIC_CTL, 0, 1, 1, 0, boost_tlv),
+        SOC_SINGLE_TLV("Bass Volume", CS42L51_TONE_CTL, 0, 0xf, 1, tone_tlv),
+        SOC_SINGLE_TLV("Treble Volume", CS42L51_TONE_CTL, 4, 0xf, 1, tone_tlv),
+        SOC_ENUM_EXT("PCM channel mixer",
+                        cs42l51_chan_mix,
+                        cs42l51_get_chan_mix, cs42l51_set_chan_mix),
+};
+
+/*
+ * to power down, one must:
+ * 1.) Enable the PDN bit
+ * 2.) enable power-down for the select channels
+ * 3.) disable the PDN bit.
+ */
+static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w,
+                struct snd_kcontrol *kcontrol, int event)
+{
+        unsigned long value;
+
+        value = snd_soc_read(w->codec, CS42L51_POWER_CTL1);
+        value &= ~CS42L51_POWER_CTL1_PDN;
+
+        switch (event) {
+        case SND_SOC_DAPM_PRE_PMD:
+                value |= CS42L51_POWER_CTL1_PDN;
+                break;
+        default:
+        case SND_SOC_DAPM_POST_PMD:
+                break;
+        }
+        snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1,
+                CS42L51_POWER_CTL1_PDN, value);
+
+        return 0;
+}
+
+static const char *cs42l51_dac_names[] = {"Direct PCM",
+        "DSP PCM", "ADC"};
+static const struct soc_enum cs42l51_dac_mux_enum =
+        SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names);
+static const struct snd_kcontrol_new cs42l51_dac_mux_controls =
+        SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum);
+
+static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left",
+        "MIC Left", "MIC+preamp Left"};
+static const struct soc_enum cs42l51_adcl_mux_enum =
+        SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names);
+static const struct snd_kcontrol_new cs42l51_adcl_mux_controls =
+        SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum);
+
+static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right",
+        "MIC Right", "MIC+preamp Right"};
+static const struct soc_enum cs42l51_adcr_mux_enum =
+        SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names);
+static const struct snd_kcontrol_new cs42l51_adcr_mux_controls =
+        SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum);
+
+static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
+        SND_SOC_DAPM_MICBIAS("Mic Bias", CS42L51_MIC_POWER_CTL, 1, 1),
+        SND_SOC_DAPM_PGA_E("Left PGA", CS42L51_POWER_CTL1, 3, 1, NULL, 0,
+                cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+        SND_SOC_DAPM_PGA_E("Right PGA", CS42L51_POWER_CTL1, 4, 1, NULL, 0,
+                cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+        SND_SOC_DAPM_ADC_E("Left ADC", "Left HiFi Capture",
+                CS42L51_POWER_CTL1, 1, 1,
+                cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+        SND_SOC_DAPM_ADC_E("Right ADC", "Right HiFi Capture",
+                CS42L51_POWER_CTL1, 2, 1,
+                cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+        SND_SOC_DAPM_DAC_E("Left DAC", "Left HiFi Playback",
+                CS42L51_POWER_CTL1, 5, 1,
+                cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+        SND_SOC_DAPM_DAC_E("Right DAC", "Right HiFi Playback",
+                CS42L51_POWER_CTL1, 6, 1,
+                cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD),
+
+        /* analog/mic */
+        SND_SOC_DAPM_INPUT("AIN1L"),
+        SND_SOC_DAPM_INPUT("AIN1R"),
+        SND_SOC_DAPM_INPUT("AIN2L"),
+        SND_SOC_DAPM_INPUT("AIN2R"),
+        SND_SOC_DAPM_INPUT("MICL"),
+        SND_SOC_DAPM_INPUT("MICR"),
+
+        SND_SOC_DAPM_MIXER("Mic Preamp Left",
+                CS42L51_MIC_POWER_CTL, 2, 1, NULL, 0),
+        SND_SOC_DAPM_MIXER("Mic Preamp Right",
+                CS42L51_MIC_POWER_CTL, 3, 1, NULL, 0),
+
+        /* HP */
+        SND_SOC_DAPM_OUTPUT("HPL"),
+        SND_SOC_DAPM_OUTPUT("HPR"),
+
+        /* mux */
+        SND_SOC_DAPM_MUX("DAC Mux", SND_SOC_NOPM, 0, 0,
+                &cs42l51_dac_mux_controls),
+        SND_SOC_DAPM_MUX("PGA-ADC Mux Left", SND_SOC_NOPM, 0, 0,
+                &cs42l51_adcl_mux_controls),
+        SND_SOC_DAPM_MUX("PGA-ADC Mux Right", SND_SOC_NOPM, 0, 0,
+                &cs42l51_adcr_mux_controls),
+};
+
+static const struct snd_soc_dapm_route cs42l51_routes[] = {
+        {"HPL", NULL, "Left DAC"},
+        {"HPR", NULL, "Right DAC"},
+
+        {"Left ADC", NULL, "Left PGA"},
+        {"Right ADC", NULL, "Right PGA"},
+
+        {"Mic Preamp Left",  NULL,  "MICL"},
+        {"Mic Preamp Right", NULL,  "MICR"},
+
+        {"PGA-ADC Mux Left",  "AIN1 Left",        "AIN1L" },
+        {"PGA-ADC Mux Left",  "AIN2 Left",        "AIN2L" },
+        {"PGA-ADC Mux Left",  "MIC Left",         "MICL"  },
+        {"PGA-ADC Mux Left",  "MIC+preamp Left",  "Mic Preamp Left" },
+        {"PGA-ADC Mux Right", "AIN1 Right",       "AIN1R" },
+        {"PGA-ADC Mux Right", "AIN2 Right",       "AIN2R" },
+        {"PGA-ADC Mux Right", "MIC Right",        "MICR" },
+        {"PGA-ADC Mux Right", "MIC+preamp Right", "Mic Preamp Right" },
+
+        {"Left PGA", NULL, "PGA-ADC Mux Left"},
+        {"Right PGA", NULL, "PGA-ADC Mux Right"},
+};
+
+static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai,
+                unsigned int format)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+        int ret = 0;
+
+        switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_I2S:
+        case SND_SOC_DAIFMT_LEFT_J:
+        case SND_SOC_DAIFMT_RIGHT_J:
+                cs42l51->audio_mode = format & SND_SOC_DAIFMT_FORMAT_MASK;
+                break;
+        default:
+                dev_err(codec->dev, "invalid DAI format\n");
+                ret = -EINVAL;
+        }
+
+        switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBM_CFM:
+                cs42l51->func = MODE_MASTER;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFS:
+                cs42l51->func = MODE_SLAVE_AUTO;
+                break;
+        default:
+                ret = -EINVAL;
+                break;
+        }
+
+        return ret;
+}
+
+struct cs42l51_ratios {
+        unsigned int ratio;
+        unsigned char speed_mode;
+        unsigned char mclk;
+};
+
+static struct cs42l51_ratios slave_ratios[] = {
+        {  512, CS42L51_QSM_MODE, 0 }, {  768, CS42L51_QSM_MODE, 0 },
+        { 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 },
+        { 2048, CS42L51_QSM_MODE, 0 }, { 3072, CS42L51_QSM_MODE, 0 },
+        {  256, CS42L51_HSM_MODE, 0 }, {  384, CS42L51_HSM_MODE, 0 },
+        {  512, CS42L51_HSM_MODE, 0 }, {  768, CS42L51_HSM_MODE, 0 },
+        { 1024, CS42L51_HSM_MODE, 0 }, { 1536, CS42L51_HSM_MODE, 0 },
+        {  128, CS42L51_SSM_MODE, 0 }, {  192, CS42L51_SSM_MODE, 0 },
+        {  256, CS42L51_SSM_MODE, 0 }, {  384, CS42L51_SSM_MODE, 0 },
+        {  512, CS42L51_SSM_MODE, 0 }, {  768, CS42L51_SSM_MODE, 0 },
+        {  128, CS42L51_DSM_MODE, 0 }, {  192, CS42L51_DSM_MODE, 0 },
+        {  256, CS42L51_DSM_MODE, 0 }, {  384, CS42L51_DSM_MODE, 0 },
+};
+
+static struct cs42l51_ratios slave_auto_ratios[] = {
+        { 1024, CS42L51_QSM_MODE, 0 }, { 1536, CS42L51_QSM_MODE, 0 },
+        { 2048, CS42L51_QSM_MODE, 1 }, { 3072, CS42L51_QSM_MODE, 1 },
+        {  512, CS42L51_HSM_MODE, 0 }, {  768, CS42L51_HSM_MODE, 0 },
+        { 1024, CS42L51_HSM_MODE, 1 }, { 1536, CS42L51_HSM_MODE, 1 },
+        {  256, CS42L51_SSM_MODE, 0 }, {  384, CS42L51_SSM_MODE, 0 },
+        {  512, CS42L51_SSM_MODE, 1 }, {  768, CS42L51_SSM_MODE, 1 },
+        {  128, CS42L51_DSM_MODE, 0 }, {  192, CS42L51_DSM_MODE, 0 },
+        {  256, CS42L51_DSM_MODE, 1 }, {  384, CS42L51_DSM_MODE, 1 },
+};
+
+static int cs42l51_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 cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+        struct cs42l51_ratios *ratios = NULL;
+        int nr_ratios = 0;
+        unsigned int rates = 0;
+        unsigned int rate_min = -1;
+        unsigned int rate_max = 0;
+        int i;
+
+        cs42l51->mclk = freq;
+
+        switch (cs42l51->func) {
+        case MODE_MASTER:
+                return -EINVAL;
+        case MODE_SLAVE:
+                ratios = slave_ratios;
+                nr_ratios = ARRAY_SIZE(slave_ratios);
+                break;
+        case MODE_SLAVE_AUTO:
+                ratios = slave_auto_ratios;
+                nr_ratios = ARRAY_SIZE(slave_auto_ratios);
+                break;
+        }
+
+        for (i = 0; i < nr_ratios; i++) {
+                unsigned int rate = freq / ratios[i].ratio;
+                rates |= snd_pcm_rate_to_rate_bit(rate);
+                if (rate < rate_min)
+                        rate_min = rate;
+                if (rate > rate_max)
+                        rate_max = rate;
+        }
+        rates &= ~SNDRV_PCM_RATE_KNOT;
+
+        if (!rates) {
+                dev_err(codec->dev, "could not find a valid sample rate\n");
+                return -EINVAL;
+        }
+
+        codec_dai->playback.rates = rates;
+        codec_dai->playback.rate_min = rate_min;
+        codec_dai->playback.rate_max = rate_max;
+
+        codec_dai->capture.rates = rates;
+        codec_dai->capture.rate_min = rate_min;
+        codec_dai->capture.rate_max = rate_max;
+
+        return 0;
+}
+
+static int cs42l51_hw_params(struct snd_pcm_substream *substream,
+                struct snd_pcm_hw_params *params,
+                struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_soc_codec *codec = socdev->card->codec;
+        struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
+        int ret;
+        unsigned int i;
+        unsigned int rate;
+        unsigned int ratio;
+        struct cs42l51_ratios *ratios = NULL;
+        int nr_ratios = 0;
+        int intf_ctl, power_ctl, fmt;
+
+        switch (cs42l51->func) {
+        case MODE_MASTER:
+                return -EINVAL;
+        case MODE_SLAVE:
+                ratios = slave_ratios;
+                nr_ratios = ARRAY_SIZE(slave_ratios);
+                break;
+        case MODE_SLAVE_AUTO:
+                ratios = slave_auto_ratios;
+                nr_ratios = ARRAY_SIZE(slave_auto_ratios);
+                break;
+        }
+
+        /* Figure out which MCLK/LRCK ratio to use */
+        rate = params_rate(params);     /* Sampling rate, in Hz */
+        ratio = cs42l51->mclk / rate;    /* MCLK/LRCK ratio */
+        for (i = 0; i < nr_ratios; i++) {
+                if (ratios[i].ratio == ratio)
+                        break;
+        }
+
+        if (i == nr_ratios) {
+                /* We did not find a matching ratio */
+                dev_err(codec->dev, "could not find matching ratio\n");
+                return -EINVAL;
+        }
+
+        intf_ctl = snd_soc_read(codec, CS42L51_INTF_CTL);
+        power_ctl = snd_soc_read(codec, CS42L51_MIC_POWER_CTL);
+
+        intf_ctl &= ~(CS42L51_INTF_CTL_MASTER | CS42L51_INTF_CTL_ADC_I2S
+                        | CS42L51_INTF_CTL_DAC_FORMAT(7));
+        power_ctl &= ~(CS42L51_MIC_POWER_CTL_SPEED(3)
+                        | CS42L51_MIC_POWER_CTL_MCLK_DIV2);
+
+        switch (cs42l51->func) {
+        case MODE_MASTER:
+                intf_ctl |= CS42L51_INTF_CTL_MASTER;
+                power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode);
+                break;
+        case MODE_SLAVE:
+                power_ctl |= CS42L51_MIC_POWER_CTL_SPEED(ratios[i].speed_mode);
+                break;
+        case MODE_SLAVE_AUTO:
+                power_ctl |= CS42L51_MIC_POWER_CTL_AUTO;
+                break;
+        }
+
+        switch (cs42l51->audio_mode) {
+        case SND_SOC_DAIFMT_I2S:
+                intf_ctl |= CS42L51_INTF_CTL_ADC_I2S;
+                intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_I2S);
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(CS42L51_DAC_DIF_LJ24);
+                break;
+        case SND_SOC_DAIFMT_RIGHT_J:
+                switch (params_format(params)) {
+                case SNDRV_PCM_FORMAT_S16_LE:
+                case SNDRV_PCM_FORMAT_S16_BE:
+                        fmt = CS42L51_DAC_DIF_RJ16;
+                        break;
+                case SNDRV_PCM_FORMAT_S18_3LE:
+                case SNDRV_PCM_FORMAT_S18_3BE:
+                        fmt = CS42L51_DAC_DIF_RJ18;
+                        break;
+                case SNDRV_PCM_FORMAT_S20_3LE:
+                case SNDRV_PCM_FORMAT_S20_3BE:
+                        fmt = CS42L51_DAC_DIF_RJ20;
+                        break;
+                case SNDRV_PCM_FORMAT_S24_LE:
+                case SNDRV_PCM_FORMAT_S24_BE:
+                        fmt = CS42L51_DAC_DIF_RJ24;
+                        break;
+                default:
+                        dev_err(codec->dev, "unknown format\n");
+                        return -EINVAL;
+                }
+                intf_ctl |= CS42L51_INTF_CTL_DAC_FORMAT(fmt);
+                break;
+        default:
+                dev_err(codec->dev, "unknown format\n");
+                return -EINVAL;
+        }
+
+        if (ratios[i].mclk)
+                power_ctl |= CS42L51_MIC_POWER_CTL_MCLK_DIV2;
+
+        ret = snd_soc_write(codec, CS42L51_INTF_CTL, intf_ctl);
+        if (ret < 0)
+                return ret;
+
+        ret = snd_soc_write(codec, CS42L51_MIC_POWER_CTL, power_ctl);
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+static int cs42l51_dai_mute(struct snd_soc_dai *dai, int mute)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        int reg;
+        int mask = CS42L51_DAC_OUT_CTL_DACA_MUTE|CS42L51_DAC_OUT_CTL_DACB_MUTE;
+
+        reg = snd_soc_read(codec, CS42L51_DAC_OUT_CTL);
+
+        if (mute)
+                reg |= mask;
+        else
+                reg &= ~mask;
+
+        return snd_soc_write(codec, CS42L51_DAC_OUT_CTL, reg);
+}
+
+static struct snd_soc_dai_ops cs42l51_dai_ops = {
+        .hw_params      = cs42l51_hw_params,
+        .set_sysclk     = cs42l51_set_dai_sysclk,
+        .set_fmt        = cs42l51_set_dai_fmt,
+        .digital_mute   = cs42l51_dai_mute,
+};
+
+struct snd_soc_dai cs42l51_dai = {
+        .name = "CS42L51 HiFi",
+        .playback = {
+                .stream_name = "Playback",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000_96000,
+                .formats = CS42L51_FORMATS,
+        },
+        .capture = {
+                .stream_name = "Capture",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000_96000,
+                .formats = CS42L51_FORMATS,
+        },
+        .ops = &cs42l51_dai_ops,
+};
+EXPORT_SYMBOL_GPL(cs42l51_dai);
+
+
+static int cs42l51_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec;
+        int ret = 0;
+
+        if (!cs42l51_codec) {
+                dev_err(&pdev->dev, "CS42L51 codec not yet registered\n");
+                return -EINVAL;
+        }
+
+        socdev->card->codec = cs42l51_codec;
+        codec = socdev->card->codec;
+
+        /* Register PCMs */
+        ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+        if (ret < 0) {
+                dev_err(&pdev->dev, "failed to create PCMs\n");
+                return ret;
+        }
+
+        snd_soc_add_controls(codec, cs42l51_snd_controls,
+                ARRAY_SIZE(cs42l51_snd_controls));
+        snd_soc_dapm_new_controls(codec, cs42l51_dapm_widgets,
+                ARRAY_SIZE(cs42l51_dapm_widgets));
+        snd_soc_dapm_add_routes(codec, cs42l51_routes,
+                ARRAY_SIZE(cs42l51_routes));
+
+        return 0;
+}
+
+
+static int cs42l51_remove(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+        snd_soc_free_pcms(socdev);
+        snd_soc_dapm_free(socdev);
+
+        return 0;
+}
+
+struct snd_soc_codec_device soc_codec_device_cs42l51 = {
+        .probe =        cs42l51_probe,
+        .remove =       cs42l51_remove
+};
+EXPORT_SYMBOL_GPL(soc_codec_device_cs42l51);
+
+static int __init cs42l51_init(void)
+{
+        int ret;
+
+        ret = i2c_add_driver(&cs42l51_i2c_driver);
+        if (ret != 0) {
+                printk(KERN_ERR "%s: can't add i2c driver\n", __func__);
+                return ret;
+        }
+        return 0;
+}
+module_init(cs42l51_init);
+
+static void __exit cs42l51_exit(void)
+{
+        i2c_del_driver(&cs42l51_i2c_driver);
+}
+module_exit(cs42l51_exit);
+
+MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>");
+MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h
new file mode 100644
index 000000000000..8f0bd9786ad2
--- /dev/null
+++ b/sound/soc/codecs/cs42l51.h
@@ -0,0 +1,163 @@
+/*
+ * cs42l51.h
+ *
+ * ASoC Driver for Cirrus Logic CS42L51 codecs
+ *
+ * Copyright (c) 2010 Arnaud Patard <apatard@mandriva.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ */
+#ifndef _CS42L51_H
+#define _CS42L51_H
+
+#define CS42L51_CHIP_ID                 0x1B
+#define CS42L51_CHIP_REV_A              0x00
+#define CS42L51_CHIP_REV_B              0x01
+
+#define CS42L51_CHIP_REV_ID             0x01
+#define CS42L51_MK_CHIP_REV(a, b)       ((a)<<3|(b))
+
+#define CS42L51_POWER_CTL1              0x02
+#define CS42L51_POWER_CTL1_PDN_DACB     (1<<6)
+#define CS42L51_POWER_CTL1_PDN_DACA     (1<<5)
+#define CS42L51_POWER_CTL1_PDN_PGAB     (1<<4)
+#define CS42L51_POWER_CTL1_PDN_PGAA     (1<<3)
+#define CS42L51_POWER_CTL1_PDN_ADCB     (1<<2)
+#define CS42L51_POWER_CTL1_PDN_ADCA     (1<<1)
+#define CS42L51_POWER_CTL1_PDN          (1<<0)
+
+#define CS42L51_MIC_POWER_CTL           0x03
+#define CS42L51_MIC_POWER_CTL_AUTO      (1<<7)
+#define CS42L51_MIC_POWER_CTL_SPEED(x)  (((x)&3)<<5)
+#define CS42L51_QSM_MODE                3
+#define CS42L51_HSM_MODE                2
+#define CS42L51_SSM_MODE                1
+#define CS42L51_DSM_MODE                0
+#define CS42L51_MIC_POWER_CTL_3ST_SP    (1<<4)
+#define CS42L51_MIC_POWER_CTL_PDN_MICB  (1<<3)
+#define CS42L51_MIC_POWER_CTL_PDN_MICA  (1<<2)
+#define CS42L51_MIC_POWER_CTL_PDN_BIAS  (1<<1)
+#define CS42L51_MIC_POWER_CTL_MCLK_DIV2 (1<<0)
+
+#define CS42L51_INTF_CTL                0x04
+#define CS42L51_INTF_CTL_LOOPBACK       (1<<7)
+#define CS42L51_INTF_CTL_MASTER         (1<<6)
+#define CS42L51_INTF_CTL_DAC_FORMAT(x)  (((x)&7)<<3)
+#define CS42L51_DAC_DIF_LJ24            0x00
+#define CS42L51_DAC_DIF_I2S             0x01
+#define CS42L51_DAC_DIF_RJ24            0x02
+#define CS42L51_DAC_DIF_RJ20            0x03
+#define CS42L51_DAC_DIF_RJ18            0x04
+#define CS42L51_DAC_DIF_RJ16            0x05
+#define CS42L51_INTF_CTL_ADC_I2S        (1<<2)
+#define CS42L51_INTF_CTL_DIGMIX         (1<<1)
+#define CS42L51_INTF_CTL_MICMIX         (1<<0)
+
+#define CS42L51_MIC_CTL                 0x05
+#define CS42L51_MIC_CTL_ADC_SNGVOL      (1<<7)
+#define CS42L51_MIC_CTL_ADCD_DBOOST     (1<<6)
+#define CS42L51_MIC_CTL_ADCA_DBOOST     (1<<5)
+#define CS42L51_MIC_CTL_MICBIAS_SEL     (1<<4)
+#define CS42L51_MIC_CTL_MICBIAS_LVL(x)  (((x)&3)<<2)
+#define CS42L51_MIC_CTL_MICB_BOOST      (1<<1)
+#define CS42L51_MIC_CTL_MICA_BOOST      (1<<0)
+
+#define CS42L51_ADC_CTL                 0x06
+#define CS42L51_ADC_CTL_ADCB_HPFEN      (1<<7)
+#define CS42L51_ADC_CTL_ADCB_HPFRZ      (1<<6)
+#define CS42L51_ADC_CTL_ADCA_HPFEN      (1<<5)
+#define CS42L51_ADC_CTL_ADCA_HPFRZ      (1<<4)
+#define CS42L51_ADC_CTL_SOFTB           (1<<3)
+#define CS42L51_ADC_CTL_ZCROSSB         (1<<2)
+#define CS42L51_ADC_CTL_SOFTA           (1<<1)
+#define CS42L51_ADC_CTL_ZCROSSA         (1<<0)
+
+#define CS42L51_ADC_INPUT               0x07
+#define CS42L51_ADC_INPUT_AINB_MUX(x)   (((x)&3)<<6)
+#define CS42L51_ADC_INPUT_AINA_MUX(x)   (((x)&3)<<4)
+#define CS42L51_ADC_INPUT_INV_ADCB      (1<<3)
+#define CS42L51_ADC_INPUT_INV_ADCA      (1<<2)
+#define CS42L51_ADC_INPUT_ADCB_MUTE     (1<<1)
+#define CS42L51_ADC_INPUT_ADCA_MUTE     (1<<0)
+
+#define CS42L51_DAC_OUT_CTL             0x08
+#define CS42L51_DAC_OUT_CTL_HP_GAIN(x)  (((x)&7)<<5)
+#define CS42L51_DAC_OUT_CTL_DAC_SNGVOL  (1<<4)
+#define CS42L51_DAC_OUT_CTL_INV_PCMB    (1<<3)
+#define CS42L51_DAC_OUT_CTL_INV_PCMA    (1<<2)
+#define CS42L51_DAC_OUT_CTL_DACB_MUTE   (1<<1)
+#define CS42L51_DAC_OUT_CTL_DACA_MUTE   (1<<0)
+
+#define CS42L51_DAC_CTL                 0x09
+#define CS42L51_DAC_CTL_DATA_SEL(x)     (((x)&3)<<6)
+#define CS42L51_DAC_CTL_FREEZE          (1<<5)
+#define CS42L51_DAC_CTL_DEEMPH          (1<<3)
+#define CS42L51_DAC_CTL_AMUTE           (1<<2)
+#define CS42L51_DAC_CTL_DACSZ(x)        (((x)&3)<<0)
+
+#define CS42L51_ALC_PGA_CTL             0x0A
+#define CS42L51_ALC_PGB_CTL             0x0B
+#define CS42L51_ALC_PGX_ALCX_SRDIS      (1<<7)
+#define CS42L51_ALC_PGX_ALCX_ZCDIS      (1<<6)
+#define CS42L51_ALC_PGX_PGX_VOL(x)      (((x)&0x1f)<<0)
+
+#define CS42L51_ADCA_ATT                0x0C
+#define CS42L51_ADCB_ATT                0x0D
+
+#define CS42L51_ADCA_VOL                0x0E
+#define CS42L51_ADCB_VOL                0x0F
+#define CS42L51_PCMA_VOL                0x10
+#define CS42L51_PCMB_VOL                0x11
+#define CS42L51_MIX_MUTE_ADCMIX         (1<<7)
+#define CS42L51_MIX_VOLUME(x)           (((x)&0x7f)<<0)
+
+#define CS42L51_BEEP_FREQ               0x12
+#define CS42L51_BEEP_VOL                0x13
+#define CS42L51_BEEP_CONF               0x14
+
+#define CS42L51_TONE_CTL                0x15
+#define CS42L51_TONE_CTL_TREB(x)        (((x)&0xf)<<4)
+#define CS42L51_TONE_CTL_BASS(x)        (((x)&0xf)<<0)
+
+#define CS42L51_AOUTA_VOL               0x16
+#define CS42L51_AOUTB_VOL               0x17
+#define CS42L51_PCM_MIXER               0x18
+#define CS42L51_LIMIT_THRES_DIS         0x19
+#define CS42L51_LIMIT_REL               0x1A
+#define CS42L51_LIMIT_ATT               0x1B
+#define CS42L51_ALC_EN                  0x1C
+#define CS42L51_ALC_REL                 0x1D
+#define CS42L51_ALC_THRES               0x1E
+#define CS42L51_NOISE_CONF              0x1F
+
+#define CS42L51_STATUS                  0x20
+#define CS42L51_STATUS_SP_CLKERR        (1<<6)
+#define CS42L51_STATUS_SPEA_OVFL        (1<<5)
+#define CS42L51_STATUS_SPEB_OVFL        (1<<4)
+#define CS42L51_STATUS_PCMA_OVFL        (1<<3)
+#define CS42L51_STATUS_PCMB_OVFL        (1<<2)
+#define CS42L51_STATUS_ADCA_OVFL        (1<<1)
+#define CS42L51_STATUS_ADCB_OVFL        (1<<0)
+
+#define CS42L51_CHARGE_FREQ             0x21
+
+#define CS42L51_FIRSTREG        0x01
+/*
+ * Hack: with register 0x21, it makes 33 registers. Looks like someone in the
+ * i2c layer doesn't like i2c smbus block read of 33 regs. Workaround by using
+ * 32 regs
+ */
+#define CS42L51_LASTREG         0x20
+#define CS42L51_NUMREGS         (CS42L51_LASTREG - CS42L51_FIRSTREG + 1)
+
+extern struct snd_soc_dai cs42l51_dai;
+extern struct snd_soc_codec_device soc_codec_device_cs42l51;
+#endif
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index 75af2d6e0e78..3c51d6a57523 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -15,23 +15,15 @@
  * 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 <linux/slab.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/tlv.h>
 #include <sound/initval.h>
-#include <asm/div64.h>
+#include <sound/tlv.h>
 
 #include "da7210.h"
 
@@ -145,6 +137,29 @@
 
 #define DA7210_VERSION "0.0.1"
 
+/*
+ * Playback Volume
+ *
+ * max          : 0x3F (+15.0 dB)
+ *                 (1.5 dB step)
+ * min          : 0x11 (-54.0 dB)
+ * mute         : 0x10
+ * reserved     : 0x00 - 0x0F
+ *
+ * ** FIXME **
+ *
+ * Reserved area are considered as "mute".
+ * -> min = -79.5 dB
+ */
+static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1);
+
+static const struct snd_kcontrol_new da7210_snd_controls[] = {
+
+        SOC_DOUBLE_R_TLV("HeadPhone Playback Volume",
+                         DA7210_HP_L_VOL, DA7210_HP_R_VOL,
+                         0, 0x3F, 0, hp_out_tlv),
+};
+
 /* Codec private data */
 struct da7210_priv {
         struct snd_soc_codec codec;
@@ -227,10 +242,6 @@ static int da7210_startup(struct snd_pcm_substream *substream,
         struct snd_soc_codec *codec = dai->codec;
 
         if (is_play) {
-                /* PlayBack Volume 40 */
-                snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40);
-                snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40);
-
                 /* Enable Out */
                 snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10);
                 snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10);
@@ -488,7 +499,7 @@ static int da7210_init(struct da7210_priv *da7210)
         ret = snd_soc_register_dai(&da7210_dai);
         if (ret) {
                 dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-                goto init_err;
+                goto codec_err;
         }
 
         /* FIXME
@@ -574,6 +585,8 @@ static int da7210_init(struct da7210_priv *da7210)
 
         return ret;
 
+codec_err:
+        snd_soc_unregister_codec(codec);
 init_err:
         kfree(codec->reg_cache);
         codec->reg_cache = NULL;
@@ -601,8 +614,10 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
         codec->control_data = i2c;
 
         ret = da7210_init(da7210);
-        if (ret < 0)
+        if (ret < 0) {
                 pr_err("Failed to initialise da7210 audio codec\n");
+                kfree(da7210);
+        }
 
         return ret;
 }
@@ -656,6 +671,9 @@ static int da7210_probe(struct platform_device *pdev)
         if (ret < 0)
                 goto pcm_err;
 
+        snd_soc_add_controls(da7210_codec, da7210_snd_controls,
+                             ARRAY_SIZE(da7210_snd_controls));
+
         dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
 
 pcm_err:
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
new file mode 100644
index 000000000000..66557de1e4fe
--- /dev/null
+++ b/sound/soc/codecs/jz4740.c
@@ -0,0 +1,511 @@
+/*
+ * Copyright (C) 2009-2010, Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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.
+ *
+ *  You should have received a copy of the  GNU General Public License along
+ *  with this program; if not, write  to the Free Software Foundation, Inc.,
+ *  675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc-dapm.h>
+#include <sound/soc.h>
+
+#define JZ4740_REG_CODEC_1 0x0
+#define JZ4740_REG_CODEC_2 0x1
+
+#define JZ4740_CODEC_1_LINE_ENABLE BIT(29)
+#define JZ4740_CODEC_1_MIC_ENABLE BIT(28)
+#define JZ4740_CODEC_1_SW1_ENABLE BIT(27)
+#define JZ4740_CODEC_1_ADC_ENABLE BIT(26)
+#define JZ4740_CODEC_1_SW2_ENABLE BIT(25)
+#define JZ4740_CODEC_1_DAC_ENABLE BIT(24)
+#define JZ4740_CODEC_1_VREF_DISABLE BIT(20)
+#define JZ4740_CODEC_1_VREF_AMP_DISABLE BIT(19)
+#define JZ4740_CODEC_1_VREF_PULLDOWN BIT(18)
+#define JZ4740_CODEC_1_VREF_LOW_CURRENT BIT(17)
+#define JZ4740_CODEC_1_VREF_HIGH_CURRENT BIT(16)
+#define JZ4740_CODEC_1_HEADPHONE_DISABLE BIT(14)
+#define JZ4740_CODEC_1_HEADPHONE_AMP_CHANGE_ANY BIT(13)
+#define JZ4740_CODEC_1_HEADPHONE_CHARGE BIT(12)
+#define JZ4740_CODEC_1_HEADPHONE_PULLDOWN (BIT(11) | BIT(10))
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M BIT(9)
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN BIT(8)
+#define JZ4740_CODEC_1_SUSPEND BIT(1)
+#define JZ4740_CODEC_1_RESET BIT(0)
+
+#define JZ4740_CODEC_1_LINE_ENABLE_OFFSET 29
+#define JZ4740_CODEC_1_MIC_ENABLE_OFFSET 28
+#define JZ4740_CODEC_1_SW1_ENABLE_OFFSET 27
+#define JZ4740_CODEC_1_ADC_ENABLE_OFFSET 26
+#define JZ4740_CODEC_1_SW2_ENABLE_OFFSET 25
+#define JZ4740_CODEC_1_DAC_ENABLE_OFFSET 24
+#define JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET 14
+#define JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET 8
+
+#define JZ4740_CODEC_2_INPUT_VOLUME_MASK                0x1f0000
+#define JZ4740_CODEC_2_SAMPLE_RATE_MASK                 0x000f00
+#define JZ4740_CODEC_2_MIC_BOOST_GAIN_MASK              0x000030
+#define JZ4740_CODEC_2_HEADPHONE_VOLUME_MASK    0x000003
+
+#define JZ4740_CODEC_2_INPUT_VOLUME_OFFSET              16
+#define JZ4740_CODEC_2_SAMPLE_RATE_OFFSET                8
+#define JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET     4
+#define JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET   0
+
+static const uint32_t jz4740_codec_regs[] = {
+        0x021b2302, 0x00170803,
+};
+
+struct jz4740_codec {
+        void __iomem *base;
+        struct resource *mem;
+
+        uint32_t reg_cache[2];
+        struct snd_soc_codec codec;
+};
+
+static inline struct jz4740_codec *codec_to_jz4740(struct snd_soc_codec *codec)
+{
+        return container_of(codec, struct jz4740_codec, codec);
+}
+
+static unsigned int jz4740_codec_read(struct snd_soc_codec *codec,
+        unsigned int reg)
+{
+        struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec);
+        return readl(jz4740_codec->base + (reg << 2));
+}
+
+static int jz4740_codec_write(struct snd_soc_codec *codec, unsigned int reg,
+        unsigned int val)
+{
+        struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec);
+
+        jz4740_codec->reg_cache[reg] = val;
+        writel(val, jz4740_codec->base + (reg << 2));
+
+        return 0;
+}
+
+static const struct snd_kcontrol_new jz4740_codec_controls[] = {
+        SOC_SINGLE("Master Playback Volume", JZ4740_REG_CODEC_2,
+                        JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET, 3, 0),
+        SOC_SINGLE("Master Capture Volume", JZ4740_REG_CODEC_2,
+                        JZ4740_CODEC_2_INPUT_VOLUME_OFFSET, 31, 0),
+        SOC_SINGLE("Master Playback Switch", JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET, 1, 1),
+        SOC_SINGLE("Mic Capture Volume", JZ4740_REG_CODEC_2,
+                        JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET, 3, 0),
+};
+
+static const struct snd_kcontrol_new jz4740_codec_output_controls[] = {
+        SOC_DAPM_SINGLE("Bypass Switch", JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_SW1_ENABLE_OFFSET, 1, 0),
+        SOC_DAPM_SINGLE("DAC Switch", JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_SW2_ENABLE_OFFSET, 1, 0),
+};
+
+static const struct snd_kcontrol_new jz4740_codec_input_controls[] = {
+        SOC_DAPM_SINGLE("Line Capture Switch", JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_LINE_ENABLE_OFFSET, 1, 0),
+        SOC_DAPM_SINGLE("Mic Capture Switch", JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_MIC_ENABLE_OFFSET, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget jz4740_codec_dapm_widgets[] = {
+        SND_SOC_DAPM_ADC("ADC", "Capture", JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_ADC_ENABLE_OFFSET, 0),
+        SND_SOC_DAPM_DAC("DAC", "Playback", JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_DAC_ENABLE_OFFSET, 0),
+
+        SND_SOC_DAPM_MIXER("Output Mixer", JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_HEADPHONE_POWERDOWN_OFFSET, 1,
+                        jz4740_codec_output_controls,
+                        ARRAY_SIZE(jz4740_codec_output_controls)),
+
+        SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0,
+                        jz4740_codec_input_controls,
+                        ARRAY_SIZE(jz4740_codec_input_controls)),
+        SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+        SND_SOC_DAPM_OUTPUT("LOUT"),
+        SND_SOC_DAPM_OUTPUT("ROUT"),
+
+        SND_SOC_DAPM_INPUT("MIC"),
+        SND_SOC_DAPM_INPUT("LIN"),
+        SND_SOC_DAPM_INPUT("RIN"),
+};
+
+static const struct snd_soc_dapm_route jz4740_codec_dapm_routes[] = {
+        {"Line Input", NULL, "LIN"},
+        {"Line Input", NULL, "RIN"},
+
+        {"Input Mixer", "Line Capture Switch", "Line Input"},
+        {"Input Mixer", "Mic Capture Switch", "MIC"},
+
+        {"ADC", NULL, "Input Mixer"},
+
+        {"Output Mixer", "Bypass Switch", "Input Mixer"},
+        {"Output Mixer", "DAC Switch", "DAC"},
+
+        {"LOUT", NULL, "Output Mixer"},
+        {"ROUT", NULL, "Output Mixer"},
+};
+
+static int jz4740_codec_hw_params(struct snd_pcm_substream *substream,
+        struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+        uint32_t val;
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_soc_codec *codec = socdev->card->codec;
+
+        switch (params_rate(params)) {
+        case 8000:
+                val = 0;
+                break;
+        case 11025:
+                val = 1;
+                break;
+        case 12000:
+                val = 2;
+                break;
+        case 16000:
+                val = 3;
+                break;
+        case 22050:
+                val = 4;
+                break;
+        case 24000:
+                val = 5;
+                break;
+        case 32000:
+                val = 6;
+                break;
+        case 44100:
+                val = 7;
+                break;
+        case 48000:
+                val = 8;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        val <<= JZ4740_CODEC_2_SAMPLE_RATE_OFFSET;
+
+        snd_soc_update_bits(codec, JZ4740_REG_CODEC_2,
+                                JZ4740_CODEC_2_SAMPLE_RATE_MASK, val);
+
+        return 0;
+}
+
+static struct snd_soc_dai_ops jz4740_codec_dai_ops = {
+        .hw_params = jz4740_codec_hw_params,
+};
+
+struct snd_soc_dai jz4740_codec_dai = {
+        .name = "jz4740",
+        .playback = {
+                .stream_name = "Playback",
+                .channels_min = 2,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000_48000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
+        },
+        .capture = {
+                .stream_name = "Capture",
+                .channels_min = 2,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000_48000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
+        },
+        .ops = &jz4740_codec_dai_ops,
+        .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(jz4740_codec_dai);
+
+static void jz4740_codec_wakeup(struct snd_soc_codec *codec)
+{
+        int i;
+        uint32_t *cache = codec->reg_cache;
+
+        snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+                JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET);
+        udelay(2);
+
+        snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+                JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0);
+
+        for (i = 0; i < ARRAY_SIZE(jz4740_codec_regs); ++i)
+                jz4740_codec_write(codec, i, cache[i]);
+}
+
+static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
+        enum snd_soc_bias_level level)
+{
+        unsigned int mask;
+        unsigned int value;
+
+        switch (level) {
+        case SND_SOC_BIAS_ON:
+                break;
+        case SND_SOC_BIAS_PREPARE:
+                mask = JZ4740_CODEC_1_VREF_DISABLE |
+                                JZ4740_CODEC_1_VREF_AMP_DISABLE |
+                                JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+                value = 0;
+
+                snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+                break;
+        case SND_SOC_BIAS_STANDBY:
+                /* The only way to clear the suspend flag is to reset the codec */
+                if (codec->bias_level == SND_SOC_BIAS_OFF)
+                        jz4740_codec_wakeup(codec);
+
+                mask = JZ4740_CODEC_1_VREF_DISABLE |
+                        JZ4740_CODEC_1_VREF_AMP_DISABLE |
+                        JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+                value = JZ4740_CODEC_1_VREF_DISABLE |
+                        JZ4740_CODEC_1_VREF_AMP_DISABLE |
+                        JZ4740_CODEC_1_HEADPHONE_POWERDOWN_M;
+
+                snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+                break;
+        case SND_SOC_BIAS_OFF:
+                mask = JZ4740_CODEC_1_SUSPEND;
+                value = JZ4740_CODEC_1_SUSPEND;
+
+                snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, mask, value);
+                break;
+        default:
+                break;
+        }
+
+        codec->bias_level = level;
+
+        return 0;
+}
+
+static struct snd_soc_codec *jz4740_codec_codec;
+
+static int jz4740_codec_dev_probe(struct platform_device *pdev)
+{
+        int ret;
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = jz4740_codec_codec;
+
+        BUG_ON(!codec);
+
+        socdev->card->codec = codec;
+
+        ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+        if (ret) {
+                dev_err(&pdev->dev, "Failed to create pcms: %d\n", ret);
+                return ret;
+        }
+
+        snd_soc_add_controls(codec, jz4740_codec_controls,
+                ARRAY_SIZE(jz4740_codec_controls));
+
+        snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets,
+                ARRAY_SIZE(jz4740_codec_dapm_widgets));
+
+        snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes,
+                ARRAY_SIZE(jz4740_codec_dapm_routes));
+
+        snd_soc_dapm_new_widgets(codec);
+
+        return 0;
+}
+
+static int jz4740_codec_dev_remove(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+        snd_soc_free_pcms(socdev);
+        snd_soc_dapm_free(socdev);
+
+        return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+
+static int jz4740_codec_suspend(struct platform_device *pdev, pm_message_t state)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+
+        return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int jz4740_codec_resume(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+
+        return jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+#else
+#define jz4740_codec_suspend NULL
+#define jz4740_codec_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_jz4740_codec = {
+        .probe = jz4740_codec_dev_probe,
+        .remove = jz4740_codec_dev_remove,
+        .suspend = jz4740_codec_suspend,
+        .resume = jz4740_codec_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_jz4740_codec);
+
+static int __devinit jz4740_codec_probe(struct platform_device *pdev)
+{
+        int ret;
+        struct jz4740_codec *jz4740_codec;
+        struct snd_soc_codec *codec;
+        struct resource *mem;
+
+        jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL);
+        if (!jz4740_codec)
+                return -ENOMEM;
+
+        mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!mem) {
+                dev_err(&pdev->dev, "Failed to get mmio memory resource\n");
+                ret = -ENOENT;
+                goto err_free_codec;
+        }
+
+        mem = request_mem_region(mem->start, resource_size(mem), pdev->name);
+        if (!mem) {
+                dev_err(&pdev->dev, "Failed to request mmio memory region\n");
+                ret = -EBUSY;
+                goto err_free_codec;
+        }
+
+        jz4740_codec->base = ioremap(mem->start, resource_size(mem));
+        if (!jz4740_codec->base) {
+                dev_err(&pdev->dev, "Failed to ioremap mmio memory\n");
+                ret = -EBUSY;
+                goto err_release_mem_region;
+        }
+        jz4740_codec->mem = mem;
+
+        jz4740_codec_dai.dev = &pdev->dev;
+
+        codec = &jz4740_codec->codec;
+
+        codec->dev              = &pdev->dev;
+        codec->name             = "jz4740";
+        codec->owner            = THIS_MODULE;
+
+        codec->read             = jz4740_codec_read;
+        codec->write            = jz4740_codec_write;
+        codec->set_bias_level   = jz4740_codec_set_bias_level;
+        codec->bias_level       = SND_SOC_BIAS_OFF;
+
+        codec->dai              = &jz4740_codec_dai;
+        codec->num_dai          = 1;
+
+        codec->reg_cache        = jz4740_codec->reg_cache;
+        codec->reg_cache_size   = 2;
+        memcpy(codec->reg_cache, jz4740_codec_regs, sizeof(jz4740_codec_regs));
+
+        mutex_init(&codec->mutex);
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        jz4740_codec_codec = codec;
+
+        snd_soc_update_bits(codec, JZ4740_REG_CODEC_1,
+                        JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE);
+
+        platform_set_drvdata(pdev, jz4740_codec);
+
+        ret = snd_soc_register_codec(codec);
+        if (ret) {
+                dev_err(&pdev->dev, "Failed to register codec\n");
+                goto err_iounmap;
+        }
+
+        ret = snd_soc_register_dai(&jz4740_codec_dai);
+        if (ret) {
+                dev_err(&pdev->dev, "Failed to register codec dai\n");
+                goto err_unregister_codec;
+        }
+
+        jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+        return 0;
+
+err_unregister_codec:
+        snd_soc_unregister_codec(codec);
+err_iounmap:
+        iounmap(jz4740_codec->base);
+err_release_mem_region:
+        release_mem_region(mem->start, resource_size(mem));
+err_free_codec:
+        kfree(jz4740_codec);
+
+        return ret;
+}
+
+static int __devexit jz4740_codec_remove(struct platform_device *pdev)
+{
+        struct jz4740_codec *jz4740_codec = platform_get_drvdata(pdev);
+        struct resource *mem = jz4740_codec->mem;
+
+        snd_soc_unregister_dai(&jz4740_codec_dai);
+        snd_soc_unregister_codec(&jz4740_codec->codec);
+
+        iounmap(jz4740_codec->base);
+        release_mem_region(mem->start, resource_size(mem));
+
+        platform_set_drvdata(pdev, NULL);
+        kfree(jz4740_codec);
+
+        return 0;
+}
+
+static struct platform_driver jz4740_codec_driver = {
+        .probe = jz4740_codec_probe,
+        .remove = __devexit_p(jz4740_codec_remove),
+        .driver = {
+                .name = "jz4740-codec",
+                .owner = THIS_MODULE,
+        },
+};
+
+static int __init jz4740_codec_init(void)
+{
+        return platform_driver_register(&jz4740_codec_driver);
+}
+module_init(jz4740_codec_init);
+
+static void __exit jz4740_codec_exit(void)
+{
+        platform_driver_unregister(&jz4740_codec_driver);
+}
+module_exit(jz4740_codec_exit);
+
+MODULE_DESCRIPTION("JZ4740 SoC internal codec driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:jz4740-codec");
diff --git a/sound/soc/codecs/jz4740.h b/sound/soc/codecs/jz4740.h
new file mode 100644
index 000000000000..b5a0691be763
--- /dev/null
+++ b/sound/soc/codecs/jz4740.h
@@ -0,0 +1,20 @@
+/*
+ * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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.
+ *
+ *  You should have received a copy of the  GNU General Public License along
+ *  with this program; if not, write  to the Free Software Foundation, Inc.,
+ *  675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#ifndef __SND_SOC_CODECS_JZ4740_CODEC_H__
+#define __SND_SOC_CODECS_JZ4740_CODEC_H__
+
+extern struct snd_soc_dai jz4740_codec_dai;
+extern struct snd_soc_codec_device soc_codec_dev_jz4740_codec;
+
+#endif
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
index a63191141052..9119836051a4 100644
--- a/sound/soc/codecs/spdif_transciever.c
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -16,8 +16,10 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
+#include <linux/slab.h>
 #include <sound/soc.h>
 #include <sound/pcm.h>
+#include <sound/initval.h>
 
 #include "spdif_transciever.h"
 
@@ -26,6 +28,48 @@ MODULE_LICENSE("GPL");
 #define STUB_RATES      SNDRV_PCM_RATE_8000_96000
 #define STUB_FORMATS    SNDRV_PCM_FMTBIT_S16_LE
 
+static struct snd_soc_codec *spdif_dit_codec;
+
+static int spdif_dit_codec_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec;
+        int ret;
+
+        if (spdif_dit_codec == NULL) {
+                dev_err(&pdev->dev, "Codec device not registered\n");
+                return -ENODEV;
+        }
+
+        socdev->card->codec = spdif_dit_codec;
+        codec = spdif_dit_codec;
+
+        ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+        if (ret < 0) {
+                dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+                goto err_create_pcms;
+        }
+
+        return 0;
+
+err_create_pcms:
+        return ret;
+}
+
+static int spdif_dit_codec_remove(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+        snd_soc_free_pcms(socdev);
+
+        return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_spdif_dit = {
+        .probe          = spdif_dit_codec_probe,
+        .remove         = spdif_dit_codec_remove,
+}; EXPORT_SYMBOL_GPL(soc_codec_dev_spdif_dit);
+
 struct snd_soc_dai dit_stub_dai = {
         .name           = "DIT",
         .playback       = {
@@ -40,13 +84,61 @@ EXPORT_SYMBOL_GPL(dit_stub_dai);
 
 static int spdif_dit_probe(struct platform_device *pdev)
 {
+        struct snd_soc_codec *codec;
+        int ret;
+
+        if (spdif_dit_codec) {
+                dev_err(&pdev->dev, "Another Codec is registered\n");
+                ret = -EINVAL;
+                goto err_reg_codec;
+        }
+
+        codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+        if (codec == NULL)
+                return -ENOMEM;
+
+        codec->dev = &pdev->dev;
+
+        mutex_init(&codec->mutex);
+
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        codec->name = "spdif-dit";
+        codec->owner = THIS_MODULE;
+        codec->dai = &dit_stub_dai;
+        codec->num_dai = 1;
+
+        spdif_dit_codec = codec;
+
+        ret = snd_soc_register_codec(codec);
+        if (ret < 0) {
+                dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+                goto err_reg_codec;
+        }
+
         dit_stub_dai.dev = &pdev->dev;
-        return snd_soc_register_dai(&dit_stub_dai);
+        ret = snd_soc_register_dai(&dit_stub_dai);
+        if (ret < 0) {
+                dev_err(codec->dev, "Failed to register dai: %d\n", ret);
+                goto err_reg_dai;
+        }
+
+        return 0;
+
+err_reg_dai:
+        snd_soc_unregister_codec(codec);
+err_reg_codec:
+        kfree(spdif_dit_codec);
+        return ret;
 }
 
 static int spdif_dit_remove(struct platform_device *pdev)
 {
         snd_soc_unregister_dai(&dit_stub_dai);
+        snd_soc_unregister_codec(spdif_dit_codec);
+        kfree(spdif_dit_codec);
+        spdif_dit_codec = NULL;
         return 0;
 }
 
diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h
index 296f2eb6c4ef..1e102124f546 100644
--- a/sound/soc/codecs/spdif_transciever.h
+++ b/sound/soc/codecs/spdif_transciever.h
@@ -12,6 +12,7 @@
 #ifndef CODEC_STUBS_H
 #define CODEC_STUBS_H
 
+extern struct snd_soc_codec_device soc_codec_dev_spdif_dit;
 extern struct snd_soc_dai dit_stub_dai;
 
 #endif /* CODEC_STUBS_H */
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index b0bae3508b29..0a4b0fef3355 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -560,13 +560,16 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
         switch (level) {
         case SND_SOC_BIAS_ON:
                 /* vref/mid, osc on, dac unmute */
+                reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
+                        TLV320AIC23_DAC_OFF);
                 tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
                 break;
         case SND_SOC_BIAS_PREPARE:
                 break;
         case SND_SOC_BIAS_STANDBY:
                 /* everything off except vref/vmid, */
-                tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
+                tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \
+                        TLV320AIC23_CLK_OFF);
                 break;
         case SND_SOC_BIAS_OFF:
                 /* everything off, dac mute, inactive */
@@ -615,7 +618,6 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_codec *codec = socdev->card->codec;
 
-        tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
         tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
         return 0;
@@ -632,7 +634,6 @@ static int tlv320aic23_resume(struct platform_device *pdev)
                 u16 val = tlv320aic23_read_reg_cache(codec, reg);
                 tlv320aic23_write(codec, reg, val);
         }
-
         tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
         return 0;
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 65adc77eada1..8651b01ed223 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -49,8 +49,6 @@
 
 #define NSAMPLE_MAX             5700
 
-#define LATENCY_TIME_MS         20
-
 #define MODE7_LTHR              10
 #define MODE7_UTHR              (DAC33_BUFFER_SIZE_SAMPLES - 10)
 
@@ -62,6 +60,9 @@
 #define US_TO_SAMPLES(rate, us) \
         (rate / (1000000 / us))
 
+#define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
+        ((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate)))
+
 static void dac33_calculate_times(struct snd_pcm_substream *substream);
 static int dac33_prepare_chip(struct snd_pcm_substream *substream);
 
@@ -107,6 +108,10 @@ struct tlv320dac33_priv {
                                          * this */
         enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */
         unsigned int nsample;           /* burst read amount from host */
+        int mode1_latency;              /* latency caused by the i2c writes in
+                                         * us */
+        int auto_fifo_config;           /* Configure the FIFO based on the
+                                         * period size */
         u8 burst_bclkdiv;               /* BCLK divider value in burst mode */
         unsigned int burst_rate;        /* Interface speed in Burst modes */
 
@@ -120,6 +125,8 @@ struct tlv320dac33_priv {
                                          * samples */
         unsigned int mode7_us_to_lthr;  /* Time to reach lthr from uthr */
 
+        unsigned int uthr;
+
         enum dac33_state state;
 };
 
@@ -442,6 +449,39 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
         return ret;
 }
 
+static int dac33_get_uthr(struct snd_kcontrol *kcontrol,
+                         struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+
+        ucontrol->value.integer.value[0] = dac33->uthr;
+
+        return 0;
+}
+
+static int dac33_set_uthr(struct snd_kcontrol *kcontrol,
+                         struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+        int ret = 0;
+
+        if (dac33->substream)
+                return -EBUSY;
+
+        if (dac33->uthr == ucontrol->value.integer.value[0])
+                return 0;
+
+        if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) ||
+            ucontrol->value.integer.value[0] > MODE7_UTHR)
+                ret = -EINVAL;
+        else
+                dac33->uthr = ucontrol->value.integer.value[0];
+
+        return ret;
+}
+
 static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol,
                          struct snd_ctl_elem_value *ucontrol)
 {
@@ -503,13 +543,18 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = {
                  DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1),
 };
 
-static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = {
-        SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
-                 dac33_get_nsample, dac33_set_nsample),
+static const struct snd_kcontrol_new dac33_mode_snd_controls[] = {
         SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum,
                  dac33_get_fifo_mode, dac33_set_fifo_mode),
 };
 
+static const struct snd_kcontrol_new dac33_fifo_snd_controls[] = {
+        SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
+                dac33_get_nsample, dac33_set_nsample),
+        SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0,
+                 dac33_get_uthr, dac33_set_uthr),
+};
+
 /* Analog bypass */
 static const struct snd_kcontrol_new dac33_dapm_abypassl_control =
         SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1);
@@ -612,7 +657,7 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
         switch (dac33->fifo_mode) {
         case DAC33_FIFO_MODE1:
                 dac33_write16(codec, DAC33_NSAMPLE_MSB,
-                        DAC33_THRREG(dac33->nsample + dac33->alarm_threshold));
+                        DAC33_THRREG(dac33->nsample));
 
                 /* Take the timestamps */
                 spin_lock_irq(&dac33->lock);
@@ -761,6 +806,10 @@ static void dac33_shutdown(struct snd_pcm_substream *substream,
         struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
 
         dac33->substream = NULL;
+
+        /* Reset the nSample restrictions */
+        dac33->nsample_min = 0;
+        dac33->nsample_max = NSAMPLE_MAX;
 }
 
 static int dac33_hw_params(struct snd_pcm_substream *substream,
@@ -985,7 +1034,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
                  * Configure the threshold levels, and leave 10 sample space
                  * at the bottom, and also at the top of the FIFO
                  */
-                dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR));
+                dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr));
                 dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR));
                 break;
         default:
@@ -1003,57 +1052,71 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream)
         struct snd_soc_device *socdev = rtd->socdev;
         struct snd_soc_codec *codec = socdev->card->codec;
         struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
+        unsigned int period_size = substream->runtime->period_size;
+        unsigned int rate = substream->runtime->rate;
         unsigned int nsample_limit;
 
         /* In bypass mode we don't need to calculate */
         if (!dac33->fifo_mode)
                 return;
 
-        /* Number of samples (16bit, stereo) in one period */
-        dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4;
-
-        /* Number of samples (16bit, stereo) in ALSA buffer */
-        dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4;
-        /* Subtract one period from the total */
-        dac33->nsample_max -= dac33->nsample_min;
-
-        /* Number of samples for LATENCY_TIME_MS / 2 */
-        dac33->alarm_threshold = substream->runtime->rate /
-                                 (1000 / (LATENCY_TIME_MS / 2));
-
-        /* Find and fix up the lowest nsmaple limit */
-        nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS);
-
-        if (dac33->nsample_min < nsample_limit)
-                dac33->nsample_min = nsample_limit;
-
-        if (dac33->nsample < dac33->nsample_min)
-                dac33->nsample = dac33->nsample_min;
-
-        /*
-         * Find and fix up the highest nsmaple limit
-         * In order to not overflow the DAC33 buffer substract the
-         * alarm_threshold value from the size of the DAC33 buffer
-         */
-        nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold;
-
-        if (dac33->nsample_max > nsample_limit)
-                dac33->nsample_max = nsample_limit;
-
-        if (dac33->nsample > dac33->nsample_max)
-                dac33->nsample = dac33->nsample_max;
-
         switch (dac33->fifo_mode) {
         case DAC33_FIFO_MODE1:
+                /* Number of samples under i2c latency */
+                dac33->alarm_threshold = US_TO_SAMPLES(rate,
+                                                dac33->mode1_latency);
+                if (dac33->auto_fifo_config) {
+                        if (period_size <= dac33->alarm_threshold)
+                                /*
+                                 * Configure nSamaple to number of periods,
+                                 * which covers the latency requironment.
+                                 */
+                                dac33->nsample = period_size *
+                                       ((dac33->alarm_threshold / period_size) +
+                                       (dac33->alarm_threshold % period_size ?
+                                       1 : 0));
+                        else
+                                dac33->nsample = period_size;
+                } else {
+                        /* nSample time shall not be shorter than i2c latency */
+                        dac33->nsample_min = dac33->alarm_threshold;
+                        /*
+                         * nSample should not be bigger than alsa buffer minus
+                         * size of one period to avoid overruns
+                         */
+                        dac33->nsample_max = substream->runtime->buffer_size -
+                                                period_size;
+                        nsample_limit = DAC33_BUFFER_SIZE_SAMPLES -
+                                        dac33->alarm_threshold;
+                        if (dac33->nsample_max > nsample_limit)
+                                dac33->nsample_max = nsample_limit;
+
+                        /* Correct the nSample if it is outside of the ranges */
+                        if (dac33->nsample < dac33->nsample_min)
+                                dac33->nsample = dac33->nsample_min;
+                        if (dac33->nsample > dac33->nsample_max)
+                                dac33->nsample = dac33->nsample_max;
+                }
+
                 dac33->mode1_us_burst = SAMPLES_TO_US(dac33->burst_rate,
                                                       dac33->nsample);
                 dac33->t_stamp1 = 0;
                 dac33->t_stamp2 = 0;
                 break;
         case DAC33_FIFO_MODE7:
+                if (dac33->auto_fifo_config) {
+                        dac33->uthr = UTHR_FROM_PERIOD_SIZE(
+                                        period_size,
+                                        rate,
+                                        dac33->burst_rate) + 9;
+                        if (dac33->uthr > MODE7_UTHR)
+                                dac33->uthr = MODE7_UTHR;
+                        if (dac33->uthr < (MODE7_LTHR + 10))
+                                dac33->uthr = (MODE7_LTHR + 10);
+                }
                 dac33->mode7_us_to_lthr =
-                                        SAMPLES_TO_US(substream->runtime->rate,
-                                                MODE7_UTHR - MODE7_LTHR + 1);
+                                SAMPLES_TO_US(substream->runtime->rate,
+                                        dac33->uthr - MODE7_LTHR + 1);
                 dac33->t_stamp1 = 0;
                 break;
         default:
@@ -1104,7 +1167,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
         struct snd_soc_codec *codec = socdev->card->codec;
         struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec);
         unsigned long long t0, t1, t_now;
-        unsigned int time_delta;
+        unsigned int time_delta, uthr;
         int samples_out, samples_in, samples;
         snd_pcm_sframes_t delay = 0;
 
@@ -1182,6 +1245,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
         case DAC33_FIFO_MODE7:
                 spin_lock(&dac33->lock);
                 t0 = dac33->t_stamp1;
+                uthr = dac33->uthr;
                 spin_unlock(&dac33->lock);
                 t_now = ktime_to_us(ktime_get());
 
@@ -1194,7 +1258,7 @@ static snd_pcm_sframes_t dac33_dai_delay(
                          * Either the timestamps are messed or equal. Report
                          * maximum delay
                          */
-                        delay = MODE7_UTHR;
+                        delay = uthr;
                         goto out;
                 }
 
@@ -1208,8 +1272,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
                                         substream->runtime->rate,
                                         time_delta);
 
-                        if (likely(MODE7_UTHR > samples_out))
-                                delay = MODE7_UTHR - samples_out;
+                        if (likely(uthr > samples_out))
+                                delay = uthr - samples_out;
                         else
                                 delay = 0;
                 } else {
@@ -1227,8 +1291,8 @@ static snd_pcm_sframes_t dac33_dai_delay(
                                         time_delta);
                         delay = MODE7_LTHR + samples_in - samples_out;
 
-                        if (unlikely(delay > MODE7_UTHR))
-                                delay = MODE7_UTHR;
+                        if (unlikely(delay > uthr))
+                                delay = uthr;
                 }
                 break;
         default:
@@ -1347,10 +1411,15 @@ static int dac33_soc_probe(struct platform_device *pdev)
 
         snd_soc_add_controls(codec, dac33_snd_controls,
                              ARRAY_SIZE(dac33_snd_controls));
-        /* Only add the nSample controls, if we have valid IRQ number */
-        if (dac33->irq >= 0)
-                snd_soc_add_controls(codec, dac33_nsample_snd_controls,
-                                     ARRAY_SIZE(dac33_nsample_snd_controls));
+        /* Only add the FIFO controls, if we have valid IRQ number */
+        if (dac33->irq >= 0) {
+                snd_soc_add_controls(codec, dac33_mode_snd_controls,
+                                     ARRAY_SIZE(dac33_mode_snd_controls));
+                /* FIFO usage controls only, if autoio config is not selected */
+                if (!dac33->auto_fifo_config)
+                        snd_soc_add_controls(codec, dac33_fifo_snd_controls,
+                                        ARRAY_SIZE(dac33_fifo_snd_controls));
+        }
 
         dac33_add_widgets(codec);
 
@@ -1481,9 +1550,14 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client,
         /* Pre calculate the burst rate */
         dac33->burst_rate = BURST_BASEFREQ_HZ / dac33->burst_bclkdiv / 32;
         dac33->keep_bclk = pdata->keep_bclk;
+        dac33->auto_fifo_config = pdata->auto_fifo_config;
+        dac33->mode1_latency = pdata->mode1_latency;
+        if (!dac33->mode1_latency)
+                dac33->mode1_latency = 10000; /* 10ms */
         dac33->irq = client->irq;
         dac33->nsample = NSAMPLE_MAX;
         dac33->nsample_max = NSAMPLE_MAX;
+        dac33->uthr = MODE7_UTHR;
         /* Disable FIFO use by default */
         dac33->fifo_mode = DAC33_FIFO_BYPASS;
 
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index b4fcdb01fc49..7b618bbff884 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -43,37 +43,37 @@
  */
 static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
         0x00, /* this register not used         */
-        0x91, /* REG_CODEC_MODE         (0x1)   */
-        0xc3, /* REG_OPTION             (0x2)   */
+        0x00, /* REG_CODEC_MODE         (0x1)   */
+        0x00, /* REG_OPTION             (0x2)   */
         0x00, /* REG_UNKNOWN            (0x3)   */
         0x00, /* REG_MICBIAS_CTL        (0x4)   */
-        0x20, /* REG_ANAMICL            (0x5)   */
+        0x00, /* REG_ANAMICL            (0x5)   */
         0x00, /* REG_ANAMICR            (0x6)   */
         0x00, /* REG_AVADC_CTL          (0x7)   */
         0x00, /* REG_ADCMICSEL          (0x8)   */
         0x00, /* REG_DIGMIXING          (0x9)   */
-        0x0c, /* REG_ATXL1PGA           (0xA)   */
-        0x0c, /* REG_ATXR1PGA           (0xB)   */
-        0x00, /* REG_AVTXL2PGA          (0xC)   */
-        0x00, /* REG_AVTXR2PGA          (0xD)   */
+        0x0f, /* REG_ATXL1PGA           (0xA)   */
+        0x0f, /* REG_ATXR1PGA           (0xB)   */
+        0x0f, /* REG_AVTXL2PGA          (0xC)   */
+        0x0f, /* REG_AVTXR2PGA          (0xD)   */
         0x00, /* REG_AUDIO_IF           (0xE)   */
         0x00, /* REG_VOICE_IF           (0xF)   */
-        0x00, /* REG_ARXR1PGA           (0x10)  */
-        0x00, /* REG_ARXL1PGA           (0x11)  */
-        0x6c, /* REG_ARXR2PGA           (0x12)  */
-        0x6c, /* REG_ARXL2PGA           (0x13)  */
-        0x00, /* REG_VRXPGA             (0x14)  */
+        0x3f, /* REG_ARXR1PGA           (0x10)  */
+        0x3f, /* REG_ARXL1PGA           (0x11)  */
+        0x3f, /* REG_ARXR2PGA           (0x12)  */
+        0x3f, /* REG_ARXL2PGA           (0x13)  */
+        0x25, /* REG_VRXPGA             (0x14)  */
         0x00, /* REG_VSTPGA             (0x15)  */
         0x00, /* REG_VRX2ARXPGA         (0x16)  */
         0x00, /* REG_AVDAC_CTL          (0x17)  */
         0x00, /* REG_ARX2VTXPGA         (0x18)  */
-        0x00, /* REG_ARXL1_APGA_CTL     (0x19)  */
-        0x00, /* REG_ARXR1_APGA_CTL     (0x1A)  */
-        0x4a, /* REG_ARXL2_APGA_CTL     (0x1B)  */
-        0x4a, /* REG_ARXR2_APGA_CTL     (0x1C)  */
+        0x32, /* REG_ARXL1_APGA_CTL     (0x19)  */
+        0x32, /* REG_ARXR1_APGA_CTL     (0x1A)  */
+        0x32, /* REG_ARXL2_APGA_CTL     (0x1B)  */
+        0x32, /* REG_ARXR2_APGA_CTL     (0x1C)  */
         0x00, /* REG_ATX2ARXPGA         (0x1D)  */
         0x00, /* REG_BT_IF              (0x1E)  */
-        0x00, /* REG_BTPGA              (0x1F)  */
+        0x55, /* REG_BTPGA              (0x1F)  */
         0x00, /* REG_BTSTPGA            (0x20)  */
         0x00, /* REG_EAR_CTL            (0x21)  */
         0x00, /* REG_HS_SEL             (0x22)  */
@@ -85,32 +85,32 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
         0x00, /* REG_PRECKR_CTL         (0x28)  */
         0x00, /* REG_HFL_CTL            (0x29)  */
         0x00, /* REG_HFR_CTL            (0x2A)  */
-        0x00, /* REG_ALC_CTL            (0x2B)  */
+        0x05, /* REG_ALC_CTL            (0x2B)  */
         0x00, /* REG_ALC_SET1           (0x2C)  */
         0x00, /* REG_ALC_SET2           (0x2D)  */
         0x00, /* REG_BOOST_CTL          (0x2E)  */
         0x00, /* REG_SOFTVOL_CTL        (0x2F)  */
-        0x00, /* REG_DTMF_FREQSEL       (0x30)  */
+        0x13, /* REG_DTMF_FREQSEL       (0x30)  */
         0x00, /* REG_DTMF_TONEXT1H      (0x31)  */
         0x00, /* REG_DTMF_TONEXT1L      (0x32)  */
         0x00, /* REG_DTMF_TONEXT2H      (0x33)  */
         0x00, /* REG_DTMF_TONEXT2L      (0x34)  */
-        0x00, /* REG_DTMF_TONOFF        (0x35)  */
-        0x00, /* REG_DTMF_WANONOFF      (0x36)  */
+        0x79, /* REG_DTMF_TONOFF        (0x35)  */
+        0x11, /* REG_DTMF_WANONOFF      (0x36)  */
         0x00, /* REG_I2S_RX_SCRAMBLE_H  (0x37)  */
         0x00, /* REG_I2S_RX_SCRAMBLE_M  (0x38)  */
         0x00, /* REG_I2S_RX_SCRAMBLE_L  (0x39)  */
         0x06, /* REG_APLL_CTL           (0x3A)  */
         0x00, /* REG_DTMF_CTL           (0x3B)  */
-        0x00, /* REG_DTMF_PGA_CTL2      (0x3C)  */
-        0x00, /* REG_DTMF_PGA_CTL1      (0x3D)  */
+        0x44, /* REG_DTMF_PGA_CTL2      (0x3C)  */
+        0x69, /* REG_DTMF_PGA_CTL1      (0x3D)  */
         0x00, /* REG_MISC_SET_1         (0x3E)  */
         0x00, /* REG_PCMBTMUX           (0x3F)  */
         0x00, /* not used               (0x40)  */
         0x00, /* not used               (0x41)  */
         0x00, /* not used               (0x42)  */
         0x00, /* REG_RX_PATH_SEL        (0x43)  */
-        0x00, /* REG_VDL_APGA_CTL       (0x44)  */
+        0x32, /* REG_VDL_APGA_CTL       (0x44)  */
         0x00, /* REG_VIBRA_CTL          (0x45)  */
         0x00, /* REG_VIBRA_SET          (0x46)  */
         0x00, /* REG_VIBRA_PWM_SET      (0x47)  */
@@ -143,6 +143,9 @@ struct twl4030_priv {
         u8 earpiece_enabled;
         u8 predrivel_enabled, predriver_enabled;
         u8 carkitl_enabled, carkitr_enabled;
+
+        /* Delay needed after enabling the digimic interface */
+        unsigned int digimic_delay;
 };
 
 /*
@@ -244,58 +247,95 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
         udelay(10);
 }
 
-static void twl4030_init_chip(struct snd_soc_codec *codec)
+static inline void twl4030_check_defaults(struct snd_soc_codec *codec)
 {
-        u8 *cache = codec->reg_cache;
-        int i;
+        int i, difference = 0;
+        u8 val;
+
+        dev_dbg(codec->dev, "Checking TWL audio default configuration\n");
+        for (i = 1; i <= TWL4030_REG_MISC_SET_2; i++) {
+                twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &val, i);
+                if (val != twl4030_reg[i]) {
+                        difference++;
+                        dev_dbg(codec->dev,
+                                 "Reg 0x%02x: chip: 0x%02x driver: 0x%02x\n",
+                                 i, val, twl4030_reg[i]);
+                }
+        }
+        dev_dbg(codec->dev, "Found %d non maching registers. %s\n",
+                 difference, difference ? "Not OK" : "OK");
+}
 
-        /* clear CODECPDZ prior to setting register defaults */
-        twl4030_codec_enable(codec, 0);
+static inline void twl4030_reset_registers(struct snd_soc_codec *codec)
+{
+        int i;
 
         /* set all audio section registers to reasonable defaults */
         for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
                 if (i != TWL4030_REG_APLL_CTL)
-                        twl4030_write(codec, i, cache[i]);
+                        twl4030_write(codec, i, twl4030_reg[i]);
 
 }
 
-static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
+static void twl4030_init_chip(struct platform_device *pdev)
 {
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct twl4030_setup_data *setup = socdev->codec_data;
+        struct snd_soc_codec *codec = socdev->card->codec;
         struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-        int status = -1;
+        u8 reg, byte;
+        int i = 0;
 
-        if (enable) {
-                twl4030->apll_enabled++;
-                if (twl4030->apll_enabled == 1)
-                        status = twl4030_codec_enable_resource(
-                                                        TWL4030_CODEC_RES_APLL);
-        } else {
-                twl4030->apll_enabled--;
-                if (!twl4030->apll_enabled)
-                        status = twl4030_codec_disable_resource(
-                                                        TWL4030_CODEC_RES_APLL);
-        }
+        /* Check defaults, if instructed before anything else */
+        if (setup && setup->check_defaults)
+                twl4030_check_defaults(codec);
 
-        if (status >= 0)
-                twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
-}
+        /* Reset registers, if no setup data or if instructed to do so */
+        if (!setup || (setup && setup->reset_registers))
+                twl4030_reset_registers(codec);
 
-static void twl4030_power_up(struct snd_soc_codec *codec)
-{
-        struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
-        u8 anamicl, regmisc1, byte;
-        int i = 0;
+        /* Refresh APLL_CTL register from HW */
+        twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+                            TWL4030_REG_APLL_CTL);
+        twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, byte);
+
+        /* anti-pop when changing analog gain */
+        reg = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+        twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+                reg | TWL4030_SMOOTH_ANAVOL_EN);
 
-        if (twl4030->codec_powered)
+        twl4030_write(codec, TWL4030_REG_OPTION,
+                TWL4030_ATXL1_EN | TWL4030_ATXR1_EN |
+                TWL4030_ARXL2_EN | TWL4030_ARXR2_EN);
+
+        /* REG_ARXR2_APGA_CTL reset according to the TRM: 0dB, DA_EN */
+        twl4030_write(codec, TWL4030_REG_ARXR2_APGA_CTL, 0x32);
+
+        /* Machine dependent setup */
+        if (!setup)
                 return;
 
-        /* set CODECPDZ to turn on codec */
-        twl4030_codec_enable(codec, 1);
+        twl4030->digimic_delay = setup->digimic_delay;
+
+        /* Configuration for headset ramp delay from setup data */
+        if (setup->sysclk != twl4030->sysclk)
+                dev_warn(codec->dev,
+                                "Mismatch in APLL mclk: %u (configured: %u)\n",
+                                setup->sysclk, twl4030->sysclk);
+
+        reg = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+        reg &= ~TWL4030_RAMP_DELAY;
+        reg |= (setup->ramp_delay_value << 2);
+        twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, reg);
 
         /* initiate offset cancellation */
-        anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+        twl4030_codec_enable(codec, 1);
+
+        reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+        reg &= ~TWL4030_OFFSET_CNCL_SEL;
+        reg |= setup->offset_cncl_path;
         twl4030_write(codec, TWL4030_REG_ANAMICL,
-                anamicl | TWL4030_CNCL_OFFSET_START);
+                reg | TWL4030_CNCL_OFFSET_START);
 
         /* wait for offset cancellation to complete */
         do {
@@ -310,23 +350,28 @@ static void twl4030_power_up(struct snd_soc_codec *codec)
         /* Make sure that the reg_cache has the same value as the HW */
         twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
 
-        /* anti-pop when changing analog gain */
-        regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
-        twl4030_write(codec, TWL4030_REG_MISC_SET_1,
-                regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
-
-        /* toggle CODECPDZ as per TRM */
         twl4030_codec_enable(codec, 0);
-        twl4030_codec_enable(codec, 1);
 }
 
-/*
- * Unconditional power down
- */
-static void twl4030_power_down(struct snd_soc_codec *codec)
+static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 {
-        /* power down */
-        twl4030_codec_enable(codec, 0);
+        struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
+        int status = -1;
+
+        if (enable) {
+                twl4030->apll_enabled++;
+                if (twl4030->apll_enabled == 1)
+                        status = twl4030_codec_enable_resource(
+                                                        TWL4030_CODEC_RES_APLL);
+        } else {
+                twl4030->apll_enabled--;
+                if (!twl4030->apll_enabled)
+                        status = twl4030_codec_disable_resource(
+                                                        TWL4030_CODEC_RES_APLL);
+        }
+
+        if (status >= 0)
+                twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
 }
 
 /* Earpiece */
@@ -500,10 +545,11 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
 static const struct snd_kcontrol_new twl4030_dapm_abypassv_control =
         SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0);
 
-/* Digital bypass gain, 0 mutes the bypass */
+/* Digital bypass gain, mute instead of -30dB */
 static const unsigned int twl4030_dapm_dbypass_tlv[] = {
-        TLV_DB_RANGE_HEAD(2),
-        0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1),
+        TLV_DB_RANGE_HEAD(3),
+        0, 1, TLV_DB_SCALE_ITEM(-3000, 600, 1),
+        2, 3, TLV_DB_SCALE_ITEM(-2400, 0, 0),
         4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0),
 };
 
@@ -531,36 +577,6 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
                         TWL4030_REG_VSTPGA, 0, 0x29, 0,
                         twl4030_dapm_dbypassv_tlv);
 
-static int micpath_event(struct snd_soc_dapm_widget *w,
-        struct snd_kcontrol *kcontrol, int event)
-{
-        struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
-        unsigned char adcmicsel, micbias_ctl;
-
-        adcmicsel = twl4030_read_reg_cache(w->codec, TWL4030_REG_ADCMICSEL);
-        micbias_ctl = twl4030_read_reg_cache(w->codec, TWL4030_REG_MICBIAS_CTL);
-        /* Prepare the bits for the given TX path:
-         * shift_l == 0: TX1 microphone path
-         * shift_l == 2: TX2 microphone path */
-        if (e->shift_l) {
-                /* TX2 microphone path */
-                if (adcmicsel & TWL4030_TX2IN_SEL)
-                        micbias_ctl |= TWL4030_MICBIAS2_CTL; /* digimic */
-                else
-                        micbias_ctl &= ~TWL4030_MICBIAS2_CTL;
-        } else {
-                /* TX1 microphone path */
-                if (adcmicsel & TWL4030_TX1IN_SEL)
-                        micbias_ctl |= TWL4030_MICBIAS1_CTL; /* digimic */
-                else
-                        micbias_ctl &= ~TWL4030_MICBIAS1_CTL;
-        }
-
-        twl4030_write(w->codec, TWL4030_REG_MICBIAS_CTL, micbias_ctl);
-
-        return 0;
-}
-
 /*
  * Output PGA builder:
  * Handle the muting and unmuting of the given output (turning off the
@@ -814,6 +830,16 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w,
         return 0;
 }
 
+static int digimic_event(struct snd_soc_dapm_widget *w,
+                struct snd_kcontrol *kcontrol, int event)
+{
+        struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec);
+
+        if (twl4030->digimic_delay)
+                mdelay(twl4030->digimic_delay);
+        return 0;
+}
+
 /*
  * Some of the gain controls in TWL (mostly those which are associated with
  * the outputs) are implemented in an interesting way:
@@ -1374,14 +1400,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
         /* Analog/Digital mic path selection.
            TX1 Left/Right: either analog Left/Right or Digimic0
            TX2 Left/Right: either analog Left/Right or Digimic1 */
-        SND_SOC_DAPM_MUX_E("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_micpathtx1_control, micpath_event,
-                SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
-                SND_SOC_DAPM_POST_REG),
-        SND_SOC_DAPM_MUX_E("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_micpathtx2_control, micpath_event,
-                SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
-                SND_SOC_DAPM_POST_REG),
+        SND_SOC_DAPM_MUX("TX1 Capture Route", SND_SOC_NOPM, 0, 0,
+                &twl4030_dapm_micpathtx1_control),
+        SND_SOC_DAPM_MUX("TX2 Capture Route", SND_SOC_NOPM, 0, 0,
+                &twl4030_dapm_micpathtx2_control),
 
         /* Analog input mixers for the capture amplifiers */
         SND_SOC_DAPM_MIXER("Analog Left",
@@ -1398,10 +1420,17 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
         SND_SOC_DAPM_PGA("ADC Physical Right",
                 TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0),
 
-        SND_SOC_DAPM_PGA("Digimic0 Enable",
-                TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0),
-        SND_SOC_DAPM_PGA("Digimic1 Enable",
-                TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0),
+        SND_SOC_DAPM_PGA_E("Digimic0 Enable",
+                TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0,
+                digimic_event, SND_SOC_DAPM_POST_PMU),
+        SND_SOC_DAPM_PGA_E("Digimic1 Enable",
+                TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0,
+                digimic_event, SND_SOC_DAPM_POST_PMU),
+
+        SND_SOC_DAPM_SUPPLY("micbias1 select", TWL4030_REG_MICBIAS_CTL, 5, 0,
+                            NULL, 0),
+        SND_SOC_DAPM_SUPPLY("micbias2 select", TWL4030_REG_MICBIAS_CTL, 6, 0,
+                            NULL, 0),
 
         SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0),
         SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0),
@@ -1419,8 +1448,11 @@ static const struct snd_soc_dapm_route intercon[] = {
         /* Supply for the digital part (APLL) */
         {"Digital Voice Playback Mixer", NULL, "APLL Enable"},
 
-        {"Digital R1 Playback Mixer", NULL, "AIF Enable"},
-        {"Digital L1 Playback Mixer", NULL, "AIF Enable"},
+        {"DAC Left1", NULL, "AIF Enable"},
+        {"DAC Right1", NULL, "AIF Enable"},
+        {"DAC Left2", NULL, "AIF Enable"},
+        {"DAC Right1", NULL, "AIF Enable"},
+
         {"Digital R2 Playback Mixer", NULL, "AIF Enable"},
         {"Digital L2 Playback Mixer", NULL, "AIF Enable"},
 
@@ -1491,10 +1523,10 @@ static const struct snd_soc_dapm_route intercon[] = {
 
         /* outputs */
         /* Must be always connected (for AIF and APLL) */
-        {"Virtual HiFi OUT", NULL, "Digital L1 Playback Mixer"},
-        {"Virtual HiFi OUT", NULL, "Digital R1 Playback Mixer"},
-        {"Virtual HiFi OUT", NULL, "Digital L2 Playback Mixer"},
-        {"Virtual HiFi OUT", NULL, "Digital R2 Playback Mixer"},
+        {"Virtual HiFi OUT", NULL, "DAC Left1"},
+        {"Virtual HiFi OUT", NULL, "DAC Right1"},
+        {"Virtual HiFi OUT", NULL, "DAC Left2"},
+        {"Virtual HiFi OUT", NULL, "DAC Right2"},
         /* Must be always connected (for APLL) */
         {"Virtual Voice OUT", NULL, "Digital Voice Playback Mixer"},
         /* Physical outputs */
@@ -1531,6 +1563,9 @@ static const struct snd_soc_dapm_route intercon[] = {
         {"Digimic0 Enable", NULL, "DIGIMIC0"},
         {"Digimic1 Enable", NULL, "DIGIMIC1"},
 
+        {"DIGIMIC0", NULL, "micbias1 select"},
+        {"DIGIMIC1", NULL, "micbias2 select"},
+
         /* TX1 Left capture path */
         {"TX1 Capture Route", "Analog", "ADC Physical Left"},
         {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
@@ -1605,10 +1640,10 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
                 break;
         case SND_SOC_BIAS_STANDBY:
                 if (codec->bias_level == SND_SOC_BIAS_OFF)
-                        twl4030_power_up(codec);
+                        twl4030_codec_enable(codec, 1);
                 break;
         case SND_SOC_BIAS_OFF:
-                twl4030_power_down(codec);
+                twl4030_codec_enable(codec, 0);
                 break;
         }
         codec->bias_level = level;
@@ -1794,13 +1829,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
                 return -EINVAL;
         }
 
-        if (mode != old_mode) {
-                /* change rate and set CODECPDZ */
-                twl4030_codec_enable(codec, 0);
-                twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
-                twl4030_codec_enable(codec, 1);
-        }
-
         /* sample size */
         old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
         format = old_format;
@@ -1818,16 +1846,20 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
                 return -EINVAL;
         }
 
-        if (format != old_format) {
-
-                /* clear CODECPDZ before changing format (codec requirement) */
-                twl4030_codec_enable(codec, 0);
-
-                /* change format */
-                twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
-
-                /* set CODECPDZ afterwards */
-                twl4030_codec_enable(codec, 1);
+        if (format != old_format || mode != old_mode) {
+                if (twl4030->codec_powered) {
+                        /*
+                         * If the codec is powered, than we need to toggle the
+                         * codec power.
+                         */
+                        twl4030_codec_enable(codec, 0);
+                        twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+                        twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+                        twl4030_codec_enable(codec, 1);
+                } else {
+                        twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+                        twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+                }
         }
 
         /* Store the important parameters for the DAI configuration and set
@@ -1877,6 +1909,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
                              unsigned int fmt)
 {
         struct snd_soc_codec *codec = codec_dai->codec;
+        struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
         u8 old_format, format;
 
         /* get format */
@@ -1911,15 +1944,17 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
         }
 
         if (format != old_format) {
-
-                /* clear CODECPDZ before changing format (codec requirement) */
-                twl4030_codec_enable(codec, 0);
-
-                /* change format */
-                twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
-
-                /* set CODECPDZ afterwards */
-                twl4030_codec_enable(codec, 1);
+                if (twl4030->codec_powered) {
+                        /*
+                         * If the codec is powered, than we need to toggle the
+                         * codec power.
+                         */
+                        twl4030_codec_enable(codec, 0);
+                        twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+                        twl4030_codec_enable(codec, 1);
+                } else {
+                        twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
+                }
         }
 
         return 0;
@@ -2011,6 +2046,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_device *socdev = rtd->socdev;
         struct snd_soc_codec *codec = socdev->card->codec;
+        struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
         u8 old_mode, mode;
 
         /* Enable voice digital filters */
@@ -2035,10 +2071,17 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
         }
 
         if (mode != old_mode) {
-                /* change rate and set CODECPDZ */
-                twl4030_codec_enable(codec, 0);
-                twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
-                twl4030_codec_enable(codec, 1);
+                if (twl4030->codec_powered) {
+                        /*
+                         * If the codec is powered, than we need to toggle the
+                         * codec power.
+                         */
+                        twl4030_codec_enable(codec, 0);
+                        twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+                        twl4030_codec_enable(codec, 1);
+                } else {
+                        twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+                }
         }
 
         return 0;
@@ -2068,6 +2111,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
                 unsigned int fmt)
 {
         struct snd_soc_codec *codec = codec_dai->codec;
+        struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
         u8 old_format, format;
 
         /* get format */
@@ -2099,10 +2143,17 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
         }
 
         if (format != old_format) {
-                /* change format and set CODECPDZ */
-                twl4030_codec_enable(codec, 0);
-                twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
-                twl4030_codec_enable(codec, 1);
+                if (twl4030->codec_powered) {
+                        /*
+                         * If the codec is powered, than we need to toggle the
+                         * codec power.
+                         */
+                        twl4030_codec_enable(codec, 0);
+                        twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+                        twl4030_codec_enable(codec, 1);
+                } else {
+                        twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+                }
         }
 
         return 0;
@@ -2202,31 +2253,15 @@ static struct snd_soc_codec *twl4030_codec;
 static int twl4030_soc_probe(struct platform_device *pdev)
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-        struct twl4030_setup_data *setup = socdev->codec_data;
         struct snd_soc_codec *codec;
-        struct twl4030_priv *twl4030;
         int ret;
 
         BUG_ON(!twl4030_codec);
 
         codec = twl4030_codec;
-        twl4030 = snd_soc_codec_get_drvdata(codec);
         socdev->card->codec = codec;
 
-        /* Configuration for headset ramp delay from setup data */
-        if (setup) {
-                unsigned char hs_pop;
-
-                if (setup->sysclk != twl4030->sysclk)
-                        dev_warn(&pdev->dev,
-                                 "Mismatch in APLL mclk: %u (configured: %u)\n",
-                                 setup->sysclk, twl4030->sysclk);
-
-                hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
-                hs_pop &= ~TWL4030_RAMP_DELAY;
-                hs_pop |= (setup->ramp_delay_value << 2);
-                twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-        }
+        twl4030_init_chip(pdev);
 
         /* register pcms */
         ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -2247,6 +2282,8 @@ static int twl4030_soc_remove(struct platform_device *pdev)
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_codec *codec = socdev->card->codec;
 
+        /* Reset registers to their chip default before leaving */
+        twl4030_reset_registers(codec);
         twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
         snd_soc_free_pcms(socdev);
         snd_soc_dapm_free(socdev);
@@ -2287,6 +2324,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
         codec->read = twl4030_read_reg_cache;
         codec->write = twl4030_write;
         codec->set_bias_level = twl4030_set_bias_level;
+        codec->idle_bias_off = 1;
         codec->dai = twl4030_dai;
         codec->num_dai = ARRAY_SIZE(twl4030_dai);
         codec->reg_cache_size = sizeof(twl4030_reg);
@@ -2302,9 +2340,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 
         /* Set the defaults, and power up the codec */
         twl4030->sysclk = twl4030_codec_get_mclk() / 1000;
-        twl4030_init_chip(codec);
         codec->bias_level = SND_SOC_BIAS_OFF;
-        twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
         ret = snd_soc_register_codec(codec);
         if (ret != 0) {
@@ -2322,7 +2358,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
         return 0;
 
 error_codec:
-        twl4030_power_down(codec);
+        twl4030_codec_enable(codec, 0);
         kfree(codec->reg_cache);
 error_cache:
         kfree(twl4030);
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index f206d242ca31..6c57430f6e24 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -41,7 +41,11 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030;
 
 struct twl4030_setup_data {
         unsigned int ramp_delay_value;
+        unsigned int digimic_delay; /* in ms */
         unsigned int sysclk;
+        unsigned int offset_cncl_path;
+        unsigned int check_defaults:1;
+        unsigned int reset_registers:1;
         unsigned int hs_extmute:1;
         void (*set_hs_extmute)(int mute);
 };
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index af36346ff336..64a807f1a8a1 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -360,6 +360,13 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf)
         return 0;
 }
 
+static int twl6040_hs_dac_event(struct snd_soc_dapm_widget *w,
+                        struct snd_kcontrol *kcontrol, int event)
+{
+        msleep(1);
+        return 0;
+}
+
 static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
                         struct snd_kcontrol *kcontrol, int event)
 {
@@ -371,6 +378,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w,
         else
                 priv->non_lp--;
 
+        msleep(1);
+
         return 0;
 }
 
@@ -471,20 +480,6 @@ static const struct snd_kcontrol_new hfdacl_switch_controls =
 static const struct snd_kcontrol_new hfdacr_switch_controls =
         SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0);
 
-/* Headset driver switches */
-static const struct snd_kcontrol_new hsl_driver_switch_controls =
-        SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0);
-
-static const struct snd_kcontrol_new hsr_driver_switch_controls =
-        SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0);
-
-/* Handsfree driver switches */
-static const struct snd_kcontrol_new hfl_driver_switch_controls =
-        SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0);
-
-static const struct snd_kcontrol_new hfr_driver_switch_controls =
-        SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0);
-
 static const struct snd_kcontrol_new ep_driver_switch_controls =
         SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0);
 
@@ -548,10 +543,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
                         TWL6040_REG_DMICBCTL, 4, 0),
 
         /* DACs */
-        SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback",
-                        TWL6040_REG_HSLCTL, 0, 0),
-        SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback",
-                        TWL6040_REG_HSRCTL, 0, 0),
+        SND_SOC_DAPM_DAC_E("HSDAC Left", "Headset Playback",
+                        TWL6040_REG_HSLCTL, 0, 0,
+                        twl6040_hs_dac_event,
+                        SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+        SND_SOC_DAPM_DAC_E("HSDAC Right", "Headset Playback",
+                        TWL6040_REG_HSRCTL, 0, 0,
+                        twl6040_hs_dac_event,
+                        SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
         SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback",
                         TWL6040_REG_HFLCTL, 0, 0,
                         twl6040_power_mode_event,
@@ -571,18 +570,19 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
         SND_SOC_DAPM_SWITCH("HFDAC Right Playback",
                         SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls),
 
-        SND_SOC_DAPM_SWITCH("Headset Left Driver",
-                        SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls),
-        SND_SOC_DAPM_SWITCH("Headset Right Driver",
-                        SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls),
-        SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver",
-                        SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls,
+        /* Analog playback drivers */
+        SND_SOC_DAPM_PGA_E("Handsfree Left Driver",
+                        TWL6040_REG_HFLCTL, 4, 0, NULL, 0,
                         twl6040_power_mode_event,
                         SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
-        SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver",
-                        SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls,
+        SND_SOC_DAPM_PGA_E("Handsfree Right Driver",
+                        TWL6040_REG_HFRCTL, 4, 0, NULL, 0,
                         twl6040_power_mode_event,
                         SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+        SND_SOC_DAPM_PGA("Headset Left Driver",
+                        TWL6040_REG_HSLCTL, 2, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("Headset Right Driver",
+                        TWL6040_REG_HSRCTL, 2, 0, NULL, 0),
         SND_SOC_DAPM_SWITCH_E("Earphone Driver",
                         SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls,
                         twl6040_power_mode_event,
@@ -616,8 +616,8 @@ static const struct snd_soc_dapm_route intercon[] = {
         {"HSDAC Left Playback", "Switch", "HSDAC Left"},
         {"HSDAC Right Playback", "Switch", "HSDAC Right"},
 
-        {"Headset Left Driver", "Switch", "HSDAC Left Playback"},
-        {"Headset Right Driver", "Switch", "HSDAC Right Playback"},
+        {"Headset Left Driver", NULL, "HSDAC Left Playback"},
+        {"Headset Right Driver", NULL, "HSDAC Right Playback"},
 
         {"HSOL", NULL, "Headset Left Driver"},
         {"HSOR", NULL, "Headset Right Driver"},
@@ -928,7 +928,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai,
                 case 19200000:
                         /* mclk input, pll disabled */
                         hppllctl |= TWL6040_MCLK_19200KHZ |
-                                    TWL6040_HPLLSQRBP |
+                                    TWL6040_HPLLSQRENA |
                                     TWL6040_HPLLBP;
                         break;
                 case 26000000:
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 28aac53c97bb..f3b4c1d6a82d 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -28,19 +28,6 @@
 #include "uda134x.h"
 
 
-#define POWER_OFF_ON_STANDBY 1
-/*
-  ALSA SOC usually puts the device in standby mode when it's not used
-  for sometime. If you define POWER_OFF_ON_STANDBY the driver will
-  turn off the ADC/DAC when this callback is invoked and turn it back
-  on when needed. Unfortunately this will result in a very light bump
-  (it can be audible only with good earphones). If this bothers you
-  just comment this line, you will have slightly higher power
-  consumption . Please note that sending the L3 command for ADC is
-  enough to make the bump, so it doesn't make difference if you
-  completely take off power from the codec.
- */
-
 #define UDA134X_RATES SNDRV_PCM_RATE_8000_48000
 #define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
                 SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE)
@@ -58,7 +45,7 @@ static const char uda134x_reg[UDA134X_REGS_NUM] = {
         /* Extended address registers */
         0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00,
         /* Status, data regs */
-        0x00, 0x83, 0x00, 0x40, 0x80, 0x00,
+        0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00,
 };
 
 /*
@@ -117,6 +104,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
         case UDA134X_DATA000:
         case UDA134X_DATA001:
         case UDA134X_DATA010:
+        case UDA134X_DATA011:
                 addr = UDA134X_DATA0_ADDR;
                 break;
         case UDA134X_DATA1:
@@ -353,8 +341,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
         switch (level) {
         case SND_SOC_BIAS_ON:
                 /* ADC, DAC on */
-                reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
-                uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+                switch (pd->model) {
+                case UDA134X_UDA1340:
+                case UDA134X_UDA1344:
+                case UDA134X_UDA1345:
+                        reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
+                        uda134x_write(codec, UDA134X_DATA011, reg | 0x03);
+                        break;
+                case UDA134X_UDA1341:
+                        reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+                        uda134x_write(codec, UDA134X_STATUS1, reg | 0x03);
+                        break;
+                default:
+                        printk(KERN_ERR "UDA134X SoC codec: "
+                               "unsupported model %d\n", pd->model);
+                        return -EINVAL;
+                }
                 break;
         case SND_SOC_BIAS_PREPARE:
                 /* power on */
@@ -367,8 +369,22 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec,
                 break;
         case SND_SOC_BIAS_STANDBY:
                 /* ADC, DAC power off */
-                reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
-                uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+                switch (pd->model) {
+                case UDA134X_UDA1340:
+                case UDA134X_UDA1344:
+                case UDA134X_UDA1345:
+                        reg = uda134x_read_reg_cache(codec, UDA134X_DATA011);
+                        uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03));
+                        break;
+                case UDA134X_UDA1341:
+                        reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1);
+                        uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03));
+                        break;
+                default:
+                        printk(KERN_ERR "UDA134X SoC codec: "
+                               "unsupported model %d\n", pd->model);
+                        return -EINVAL;
+                }
                 break;
         case SND_SOC_BIAS_OFF:
                 /* power off */
@@ -531,9 +547,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
         codec->num_dai = 1;
         codec->read = uda134x_read_reg_cache;
         codec->write = uda134x_write;
-#ifdef POWER_OFF_ON_STANDBY
-        codec->set_bias_level = uda134x_set_bias_level;
-#endif
+
         INIT_LIST_HEAD(&codec->dapm_widgets);
         INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -544,6 +558,14 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 
         uda134x_reset(codec);
 
+        if (pd->is_powered_on_standby) {
+                codec->set_bias_level = NULL;
+                uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
+        } else {
+                codec->set_bias_level = uda134x_set_bias_level;
+                uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+        }
+
         /* register pcms */
         ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
         if (ret < 0) {
diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h
index 94f440490b31..205f03b3eaf8 100644
--- a/sound/soc/codecs/uda134x.h
+++ b/sound/soc/codecs/uda134x.h
@@ -23,9 +23,10 @@
 #define UDA134X_DATA000 10
 #define UDA134X_DATA001 11
 #define UDA134X_DATA010 12
-#define UDA134X_DATA1   13
+#define UDA134X_DATA011 13
+#define UDA134X_DATA1   14
 
-#define UDA134X_REGS_NUM 14
+#define UDA134X_REGS_NUM 15
 
 #define STATUS0_DAIFMT_MASK (~(7<<1))
 #define STATUS0_SYSCLK_MASK (~(3<<4))
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 002e289d1255..4bcd168794e1 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -795,6 +795,8 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
 
         dev_set_drvdata(&i2c->dev, wm2000);
         wm2000->anc_eng_ena = 1;
+        wm2000->anc_active = 1;
+        wm2000->spk_ena = 1;
         wm2000->i2c = i2c;
 
         wm2000_reset(wm2000);
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 37242a7d3077..0ad039b4adf5 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -482,7 +482,8 @@ static int wm8523_register(struct wm8523_priv *wm8523,
 
         if (wm8523_codec) {
                 dev_err(codec->dev, "Another WM8523 is registered\n");
-                return -EINVAL;
+                ret = -EINVAL;
+                goto err;
         }
 
         mutex_init(&codec->mutex);
@@ -570,18 +571,19 @@ static int wm8523_register(struct wm8523_priv *wm8523,
         ret = snd_soc_register_codec(codec);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-                return ret;
+                goto err_enable;
         }
 
         ret = snd_soc_register_dai(&wm8523_dai);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-                snd_soc_unregister_codec(codec);
-                return ret;
+                goto err_codec;
         }
 
         return 0;
 
+err_codec:
+        snd_soc_unregister_codec(codec);
 err_enable:
         regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
 err_get:
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index effb14eee7d4..e2dba07f0260 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -439,7 +439,8 @@ static int wm8711_register(struct wm8711_priv *wm8711,
 
         if (wm8711_codec) {
                 dev_err(codec->dev, "Another WM8711 is registered\n");
-                return -EINVAL;
+                ret = -EINVAL;
+                goto err;
         }
 
         mutex_init(&codec->mutex);
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
new file mode 100644
index 000000000000..b9ea8904ad4b
--- /dev/null
+++ b/sound/soc/codecs/wm8741.c
@@ -0,0 +1,579 @@
+/*
+ * wm8741.c  --  WM8741 ALSA SoC Audio driver
+ *
+ * Copyright 2010 Wolfson Microelectronics plc
+ *
+ * Author: Ian Lartey <ian@opensource.wolfsonmicro.com>
+ *
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/slab.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 "wm8741.h"
+
+static struct snd_soc_codec *wm8741_codec;
+struct snd_soc_codec_device soc_codec_dev_wm8741;
+
+#define WM8741_NUM_SUPPLIES 2
+static const char *wm8741_supply_names[WM8741_NUM_SUPPLIES] = {
+        "AVDD",
+        "DVDD",
+};
+
+#define WM8741_NUM_RATES 4
+
+/* codec private data */
+struct wm8741_priv {
+        struct snd_soc_codec codec;
+        u16 reg_cache[WM8741_REGISTER_COUNT];
+        struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES];
+        unsigned int sysclk;
+        unsigned int rate_constraint_list[WM8741_NUM_RATES];
+        struct snd_pcm_hw_constraint_list rate_constraint;
+};
+
+static const u16 wm8741_reg_defaults[WM8741_REGISTER_COUNT] = {
+        0x0000,     /* R0  - DACLLSB Attenuation */
+        0x0000,     /* R1  - DACLMSB Attenuation */
+        0x0000,     /* R2  - DACRLSB Attenuation */
+        0x0000,     /* R3  - DACRMSB Attenuation */
+        0x0000,     /* R4  - Volume Control */
+        0x000A,     /* R5  - Format Control */
+        0x0000,     /* R6  - Filter Control */
+        0x0000,     /* R7  - Mode Control 1 */
+        0x0002,     /* R8  - Mode Control 2 */
+        0x0000,     /* R9  - Reset */
+        0x0002,     /* R32 - ADDITONAL_CONTROL_1 */
+};
+
+
+static int wm8741_reset(struct snd_soc_codec *codec)
+{
+        return snd_soc_write(codec, WM8741_RESET, 0);
+}
+
+static const DECLARE_TLV_DB_SCALE(dac_tlv_fine, -12700, 13, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 400, 0);
+
+static const struct snd_kcontrol_new wm8741_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Fine Playback Volume", WM8741_DACLLSB_ATTENUATION,
+                 WM8741_DACRLSB_ATTENUATION, 1, 255, 1, dac_tlv_fine),
+SOC_DOUBLE_R_TLV("Playback Volume", WM8741_DACLMSB_ATTENUATION,
+                 WM8741_DACRMSB_ATTENUATION, 0, 511, 1, dac_tlv),
+};
+
+static const struct snd_soc_dapm_widget wm8741_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DACL", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_DAC("DACR", "Playback", SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_OUTPUT("VOUTLP"),
+SND_SOC_DAPM_OUTPUT("VOUTLN"),
+SND_SOC_DAPM_OUTPUT("VOUTRP"),
+SND_SOC_DAPM_OUTPUT("VOUTRN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+        { "VOUTLP", NULL, "DACL" },
+        { "VOUTLN", NULL, "DACL" },
+        { "VOUTRP", NULL, "DACR" },
+        { "VOUTRN", NULL, "DACR" },
+};
+
+static int wm8741_add_widgets(struct snd_soc_codec *codec)
+{
+        snd_soc_dapm_new_controls(codec, wm8741_dapm_widgets,
+                                  ARRAY_SIZE(wm8741_dapm_widgets));
+
+        snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+        return 0;
+}
+
+static struct {
+        int value;
+        int ratio;
+} lrclk_ratios[WM8741_NUM_RATES] = {
+        { 1, 256 },
+        { 2, 384 },
+        { 3, 512 },
+        { 4, 768 },
+};
+
+
+static int wm8741_startup(struct snd_pcm_substream *substream,
+                          struct snd_soc_dai *dai)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
+
+        /* The set of sample rates that can be supported depends on the
+         * MCLK supplied to the CODEC - enforce this.
+         */
+        if (!wm8741->sysclk) {
+                dev_err(codec->dev,
+                        "No MCLK configured, call set_sysclk() on init\n");
+                return -EINVAL;
+        }
+
+        snd_pcm_hw_constraint_list(substream->runtime, 0,
+                                   SNDRV_PCM_HW_PARAM_RATE,
+                                   &wm8741->rate_constraint);
+
+        return 0;
+}
+
+static int wm8741_hw_params(struct snd_pcm_substream *substream,
+                            struct snd_pcm_hw_params *params,
+                            struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_soc_codec *codec = socdev->card->codec;
+        struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
+        u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC;
+        int i;
+
+        /* Find a supported LRCLK ratio */
+        for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
+                if (wm8741->sysclk / params_rate(params) ==
+                    lrclk_ratios[i].ratio)
+                        break;
+        }
+
+        /* Should never happen, should be handled by constraints */
+        if (i == ARRAY_SIZE(lrclk_ratios)) {
+                dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n",
+                        wm8741->sysclk / params_rate(params));
+                return -EINVAL;
+        }
+
+        /* bit size */
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                break;
+        case SNDRV_PCM_FORMAT_S20_3LE:
+                iface |= 0x0001;
+                break;
+        case SNDRV_PCM_FORMAT_S24_LE:
+                iface |= 0x0002;
+                break;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                iface |= 0x0003;
+                break;
+        default:
+                dev_dbg(codec->dev, "wm8741_hw_params:    Unsupported bit size param = %d",
+                        params_format(params));
+                return -EINVAL;
+        }
+
+        dev_dbg(codec->dev, "wm8741_hw_params:    bit size param = %d",
+                params_format(params));
+
+        snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface);
+        return 0;
+}
+
+static int wm8741_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 wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
+        unsigned int val;
+        int i;
+
+        dev_dbg(codec->dev, "wm8741_set_dai_sysclk info: freq=%dHz\n", freq);
+
+        wm8741->sysclk = freq;
+
+        wm8741->rate_constraint.count = 0;
+
+        for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) {
+                dev_dbg(codec->dev, "index = %d, ratio = %d, freq = %d",
+                                i, lrclk_ratios[i].ratio, freq);
+
+                val = freq / lrclk_ratios[i].ratio;
+                /* Check that it's a standard rate since core can't
+                 * cope with others and having the odd rates confuses
+                 * constraint matching.
+                 */
+                switch (val) {
+                case 32000:
+                case 44100:
+                case 48000:
+                case 64000:
+                case 88200:
+                case 96000:
+                        dev_dbg(codec->dev, "Supported sample rate: %dHz\n",
+                                val);
+                        wm8741->rate_constraint_list[i] = val;
+                        wm8741->rate_constraint.count++;
+                        break;
+                default:
+                        dev_dbg(codec->dev, "Skipping sample rate: %dHz\n",
+                                val);
+                }
+        }
+
+        /* Need at least one supported rate... */
+        if (wm8741->rate_constraint.count == 0)
+                return -EINVAL;
+
+        return 0;
+}
+
+static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai,
+                unsigned int fmt)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1C3;
+
+        /* check master/slave audio interface */
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        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 |= 0x0008;
+                break;
+        case SND_SOC_DAIFMT_RIGHT_J:
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                iface |= 0x0004;
+                break;
+        case SND_SOC_DAIFMT_DSP_A:
+                iface |= 0x0003;
+                break;
+        case SND_SOC_DAIFMT_DSP_B:
+                iface |= 0x0013;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /* clock inversion */
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                break;
+        case SND_SOC_DAIFMT_IB_IF:
+                iface |= 0x0010;
+                break;
+        case SND_SOC_DAIFMT_IB_NF:
+                iface |= 0x0020;
+                break;
+        case SND_SOC_DAIFMT_NB_IF:
+                iface |= 0x0030;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+
+        dev_dbg(codec->dev, "wm8741_set_dai_fmt:    Format=%x, Clock Inv=%x\n",
+                                fmt & SND_SOC_DAIFMT_FORMAT_MASK,
+                                ((fmt & SND_SOC_DAIFMT_INV_MASK)));
+
+        snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface);
+        return 0;
+}
+
+#define WM8741_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+                        SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \
+                        SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \
+                        SNDRV_PCM_RATE_192000)
+
+#define WM8741_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+                        SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm8741_dai_ops = {
+        .startup        = wm8741_startup,
+        .hw_params      = wm8741_hw_params,
+        .set_sysclk     = wm8741_set_dai_sysclk,
+        .set_fmt        = wm8741_set_dai_fmt,
+};
+
+struct snd_soc_dai wm8741_dai = {
+        .name = "WM8741",
+        .playback = {
+                .stream_name = "Playback",
+                .channels_min = 2,  /* Mono modes not yet supported */
+                .channels_max = 2,
+                .rates = WM8741_RATES,
+                .formats = WM8741_FORMATS,
+        },
+        .ops = &wm8741_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm8741_dai);
+
+#ifdef CONFIG_PM
+static int wm8741_resume(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+        u16 *cache = codec->reg_cache;
+        int i;
+
+        /* RESTORE REG Cache */
+        for (i = 0; i < WM8741_REGISTER_COUNT; i++) {
+                if (cache[i] == wm8741_reg_defaults[i] || WM8741_RESET == i)
+                        continue;
+                snd_soc_write(codec, i, cache[i]);
+        }
+        return 0;
+}
+#else
+#define wm8741_suspend NULL
+#define wm8741_resume NULL
+#endif
+
+static int wm8741_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec;
+        int ret = 0;
+
+        if (wm8741_codec == NULL) {
+                dev_err(&pdev->dev, "Codec device not registered\n");
+                return -ENODEV;
+        }
+
+        socdev->card->codec = wm8741_codec;
+        codec = wm8741_codec;
+
+        /* register pcms */
+        ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+        if (ret < 0) {
+                dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+                goto pcm_err;
+        }
+
+        snd_soc_add_controls(codec, wm8741_snd_controls,
+                             ARRAY_SIZE(wm8741_snd_controls));
+        wm8741_add_widgets(codec);
+
+        return ret;
+
+pcm_err:
+        return ret;
+}
+
+static int wm8741_remove(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+        snd_soc_free_pcms(socdev);
+        snd_soc_dapm_free(socdev);
+
+        return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8741 = {
+        .probe =        wm8741_probe,
+        .remove =       wm8741_remove,
+        .resume =       wm8741_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8741);
+
+static int wm8741_register(struct wm8741_priv *wm8741,
+                           enum snd_soc_control_type control)
+{
+        int ret;
+        struct snd_soc_codec *codec = &wm8741->codec;
+        int i;
+
+        if (wm8741_codec) {
+                dev_err(codec->dev, "Another WM8741 is registered\n");
+                return -EINVAL;
+        }
+
+        mutex_init(&codec->mutex);
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        snd_soc_codec_set_drvdata(codec, wm8741);
+        codec->name = "WM8741";
+        codec->owner = THIS_MODULE;
+        codec->bias_level = SND_SOC_BIAS_OFF;
+        codec->set_bias_level = NULL;
+        codec->dai = &wm8741_dai;
+        codec->num_dai = 1;
+        codec->reg_cache_size = WM8741_REGISTER_COUNT;
+        codec->reg_cache = &wm8741->reg_cache;
+
+        wm8741->rate_constraint.list = &wm8741->rate_constraint_list[0];
+        wm8741->rate_constraint.count =
+                ARRAY_SIZE(wm8741->rate_constraint_list);
+
+        memcpy(codec->reg_cache, wm8741_reg_defaults,
+                sizeof(wm8741->reg_cache));
+
+        ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+                goto err;
+        }
+
+        for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++)
+                wm8741->supplies[i].supply = wm8741_supply_names[i];
+
+        ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies),
+                                 wm8741->supplies);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+                goto err;
+        }
+
+        ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies),
+                                    wm8741->supplies);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+                goto err_get;
+        }
+
+        ret = wm8741_reset(codec);
+        if (ret < 0) {
+                dev_err(codec->dev, "Failed to issue reset\n");
+                goto err_enable;
+        }
+
+        wm8741_dai.dev = codec->dev;
+
+        /* Change some default settings - latch VU */
+        wm8741->reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL;
+        wm8741->reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM;
+        wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL;
+        wm8741->reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM;
+
+        wm8741_codec = codec;
+
+        ret = snd_soc_register_codec(codec);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+                return ret;
+        }
+
+        ret = snd_soc_register_dai(&wm8741_dai);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+                snd_soc_unregister_codec(codec);
+                return ret;
+        }
+
+        dev_dbg(codec->dev, "Successful registration\n");
+        return 0;
+
+err_enable:
+        regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+
+err_get:
+        regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+
+err:
+        kfree(wm8741);
+        return ret;
+}
+
+static void wm8741_unregister(struct wm8741_priv *wm8741)
+{
+        regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+
+        snd_soc_unregister_dai(&wm8741_dai);
+        snd_soc_unregister_codec(&wm8741->codec);
+        kfree(wm8741);
+        wm8741_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int wm8741_i2c_probe(struct i2c_client *i2c,
+                                      const struct i2c_device_id *id)
+{
+        struct wm8741_priv *wm8741;
+        struct snd_soc_codec *codec;
+
+        wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL);
+        if (wm8741 == NULL)
+                return -ENOMEM;
+
+        codec = &wm8741->codec;
+        codec->hw_write = (hw_write_t)i2c_master_send;
+
+        i2c_set_clientdata(i2c, wm8741);
+        codec->control_data = i2c;
+
+        codec->dev = &i2c->dev;
+
+        return wm8741_register(wm8741, SND_SOC_I2C);
+}
+
+static __devexit int wm8741_i2c_remove(struct i2c_client *client)
+{
+        struct wm8741_priv *wm8741 = i2c_get_clientdata(client);
+        wm8741_unregister(wm8741);
+        return 0;
+}
+
+static const struct i2c_device_id wm8741_i2c_id[] = {
+        { "wm8741", 0 },
+        { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id);
+
+
+static struct i2c_driver wm8741_i2c_driver = {
+        .driver = {
+                .name = "WM8741",
+                .owner = THIS_MODULE,
+        },
+        .probe =    wm8741_i2c_probe,
+        .remove =   __devexit_p(wm8741_i2c_remove),
+        .id_table = wm8741_i2c_id,
+};
+#endif
+
+static int __init wm8741_modinit(void)
+{
+        int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+        ret = i2c_add_driver(&wm8741_i2c_driver);
+        if (ret != 0) {
+                printk(KERN_ERR "Failed to register WM8741 I2C driver: %d\n",
+                       ret);
+        }
+#endif
+        return 0;
+}
+module_init(wm8741_modinit);
+
+static void __exit wm8741_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+        i2c_del_driver(&wm8741_i2c_driver);
+#endif
+}
+module_exit(wm8741_exit);
+
+MODULE_DESCRIPTION("ASoC WM8741 driver");
+MODULE_AUTHOR("Ian Lartey <ian@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8741.h b/sound/soc/codecs/wm8741.h
new file mode 100644
index 000000000000..fdef6ecd1f6f
--- /dev/null
+++ b/sound/soc/codecs/wm8741.h
@@ -0,0 +1,214 @@
+/*
+ * wm8741.h  --  WM8423 ASoC driver
+ *
+ * Copyright 2010 Wolfson Microelectronics, plc
+ *
+ * Author: Ian Lartey <ian@opensource.wolfsonmicro.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 _WM8741_H
+#define _WM8741_H
+
+/*
+ * Register values.
+ */
+#define WM8741_DACLLSB_ATTENUATION              0x00
+#define WM8741_DACLMSB_ATTENUATION              0x01
+#define WM8741_DACRLSB_ATTENUATION              0x02
+#define WM8741_DACRMSB_ATTENUATION              0x03
+#define WM8741_VOLUME_CONTROL                   0x04
+#define WM8741_FORMAT_CONTROL                   0x05
+#define WM8741_FILTER_CONTROL                   0x06
+#define WM8741_MODE_CONTROL_1                   0x07
+#define WM8741_MODE_CONTROL_2                   0x08
+#define WM8741_RESET                            0x09
+#define WM8741_ADDITIONAL_CONTROL_1             0x20
+
+#define WM8741_REGISTER_COUNT                   11
+#define WM8741_MAX_REGISTER                     0x20
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - DACLLSB_ATTENUATION
+ */
+#define WM8741_UPDATELL                         0x0020  /* UPDATELL */
+#define WM8741_UPDATELL_MASK                    0x0020  /* UPDATELL */
+#define WM8741_UPDATELL_SHIFT                        5  /* UPDATELL */
+#define WM8741_UPDATELL_WIDTH                        1  /* UPDATELL */
+#define WM8741_LAT_4_0_MASK                     0x001F  /* LAT[4:0] - [4:0] */
+#define WM8741_LAT_4_0_SHIFT                         0  /* LAT[4:0] - [4:0] */
+#define WM8741_LAT_4_0_WIDTH                         5  /* LAT[4:0] - [4:0] */
+
+/*
+ * R1 (0x01) - DACLMSB_ATTENUATION
+ */
+#define WM8741_UPDATELM                         0x0020  /* UPDATELM */
+#define WM8741_UPDATELM_MASK                    0x0020  /* UPDATELM */
+#define WM8741_UPDATELM_SHIFT                        5  /* UPDATELM */
+#define WM8741_UPDATELM_WIDTH                        1  /* UPDATELM */
+#define WM8741_LAT_9_5_0_MASK                   0x001F  /* LAT[9:5] - [4:0] */
+#define WM8741_LAT_9_5_0_SHIFT                       0  /* LAT[9:5] - [4:0] */
+#define WM8741_LAT_9_5_0_WIDTH                       5  /* LAT[9:5] - [4:0] */
+
+/*
+ * R2 (0x02) - DACRLSB_ATTENUATION
+ */
+#define WM8741_UPDATERL                         0x0020  /* UPDATERL */
+#define WM8741_UPDATERL_MASK                    0x0020  /* UPDATERL */
+#define WM8741_UPDATERL_SHIFT                        5  /* UPDATERL */
+#define WM8741_UPDATERL_WIDTH                        1  /* UPDATERL */
+#define WM8741_RAT_4_0_MASK                     0x001F  /* RAT[4:0] - [4:0] */
+#define WM8741_RAT_4_0_SHIFT                         0  /* RAT[4:0] - [4:0] */
+#define WM8741_RAT_4_0_WIDTH                         5  /* RAT[4:0] - [4:0] */
+
+/*
+ * R3 (0x03) - DACRMSB_ATTENUATION
+ */
+#define WM8741_UPDATERM                         0x0020  /* UPDATERM */
+#define WM8741_UPDATERM_MASK                    0x0020  /* UPDATERM */
+#define WM8741_UPDATERM_SHIFT                        5  /* UPDATERM */
+#define WM8741_UPDATERM_WIDTH                        1  /* UPDATERM */
+#define WM8741_RAT_9_5_0_MASK                   0x001F  /* RAT[9:5] - [4:0] */
+#define WM8741_RAT_9_5_0_SHIFT                       0  /* RAT[9:5] - [4:0] */
+#define WM8741_RAT_9_5_0_WIDTH                       5  /* RAT[9:5] - [4:0] */
+
+/*
+ * R4 (0x04) - VOLUME_CONTROL
+ */
+#define WM8741_AMUTE                            0x0080  /* AMUTE */
+#define WM8741_AMUTE_MASK                       0x0080  /* AMUTE */
+#define WM8741_AMUTE_SHIFT                           7  /* AMUTE */
+#define WM8741_AMUTE_WIDTH                           1  /* AMUTE */
+#define WM8741_ZFLAG_MASK                       0x0060  /* ZFLAG - [6:5] */
+#define WM8741_ZFLAG_SHIFT                           5  /* ZFLAG - [6:5] */
+#define WM8741_ZFLAG_WIDTH                           2  /* ZFLAG - [6:5] */
+#define WM8741_IZD                              0x0010  /* IZD */
+#define WM8741_IZD_MASK                         0x0010  /* IZD */
+#define WM8741_IZD_SHIFT                             4  /* IZD */
+#define WM8741_IZD_WIDTH                             1  /* IZD */
+#define WM8741_SOFT                             0x0008  /* SOFT MUTE */
+#define WM8741_SOFT_MASK                        0x0008  /* SOFT MUTE */
+#define WM8741_SOFT_SHIFT                            3  /* SOFT MUTE */
+#define WM8741_SOFT_WIDTH                            1  /* SOFT MUTE */
+#define WM8741_ATC                              0x0004  /* ATC */
+#define WM8741_ATC_MASK                         0x0004  /* ATC */
+#define WM8741_ATC_SHIFT                             2  /* ATC */
+#define WM8741_ATC_WIDTH                             1  /* ATC */
+#define WM8741_ATT2DB                           0x0002  /* ATT2DB */
+#define WM8741_ATT2DB_MASK                      0x0002  /* ATT2DB */
+#define WM8741_ATT2DB_SHIFT                          1  /* ATT2DB */
+#define WM8741_ATT2DB_WIDTH                          1  /* ATT2DB */
+#define WM8741_VOL_RAMP                         0x0001  /* VOL_RAMP */
+#define WM8741_VOL_RAMP_MASK                    0x0001  /* VOL_RAMP */
+#define WM8741_VOL_RAMP_SHIFT                        0  /* VOL_RAMP */
+#define WM8741_VOL_RAMP_WIDTH                        1  /* VOL_RAMP */
+
+/*
+ * R5 (0x05) - FORMAT_CONTROL
+ */
+#define WM8741_PWDN                             0x0080  /* PWDN */
+#define WM8741_PWDN_MASK                        0x0080  /* PWDN */
+#define WM8741_PWDN_SHIFT                            7  /* PWDN */
+#define WM8741_PWDN_WIDTH                            1  /* PWDN */
+#define WM8741_REV                              0x0040  /* REV */
+#define WM8741_REV_MASK                         0x0040  /* REV */
+#define WM8741_REV_SHIFT                             6  /* REV */
+#define WM8741_REV_WIDTH                             1  /* REV */
+#define WM8741_BCP                              0x0020  /* BCP */
+#define WM8741_BCP_MASK                         0x0020  /* BCP */
+#define WM8741_BCP_SHIFT                             5  /* BCP */
+#define WM8741_BCP_WIDTH                             1  /* BCP */
+#define WM8741_LRP                              0x0010  /* LRP */
+#define WM8741_LRP_MASK                         0x0010  /* LRP */
+#define WM8741_LRP_SHIFT                             4  /* LRP */
+#define WM8741_LRP_WIDTH                             1  /* LRP */
+#define WM8741_FMT_MASK                         0x000C  /* FMT - [3:2] */
+#define WM8741_FMT_SHIFT                             2  /* FMT - [3:2] */
+#define WM8741_FMT_WIDTH                             2  /* FMT - [3:2] */
+#define WM8741_IWL_MASK                         0x0003  /* IWL - [1:0] */
+#define WM8741_IWL_SHIFT                             0  /* IWL - [1:0] */
+#define WM8741_IWL_WIDTH                             2  /* IWL - [1:0] */
+
+/*
+ * R6 (0x06) - FILTER_CONTROL
+ */
+#define WM8741_ZFLAG_HI                         0x0080  /* ZFLAG_HI */
+#define WM8741_ZFLAG_HI_MASK                    0x0080  /* ZFLAG_HI */
+#define WM8741_ZFLAG_HI_SHIFT                        7  /* ZFLAG_HI */
+#define WM8741_ZFLAG_HI_WIDTH                        1  /* ZFLAG_HI */
+#define WM8741_DEEMPH_MASK                      0x0060  /* DEEMPH - [6:5] */
+#define WM8741_DEEMPH_SHIFT                          5  /* DEEMPH - [6:5] */
+#define WM8741_DEEMPH_WIDTH                          2  /* DEEMPH - [6:5] */
+#define WM8741_DSDFILT_MASK                     0x0018  /* DSDFILT - [4:3] */
+#define WM8741_DSDFILT_SHIFT                         3  /* DSDFILT - [4:3] */
+#define WM8741_DSDFILT_WIDTH                         2  /* DSDFILT - [4:3] */
+#define WM8741_FIRSEL_MASK                      0x0007  /* FIRSEL - [2:0] */
+#define WM8741_FIRSEL_SHIFT                          0  /* FIRSEL - [2:0] */
+#define WM8741_FIRSEL_WIDTH                          3  /* FIRSEL - [2:0] */
+
+/*
+ * R7 (0x07) - MODE_CONTROL_1
+ */
+#define WM8741_MODE8X                           0x0080  /* MODE8X */
+#define WM8741_MODE8X_MASK                      0x0080  /* MODE8X */
+#define WM8741_MODE8X_SHIFT                          7  /* MODE8X */
+#define WM8741_MODE8X_WIDTH                          1  /* MODE8X */
+#define WM8741_OSR_MASK                         0x0060  /* OSR - [6:5] */
+#define WM8741_OSR_SHIFT                             5  /* OSR - [6:5] */
+#define WM8741_OSR_WIDTH                             2  /* OSR - [6:5] */
+#define WM8741_SR_MASK                          0x001C  /* SR - [4:2] */
+#define WM8741_SR_SHIFT                              2  /* SR - [4:2] */
+#define WM8741_SR_WIDTH                              3  /* SR - [4:2] */
+#define WM8741_MODESEL_MASK                     0x0003  /* MODESEL - [1:0] */
+#define WM8741_MODESEL_SHIFT                         0  /* MODESEL - [1:0] */
+#define WM8741_MODESEL_WIDTH                         2  /* MODESEL - [1:0] */
+
+/*
+ * R8 (0x08) - MODE_CONTROL_2
+ */
+#define WM8741_DSD_GAIN                         0x0040  /* DSD_GAIN */
+#define WM8741_DSD_GAIN_MASK                    0x0040  /* DSD_GAIN */
+#define WM8741_DSD_GAIN_SHIFT                        6  /* DSD_GAIN */
+#define WM8741_DSD_GAIN_WIDTH                        1  /* DSD_GAIN */
+#define WM8741_SDOUT                            0x0020  /* SDOUT */
+#define WM8741_SDOUT_MASK                       0x0020  /* SDOUT */
+#define WM8741_SDOUT_SHIFT                           5  /* SDOUT */
+#define WM8741_SDOUT_WIDTH                           1  /* SDOUT */
+#define WM8741_DOUT                             0x0010  /* DOUT */
+#define WM8741_DOUT_MASK                        0x0010  /* DOUT */
+#define WM8741_DOUT_SHIFT                            4  /* DOUT */
+#define WM8741_DOUT_WIDTH                            1  /* DOUT */
+#define WM8741_DIFF_MASK                        0x000C  /* DIFF - [3:2] */
+#define WM8741_DIFF_SHIFT                            2  /* DIFF - [3:2] */
+#define WM8741_DIFF_WIDTH                            2  /* DIFF - [3:2] */
+#define WM8741_DITHER_MASK                      0x0003  /* DITHER - [1:0] */
+#define WM8741_DITHER_SHIFT                          0  /* DITHER - [1:0] */
+#define WM8741_DITHER_WIDTH                          2  /* DITHER - [1:0] */
+
+/*
+ * R32 (0x20) - ADDITONAL_CONTROL_1
+ */
+#define WM8741_DSD_LEVEL                        0x0002  /* DSD_LEVEL */
+#define WM8741_DSD_LEVEL_MASK                   0x0002  /* DSD_LEVEL */
+#define WM8741_DSD_LEVEL_SHIFT                       1  /* DSD_LEVEL */
+#define WM8741_DSD_LEVEL_WIDTH                       1  /* DSD_LEVEL */
+#define WM8741_DSD_NO_NOTCH                     0x0001  /* DSD_NO_NOTCH */
+#define WM8741_DSD_NO_NOTCH_MASK                0x0001  /* DSD_NO_NOTCH */
+#define WM8741_DSD_NO_NOTCH_SHIFT                    0  /* DSD_NO_NOTCH */
+#define WM8741_DSD_NO_NOTCH_WIDTH                    1  /* DSD_NO_NOTCH */
+
+#define  WM8741_SYSCLK 0
+
+extern struct snd_soc_dai wm8741_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8741;
+
+#endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 9407e193fcc3..e2c05e3e323a 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -884,6 +884,7 @@ static int wm8750_i2c_remove(struct i2c_client *client)
 
 static const struct i2c_device_id wm8750_i2c_id[] = {
         { "wm8750", 0 },
+        { "wm8987", 0 }, /* WM8987 is register compatible with WM8750 */
         { }
 };
 MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id);
@@ -925,14 +926,22 @@ static int __devexit wm8750_spi_remove(struct spi_device *spi)
         return 0;
 }
 
+static const struct spi_device_id wm8750_spi_id[] = {
+        { "wm8750", 0 },
+        { "wm8987", 0 },
+        { }
+};
+MODULE_DEVICE_TABLE(spi, wm8750_spi_id);
+
 static struct spi_driver wm8750_spi_driver = {
         .driver = {
-                .name   = "wm8750",
+                .name   = "WM8750 SPI Codec",
                 .bus    = &spi_bus_type,
                 .owner  = THIS_MODULE,
         },
         .probe          = wm8750_spi_probe,
         .remove         = __devexit_p(wm8750_spi_remove),
+        .id_table       = wm8750_spi_id,
 };
 #endif
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 87f14f8675fa..f7dcabf6283c 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2433,7 +2433,8 @@ static int wm8904_register(struct wm8904_priv *wm8904,
 
         if (wm8904_codec) {
                 dev_err(codec->dev, "Another WM8904 is registered\n");
-                return -EINVAL;
+                ret = -EINVAL;
+                goto err;
         }
 
         mutex_init(&codec->mutex);
@@ -2462,7 +2463,8 @@ static int wm8904_register(struct wm8904_priv *wm8904,
         default:
                 dev_err(codec->dev, "Unknown device type %d\n",
                         wm8904->devtype);
-                return -EINVAL;
+                ret = -EINVAL;
+                goto err;
         }
 
         memcpy(codec->reg_cache, wm8904_reg, sizeof(wm8904_reg));
@@ -2566,18 +2568,19 @@ static int wm8904_register(struct wm8904_priv *wm8904,
         ret = snd_soc_register_codec(codec);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-                return ret;
+                goto err_enable;
         }
 
         ret = snd_soc_register_dai(&wm8904_dai);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-                snd_soc_unregister_codec(codec);
-                return ret;
+                goto err_codec;
         }
 
         return 0;
 
+err_codec:
+        snd_soc_unregister_codec(codec);
 err_enable:
         regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
 err_get:
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index e3c4bbfaae27..f0c11138e610 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -845,6 +845,7 @@ static void wm8940_unregister(struct wm8940_priv *wm8940)
 static int wm8940_i2c_probe(struct i2c_client *i2c,
                             const struct i2c_device_id *id)
 {
+        int ret;
         struct wm8940_priv *wm8940;
         struct snd_soc_codec *codec;
 
@@ -858,7 +859,11 @@ static int wm8940_i2c_probe(struct i2c_client *i2c,
         codec->control_data = i2c;
         codec->dev = &i2c->dev;
 
-        return wm8940_register(wm8940, SND_SOC_I2C);
+        ret = wm8940_register(wm8940, SND_SOC_I2C);
+        if (ret < 0)
+                kfree(wm8940);
+
+        return ret;
 }
 
 static int __devexit wm8940_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c
index fedb76452f1b..5f025593d84d 100644
--- a/sound/soc/codecs/wm8955.c
+++ b/sound/soc/codecs/wm8955.c
@@ -964,7 +964,8 @@ static int wm8955_register(struct wm8955_priv *wm8955,
 
         if (wm8955_codec) {
                 dev_err(codec->dev, "Another WM8955 is registered\n");
-                return -EINVAL;
+                ret = -EINVAL;
+                goto err;
         }
 
         mutex_init(&codec->mutex);
@@ -1047,18 +1048,19 @@ static int wm8955_register(struct wm8955_priv *wm8955,
         ret = snd_soc_register_codec(codec);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-                return ret;
+                goto err_enable;
         }
 
         ret = snd_soc_register_dai(&wm8955_dai);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-                snd_soc_unregister_codec(codec);
-                return ret;
+                goto err_codec;
         }
 
         return 0;
 
+err_codec:
+        snd_soc_unregister_codec(codec);
 err_enable:
         regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies);
 err_get:
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 7233cc68435a..3c6ee61f6c95 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -79,12 +79,13 @@ struct wm8960_priv {
         struct snd_soc_dapm_widget *lout1;
         struct snd_soc_dapm_widget *rout1;
         struct snd_soc_dapm_widget *out3;
+        bool deemph;
+        int playback_fs;
 };
 
 #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0)
 
 /* enumerated controls */
-static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
 static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
         "Right Inverted", "Stereo Inversion"};
 static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
@@ -93,7 +94,6 @@ static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
 static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
 
 static const struct soc_enum wm8960_enum[] = {
-        SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph),
         SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
         SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
         SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
@@ -102,6 +102,59 @@ static const struct soc_enum wm8960_enum[] = {
         SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
 };
 
+static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
+
+static int wm8960_set_deemph(struct snd_soc_codec *codec)
+{
+        struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+        int val, i, best;
+
+        /* If we're using deemphasis select the nearest available sample
+         * rate.
+         */
+        if (wm8960->deemph) {
+                best = 1;
+                for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
+                        if (abs(deemph_settings[i] - wm8960->playback_fs) <
+                            abs(deemph_settings[best] - wm8960->playback_fs))
+                                best = i;
+                }
+
+                val = best << 1;
+        } else {
+                val = 0;
+        }
+
+        dev_dbg(codec->dev, "Set deemphasis %d\n", val);
+
+        return snd_soc_update_bits(codec, WM8960_DACCTL1,
+                                   0x6, val);
+}
+
+static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+
+        return wm8960->deemph;
+}
+
+static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+        int deemph = ucontrol->value.enumerated.item[0];
+
+        if (deemph > 1)
+                return -EINVAL;
+
+        wm8960->deemph = deemph;
+
+        return wm8960_set_deemph(codec);
+}
+
 static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
 static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
@@ -131,23 +184,24 @@ SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0),
 SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0),
 
 SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
-SOC_ENUM("ADC Polarity", wm8960_enum[1]),
-SOC_ENUM("Playback De-emphasis", wm8960_enum[0]),
+SOC_ENUM("ADC Polarity", wm8960_enum[0]),
 SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
 
 SOC_ENUM("DAC Polarity", wm8960_enum[2]),
+SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
+                    wm8960_get_deemph, wm8960_put_deemph),
 
-SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
-SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
+SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[2]),
+SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[3]),
 SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
 SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
 
-SOC_ENUM("ALC Function", wm8960_enum[5]),
+SOC_ENUM("ALC Function", wm8960_enum[4]),
 SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
 SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
 SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
 SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
-SOC_ENUM("ALC Mode", wm8960_enum[6]),
+SOC_ENUM("ALC Mode", wm8960_enum[5]),
 SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
 SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
 
@@ -433,6 +487,21 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
         return 0;
 }
 
+static struct {
+        int rate;
+        unsigned int val;
+} alc_rates[] = {
+        { 48000, 0 },
+        { 44100, 0 },
+        { 32000, 1 },
+        { 22050, 2 },
+        { 24000, 2 },
+        { 16000, 3 },
+        { 11250, 4 },
+        { 12000, 4 },
+        {  8000, 5 },
+};
+
 static int wm8960_hw_params(struct snd_pcm_substream *substream,
                             struct snd_pcm_hw_params *params,
                             struct snd_soc_dai *dai)
@@ -440,7 +509,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_device *socdev = rtd->socdev;
         struct snd_soc_codec *codec = socdev->card->codec;
+        struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
         u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3;
+        int i;
 
         /* bit size */
         switch (params_format(params)) {
@@ -454,6 +525,18 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream,
                 break;
         }
 
+        /* Update filters for the new rate */
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                wm8960->playback_fs = params_rate(params);
+                wm8960_set_deemph(codec);
+        } else {
+                for (i = 0; i < ARRAY_SIZE(alc_rates); i++)
+                        if (alc_rates[i].rate == params_rate(params))
+                                snd_soc_update_bits(codec,
+                                                    WM8960_ADDCTL3, 0x7,
+                                                    alc_rates[i].val);
+        }
+
         /* set iface */
         snd_soc_write(codec, WM8960_IFACE1, iface);
         return 0;
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 5b9a756242f1..2549d3a297ab 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -1102,7 +1102,7 @@ static int wm8961_register(struct wm8961_priv *wm8961)
         ret = wm8961_reset(codec);
         if (ret < 0) {
                 dev_err(codec->dev, "Failed to issue reset\n");
-                return ret;
+                goto err;
         }
 
         /* Enable class W */
@@ -1147,18 +1147,19 @@ static int wm8961_register(struct wm8961_priv *wm8961)
         ret = snd_soc_register_codec(codec);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-                return ret;
+                goto err;
         }
 
         ret = snd_soc_register_dai(&wm8961_dai);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-                snd_soc_unregister_codec(codec);
-                return ret;
+                goto err_codec;
         }
 
         return 0;
 
+err_codec:
+        snd_soc_unregister_codec(codec);
 err:
         kfree(wm8961);
         return ret;
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index a2c4b2f37cca..1468fe10cbbe 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -670,7 +670,8 @@ static __devinit int wm8974_register(struct wm8974_priv *wm8974)
 
         if (wm8974_codec) {
                 dev_err(codec->dev, "Another WM8974 is registered\n");
-                return -EINVAL;
+                ret = -EINVAL;
+                goto err;
         }
 
         mutex_init(&codec->mutex);
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 51d5f433215c..8a1ad778e7e3 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -1076,7 +1076,6 @@ static __devinit int wm8978_register(struct wm8978_priv *wm8978)
 err_codec:
         snd_soc_unregister_codec(codec);
 err:
-        kfree(wm8978);
         return ret;
 }
 
@@ -1085,13 +1084,13 @@ static __devexit void wm8978_unregister(struct wm8978_priv *wm8978)
         wm8978_set_bias_level(&wm8978->codec, SND_SOC_BIAS_OFF);
         snd_soc_unregister_dai(&wm8978_dai);
         snd_soc_unregister_codec(&wm8978->codec);
-        kfree(wm8978);
         wm8978_codec = NULL;
 }
 
 static __devinit int wm8978_i2c_probe(struct i2c_client *i2c,
                                       const struct i2c_device_id *id)
 {
+        int ret;
         struct wm8978_priv *wm8978;
         struct snd_soc_codec *codec;
 
@@ -1107,13 +1106,18 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c,
 
         codec->dev = &i2c->dev;
 
-        return wm8978_register(wm8978);
+        ret = wm8978_register(wm8978);
+        if (ret < 0)
+                kfree(wm8978);
+
+        return ret;
 }
 
 static __devexit int wm8978_i2c_remove(struct i2c_client *client)
 {
         struct wm8978_priv *wm8978 = i2c_get_clientdata(client);
         wm8978_unregister(wm8978);
+        kfree(wm8978);
         return 0;
 }
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c018772cc430..dd8d909788c1 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -30,8 +30,6 @@
 
 #include "wm8990.h"
 
-#define WM8990_VERSION "0.2"
-
 /* codec private data */
 struct wm8990_priv {
         unsigned int sysclk;
@@ -1511,8 +1509,6 @@ static int wm8990_probe(struct platform_device *pdev)
         struct wm8990_priv *wm8990;
         int ret;
 
-        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)
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index e84a1177f350..a87046a96f2a 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1677,6 +1677,26 @@ static struct {
 
 static int wm8994_readable(unsigned int reg)
 {
+        switch (reg) {
+        case WM8994_GPIO_1:
+        case WM8994_GPIO_2:
+        case WM8994_GPIO_3:
+        case WM8994_GPIO_4:
+        case WM8994_GPIO_5:
+        case WM8994_GPIO_6:
+        case WM8994_GPIO_7:
+        case WM8994_GPIO_8:
+        case WM8994_GPIO_9:
+        case WM8994_GPIO_10:
+        case WM8994_GPIO_11:
+        case WM8994_INTERRUPT_STATUS_1:
+        case WM8994_INTERRUPT_STATUS_2:
+        case WM8994_INTERRUPT_RAW_STATUS_2:
+                return 1;
+        default:
+                break;
+        }
+
         if (reg >= ARRAY_SIZE(access_masks))
                 return 0;
         return access_masks[reg].readable != 0;
@@ -2341,6 +2361,20 @@ SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING,
                 0, 1, 0),
 };
 
+static const struct snd_kcontrol_new aif1adc2l_mix[] = {
+SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING,
+                1, 1, 0),
+SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING,
+                0, 1, 0),
+};
+
+static const struct snd_kcontrol_new aif1adc2r_mix[] = {
+SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING,
+                1, 1, 0),
+SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING,
+                0, 1, 0),
+};
+
 static const struct snd_kcontrol_new aif2dac2l_mix[] = {
 SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING,
                 5, 1, 0),
@@ -2472,6 +2506,7 @@ static const struct snd_kcontrol_new aif3adc_mux =
 static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("DMIC1DAT"),
 SND_SOC_DAPM_INPUT("DMIC2DAT"),
+SND_SOC_DAPM_INPUT("Clock"),
 
 SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
                     SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -2506,6 +2541,11 @@ SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0,
 SND_SOC_DAPM_MIXER("AIF1ADC1R Mixer", SND_SOC_NOPM, 0, 0,
                    aif1adc1r_mix, ARRAY_SIZE(aif1adc1r_mix)),
 
+SND_SOC_DAPM_MIXER("AIF1ADC2L Mixer", SND_SOC_NOPM, 0, 0,
+                   aif1adc2l_mix, ARRAY_SIZE(aif1adc2l_mix)),
+SND_SOC_DAPM_MIXER("AIF1ADC2R Mixer", SND_SOC_NOPM, 0, 0,
+                   aif1adc2r_mix, ARRAY_SIZE(aif1adc2r_mix)),
+
 SND_SOC_DAPM_MIXER("AIF2DAC2L Mixer", SND_SOC_NOPM, 0, 0,
                    aif2dac2l_mix, ARRAY_SIZE(aif2dac2l_mix)),
 SND_SOC_DAPM_MIXER("AIF2DAC2R Mixer", SND_SOC_NOPM, 0, 0,
@@ -2668,6 +2708,14 @@ static const struct snd_soc_dapm_route intercon[] = {
         { "AIF1ADC1R Mixer", "ADC/DMIC Switch", "ADCR Mux" },
         { "AIF1ADC1R Mixer", "AIF2 Switch", "AIF2DACR" },
 
+        { "AIF1ADC2L", NULL, "AIF1ADC2L Mixer" },
+        { "AIF1ADC2L Mixer", "DMIC Switch", "DMIC2L" },
+        { "AIF1ADC2L Mixer", "AIF2 Switch", "AIF2DACL" },
+
+        { "AIF1ADC2R", NULL, "AIF1ADC2R Mixer" },
+        { "AIF1ADC2R Mixer", "DMIC Switch", "DMIC2R" },
+        { "AIF1ADC2R Mixer", "AIF2 Switch", "AIF2DACR" },
+
         /* Pin level routing for AIF3 */
         { "AIF1DAC1L", NULL, "AIF1DAC Mux" },
         { "AIF1DAC1R", NULL, "AIF1DAC Mux" },
@@ -2946,11 +2994,14 @@ static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
         return 0;
 }
 
+static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 };
+
 static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
                 int clk_id, unsigned int freq, int dir)
 {
         struct snd_soc_codec *codec = dai->codec;
         struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+        int i;
 
         switch (dai->id) {
         case 1:
@@ -2988,6 +3039,25 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
                 dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id);
                 break;
 
+        case WM8994_SYSCLK_OPCLK:
+                /* Special case - a division (times 10) is given and
+                 * no effect on main clocking. 
+                 */
+                if (freq) {
+                        for (i = 0; i < ARRAY_SIZE(opclk_divs); i++)
+                                if (opclk_divs[i] == freq)
+                                        break;
+                        if (i == ARRAY_SIZE(opclk_divs))
+                                return -EINVAL;
+                        snd_soc_update_bits(codec, WM8994_CLOCKING_2,
+                                            WM8994_OPCLK_DIV_MASK, i);
+                        snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
+                                            WM8994_OPCLK_ENA, WM8994_OPCLK_ENA);
+                } else {
+                        snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
+                                            WM8994_OPCLK_ENA, 0);
+                }
+
         default:
                 return -EINVAL;
         }
@@ -4004,6 +4074,11 @@ static int wm8994_codec_probe(struct platform_device *pdev)
                             1 << WM8994_AIF2DAC_3D_GAIN_SHIFT,
                             1 << WM8994_AIF2DAC_3D_GAIN_SHIFT);
 
+        /* Unconditionally enable AIF1 ADC TDM mode; it only affects
+         * behaviour on idle TDM clock cycles. */
+        snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1,
+                            WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM);
+
         wm8994_update_class_w(codec);
 
         ret = snd_soc_register_codec(codec);
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 7072dc539354..2e0ca67a8df7 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -20,6 +20,9 @@ extern struct snd_soc_dai wm8994_dai[];
 #define WM8994_SYSCLK_FLL1  3
 #define WM8994_SYSCLK_FLL2  4
 
+/* OPCLK is also configured with set_dai_sysclk, specify division*10 as rate. */
+#define WM8994_SYSCLK_OPCLK 5
+
 #define WM8994_FLL1 1
 #define WM8994_FLL2 2
 
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 13186fb4dcb4..76b37ff6c264 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1356,7 +1356,7 @@ static int wm9081_register(struct wm9081_priv *wm9081,
         ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-                return ret;
+                goto err;
         }
 
         reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET);
@@ -1369,7 +1369,7 @@ static int wm9081_register(struct wm9081_priv *wm9081,
         ret = wm9081_reset(codec);
         if (ret < 0) {
                 dev_err(codec->dev, "Failed to issue reset\n");
-                return ret;
+                goto err;
         }
 
         wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1388,18 +1388,19 @@ static int wm9081_register(struct wm9081_priv *wm9081,
         ret = snd_soc_register_codec(codec);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-                return ret;
+                goto err;
         }
 
         ret = snd_soc_register_dai(&wm9081_dai);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-                snd_soc_unregister_codec(codec);
-                return ret;
+                goto err_codec;
         }
 
         return 0;
 
+err_codec:
+        snd_soc_unregister_codec(codec);
 err:
         kfree(wm9081);
         return ret;
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 16f1a57da08a..2cb81538cd91 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -410,6 +410,8 @@ static int hp_event(struct snd_soc_dapm_widget *w,
                                     WM8993_HPOUT1L_DLY |
                                     WM8993_HPOUT1R_DLY, 0);
 
+                snd_soc_write(codec, WM8993_DC_SERVO_0, 0);
+
                 snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
                                     WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
                                     0);
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index adadcd3aa1b1..9e8932abf158 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -26,6 +26,7 @@
 #include <mach/asp.h>
 
 #include "davinci-pcm.h"
+#include "davinci-i2s.h"
 
 
 /*
@@ -68,16 +69,21 @@
 #define DAVINCI_MCBSP_RCR_RDATDLY(v)    ((v) << 16)
 #define DAVINCI_MCBSP_RCR_RFIG          (1 << 18)
 #define DAVINCI_MCBSP_RCR_RWDLEN2(v)    ((v) << 21)
+#define DAVINCI_MCBSP_RCR_RFRLEN2(v)    ((v) << 24)
+#define DAVINCI_MCBSP_RCR_RPHASE        BIT(31)
 
 #define DAVINCI_MCBSP_XCR_XWDLEN1(v)    ((v) << 5)
 #define DAVINCI_MCBSP_XCR_XFRLEN1(v)    ((v) << 8)
 #define DAVINCI_MCBSP_XCR_XDATDLY(v)    ((v) << 16)
 #define DAVINCI_MCBSP_XCR_XFIG          (1 << 18)
 #define DAVINCI_MCBSP_XCR_XWDLEN2(v)    ((v) << 21)
+#define DAVINCI_MCBSP_XCR_XFRLEN2(v)    ((v) << 24)
+#define DAVINCI_MCBSP_XCR_XPHASE        BIT(31)
 
 #define DAVINCI_MCBSP_SRGR_FWID(v)      ((v) << 8)
 #define DAVINCI_MCBSP_SRGR_FPER(v)      ((v) << 16)
 #define DAVINCI_MCBSP_SRGR_FSGM         (1 << 28)
+#define DAVINCI_MCBSP_SRGR_CLKSM        BIT(29)
 
 #define DAVINCI_MCBSP_PCR_CLKRP         (1 << 0)
 #define DAVINCI_MCBSP_PCR_CLKXP         (1 << 1)
@@ -116,6 +122,7 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = {
 };
 
 struct davinci_mcbsp_dev {
+        struct device *dev;
         struct davinci_pcm_dma_params   dma_params[2];
         void __iomem                    *base;
 #define MOD_DSP_A       0
@@ -144,6 +151,11 @@ struct davinci_mcbsp_dev {
          * won't end up being swapped because of the underrun.
          */
         unsigned enable_channel_combine:1;
+
+        unsigned int fmt;
+        int clk_div;
+        int clk_input_pin;
+        bool i2s_accurate_sck;
 };
 
 static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -254,10 +266,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
         struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
         unsigned int pcr;
         unsigned int srgr;
+        /* Attention srgr is updated by hw_params! */
         srgr = DAVINCI_MCBSP_SRGR_FSGM |
                 DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) |
                 DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1);
 
+        dev->fmt = fmt;
         /* set master/slave audio interface */
         switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
         case SND_SOC_DAIFMT_CBS_CFS:
@@ -268,11 +282,26 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
                         DAVINCI_MCBSP_PCR_CLKRM;
                 break;
         case SND_SOC_DAIFMT_CBM_CFS:
-                /* McBSP CLKR pin is the input for the Sample Rate Generator.
-                 * McBSP FSR and FSX are driven by the Sample Rate Generator. */
-                pcr = DAVINCI_MCBSP_PCR_SCLKME |
-                        DAVINCI_MCBSP_PCR_FSXM |
-                        DAVINCI_MCBSP_PCR_FSRM;
+                pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM;
+                /*
+                 * Selection of the clock input pin that is the
+                 * input for the Sample Rate Generator.
+                 * McBSP FSR and FSX are driven by the Sample Rate
+                 * Generator.
+                 */
+                switch (dev->clk_input_pin) {
+                case MCBSP_CLKS:
+                        pcr |= DAVINCI_MCBSP_PCR_CLKXM |
+                                DAVINCI_MCBSP_PCR_CLKRM;
+                        break;
+                case MCBSP_CLKR:
+                        pcr |= DAVINCI_MCBSP_PCR_SCLKME;
+                        break;
+                default:
+                        dev_err(dev->dev, "bad clk_input_pin\n");
+                        return -EINVAL;
+                }
+
                 break;
         case SND_SOC_DAIFMT_CBM_CFM:
                 /* codec is master */
@@ -372,6 +401,18 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
         return 0;
 }
 
+static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
+                                int div_id, int div)
+{
+        struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
+
+        if (div_id != DAVINCI_MCBSP_CLKGDV)
+                return -ENODEV;
+
+        dev->clk_div = div;
+        return 0;
+}
+
 static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
                                  struct snd_pcm_hw_params *params,
                                  struct snd_soc_dai *dai)
@@ -380,8 +421,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
         struct davinci_pcm_dma_params *dma_params =
                                         &dev->dma_params[substream->stream];
         struct snd_interval *i = NULL;
-        int mcbsp_word_length;
-        unsigned int rcr, xcr, srgr;
+        int mcbsp_word_length, master;
+        unsigned int rcr, xcr, srgr, clk_div, freq, framesize;
         u32 spcr;
         snd_pcm_format_t fmt;
         unsigned element_cnt = 1;
@@ -396,12 +437,59 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
                 davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
         }
 
-        i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
-        srgr = DAVINCI_MCBSP_SRGR_FSGM;
-        srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1);
+        master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK;
+        fmt = params_format(params);
+        mcbsp_word_length = asp_word_length[fmt];
 
-        i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
-        srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1);
+        switch (master) {
+        case SND_SOC_DAIFMT_CBS_CFS:
+                freq = clk_get_rate(dev->clk);
+                srgr = DAVINCI_MCBSP_SRGR_FSGM |
+                       DAVINCI_MCBSP_SRGR_CLKSM;
+                srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length *
+                                                8 - 1);
+                if (dev->i2s_accurate_sck) {
+                        clk_div = 256;
+                        do {
+                                framesize = (freq / (--clk_div)) /
+                                params->rate_num *
+                                        params->rate_den;
+                        } while (((framesize < 33) || (framesize > 4095)) &&
+                                 (clk_div));
+                        clk_div--;
+                        srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1);
+                } else {
+                        /* symmetric waveforms */
+                        clk_div = freq / (mcbsp_word_length * 16) /
+                                  params->rate_num * params->rate_den;
+                        srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length *
+                                                        16 - 1);
+                }
+                clk_div &= 0xFF;
+                srgr |= clk_div;
+                break;
+        case SND_SOC_DAIFMT_CBM_CFS:
+                srgr = DAVINCI_MCBSP_SRGR_FSGM;
+                clk_div = dev->clk_div - 1;
+                srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1);
+                srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1);
+                clk_div &= 0xFF;
+                srgr |= clk_div;
+                break;
+        case SND_SOC_DAIFMT_CBM_CFM:
+                /* Clock and frame sync given from external sources */
+                i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
+                srgr = DAVINCI_MCBSP_SRGR_FSGM;
+                srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1);
+                pr_debug("%s - %d  FWID set: re-read srgr = %X\n",
+                        __func__, __LINE__, snd_interval_value(i) - 1);
+
+                i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
+                srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1);
+                break;
+        default:
+                return -EINVAL;
+        }
         davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
 
         rcr = DAVINCI_MCBSP_RCR_RFIG;
@@ -426,12 +514,41 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
                         element_cnt = 1;
                         fmt = double_fmt[fmt];
                 }
+                switch (master) {
+                case SND_SOC_DAIFMT_CBS_CFS:
+                case SND_SOC_DAIFMT_CBS_CFM:
+                        rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0);
+                        xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0);
+                        rcr |= DAVINCI_MCBSP_RCR_RPHASE;
+                        xcr |= DAVINCI_MCBSP_XCR_XPHASE;
+                        break;
+                case SND_SOC_DAIFMT_CBM_CFM:
+                case SND_SOC_DAIFMT_CBM_CFS:
+                        rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1);
+                        xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1);
+                        break;
+                default:
+                        return -EINVAL;
+                }
         }
         dma_params->acnt = dma_params->data_type = data_type[fmt];
         dma_params->fifo_level = 0;
         mcbsp_word_length = asp_word_length[fmt];
-        rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1);
-        xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1);
+
+        switch (master) {
+        case SND_SOC_DAIFMT_CBS_CFS:
+        case SND_SOC_DAIFMT_CBS_CFM:
+                rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0);
+                xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0);
+                break;
+        case SND_SOC_DAIFMT_CBM_CFM:
+        case SND_SOC_DAIFMT_CBM_CFS:
+                rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1);
+                xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1);
+                break;
+        default:
+                return -EINVAL;
+        }
 
         rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
                 DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length);
@@ -442,6 +559,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
                 davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr);
         else
                 davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr);
+
+        pr_debug("%s - %d  srgr=%X\n", __func__, __LINE__, srgr);
+        pr_debug("%s - %d  xcr=%X\n", __func__, __LINE__, xcr);
+        pr_debug("%s - %d  rcr=%X\n", __func__, __LINE__, rcr);
         return 0;
 }
 
@@ -500,6 +621,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
         .trigger        = davinci_i2s_trigger,
         .hw_params      = davinci_i2s_hw_params,
         .set_fmt        = davinci_i2s_set_dai_fmt,
+        .set_clkdiv     = davinci_i2s_dai_set_clkdiv,
 
 };
 
@@ -526,6 +648,8 @@ static int davinci_i2s_probe(struct platform_device *pdev)
         struct snd_platform_data *pdata = pdev->dev.platform_data;
         struct davinci_mcbsp_dev *dev;
         struct resource *mem, *ioarea, *res;
+        enum dma_event_q asp_chan_q = EVENTQ_0;
+        enum dma_event_q ram_chan_q = EVENTQ_1;
         int ret;
 
         mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -552,7 +676,17 @@ static int davinci_i2s_probe(struct platform_device *pdev)
                         pdata->sram_size_playback;
                 dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size =
                         pdata->sram_size_capture;
+                dev->clk_input_pin = pdata->clk_input_pin;
+                dev->i2s_accurate_sck = pdata->i2s_accurate_sck;
+                asp_chan_q = pdata->asp_chan_q;
+                ram_chan_q = pdata->ram_chan_q;
         }
+
+        dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q   = asp_chan_q;
+        dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q   = ram_chan_q;
+        dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q    = asp_chan_q;
+        dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q    = ram_chan_q;
+
         dev->clk = clk_get(&pdev->dev, NULL);
         if (IS_ERR(dev->clk)) {
                 ret = -ENODEV;
@@ -584,6 +718,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
                 goto err_free_mem;
         }
         dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
+        dev->dev = &pdev->dev;
 
         davinci_i2s_dai.private_data = dev;
         davinci_i2s_dai.capture.dma_data = dev->dma_params;
diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h
index 241648ce8873..0b1e77b8c279 100644
--- a/sound/soc/davinci/davinci-i2s.h
+++ b/sound/soc/davinci/davinci-i2s.h
@@ -12,6 +12,11 @@
 #ifndef _DAVINCI_I2S_H
 #define _DAVINCI_I2S_H
 
+/* McBSP dividers */
+enum davinci_mcbsp_div {
+        DAVINCI_MCBSP_CLKGDV,              /* Sample rate generator divider */
+};
+
 extern struct snd_soc_dai davinci_i2s_dai;
 
 #endif
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index d3955096d872..b24720894af6 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -890,7 +890,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
         dev->rxnumevt = pdata->rxnumevt;
 
         dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
-        dma_data->eventq_no = pdata->eventq_no;
+        dma_data->asp_chan_q = pdata->asp_chan_q;
+        dma_data->ram_chan_q = pdata->ram_chan_q;
         dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
                                                         io_v2p(dev->base));
 
@@ -904,7 +905,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
         dma_data->channel = res->start;
 
         dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
-        dma_data->eventq_no = pdata->eventq_no;
+        dma_data->asp_chan_q = pdata->asp_chan_q;
+        dma_data->ram_chan_q = pdata->ram_chan_q;
         dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
                                                         io_v2p(dev->base));
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 2dc406f42fe7..a7124116d2e0 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -381,7 +381,7 @@ static int request_ping_pong(struct snd_pcm_substream *substream,
         /* Request ram master channel */
         link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY,
                                   davinci_pcm_dma_irq, substream,
-                                  EVENTQ_1);
+                                  prtd->params->ram_chan_q);
         if (link < 0)
                 goto exit1;
 
@@ -477,7 +477,8 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 
         /* Request asp master DMA channel */
         link = prtd->asp_channel = edma_alloc_channel(params->channel,
-                        davinci_pcm_dma_irq, substream, EVENTQ_0);
+                        davinci_pcm_dma_irq, substream,
+                        prtd->params->asp_chan_q);
         if (link < 0)
                 goto exit1;
 
@@ -800,7 +801,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm)
                 dma_free_writecombine(pcm->card->dev, buf->bytes,
                                       buf->area, buf->addr);
                 buf->area = NULL;
-                iram_dma = (struct snd_dma_buffer *)buf->private_data;
+                iram_dma = buf->private_data;
                 if (iram_dma) {
                         sram_free(iram_dma->area, iram_dma->bytes);
                         kfree(iram_dma);
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 0764944cf10f..b799a02333d8 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -21,7 +21,8 @@ struct davinci_pcm_dma_params {
         unsigned short acnt;
         dma_addr_t dma_addr;            /* device physical address for DMA */
         unsigned sram_size;
-        enum dma_event_q eventq_no;     /* event queue number */
+        enum dma_event_q asp_chan_q;    /* event queue number for ASP channel */
+        enum dma_event_q ram_chan_q;    /* event queue number for RAM channel */
         unsigned char data_type;        /* xfer data type */
         unsigned char convert_mono_stereo;
         unsigned int fifo_level;
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index 9aa980d38231..48678533da7a 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -203,7 +203,7 @@ static int davinci_vcif_probe(struct platform_device *pdev)
         int ret;
 
         davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL);
-        if (!davinci_vc) {
+        if (!davinci_vcif_dev) {
                 dev_dbg(&pdev->dev,
                         "could not allocate memory for private data\n");
                 return -ENOMEM;
diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig
new file mode 100644
index 000000000000..f617f560f46b
--- /dev/null
+++ b/sound/soc/ep93xx/Kconfig
@@ -0,0 +1,18 @@
+config SND_EP93XX_SOC
+        tristate "SoC Audio support for the Cirrus Logic EP93xx series"
+        depends on ARCH_EP93XX && SND_SOC
+        help
+          Say Y or M if you want to add support for codecs attached to
+          the EP93xx I2S interface.
+
+config SND_EP93XX_SOC_I2S
+        tristate
+
+config SND_EP93XX_SOC_SNAPPERCL15
+        tristate "SoC Audio support for Bluewater Systems Snapper CL15 module"
+        depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15
+        select SND_EP93XX_SOC_I2S
+        select SND_SOC_TLV320AIC23
+        help
+          Say Y or M here if you want to add support for I2S audio on the
+          Bluewater Systems Snapper CL15 module.
diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile
new file mode 100644
index 000000000000..272e60f57b9a
--- /dev/null
+++ b/sound/soc/ep93xx/Makefile
@@ -0,0 +1,11 @@
+# EP93xx Platform Support
+snd-soc-ep93xx-objs                             := ep93xx-pcm.o
+snd-soc-ep93xx-i2s-objs                         := ep93xx-i2s.o
+
+obj-$(CONFIG_SND_EP93XX_SOC)                    += snd-soc-ep93xx.o
+obj-$(CONFIG_SND_EP93XX_SOC_I2S)                += snd-soc-ep93xx-i2s.o
+
+# EP93XX Machine Support
+snd-soc-snappercl15-objs                        := snappercl15.o
+
+obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15)        += snd-soc-snappercl15.o
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
new file mode 100644
index 000000000000..00b946632184
--- /dev/null
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -0,0 +1,487 @@
+/*
+ * linux/sound/soc/ep93xx-i2s.c
+ * EP93xx I2S driver
+ *
+ * Copyright (C) 2010 Ryan Mallon <ryan@bluewatersys.com>
+ *
+ * Based on the original driver by:
+ *   Copyright (C) 2007 Chase Douglas <chasedouglas@gmail>
+ *   Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <mach/hardware.h>
+#include <mach/ep93xx-regs.h>
+#include <mach/dma.h>
+
+#include "ep93xx-pcm.h"
+#include "ep93xx-i2s.h"
+
+#define EP93XX_I2S_TXCLKCFG             0x00
+#define EP93XX_I2S_RXCLKCFG             0x04
+#define EP93XX_I2S_GLCTRL               0x0C
+
+#define EP93XX_I2S_TXLINCTRLDATA        0x28
+#define EP93XX_I2S_TXCTRL               0x2C
+#define EP93XX_I2S_TXWRDLEN             0x30
+#define EP93XX_I2S_TX0EN                0x34
+
+#define EP93XX_I2S_RXLINCTRLDATA        0x58
+#define EP93XX_I2S_RXCTRL               0x5C
+#define EP93XX_I2S_RXWRDLEN             0x60
+#define EP93XX_I2S_RX0EN                0x64
+
+#define EP93XX_I2S_WRDLEN_16            (0 << 0)
+#define EP93XX_I2S_WRDLEN_24            (1 << 0)
+#define EP93XX_I2S_WRDLEN_32            (2 << 0)
+
+#define EP93XX_I2S_LINCTRLDATA_R_JUST   (1 << 2) /* Right justify */
+
+#define EP93XX_I2S_CLKCFG_LRS           (1 << 0) /* lrclk polarity */
+#define EP93XX_I2S_CLKCFG_CKP           (1 << 1) /* Bit clock polarity */
+#define EP93XX_I2S_CLKCFG_REL           (1 << 2) /* First bit transition */
+#define EP93XX_I2S_CLKCFG_MASTER        (1 << 3) /* Master mode */
+#define EP93XX_I2S_CLKCFG_NBCG          (1 << 4) /* Not bit clock gating */
+
+struct ep93xx_i2s_info {
+        struct clk                      *mclk;
+        struct clk                      *sclk;
+        struct clk                      *lrclk;
+        struct ep93xx_pcm_dma_params    *dma_params;
+        struct resource                 *mem;
+        void __iomem                    *regs;
+};
+
+struct ep93xx_pcm_dma_params ep93xx_i2s_dma_params[] = {
+        [SNDRV_PCM_STREAM_PLAYBACK] = {
+                .name           = "i2s-pcm-out",
+                .dma_port       = EP93XX_DMA_M2P_PORT_I2S1,
+        },
+        [SNDRV_PCM_STREAM_CAPTURE] = {
+                .name           = "i2s-pcm-in",
+                .dma_port       = EP93XX_DMA_M2P_PORT_I2S1,
+        },
+};
+
+static inline void ep93xx_i2s_write_reg(struct ep93xx_i2s_info *info,
+                                        unsigned reg, unsigned val)
+{
+        __raw_writel(val, info->regs + reg);
+}
+
+static inline unsigned ep93xx_i2s_read_reg(struct ep93xx_i2s_info *info,
+                                           unsigned reg)
+{
+        return __raw_readl(info->regs + reg);
+}
+
+static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream)
+{
+        unsigned base_reg;
+        int i;
+
+        if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 &&
+            (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) {
+                /* Enable clocks */
+                clk_enable(info->mclk);
+                clk_enable(info->sclk);
+                clk_enable(info->lrclk);
+
+                /* Enable i2s */
+                ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 1);
+        }
+
+        /* Enable fifos */
+        if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+                base_reg = EP93XX_I2S_TX0EN;
+        else
+                base_reg = EP93XX_I2S_RX0EN;
+        for (i = 0; i < 3; i++)
+                ep93xx_i2s_write_reg(info, base_reg + (i * 4), 1);
+}
+
+static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream)
+{
+        unsigned base_reg;
+        int i;
+
+        /* Disable fifos */
+        if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+                base_reg = EP93XX_I2S_TX0EN;
+        else
+                base_reg = EP93XX_I2S_RX0EN;
+        for (i = 0; i < 3; i++)
+                ep93xx_i2s_write_reg(info, base_reg + (i * 4), 0);
+
+        if ((ep93xx_i2s_read_reg(info, EP93XX_I2S_TX0EN) & 0x1) == 0 &&
+            (ep93xx_i2s_read_reg(info, EP93XX_I2S_RX0EN) & 0x1) == 0) {
+                /* Disable i2s */
+                ep93xx_i2s_write_reg(info, EP93XX_I2S_GLCTRL, 0);
+
+                /* Disable clocks */
+                clk_disable(info->lrclk);
+                clk_disable(info->sclk);
+                clk_disable(info->mclk);
+        }
+}
+
+static int ep93xx_i2s_startup(struct snd_pcm_substream *substream,
+                              struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        struct ep93xx_i2s_info *info = rtd->dai->cpu_dai->private_data;
+
+        snd_soc_dai_set_dma_data(cpu_dai, substream,
+                                 &info->dma_params[substream->stream]);
+        return 0;
+}
+
+static void ep93xx_i2s_shutdown(struct snd_pcm_substream *substream,
+                                struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct ep93xx_i2s_info *info = rtd->dai->cpu_dai->private_data;
+
+        ep93xx_i2s_disable(info, substream->stream);
+}
+
+static int ep93xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+                                  unsigned int fmt)
+{
+        struct ep93xx_i2s_info *info = cpu_dai->private_data;
+        unsigned int clk_cfg, lin_ctrl;
+
+        clk_cfg  = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXCLKCFG);
+        lin_ctrl = ep93xx_i2s_read_reg(info, EP93XX_I2S_RXLINCTRLDATA);
+
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_I2S:
+                clk_cfg |= EP93XX_I2S_CLKCFG_REL;
+                lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST;
+                break;
+
+        case SND_SOC_DAIFMT_LEFT_J:
+                clk_cfg &= ~EP93XX_I2S_CLKCFG_REL;
+                lin_ctrl &= ~EP93XX_I2S_LINCTRLDATA_R_JUST;
+                break;
+
+        case SND_SOC_DAIFMT_RIGHT_J:
+                clk_cfg &= ~EP93XX_I2S_CLKCFG_REL;
+                lin_ctrl |= EP93XX_I2S_LINCTRLDATA_R_JUST;
+                break;
+
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBS_CFS:
+                /* CPU is master */
+                clk_cfg |= EP93XX_I2S_CLKCFG_MASTER;
+                break;
+
+        case SND_SOC_DAIFMT_CBM_CFM:
+                /* Codec is master */
+                clk_cfg &= ~EP93XX_I2S_CLKCFG_MASTER;
+                break;
+
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                /* Negative bit clock, lrclk low on left word */
+                clk_cfg &= ~(EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL);
+                break;
+
+        case SND_SOC_DAIFMT_NB_IF:
+                /* Negative bit clock, lrclk low on right word */
+                clk_cfg &= ~EP93XX_I2S_CLKCFG_CKP;
+                clk_cfg |= EP93XX_I2S_CLKCFG_REL;
+                break;
+
+        case SND_SOC_DAIFMT_IB_NF:
+                /* Positive bit clock, lrclk low on left word */
+                clk_cfg |= EP93XX_I2S_CLKCFG_CKP;
+                clk_cfg &= ~EP93XX_I2S_CLKCFG_REL;
+                break;
+
+        case SND_SOC_DAIFMT_IB_IF:
+                /* Positive bit clock, lrclk low on right word */
+                clk_cfg |= EP93XX_I2S_CLKCFG_CKP | EP93XX_I2S_CLKCFG_REL;
+                break;
+        }
+
+        /* Write new register values */
+        ep93xx_i2s_write_reg(info, EP93XX_I2S_RXCLKCFG, clk_cfg);
+        ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCLKCFG, clk_cfg);
+        ep93xx_i2s_write_reg(info, EP93XX_I2S_RXLINCTRLDATA, lin_ctrl);
+        ep93xx_i2s_write_reg(info, EP93XX_I2S_TXLINCTRLDATA, lin_ctrl);
+        return 0;
+}
+
+static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
+                                struct snd_pcm_hw_params *params,
+                                struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        struct ep93xx_i2s_info *info = cpu_dai->private_data;
+        unsigned word_len, div, sdiv, lrdiv;
+        int found = 0, err;
+
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                word_len = EP93XX_I2S_WRDLEN_16;
+                break;
+
+        case SNDRV_PCM_FORMAT_S24_LE:
+                word_len = EP93XX_I2S_WRDLEN_24;
+                break;
+
+        case SNDRV_PCM_FORMAT_S32_LE:
+                word_len = EP93XX_I2S_WRDLEN_32;
+                break;
+
+        default:
+                return -EINVAL;
+        }
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                ep93xx_i2s_write_reg(info, EP93XX_I2S_TXWRDLEN, word_len);
+        else
+                ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len);
+
+        /*
+         * Calculate the sdiv (bit clock) and lrdiv (left/right clock) values.
+         * If the lrclk is pulse length is larger than the word size, then the
+         * bit clock will be gated for the unused bits.
+         */
+        div = (clk_get_rate(info->mclk) / params_rate(params)) *
+                params_channels(params);
+        for (sdiv = 2; sdiv <= 4; sdiv += 2)
+                for (lrdiv = 32; lrdiv <= 128; lrdiv <<= 1)
+                        if (sdiv * lrdiv == div) {
+                                found = 1;
+                                goto out;
+                        }
+out:
+        if (!found)
+                return -EINVAL;
+
+        err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv);
+        if (err)
+                return err;
+
+        err = clk_set_rate(info->lrclk, clk_get_rate(info->sclk) / lrdiv);
+        if (err)
+                return err;
+
+        ep93xx_i2s_enable(info, substream->stream);
+        return 0;
+}
+
+static int ep93xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
+                                 unsigned int freq, int dir)
+{
+        struct ep93xx_i2s_info *info = cpu_dai->private_data;
+
+        if (dir == SND_SOC_CLOCK_IN || clk_id != 0)
+                return -EINVAL;
+
+        return clk_set_rate(info->mclk, freq);
+}
+
+#ifdef CONFIG_PM
+static int ep93xx_i2s_suspend(struct snd_soc_dai *dai)
+{
+        struct ep93xx_i2s_info *info = dai->private_data;
+
+        if (!dai->active)
+                return;
+
+        ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK);
+        ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE);
+}
+
+static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
+{
+        struct ep93xx_i2s_info *info = dai->private_data;
+
+        if (!dai->active)
+                return;
+
+        ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK);
+        ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE);
+}
+#else
+#define ep93xx_i2s_suspend      NULL
+#define ep93xx_i2s_resume       NULL
+#endif
+
+static struct snd_soc_dai_ops ep93xx_i2s_dai_ops = {
+        .startup        = ep93xx_i2s_startup,
+        .shutdown       = ep93xx_i2s_shutdown,
+        .hw_params      = ep93xx_i2s_hw_params,
+        .set_sysclk     = ep93xx_i2s_set_sysclk,
+        .set_fmt        = ep93xx_i2s_set_dai_fmt,
+};
+
+#define EP93XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+                            SNDRV_PCM_FMTBIT_S24_LE | \
+                            SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai ep93xx_i2s_dai = {
+        .name           = "ep93xx-i2s",
+        .id             = 0,
+        .symmetric_rates= 1,
+        .suspend        = ep93xx_i2s_suspend,
+        .resume         = ep93xx_i2s_resume,
+        .playback       = {
+                .channels_min   = 2,
+                .channels_max   = 2,
+                .rates          = SNDRV_PCM_RATE_8000_48000,
+                .formats        = EP93XX_I2S_FORMATS,
+        },
+        .capture        = {
+                 .channels_min  = 2,
+                 .channels_max  = 2,
+                 .rates         = SNDRV_PCM_RATE_8000_48000,
+                 .formats       = EP93XX_I2S_FORMATS,
+        },
+        .ops            = &ep93xx_i2s_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ep93xx_i2s_dai);
+
+static int ep93xx_i2s_probe(struct platform_device *pdev)
+{
+        struct ep93xx_i2s_info *info;
+        struct resource *res;
+        int err;
+
+        info = kzalloc(sizeof(struct ep93xx_i2s_info), GFP_KERNEL);
+        if (!info) {
+                err = -ENOMEM;
+                goto fail;
+        }
+
+        ep93xx_i2s_dai.dev = &pdev->dev;
+        ep93xx_i2s_dai.private_data = info;
+        info->dma_params = ep93xx_i2s_dma_params;
+
+        res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!res) {
+                err = -ENODEV;
+                goto fail;
+        }
+
+        info->mem = request_mem_region(res->start, resource_size(res),
+                                       pdev->name);
+        if (!info->mem) {
+                err = -EBUSY;
+                goto fail;
+        }
+
+        info->regs = ioremap(info->mem->start, resource_size(info->mem));
+        if (!info->regs) {
+                err = -ENXIO;
+                goto fail_release_mem;
+        }
+
+        info->mclk = clk_get(&pdev->dev, "mclk");
+        if (IS_ERR(info->mclk)) {
+                err = PTR_ERR(info->mclk);
+                goto fail_unmap_mem;
+        }
+
+        info->sclk = clk_get(&pdev->dev, "sclk");
+        if (IS_ERR(info->sclk)) {
+                err = PTR_ERR(info->sclk);
+                goto fail_put_mclk;
+        }
+
+        info->lrclk = clk_get(&pdev->dev, "lrclk");
+        if (IS_ERR(info->lrclk)) {
+                err = PTR_ERR(info->lrclk);
+                goto fail_put_sclk;
+        }
+
+        err = snd_soc_register_dai(&ep93xx_i2s_dai);
+        if (err)
+                goto fail_put_lrclk;
+
+        return 0;
+
+fail_put_lrclk:
+        clk_put(info->lrclk);
+fail_put_sclk:
+        clk_put(info->sclk);
+fail_put_mclk:
+        clk_put(info->mclk);
+fail_unmap_mem:
+        iounmap(info->regs);
+fail_release_mem:
+        release_mem_region(info->mem->start, resource_size(info->mem));
+        kfree(info);
+fail:
+        return err;
+}
+
+static int __devexit ep93xx_i2s_remove(struct platform_device *pdev)
+{
+        struct ep93xx_i2s_info *info = ep93xx_i2s_dai.private_data;
+
+        snd_soc_unregister_dai(&ep93xx_i2s_dai);
+        clk_put(info->lrclk);
+        clk_put(info->sclk);
+        clk_put(info->mclk);
+        iounmap(info->regs);
+        release_mem_region(info->mem->start, resource_size(info->mem));
+        kfree(info);
+        return 0;
+}
+
+static struct platform_driver ep93xx_i2s_driver = {
+        .probe  = ep93xx_i2s_probe,
+        .remove = __devexit_p(ep93xx_i2s_remove),
+        .driver = {
+                .name   = "ep93xx-i2s",
+                .owner  = THIS_MODULE,
+        },
+};
+
+static int __init ep93xx_i2s_init(void)
+{
+        return platform_driver_register(&ep93xx_i2s_driver);
+}
+
+static void __exit ep93xx_i2s_exit(void)
+{
+        platform_driver_unregister(&ep93xx_i2s_driver);
+}
+
+module_init(ep93xx_i2s_init);
+module_exit(ep93xx_i2s_exit);
+
+MODULE_ALIAS("platform:ep93xx-i2s");
+MODULE_AUTHOR("Ryan Mallon <ryan@bluewatersys.com>");
+MODULE_DESCRIPTION("EP93XX I2S driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ep93xx/ep93xx-i2s.h b/sound/soc/ep93xx/ep93xx-i2s.h
new file mode 100644
index 000000000000..3bd4ebfaa1de
--- /dev/null
+++ b/sound/soc/ep93xx/ep93xx-i2s.h
@@ -0,0 +1,18 @@
+/*
+ * linux/sound/soc/ep93xx-i2s.h
+ * EP93xx I2S driver
+ *
+ * Copyright (C) 2010 Ryan Mallon <ryan@bluewatersys.com>
+ *  
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _EP93XX_SND_SOC_I2S_H
+#define _EP93XX_SND_SOC_I2S_H
+
+extern struct snd_soc_dai ep93xx_i2s_dai;
+
+#endif /* _EP93XX_SND_SOC_I2S_H */
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
new file mode 100644
index 000000000000..4ba938400791
--- /dev/null
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -0,0 +1,319 @@
+/*
+ * linux/sound/arm/ep93xx-pcm.c - EP93xx ALSA PCM interface
+ *
+ * Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org>
+ * Copyright (C) 2006 Applied Data Systems
+ *
+ * Rewritten for the SoC audio subsystem (Based on PXA2xx code):
+ *   Copyright (c) 2008 Ryan Mallon <ryan@bluewatersys.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/init.h>
+#include <linux/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 <mach/dma.h>
+#include <mach/hardware.h>
+#include <mach/ep93xx-regs.h>
+
+#include "ep93xx-pcm.h"
+
+static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
+        .info                   = (SNDRV_PCM_INFO_MMAP          |
+                                   SNDRV_PCM_INFO_MMAP_VALID    |
+                                   SNDRV_PCM_INFO_INTERLEAVED   |
+                                   SNDRV_PCM_INFO_BLOCK_TRANSFER),
+                                   
+        .rates                  = SNDRV_PCM_RATE_8000_48000,
+        .rate_min               = SNDRV_PCM_RATE_8000,
+        .rate_max               = SNDRV_PCM_RATE_48000,
+        
+        .formats                = (SNDRV_PCM_FMTBIT_S16_LE |
+                                   SNDRV_PCM_FMTBIT_S24_LE |
+                                   SNDRV_PCM_FMTBIT_S32_LE),
+        
+        .buffer_bytes_max       = 131072,
+        .period_bytes_min       = 32,
+        .period_bytes_max       = 32768,
+        .periods_min            = 1,
+        .periods_max            = 32,
+        .fifo_size              = 32,
+};
+
+struct ep93xx_runtime_data
+{
+        struct ep93xx_dma_m2p_client    cl;
+        struct ep93xx_pcm_dma_params    *params;
+        int                             pointer_bytes;
+        struct tasklet_struct           period_tasklet;
+        int                             periods;
+        struct ep93xx_dma_buffer        buf[32];
+};
+
+static void ep93xx_pcm_period_elapsed(unsigned long data)
+{
+        struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
+        snd_pcm_period_elapsed(substream);
+}
+
+static void ep93xx_pcm_buffer_started(void *cookie,
+                                      struct ep93xx_dma_buffer *buf)
+{
+}
+
+static void ep93xx_pcm_buffer_finished(void *cookie, 
+                                       struct ep93xx_dma_buffer *buf, 
+                                       int bytes, int error)
+{
+        struct snd_pcm_substream *substream = cookie;
+        struct ep93xx_runtime_data *rtd = substream->runtime->private_data;
+
+        if (buf == rtd->buf + rtd->periods - 1)
+                rtd->pointer_bytes = 0;
+        else
+                rtd->pointer_bytes += buf->size;
+
+        if (!error) {
+                ep93xx_dma_m2p_submit_recursive(&rtd->cl, buf);
+                tasklet_schedule(&rtd->period_tasklet);
+        } else {
+                snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+        }
+}
+
+static int ep93xx_pcm_open(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *soc_rtd = substream->private_data;
+        struct snd_soc_dai *cpu_dai = soc_rtd->dai->cpu_dai;
+        struct ep93xx_pcm_dma_params *dma_params;
+        struct ep93xx_runtime_data *rtd;    
+        int ret;
+
+        dma_params = snd_soc_dai_get_dma_data(cpu_dai, substream);
+        snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware);
+
+        rtd = kmalloc(sizeof(*rtd), GFP_KERNEL);
+        if (!rtd) 
+                return -ENOMEM;
+
+        memset(&rtd->period_tasklet, 0, sizeof(rtd->period_tasklet));
+        rtd->period_tasklet.func = ep93xx_pcm_period_elapsed;
+        rtd->period_tasklet.data = (unsigned long)substream;
+
+        rtd->cl.name = dma_params->name;
+        rtd->cl.flags = dma_params->dma_port | EP93XX_DMA_M2P_IGNORE_ERROR |
+                ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+                 EP93XX_DMA_M2P_TX : EP93XX_DMA_M2P_RX);
+        rtd->cl.cookie = substream;
+        rtd->cl.buffer_started = ep93xx_pcm_buffer_started;
+        rtd->cl.buffer_finished = ep93xx_pcm_buffer_finished;
+        ret = ep93xx_dma_m2p_client_register(&rtd->cl);
+        if (ret < 0) {
+                kfree(rtd);
+                return ret;
+        }
+        
+        substream->runtime->private_data = rtd;
+        return 0;
+}
+
+static int ep93xx_pcm_close(struct snd_pcm_substream *substream)
+{
+        struct ep93xx_runtime_data *rtd = substream->runtime->private_data;
+
+        ep93xx_dma_m2p_client_unregister(&rtd->cl);
+        kfree(rtd);
+        return 0;
+}
+
+static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream,
+                                struct snd_pcm_hw_params *params)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct ep93xx_runtime_data *rtd = runtime->private_data;
+        size_t totsize = params_buffer_bytes(params);
+        size_t period = params_period_bytes(params);
+        int i;
+
+        snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+        runtime->dma_bytes = totsize;
+
+        rtd->periods = (totsize + period - 1) / period;
+        for (i = 0; i < rtd->periods; i++) {
+                rtd->buf[i].bus_addr = runtime->dma_addr + (i * period);
+                rtd->buf[i].size = period;
+                if ((i + 1) * period > totsize)
+                        rtd->buf[i].size = totsize - (i * period);
+        }
+
+        return 0;
+}
+
+static int ep93xx_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        snd_pcm_set_runtime_buffer(substream, NULL);
+        return 0;
+}
+
+static int ep93xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct ep93xx_runtime_data *rtd = substream->runtime->private_data;
+        int ret;
+        int i;
+
+        ret = 0;
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                rtd->pointer_bytes = 0;
+                for (i = 0; i < rtd->periods; i++)
+                        ep93xx_dma_m2p_submit(&rtd->cl, rtd->buf + i);
+                break;
+
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                ep93xx_dma_m2p_flush(&rtd->cl);
+                break;
+
+        default:
+                ret = -EINVAL;
+                break;
+        }
+
+        return ret;
+}
+
+static snd_pcm_uframes_t ep93xx_pcm_pointer(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct ep93xx_runtime_data *rtd = substream->runtime->private_data;
+
+        /* FIXME: implement this with sub-period granularity */
+        return bytes_to_frames(runtime, rtd->pointer_bytes);
+}
+
+static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream,
+                           struct vm_area_struct *vma)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+
+        return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+                                     runtime->dma_area,
+                                     runtime->dma_addr,
+                                     runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops ep93xx_pcm_ops = {
+        .open           = ep93xx_pcm_open,
+        .close          = ep93xx_pcm_close,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .hw_params      = ep93xx_pcm_hw_params,
+        .hw_free        = ep93xx_pcm_hw_free,
+        .trigger        = ep93xx_pcm_trigger,
+        .pointer        = ep93xx_pcm_pointer,
+        .mmap           = ep93xx_pcm_mmap,
+};
+
+static int ep93xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+        struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+        struct snd_dma_buffer *buf = &substream->dma_buffer;
+        size_t size = ep93xx_pcm_hardware.buffer_bytes_max;
+
+        buf->dev.type = SNDRV_DMA_TYPE_DEV;
+        buf->dev.dev = pcm->card->dev;
+        buf->private_data = NULL;
+        buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+                                           &buf->addr, GFP_KERNEL);
+        buf->bytes = size;
+
+        return (buf->area == NULL) ? -ENOMEM : 0;
+}
+
+static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+        struct snd_pcm_substream *substream;
+        struct snd_dma_buffer *buf;
+        int stream;
+
+        for (stream = 0; stream < 2; stream++) {                
+                substream = pcm->streams[stream].substream;
+                if (!substream)
+                        continue;
+                
+                buf = &substream->dma_buffer;
+                if (!buf->area)
+                        continue;
+
+                dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area,
+                                      buf->addr);
+                buf->area = NULL;
+        }
+}
+
+static u64 ep93xx_pcm_dmamask = 0xffffffff;
+
+static int ep93xx_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 = &ep93xx_pcm_dmamask;
+        if (!card->dev->coherent_dma_mask)
+                card->dev->coherent_dma_mask = 0xffffffff;
+
+        if (dai->playback.channels_min) {
+                ret = ep93xx_pcm_preallocate_dma_buffer(pcm,
+                                        SNDRV_PCM_STREAM_PLAYBACK);
+                if (ret)
+                        return ret;
+        }
+
+        if (dai->capture.channels_min) {
+                ret = ep93xx_pcm_preallocate_dma_buffer(pcm,
+                                        SNDRV_PCM_STREAM_CAPTURE);
+                if (ret)
+                        return ret;
+        }
+
+        return 0;
+}
+
+struct snd_soc_platform ep93xx_soc_platform = {
+        .name           = "ep93xx-audio",
+        .pcm_ops        = &ep93xx_pcm_ops,
+        .pcm_new        = &ep93xx_pcm_new,
+        .pcm_free       = &ep93xx_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(ep93xx_soc_platform);
+
+static int __init ep93xx_soc_platform_init(void)
+{
+        return snd_soc_register_platform(&ep93xx_soc_platform);
+}
+
+static void __exit ep93xx_soc_platform_exit(void)
+{
+        snd_soc_unregister_platform(&ep93xx_soc_platform);
+}
+
+module_init(ep93xx_soc_platform_init);
+module_exit(ep93xx_soc_platform_exit);
+
+MODULE_AUTHOR("Ryan Mallon <ryan@bluewatersys.com>");
+MODULE_DESCRIPTION("EP93xx ALSA PCM interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ep93xx/ep93xx-pcm.h b/sound/soc/ep93xx/ep93xx-pcm.h
new file mode 100644
index 000000000000..4ffdd3f62fe9
--- /dev/null
+++ b/sound/soc/ep93xx/ep93xx-pcm.h
@@ -0,0 +1,22 @@
+/*
+ * sound/soc/ep93xx/ep93xx-pcm.h - EP93xx ALSA PCM interface
+ *
+ * Copyright (C) 2006 Lennert Buytenhek <buytenh@wantstofly.org>
+ * Copyright (C) 2006 Applied Data Systems
+ *
+ * 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 _EP93XX_SND_SOC_PCM_H
+#define _EP93XX_SND_SOC_PCM_H
+
+struct ep93xx_pcm_dma_params {
+        char    *name;
+        int     dma_port;
+};
+
+extern struct snd_soc_platform ep93xx_soc_platform;
+
+#endif /* _EP93XX_SND_SOC_PCM_H */
diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c
new file mode 100644
index 000000000000..64955340ff75
--- /dev/null
+++ b/sound/soc/ep93xx/snappercl15.c
@@ -0,0 +1,150 @@
+/*
+ * snappercl15.c -- SoC audio for Bluewater Systems Snapper CL15 module
+ *
+ * Copyright (C) 2008 Bluewater Systems Ltd
+ * Author: Ryan Mallon <ryan@bluewatersys.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/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "ep93xx-pcm.h"
+#include "ep93xx-i2s.h"
+
+#define CODEC_CLOCK 5644800
+
+static int snappercl15_hw_params(struct snd_pcm_substream *substream,
+                                 struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        int err;
+
+        err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_CBS_CFS);
+
+        err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | 
+                                  SND_SOC_DAIFMT_NB_IF |                  
+                                  SND_SOC_DAIFMT_CBS_CFS);
+        if (err)
+                return err;
+
+        err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, 
+                                     SND_SOC_CLOCK_IN);
+        if (err)
+                return err;
+
+        err = snd_soc_dai_set_sysclk(cpu_dai, 0, CODEC_CLOCK, 
+                                     SND_SOC_CLOCK_OUT);
+        if (err)
+                return err;
+
+        return 0;
+}
+
+static struct snd_soc_ops snappercl15_ops = {
+        .hw_params      = snappercl15_hw_params,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+        SND_SOC_DAPM_HP("Headphone Jack", NULL),
+        SND_SOC_DAPM_LINE("Line In", NULL),
+        SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+        {"Headphone Jack", NULL, "LHPOUT"},
+        {"Headphone Jack", NULL, "RHPOUT"},
+
+        {"LLINEIN", NULL, "Line In"},
+        {"RLINEIN", NULL, "Line In"},
+
+        {"MICIN", NULL, "Mic Jack"},
+};
+
+static int snappercl15_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+        snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+                                  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+        return 0;
+}
+
+static struct snd_soc_dai_link snappercl15_dai = {
+        .name           = "tlv320aic23",
+        .stream_name    = "AIC23",
+        .cpu_dai        = &ep93xx_i2s_dai,
+        .codec_dai      = &tlv320aic23_dai,
+        .init           = snappercl15_tlv320aic23_init,
+        .ops            = &snappercl15_ops,
+};
+
+static struct snd_soc_card snd_soc_snappercl15 = {
+        .name           = "Snapper CL15",
+        .platform       = &ep93xx_soc_platform,
+        .dai_link       = &snappercl15_dai,
+        .num_links      = 1,
+};
+
+static struct snd_soc_device snappercl15_snd_devdata = {
+        .card           = &snd_soc_snappercl15,
+        .codec_dev      = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *snappercl15_snd_device;
+
+static int __init snappercl15_init(void)
+{
+        int ret;
+
+        if (!machine_is_snapper_cl15())
+                return -ENODEV;
+
+        ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
+                                 EP93XX_SYSCON_I2SCLKDIV_ORIDE |
+                                 EP93XX_SYSCON_I2SCLKDIV_SPOL);
+        if (ret)
+                return ret;
+
+        snappercl15_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!snappercl15_snd_device)
+                return -ENOMEM;
+        
+        platform_set_drvdata(snappercl15_snd_device, &snappercl15_snd_devdata);
+        snappercl15_snd_devdata.dev = &snappercl15_snd_device->dev;
+        ret = platform_device_add(snappercl15_snd_device);
+        if (ret)
+                platform_device_put(snappercl15_snd_device);
+
+        return ret;
+}
+
+static void __exit snappercl15_exit(void)
+{
+        platform_device_unregister(snappercl15_snd_device);
+        ep93xx_i2s_release();
+}
+
+module_init(snappercl15_init);
+module_exit(snappercl15_exit);
+
+MODULE_AUTHOR("Ryan Mallon <ryan@bluewatersys.com>");
+MODULE_DESCRIPTION("ALSA SoC Snapper CL15");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 4f455bd6851f..676841cbae98 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -16,7 +16,6 @@
 
 #include <asm/mpc52xx_psc.h>
 
-#include "mpc5200_psc_i2s.h"
 #include "mpc5200_dma.h"
 
 /**
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h
deleted file mode 100644
index ce55e070fdf3..000000000000
--- a/sound/soc/fsl/mpc5200_psc_i2s.h
+++ /dev/null
@@ -1,12 +0,0 @@
-/*
- * Freescale MPC5200 PSC in I2S mode
- * ALSA SoC Digital Audio Interface (DAI) driver
- *
- */
-
-#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
-#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__
-
-extern struct snd_soc_dai psc_i2s_dai[];
-
-#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index 252defea93b5..52dac5e3874c 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -1,4 +1,4 @@
-config SND_IMX_SOC
+menuconfig SND_IMX_SOC
         tristate "SoC Audio for Freescale i.MX CPUs"
         depends on ARCH_MXC
         select SND_PCM
@@ -8,14 +8,12 @@ config SND_IMX_SOC
           Say Y or M if you want to add support for codecs attached to
           the i.MX SSI interface.
 
-config SND_MXC_SOC_SSI
-        tristate
+if SND_IMX_SOC
 
 config SND_MXC_SOC_WM1133_EV1
         tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
-        depends on SND_IMX_SOC && MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
+        depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
         select SND_SOC_WM8350
-        select SND_MXC_SOC_SSI
         help
           Enable support for audio on the i.MX31ADS with the WM1133-EV1
           PMIC board with WM8835x fitted.
@@ -23,8 +21,17 @@ config SND_MXC_SOC_WM1133_EV1
 config SND_SOC_PHYCORE_AC97
         tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
         depends on MACH_PCM043 || MACH_PCA100
-        select SND_MXC_SOC_SSI
         select SND_SOC_WM9712
         help
           Say Y if you want to add support for SoC audio on Phytec phyCORE
           and phyCARD boards in AC97 mode
+
+config SND_SOC_EUKREA_TLV320
+        tristate "Eukrea TLV320"
+        depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD
+        select SND_SOC_TLV320AIC23
+        help
+          Enable I2S based access to the TLV320AIC23B codec attached
+          to the SSI interface
+
+endif   # SND_IMX_SOC
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
index 2d203635ac11..7bc57baf2b0e 100644
--- a/sound/soc/imx/Makefile
+++ b/sound/soc/imx/Makefile
@@ -8,8 +8,10 @@ endif
 obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o
 
 # i.MX Machine Support
+snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
 snd-soc-phycore-ac97-objs := phycore-ac97.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
 
+obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
new file mode 100644
index 000000000000..f15dfbdc47ee
--- /dev/null
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -0,0 +1,137 @@
+/*
+ * eukrea-tlv320.c  --  SoC audio for eukrea_cpuimxXX in I2S mode
+ *
+ * Copyright 2010 Eric Bénard, Eukréa Electromatique <eric@eukrea.com>
+ *
+ * based on sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+ * which is Copyright 2009 Simtec Electronics
+ * and on sound/soc/imx/phycore-ac97.c which is
+ * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer@pengutronix.de>
+ * 
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "imx-ssi.h"
+
+#define CODEC_CLOCK 12000000
+
+static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream,
+                            struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        int ret;
+
+        ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_NF |
+                                  SND_SOC_DAIFMT_CBM_CFM);
+        if (ret) {
+                pr_err("%s: failed set cpu dai format\n", __func__);
+                return ret;
+        }
+
+        ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_NF |
+                                  SND_SOC_DAIFMT_CBM_CFM);
+        if (ret) {
+                pr_err("%s: failed set codec dai format\n", __func__);
+                return ret;
+        }
+
+        ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+                                     CODEC_CLOCK, SND_SOC_CLOCK_OUT);
+        if (ret) {
+                pr_err("%s: failed setting codec sysclk\n", __func__);
+                return ret;
+        }
+        snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
+
+        ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
+                                SND_SOC_CLOCK_IN);
+        if (ret) {
+                pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n");
+                return ret;
+        }
+
+        return 0;
+}
+
+static struct snd_soc_ops eukrea_tlv320_snd_ops = {
+        .hw_params      = eukrea_tlv320_hw_params,
+};
+
+static struct snd_soc_dai_link eukrea_tlv320_dai = {
+        .name           = "tlv320aic23",
+        .stream_name    = "TLV320AIC23",
+        .codec_dai      = &tlv320aic23_dai,
+        .ops            = &eukrea_tlv320_snd_ops,
+};
+
+static struct snd_soc_card eukrea_tlv320 = {
+        .name           = "cpuimx-audio",
+        .platform       = &imx_soc_platform,
+        .dai_link       = &eukrea_tlv320_dai,
+        .num_links      = 1,
+};
+
+static struct snd_soc_device eukrea_tlv320_snd_devdata = {
+        .card           = &eukrea_tlv320,
+        .codec_dev      = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *eukrea_tlv320_snd_device;
+
+static int __init eukrea_tlv320_init(void)
+{
+        int ret;
+
+        if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()
+                && !machine_is_eukrea_cpuimx35sd())
+                /* return happy. We might run on a totally different machine */
+                return 0;
+
+        eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!eukrea_tlv320_snd_device)
+                return -ENOMEM;
+
+        eukrea_tlv320_dai.cpu_dai = &imx_ssi_pcm_dai[0];
+
+        platform_set_drvdata(eukrea_tlv320_snd_device, &eukrea_tlv320_snd_devdata);
+        eukrea_tlv320_snd_devdata.dev = &eukrea_tlv320_snd_device->dev;
+        ret = platform_device_add(eukrea_tlv320_snd_device);
+
+        if (ret) {
+                printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+                platform_device_put(eukrea_tlv320_snd_device);
+        }
+
+        return ret;
+}
+
+static void __exit eukrea_tlv320_exit(void)
+{
+        platform_device_unregister(eukrea_tlv320_snd_device);
+}
+
+module_init(eukrea_tlv320_init);
+module_exit(eukrea_tlv320_exit);
+
+MODULE_AUTHOR("Eric Bénard <eric@eukrea.com>");
+MODULE_DESCRIPTION("CPUIMX ALSA SoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 05f19c9284f4..0a595da4811d 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -292,12 +292,16 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
         int ret;
 
         iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
+        if (iprtd == NULL)
+                return -ENOMEM;
         runtime->private_data = iprtd;
 
         ret = snd_pcm_hw_constraint_integer(substream->runtime,
                         SNDRV_PCM_HW_PARAM_PERIODS);
-        if (ret < 0)
+        if (ret < 0) {
+                kfree(iprtd);
                 return ret;
+        }
 
         snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
         return 0;
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 6b518e07eea9..b2bf27282cd2 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -192,6 +192,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
         int ret;
 
         iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
+        if (iprtd == NULL)
+                return -ENOMEM;
         runtime->private_data = iprtd;
 
         iprtd->substream = substream;
@@ -202,8 +204,10 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
 
         ret = snd_pcm_hw_constraint_integer(substream->runtime,
                         SNDRV_PCM_HW_PARAM_PERIODS);
-        if (ret < 0)
+        if (ret < 0) {
+                kfree(iprtd);
                 return ret;
+        }
 
         snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
         return 0;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 80b4fee2442b..50f51624c535 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -83,8 +83,6 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
 /*
  * SSI DAI format configuration.
  * Should only be called when port is inactive (i.e. SSIEN = 0).
- * Note: We don't use the I2S modes but instead manually configure the
- * SSI for I2S because the I2S mode is only a register preset.
  */
 static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 {
@@ -99,6 +97,10 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
                 /* data on rising edge of bclk, frame low 1clk before data */
                 strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0;
                 scr |= SSI_SCR_NET;
+                if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) {
+                        scr &= ~SSI_I2S_MODE_MASK;
+                        scr |= SSI_SCR_I2S_MODE_SLAVE;
+                }
                 break;
         case SND_SOC_DAIFMT_LEFT_J:
                 /* data on rising edge of bclk, frame high with data */
@@ -143,6 +145,11 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
 
         strcr |= SSI_STCR_TFEN0;
 
+        if (ssi->flags & IMX_SSI_NET)
+                scr |= SSI_SCR_NET;
+        if (ssi->flags & IMX_SSI_SYN)
+                scr |= SSI_SCR_SYN;
+
         writel(strcr, ssi->base + SSI_STCR);
         writel(strcr, ssi->base + SSI_SRCR);
         writel(scr, ssi->base + SSI_SCR);
diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig
new file mode 100644
index 000000000000..5351cba66c9e
--- /dev/null
+++ b/sound/soc/jz4740/Kconfig
@@ -0,0 +1,23 @@
+config SND_JZ4740_SOC
+        tristate "SoC Audio for Ingenic JZ4740 SoC"
+        depends on MACH_JZ4740 && SND_SOC
+        help
+          Say Y or M if you want to add support for codecs attached to
+          the JZ4740 I2S interface. You will also need to select the audio
+          interfaces to support below.
+
+config SND_JZ4740_SOC_I2S
+        depends on SND_JZ4740_SOC
+        tristate "SoC Audio (I2S protocol) for Ingenic JZ4740 SoC"
+        help
+          Say Y if you want to use I2S protocol and I2S codec on Ingenic JZ4740
+          based boards.
+
+config SND_JZ4740_SOC_QI_LB60
+        tristate "SoC Audio support for Qi LB60"
+        depends on SND_JZ4740_SOC && JZ4740_QI_LB60
+        select SND_JZ4740_SOC_I2S
+    select SND_SOC_JZ4740_CODEC
+        help
+          Say Y if you want to add support for ASoC audio on the Qi LB60 board
+          a.k.a Qi Ben NanoNote.
diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile
new file mode 100644
index 000000000000..be873c1b0c20
--- /dev/null
+++ b/sound/soc/jz4740/Makefile
@@ -0,0 +1,13 @@
+#
+# Jz4740 Platform Support
+#
+snd-soc-jz4740-objs := jz4740-pcm.o
+snd-soc-jz4740-i2s-objs := jz4740-i2s.o
+
+obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o
+obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o
+
+# Jz4740 Machine Support
+snd-soc-qi-lb60-objs := qi_lb60.o
+
+obj-$(CONFIG_SND_JZ4740_SOC_QI_LB60) += snd-soc-qi-lb60.o
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
new file mode 100644
index 000000000000..eb518f0c5e01
--- /dev/null
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -0,0 +1,540 @@
+/*
+ *  Copyright (C) 2010, Lars-Peter Clausen <lars@metafoo.de>
+ *
+ *  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.
+ *
+ *  You should have received a copy of the GNU General Public License along
+ *  with this program; if not, write to the Free Software Foundation, Inc.,
+ *  675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+
+#include <linux/dma-mapping.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 "jz4740-i2s.h"
+#include "jz4740-pcm.h"
+
+#define JZ_REG_AIC_CONF         0x00
+#define JZ_REG_AIC_CTRL         0x04
+#define JZ_REG_AIC_I2S_FMT      0x10
+#define JZ_REG_AIC_FIFO_STATUS  0x14
+#define JZ_REG_AIC_I2S_STATUS   0x1c
+#define JZ_REG_AIC_CLK_DIV      0x30
+#define JZ_REG_AIC_FIFO         0x34
+
+#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_MASK (0xf << 12)
+#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_MASK (0xf <<  8)
+#define JZ_AIC_CONF_OVERFLOW_PLAY_LAST BIT(6)
+#define JZ_AIC_CONF_INTERNAL_CODEC BIT(5)
+#define JZ_AIC_CONF_I2S BIT(4)
+#define JZ_AIC_CONF_RESET BIT(3)
+#define JZ_AIC_CONF_BIT_CLK_MASTER BIT(2)
+#define JZ_AIC_CONF_SYNC_CLK_MASTER BIT(1)
+#define JZ_AIC_CONF_ENABLE BIT(0)
+
+#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12
+#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8
+
+#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19)
+#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16)
+#define JZ_AIC_CTRL_ENABLE_RX_DMA BIT(15)
+#define JZ_AIC_CTRL_ENABLE_TX_DMA BIT(14)
+#define JZ_AIC_CTRL_MONO_TO_STEREO BIT(11)
+#define JZ_AIC_CTRL_SWITCH_ENDIANNESS BIT(10)
+#define JZ_AIC_CTRL_SIGNED_TO_UNSIGNED BIT(9)
+#define JZ_AIC_CTRL_FLUSH               BIT(8)
+#define JZ_AIC_CTRL_ENABLE_ROR_INT BIT(6)
+#define JZ_AIC_CTRL_ENABLE_TUR_INT BIT(5)
+#define JZ_AIC_CTRL_ENABLE_RFS_INT BIT(4)
+#define JZ_AIC_CTRL_ENABLE_TFS_INT BIT(3)
+#define JZ_AIC_CTRL_ENABLE_LOOPBACK BIT(2)
+#define JZ_AIC_CTRL_ENABLE_PLAYBACK BIT(1)
+#define JZ_AIC_CTRL_ENABLE_CAPTURE BIT(0)
+
+#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET 19
+#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET  16
+
+#define JZ_AIC_I2S_FMT_DISABLE_BIT_CLK BIT(12)
+#define JZ_AIC_I2S_FMT_ENABLE_SYS_CLK BIT(4)
+#define JZ_AIC_I2S_FMT_MSB BIT(0)
+
+#define JZ_AIC_I2S_STATUS_BUSY BIT(2)
+
+#define JZ_AIC_CLK_DIV_MASK 0xf
+
+struct jz4740_i2s {
+        struct resource *mem;
+        void __iomem *base;
+        dma_addr_t phys_base;
+
+        struct clk *clk_aic;
+        struct clk *clk_i2s;
+
+        struct jz4740_pcm_config pcm_config_playback;
+        struct jz4740_pcm_config pcm_config_capture;
+};
+
+static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s,
+        unsigned int reg)
+{
+        return readl(i2s->base + reg);
+}
+
+static inline void jz4740_i2s_write(const struct jz4740_i2s *i2s,
+        unsigned int reg, uint32_t value)
+{
+        writel(value, i2s->base + reg);
+}
+
+static inline struct jz4740_i2s *jz4740_dai_to_i2s(struct snd_soc_dai *dai)
+{
+        return dai->private_data;
+}
+
+static int jz4740_i2s_startup(struct snd_pcm_substream *substream,
+        struct snd_soc_dai *dai)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+        uint32_t conf, ctrl;
+
+        if (dai->active)
+                return 0;
+
+        ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
+        ctrl |= JZ_AIC_CTRL_FLUSH;
+        jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
+
+        clk_enable(i2s->clk_i2s);
+
+        conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
+        conf |= JZ_AIC_CONF_ENABLE;
+        jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
+
+        return 0;
+}
+
+static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream,
+        struct snd_soc_dai *dai)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+        uint32_t conf;
+
+        if (!dai->active)
+                return;
+
+        conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
+        conf &= ~JZ_AIC_CONF_ENABLE;
+        jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
+
+        clk_disable(i2s->clk_i2s);
+}
+
+static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+        struct snd_soc_dai *dai)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+
+        uint32_t ctrl;
+        uint32_t mask;
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                mask = JZ_AIC_CTRL_ENABLE_PLAYBACK | JZ_AIC_CTRL_ENABLE_TX_DMA;
+        else
+                mask = JZ_AIC_CTRL_ENABLE_CAPTURE | JZ_AIC_CTRL_ENABLE_RX_DMA;
+
+        ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                ctrl |= mask;
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                ctrl &= ~mask;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
+
+        return 0;
+}
+
+static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+
+        uint32_t format = 0;
+        uint32_t conf;
+
+        conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
+
+        conf &= ~(JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER);
+
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBS_CFS:
+                conf |= JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER;
+                format |= JZ_AIC_I2S_FMT_ENABLE_SYS_CLK;
+                break;
+        case SND_SOC_DAIFMT_CBM_CFS:
+                conf |= JZ_AIC_CONF_SYNC_CLK_MASTER;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFM:
+                conf |= JZ_AIC_CONF_BIT_CLK_MASTER;
+                break;
+        case SND_SOC_DAIFMT_CBM_CFM:
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_MSB:
+                format |= JZ_AIC_I2S_FMT_MSB;
+                break;
+        case SND_SOC_DAIFMT_I2S:
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
+        jz4740_i2s_write(i2s, JZ_REG_AIC_I2S_FMT, format);
+
+        return 0;
+}
+
+static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream,
+        struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+        enum jz4740_dma_width dma_width;
+        struct jz4740_pcm_config *pcm_config;
+        unsigned int sample_size;
+        uint32_t ctrl;
+
+        ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL);
+
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S8:
+                sample_size = 0;
+                dma_width = JZ4740_DMA_WIDTH_8BIT;
+                break;
+        case SNDRV_PCM_FORMAT_S16:
+                sample_size = 1;
+                dma_width = JZ4740_DMA_WIDTH_16BIT;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                ctrl &= ~JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK;
+                ctrl |= sample_size << JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET;
+                if (params_channels(params) == 1)
+                        ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO;
+                else
+                        ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO;
+
+                pcm_config = &i2s->pcm_config_playback;
+                pcm_config->dma_config.dst_width = dma_width;
+
+        } else {
+                ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK;
+                ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET;
+
+                pcm_config = &i2s->pcm_config_capture;
+                pcm_config->dma_config.src_width = dma_width;
+        }
+
+        jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl);
+
+        snd_soc_dai_set_dma_data(dai, substream, pcm_config);
+
+        return 0;
+}
+
+static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+        unsigned int freq, int dir)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+        struct clk *parent;
+        int ret = 0;
+
+        switch (clk_id) {
+        case JZ4740_I2S_CLKSRC_EXT:
+                parent = clk_get(NULL, "ext");
+                clk_set_parent(i2s->clk_i2s, parent);
+                break;
+        case JZ4740_I2S_CLKSRC_PLL:
+                parent = clk_get(NULL, "pll half");
+                clk_set_parent(i2s->clk_i2s, parent);
+                ret = clk_set_rate(i2s->clk_i2s, freq);
+                break;
+        default:
+                return -EINVAL;
+        }
+        clk_put(parent);
+
+        return ret;
+}
+
+static int jz4740_i2s_suspend(struct snd_soc_dai *dai)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+        uint32_t conf;
+
+        if (dai->active) {
+                conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
+                conf &= ~JZ_AIC_CONF_ENABLE;
+                jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
+
+                clk_disable(i2s->clk_i2s);
+        }
+
+        clk_disable(i2s->clk_aic);
+
+        return 0;
+}
+
+static int jz4740_i2s_resume(struct snd_soc_dai *dai)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+        uint32_t conf;
+
+        clk_enable(i2s->clk_aic);
+
+        if (dai->active) {
+                clk_enable(i2s->clk_i2s);
+
+                conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF);
+                conf |= JZ_AIC_CONF_ENABLE;
+                jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
+        }
+
+        return 0;
+}
+
+static int jz4740_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai)
+{
+        struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai);
+        uint32_t conf;
+
+        conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) |
+                (8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) |
+                JZ_AIC_CONF_OVERFLOW_PLAY_LAST |
+                JZ_AIC_CONF_I2S |
+                JZ_AIC_CONF_INTERNAL_CODEC;
+
+        jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, JZ_AIC_CONF_RESET);
+        jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf);
+
+        return 0;
+}
+
+static struct snd_soc_dai_ops jz4740_i2s_dai_ops = {
+        .startup = jz4740_i2s_startup,
+        .shutdown = jz4740_i2s_shutdown,
+        .trigger = jz4740_i2s_trigger,
+        .hw_params = jz4740_i2s_hw_params,
+        .set_fmt = jz4740_i2s_set_fmt,
+        .set_sysclk = jz4740_i2s_set_sysclk,
+};
+
+#define JZ4740_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \
+                SNDRV_PCM_FMTBIT_S16_LE)
+
+struct snd_soc_dai jz4740_i2s_dai = {
+        .name = "jz4740-i2s",
+        .probe = jz4740_i2s_probe,
+        .playback = {
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000_48000,
+                .formats = JZ4740_I2S_FMTS,
+        },
+        .capture = {
+                .channels_min = 2,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000_48000,
+                .formats = JZ4740_I2S_FMTS,
+        },
+        .symmetric_rates = 1,
+        .ops = &jz4740_i2s_dai_ops,
+        .suspend = jz4740_i2s_suspend,
+        .resume = jz4740_i2s_resume,
+};
+EXPORT_SYMBOL_GPL(jz4740_i2s_dai);
+
+static void __devinit jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s)
+{
+        struct jz4740_dma_config *dma_config;
+
+        /* Playback */
+        dma_config = &i2s->pcm_config_playback.dma_config;
+        dma_config->src_width = JZ4740_DMA_WIDTH_32BIT,
+        dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
+        dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT;
+        dma_config->flags = JZ4740_DMA_SRC_AUTOINC;
+        dma_config->mode = JZ4740_DMA_MODE_SINGLE;
+        i2s->pcm_config_playback.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO;
+
+        /* Capture */
+        dma_config = &i2s->pcm_config_capture.dma_config;
+        dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT,
+        dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE;
+        dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE;
+        dma_config->flags = JZ4740_DMA_DST_AUTOINC;
+        dma_config->mode = JZ4740_DMA_MODE_SINGLE;
+        i2s->pcm_config_capture.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO;
+}
+
+static int __devinit jz4740_i2s_dev_probe(struct platform_device *pdev)
+{
+        struct jz4740_i2s *i2s;
+        int ret;
+
+        i2s = kzalloc(sizeof(*i2s), GFP_KERNEL);
+
+        if (!i2s)
+                return -ENOMEM;
+
+        i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!i2s->mem) {
+                ret = -ENOENT;
+                goto err_free;
+        }
+
+        i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem),
+                                pdev->name);
+        if (!i2s->mem) {
+                ret = -EBUSY;
+                goto err_free;
+        }
+
+        i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem));
+        if (!i2s->base) {
+                ret = -EBUSY;
+                goto err_release_mem_region;
+        }
+
+        i2s->phys_base = i2s->mem->start;
+
+        i2s->clk_aic = clk_get(&pdev->dev, "aic");
+        if (IS_ERR(i2s->clk_aic)) {
+                ret = PTR_ERR(i2s->clk_aic);
+                goto err_iounmap;
+        }
+
+        i2s->clk_i2s = clk_get(&pdev->dev, "i2s");
+        if (IS_ERR(i2s->clk_i2s)) {
+                ret = PTR_ERR(i2s->clk_i2s);
+                goto err_clk_put_aic;
+        }
+
+        clk_enable(i2s->clk_aic);
+
+        jz4740_i2c_init_pcm_config(i2s);
+
+        jz4740_i2s_dai.private_data = i2s;
+        ret = snd_soc_register_dai(&jz4740_i2s_dai);
+
+        if (ret) {
+                dev_err(&pdev->dev, "Failed to register DAI\n");
+                goto err_clk_put_i2s;
+        }
+
+        platform_set_drvdata(pdev, i2s);
+
+        return 0;
+
+err_clk_put_i2s:
+        clk_disable(i2s->clk_aic);
+        clk_put(i2s->clk_i2s);
+err_clk_put_aic:
+        clk_put(i2s->clk_aic);
+err_iounmap:
+        iounmap(i2s->base);
+err_release_mem_region:
+        release_mem_region(i2s->mem->start, resource_size(i2s->mem));
+err_free:
+        kfree(i2s);
+
+        return ret;
+}
+
+static int __devexit jz4740_i2s_dev_remove(struct platform_device *pdev)
+{
+        struct jz4740_i2s *i2s = platform_get_drvdata(pdev);
+
+        snd_soc_unregister_dai(&jz4740_i2s_dai);
+
+        clk_disable(i2s->clk_aic);
+        clk_put(i2s->clk_i2s);
+        clk_put(i2s->clk_aic);
+
+        iounmap(i2s->base);
+        release_mem_region(i2s->mem->start, resource_size(i2s->mem));
+
+        platform_set_drvdata(pdev, NULL);
+        kfree(i2s);
+
+        return 0;
+}
+
+static struct platform_driver jz4740_i2s_driver = {
+        .probe = jz4740_i2s_dev_probe,
+        .remove = __devexit_p(jz4740_i2s_dev_remove),
+        .driver = {
+                .name = "jz4740-i2s",
+                .owner = THIS_MODULE,
+        },
+};
+
+static int __init jz4740_i2s_init(void)
+{
+        return platform_driver_register(&jz4740_i2s_driver);
+}
+module_init(jz4740_i2s_init);
+
+static void __exit jz4740_i2s_exit(void)
+{
+        platform_driver_unregister(&jz4740_i2s_driver);
+}
+module_exit(jz4740_i2s_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen, <lars@metafoo.de>");
+MODULE_DESCRIPTION("Ingenic JZ4740 SoC I2S driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:jz4740-i2s");
diff --git a/sound/soc/jz4740/jz4740-i2s.h b/sound/soc/jz4740/jz4740-i2s.h
new file mode 100644
index 000000000000..da22ed88a589
--- /dev/null
+++ b/sound/soc/jz4740/jz4740-i2s.h
@@ -0,0 +1,18 @@
+/*
+ * 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 _JZ4740_I2S_H
+#define _JZ4740_I2S_H
+
+/* I2S clock source */
+#define JZ4740_I2S_CLKSRC_EXT 0
+#define JZ4740_I2S_CLKSRC_PLL 1
+
+#define JZ4740_I2S_BIT_CLK              0
+
+extern struct snd_soc_dai jz4740_i2s_dai;
+
+#endif
diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c
new file mode 100644
index 000000000000..ee68d850c8dd
--- /dev/null
+++ b/sound/soc/jz4740/jz4740-pcm.c
@@ -0,0 +1,373 @@
+/*
+ *  Copyright (C) 2010, Lars-Peter Clausen <lars@metafoo.de>
+ *
+ *  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.
+ *
+ *  You should have received a copy of the  GNU General Public License along
+ *  with this program; if not, write  to the Free Software Foundation, Inc.,
+ *  675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/kernel.h>
+#include <linux/module.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-jz4740/dma.h>
+#include "jz4740-pcm.h"
+
+struct jz4740_runtime_data {
+        unsigned long dma_period;
+        dma_addr_t dma_start;
+        dma_addr_t dma_pos;
+        dma_addr_t dma_end;
+
+        struct jz4740_dma_chan *dma;
+
+        dma_addr_t fifo_addr;
+};
+
+/* identify hardware playback capabilities */
+static const struct snd_pcm_hardware jz4740_pcm_hardware = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER,
+        .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8,
+
+        .rates                  = SNDRV_PCM_RATE_8000_48000,
+        .channels_min           = 1,
+        .channels_max           = 2,
+        .period_bytes_min       = 16,
+        .period_bytes_max       = 2 * PAGE_SIZE,
+        .periods_min            = 2,
+        .periods_max            = 128,
+        .buffer_bytes_max       = 128 * 2 * PAGE_SIZE,
+        .fifo_size              = 32,
+};
+
+static void jz4740_pcm_start_transfer(struct jz4740_runtime_data *prtd,
+        struct snd_pcm_substream *substream)
+{
+        unsigned long count;
+
+        if (prtd->dma_pos == prtd->dma_end)
+                prtd->dma_pos = prtd->dma_start;
+
+        if (prtd->dma_pos + prtd->dma_period > prtd->dma_end)
+                count = prtd->dma_end - prtd->dma_pos;
+        else
+                count = prtd->dma_period;
+
+        jz4740_dma_disable(prtd->dma);
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                jz4740_dma_set_src_addr(prtd->dma, prtd->dma_pos);
+                jz4740_dma_set_dst_addr(prtd->dma, prtd->fifo_addr);
+        } else {
+                jz4740_dma_set_src_addr(prtd->dma, prtd->fifo_addr);
+                jz4740_dma_set_dst_addr(prtd->dma, prtd->dma_pos);
+        }
+
+        jz4740_dma_set_transfer_count(prtd->dma, count);
+
+        prtd->dma_pos += count;
+
+        jz4740_dma_enable(prtd->dma);
+}
+
+static void jz4740_pcm_dma_transfer_done(struct jz4740_dma_chan *dma, int err,
+        void *dev_id)
+{
+        struct snd_pcm_substream *substream = dev_id;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct jz4740_runtime_data *prtd = runtime->private_data;
+
+        snd_pcm_period_elapsed(substream);
+
+        jz4740_pcm_start_transfer(prtd, substream);
+}
+
+static int jz4740_pcm_hw_params(struct snd_pcm_substream *substream,
+        struct snd_pcm_hw_params *params)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct jz4740_runtime_data *prtd = runtime->private_data;
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct jz4740_pcm_config *config;
+
+        config = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+
+        if (!config)
+                return 0;
+
+        if (!prtd->dma) {
+                if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+                        prtd->dma = jz4740_dma_request(substream, "PCM Capture");
+                else
+                        prtd->dma = jz4740_dma_request(substream, "PCM Playback");
+        }
+
+        if (!prtd->dma)
+                return -EBUSY;
+
+        jz4740_dma_configure(prtd->dma, &config->dma_config);
+        prtd->fifo_addr = config->fifo_addr;
+
+        jz4740_dma_set_complete_cb(prtd->dma, jz4740_pcm_dma_transfer_done);
+
+        snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+        runtime->dma_bytes = params_buffer_bytes(params);
+
+        prtd->dma_period = params_period_bytes(params);
+        prtd->dma_start = runtime->dma_addr;
+        prtd->dma_pos = prtd->dma_start;
+        prtd->dma_end = prtd->dma_start + runtime->dma_bytes;
+
+        return 0;
+}
+
+static int jz4740_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        struct jz4740_runtime_data *prtd = substream->runtime->private_data;
+
+        snd_pcm_set_runtime_buffer(substream, NULL);
+        if (prtd->dma) {
+                jz4740_dma_free(prtd->dma);
+                prtd->dma = NULL;
+        }
+
+        return 0;
+}
+
+static int jz4740_pcm_prepare(struct snd_pcm_substream *substream)
+{
+        struct jz4740_runtime_data *prtd = substream->runtime->private_data;
+
+        if (!prtd->dma)
+                return -EBUSY;
+
+        prtd->dma_pos = prtd->dma_start;
+
+        return 0;
+}
+
+static int jz4740_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct jz4740_runtime_data *prtd = runtime->private_data;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                jz4740_pcm_start_transfer(prtd, substream);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                jz4740_dma_disable(prtd->dma);
+                break;
+        default:
+                break;
+        }
+
+        return 0;
+}
+
+static snd_pcm_uframes_t jz4740_pcm_pointer(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct jz4740_runtime_data *prtd = runtime->private_data;
+        unsigned long byte_offset;
+        snd_pcm_uframes_t offset;
+        struct jz4740_dma_chan *dma = prtd->dma;
+
+        /* prtd->dma_pos points to the end of the current transfer. So by
+         * subtracting prdt->dma_start we get the offset to the end of the
+         * current period in bytes. By subtracting the residue of the transfer
+         * we get the current offset in bytes. */
+        byte_offset = prtd->dma_pos - prtd->dma_start;
+        byte_offset -= jz4740_dma_get_residue(dma);
+
+        offset = bytes_to_frames(runtime, byte_offset);
+        if (offset >= runtime->buffer_size)
+                offset = 0;
+
+        return offset;
+}
+
+static int jz4740_pcm_open(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct jz4740_runtime_data *prtd;
+
+        prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+        if (prtd == NULL)
+                return -ENOMEM;
+
+        snd_soc_set_runtime_hwparams(substream, &jz4740_pcm_hardware);
+
+        runtime->private_data = prtd;
+
+        return 0;
+}
+
+static int jz4740_pcm_close(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct jz4740_runtime_data *prtd = runtime->private_data;
+
+        kfree(prtd);
+
+        return 0;
+}
+
+static int jz4740_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 jz4740_pcm_ops = {
+        .open           = jz4740_pcm_open,
+        .close          = jz4740_pcm_close,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .hw_params      = jz4740_pcm_hw_params,
+        .hw_free        = jz4740_pcm_hw_free,
+        .prepare        = jz4740_pcm_prepare,
+        .trigger        = jz4740_pcm_trigger,
+        .pointer        = jz4740_pcm_pointer,
+        .mmap           = jz4740_pcm_mmap,
+};
+
+static int jz4740_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+        struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+        struct snd_dma_buffer *buf = &substream->dma_buffer;
+        size_t size = jz4740_pcm_hardware.buffer_bytes_max;
+
+        buf->dev.type = SNDRV_DMA_TYPE_DEV;
+        buf->dev.dev = pcm->card->dev;
+        buf->private_data = NULL;
+
+        buf->area = dma_alloc_noncoherent(pcm->card->dev, size,
+                                          &buf->addr, GFP_KERNEL);
+        if (!buf->area)
+                return -ENOMEM;
+
+        buf->bytes = size;
+
+        return 0;
+}
+
+static void jz4740_pcm_free(struct snd_pcm *pcm)
+{
+        struct snd_pcm_substream *substream;
+        struct snd_dma_buffer *buf;
+        int stream;
+
+        for (stream = 0; stream < SNDRV_PCM_STREAM_LAST; ++stream) {
+                substream = pcm->streams[stream].substream;
+                if (!substream)
+                        continue;
+
+                buf = &substream->dma_buffer;
+                if (!buf->area)
+                        continue;
+
+                dma_free_noncoherent(pcm->card->dev, buf->bytes, buf->area,
+                                buf->addr);
+                buf->area = NULL;
+        }
+}
+
+static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32);
+
+int jz4740_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 = &jz4740_pcm_dmamask;
+
+        if (!card->dev->coherent_dma_mask)
+                card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+        if (dai->playback.channels_min) {
+                ret = jz4740_pcm_preallocate_dma_buffer(pcm,
+                        SNDRV_PCM_STREAM_PLAYBACK);
+                if (ret)
+                        goto err;
+        }
+
+        if (dai->capture.channels_min) {
+                ret = jz4740_pcm_preallocate_dma_buffer(pcm,
+                        SNDRV_PCM_STREAM_CAPTURE);
+                if (ret)
+                        goto err;
+        }
+
+err:
+        return ret;
+}
+
+struct snd_soc_platform jz4740_soc_platform = {
+                .name           = "jz4740-pcm",
+                .pcm_ops        = &jz4740_pcm_ops,
+                .pcm_new        = jz4740_pcm_new,
+                .pcm_free       = jz4740_pcm_free,
+};
+EXPORT_SYMBOL_GPL(jz4740_soc_platform);
+
+static int __devinit jz4740_pcm_probe(struct platform_device *pdev)
+{
+        return snd_soc_register_platform(&jz4740_soc_platform);
+}
+
+static int __devexit jz4740_pcm_remove(struct platform_device *pdev)
+{
+        snd_soc_unregister_platform(&jz4740_soc_platform);
+        return 0;
+}
+
+static struct platform_driver jz4740_pcm_driver = {
+        .probe = jz4740_pcm_probe,
+        .remove = __devexit_p(jz4740_pcm_remove),
+        .driver = {
+                .name = "jz4740-pcm",
+                .owner = THIS_MODULE,
+        },
+};
+
+static int __init jz4740_soc_platform_init(void)
+{
+        return platform_driver_register(&jz4740_pcm_driver);
+}
+module_init(jz4740_soc_platform_init);
+
+static void __exit jz4740_soc_platform_exit(void)
+{
+        return platform_driver_unregister(&jz4740_pcm_driver);
+}
+module_exit(jz4740_soc_platform_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/jz4740/jz4740-pcm.h b/sound/soc/jz4740/jz4740-pcm.h
new file mode 100644
index 000000000000..e3f221e2779c
--- /dev/null
+++ b/sound/soc/jz4740/jz4740-pcm.h
@@ -0,0 +1,22 @@
+/*
+ *
+ * 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 _JZ4740_PCM_H
+#define _JZ4740_PCM_H
+
+#include <linux/dma-mapping.h>
+#include <asm/mach-jz4740/dma.h>
+
+/* platform data */
+extern struct snd_soc_platform jz4740_soc_platform;
+
+struct jz4740_pcm_config {
+        struct jz4740_dma_config dma_config;
+        phys_addr_t fifo_addr;
+};
+
+#endif
diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c
new file mode 100644
index 000000000000..f15f4918f15f
--- /dev/null
+++ b/sound/soc/jz4740/qi_lb60.c
@@ -0,0 +1,166 @@
+/*
+ * Copyright (C) 2009, Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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.
+ *
+ *  You should have received a copy of the  GNU General Public License along
+ *  with this program; if not, write  to the Free Software Foundation, Inc.,
+ *  675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#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 <linux/gpio.h>
+
+#include "../codecs/jz4740.h"
+#include "jz4740-pcm.h"
+#include "jz4740-i2s.h"
+
+
+#define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29)
+#define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4)
+
+static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget,
+                             struct snd_kcontrol *ctrl, int event)
+{
+        int on = 0;
+        if (event & SND_SOC_DAPM_POST_PMU)
+                on = 1;
+        else if (event & SND_SOC_DAPM_PRE_PMD)
+                on = 0;
+
+        gpio_set_value(QI_LB60_SND_GPIO, on);
+        gpio_set_value(QI_LB60_AMP_GPIO, on);
+
+        return 0;
+}
+
+static const struct snd_soc_dapm_widget qi_lb60_widgets[] = {
+        SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event),
+        SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route qi_lb60_routes[] = {
+        {"Mic", NULL, "MIC"},
+        {"Speaker", NULL, "LOUT"},
+        {"Speaker", NULL, "ROUT"},
+};
+
+#define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \
+                        SND_SOC_DAIFMT_NB_NF | \
+                        SND_SOC_DAIFMT_CBM_CFM)
+
+static int qi_lb60_codec_init(struct snd_soc_codec *codec)
+{
+        int ret;
+        struct snd_soc_dai *cpu_dai = codec->socdev->card->dai_link->cpu_dai;
+
+        snd_soc_dapm_nc_pin(codec, "LIN");
+        snd_soc_dapm_nc_pin(codec, "RIN");
+
+        ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT);
+        if (ret < 0) {
+                dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret);
+                return ret;
+        }
+
+        snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets));
+        snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes));
+        snd_soc_dapm_sync(codec);
+
+        return 0;
+}
+
+static struct snd_soc_dai_link qi_lb60_dai = {
+        .name = "jz4740",
+        .stream_name = "jz4740",
+        .cpu_dai = &jz4740_i2s_dai,
+        .codec_dai = &jz4740_codec_dai,
+        .init = qi_lb60_codec_init,
+};
+
+static struct snd_soc_card qi_lb60 = {
+        .name = "QI LB60",
+        .dai_link = &qi_lb60_dai,
+        .num_links = 1,
+        .platform = &jz4740_soc_platform,
+};
+
+static struct snd_soc_device qi_lb60_snd_devdata = {
+        .card = &qi_lb60,
+        .codec_dev = &soc_codec_dev_jz4740_codec,
+};
+
+static struct platform_device *qi_lb60_snd_device;
+
+static int __init qi_lb60_init(void)
+{
+        int ret;
+
+        qi_lb60_snd_device = platform_device_alloc("soc-audio", -1);
+
+        if (!qi_lb60_snd_device)
+                return -ENOMEM;
+
+        ret = gpio_request(QI_LB60_SND_GPIO, "SND");
+        if (ret) {
+                pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n",
+                                QI_LB60_SND_GPIO, ret);
+                goto err_device_put;
+        }
+
+        ret = gpio_request(QI_LB60_AMP_GPIO, "AMP");
+        if (ret) {
+                pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n",
+                                QI_LB60_AMP_GPIO, ret);
+                goto err_gpio_free_snd;
+        }
+
+        gpio_direction_output(QI_LB60_SND_GPIO, 0);
+        gpio_direction_output(QI_LB60_AMP_GPIO, 0);
+
+        platform_set_drvdata(qi_lb60_snd_device, &qi_lb60_snd_devdata);
+        qi_lb60_snd_devdata.dev = &qi_lb60_snd_device->dev;
+
+        ret = platform_device_add(qi_lb60_snd_device);
+        if (ret) {
+                pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret);
+                goto err_unset_pdata;
+        }
+
+         return 0;
+
+err_unset_pdata:
+        platform_set_drvdata(qi_lb60_snd_device, NULL);
+/*err_gpio_free_amp:*/
+        gpio_free(QI_LB60_AMP_GPIO);
+err_gpio_free_snd:
+        gpio_free(QI_LB60_SND_GPIO);
+err_device_put:
+        platform_device_put(qi_lb60_snd_device);
+
+        return ret;
+}
+module_init(qi_lb60_init);
+
+static void __exit qi_lb60_exit(void)
+{
+        gpio_free(QI_LB60_AMP_GPIO);
+        gpio_free(QI_LB60_SND_GPIO);
+        platform_device_unregister(qi_lb60_snd_device);
+}
+module_exit(qi_lb60_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
new file mode 100644
index 000000000000..16ec2a2dba4d
--- /dev/null
+++ b/sound/soc/kirkwood/Kconfig
@@ -0,0 +1,20 @@
+config SND_KIRKWOOD_SOC
+        tristate "SoC Audio for the Marvell Kirkwood chip"
+        depends on ARCH_KIRKWOOD
+        help
+          Say Y or M if you want to add support for codecs attached to
+          the Kirkwood I2S interface. You will also need to select the
+          audio interfaces to support below.
+
+config SND_KIRKWOOD_SOC_I2S
+        tristate
+
+config SND_KIRKWOOD_SOC_OPENRD
+        tristate "SoC Audio support for Kirkwood Openrd Client"
+        depends on SND_KIRKWOOD_SOC && MACH_OPENRD_CLIENT
+        select SND_KIRKWOOD_SOC_I2S
+        select SND_SOC_CS42L51
+        help
+          Say Y if you want to add support for SoC audio on
+          Openrd Client.
+
diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile
new file mode 100644
index 000000000000..33a16dcab5b5
--- /dev/null
+++ b/sound/soc/kirkwood/Makefile
@@ -0,0 +1,9 @@
+snd-soc-kirkwood-objs := kirkwood-dma.o
+snd-soc-kirkwood-i2s-objs := kirkwood-i2s.o
+
+obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o
+obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o
+
+snd-soc-openrd-objs := kirkwood-openrd.o
+
+obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
new file mode 100644
index 000000000000..a30205be3e2b
--- /dev/null
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -0,0 +1,383 @@
+/*
+ * kirkwood-dma.c
+ *
+ * (c) 2010 Arnaud Patard <apatard@mandriva.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+#include <linux/mbus.h>
+#include <sound/soc.h>
+#include "kirkwood-dma.h"
+#include "kirkwood.h"
+
+#define KIRKWOOD_RATES \
+        (SNDRV_PCM_RATE_44100 | \
+         SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+#define KIRKWOOD_FORMATS \
+        (SNDRV_PCM_FMTBIT_S16_LE | \
+         SNDRV_PCM_FMTBIT_S24_LE | \
+         SNDRV_PCM_FMTBIT_S32_LE)
+
+struct kirkwood_dma_priv {
+        struct snd_pcm_substream *play_stream;
+        struct snd_pcm_substream *rec_stream;
+        struct kirkwood_dma_data *data;
+};
+
+static struct snd_pcm_hardware kirkwood_dma_snd_hw = {
+        .info = (SNDRV_PCM_INFO_INTERLEAVED |
+                 SNDRV_PCM_INFO_MMAP |
+                 SNDRV_PCM_INFO_MMAP_VALID |
+                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                 SNDRV_PCM_INFO_PAUSE),
+        .formats                = KIRKWOOD_FORMATS,
+        .rates                  = KIRKWOOD_RATES,
+        .rate_min               = 44100,
+        .rate_max               = 96000,
+        .channels_min           = 1,
+        .channels_max           = 2,
+        .buffer_bytes_max       = KIRKWOOD_SND_MAX_PERIOD_BYTES * KIRKWOOD_SND_MAX_PERIODS,
+        .period_bytes_min       = KIRKWOOD_SND_MIN_PERIOD_BYTES,
+        .period_bytes_max       = KIRKWOOD_SND_MAX_PERIOD_BYTES,
+        .periods_min            = KIRKWOOD_SND_MIN_PERIODS,
+        .periods_max            = KIRKWOOD_SND_MAX_PERIODS,
+        .fifo_size              = 0,
+};
+
+static u64 kirkwood_dma_dmamask = 0xFFFFFFFFUL;
+
+static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id)
+{
+        struct kirkwood_dma_priv *prdata = dev_id;
+        struct kirkwood_dma_data *priv = prdata->data;
+        unsigned long mask, status, cause;
+
+        mask = readl(priv->io + KIRKWOOD_INT_MASK);
+        status = readl(priv->io + KIRKWOOD_INT_CAUSE) & mask;
+
+        cause = readl(priv->io + KIRKWOOD_ERR_CAUSE);
+        if (unlikely(cause)) {
+                printk(KERN_WARNING "%s: got err interrupt 0x%lx\n",
+                                __func__, cause);
+                writel(cause, priv->io + KIRKWOOD_ERR_CAUSE);
+                return IRQ_HANDLED;
+        }
+
+        /* we've enabled only bytes interrupts ... */
+        if (status & ~(KIRKWOOD_INT_CAUSE_PLAY_BYTES | \
+                        KIRKWOOD_INT_CAUSE_REC_BYTES)) {
+                printk(KERN_WARNING "%s: unexpected interrupt %lx\n",
+                        __func__, status);
+                return IRQ_NONE;
+        }
+
+        /* ack int */
+        writel(status, priv->io + KIRKWOOD_INT_CAUSE);
+
+        if (status & KIRKWOOD_INT_CAUSE_PLAY_BYTES)
+                snd_pcm_period_elapsed(prdata->play_stream);
+
+        if (status & KIRKWOOD_INT_CAUSE_REC_BYTES)
+                snd_pcm_period_elapsed(prdata->rec_stream);
+
+        return IRQ_HANDLED;
+}
+
+static void kirkwood_dma_conf_mbus_windows(void __iomem *base, int win,
+                                        unsigned long dma,
+                                        struct mbus_dram_target_info *dram)
+{
+        int i;
+
+        /* First disable and clear windows */
+        writel(0, base + KIRKWOOD_AUDIO_WIN_CTRL_REG(win));
+        writel(0, base + KIRKWOOD_AUDIO_WIN_BASE_REG(win));
+
+        /* try to find matching cs for current dma address */
+        for (i = 0; i < dram->num_cs; i++) {
+                struct mbus_dram_window *cs = dram->cs + i;
+                if ((cs->base & 0xffff0000) < (dma & 0xffff0000)) {
+                        writel(cs->base & 0xffff0000,
+                                base + KIRKWOOD_AUDIO_WIN_BASE_REG(win));
+                        writel(((cs->size - 1) & 0xffff0000) |
+                                (cs->mbus_attr << 8) |
+                                (dram->mbus_dram_target_id << 4) | 1,
+                                base + KIRKWOOD_AUDIO_WIN_CTRL_REG(win));
+                }
+        }
+}
+
+static int kirkwood_dma_open(struct snd_pcm_substream *substream)
+{
+        int err;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai;
+        struct kirkwood_dma_data *priv;
+        struct kirkwood_dma_priv *prdata = cpu_dai->private_data;
+        unsigned long addr;
+
+        priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
+        snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw);
+
+        /* Ensure that all constraints linked to dma burst are fullfilled */
+        err = snd_pcm_hw_constraint_minmax(runtime,
+                        SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+                        priv->burst * 2,
+                        KIRKWOOD_AUDIO_BUF_MAX-1);
+        if (err < 0)
+                return err;
+
+        err = snd_pcm_hw_constraint_step(runtime, 0,
+                        SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+                        priv->burst);
+        if (err < 0)
+                return err;
+
+        err = snd_pcm_hw_constraint_step(substream->runtime, 0,
+                         SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+                         priv->burst);
+        if (err < 0)
+                return err;
+
+        if (soc_runtime->dai->cpu_dai->private_data == NULL) {
+                prdata = kzalloc(sizeof(struct kirkwood_dma_priv), GFP_KERNEL);
+                if (prdata == NULL)
+                        return -ENOMEM;
+
+                prdata->data = priv;
+
+                err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED,
+                                  "kirkwood-i2s", prdata);
+                if (err) {
+                        kfree(prdata);
+                        return -EBUSY;
+                }
+
+                soc_runtime->dai->cpu_dai->private_data = prdata;
+
+                /*
+                 * Enable Error interrupts. We're only ack'ing them but
+                 * it's usefull for diagnostics
+                 */
+                writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK);
+        }
+
+        addr = virt_to_phys(substream->dma_buffer.area);
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                prdata->play_stream = substream;
+                kirkwood_dma_conf_mbus_windows(priv->io,
+                        KIRKWOOD_PLAYBACK_WIN, addr, priv->dram);
+        } else {
+                prdata->rec_stream = substream;
+                kirkwood_dma_conf_mbus_windows(priv->io,
+                        KIRKWOOD_RECORD_WIN, addr, priv->dram);
+        }
+
+        return 0;
+}
+
+static int kirkwood_dma_close(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai;
+        struct kirkwood_dma_priv *prdata = cpu_dai->private_data;
+        struct kirkwood_dma_data *priv;
+
+        priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
+
+        if (!prdata || !priv)
+                return 0;
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                prdata->play_stream = NULL;
+        else
+                prdata->rec_stream = NULL;
+
+        if (!prdata->play_stream && !prdata->rec_stream) {
+                writel(0, priv->io + KIRKWOOD_ERR_MASK);
+                free_irq(priv->irq, prdata);
+                kfree(prdata);
+                soc_runtime->dai->cpu_dai->private_data = NULL;
+        }
+
+        return 0;
+}
+
+static int kirkwood_dma_hw_params(struct snd_pcm_substream *substream,
+                                struct snd_pcm_hw_params *params)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+
+        snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+        runtime->dma_bytes = params_buffer_bytes(params);
+
+        return 0;
+}
+
+static int kirkwood_dma_hw_free(struct snd_pcm_substream *substream)
+{
+        snd_pcm_set_runtime_buffer(substream, NULL);
+        return 0;
+}
+
+static int kirkwood_dma_prepare(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai;
+        struct kirkwood_dma_data *priv;
+        unsigned long size, count;
+
+        priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
+
+        /* compute buffer size in term of "words" as requested in specs */
+        size = frames_to_bytes(runtime, runtime->buffer_size);
+        size = (size>>2)-1;
+        count = snd_pcm_lib_period_bytes(substream);
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                writel(count, priv->io + KIRKWOOD_PLAY_BYTE_INT_COUNT);
+                writel(runtime->dma_addr, priv->io + KIRKWOOD_PLAY_BUF_ADDR);
+                writel(size, priv->io + KIRKWOOD_PLAY_BUF_SIZE);
+        } else {
+                writel(count, priv->io + KIRKWOOD_REC_BYTE_INT_COUNT);
+                writel(runtime->dma_addr, priv->io + KIRKWOOD_REC_BUF_ADDR);
+                writel(size, priv->io + KIRKWOOD_REC_BUF_SIZE);
+        }
+
+
+        return 0;
+}
+
+static snd_pcm_uframes_t kirkwood_dma_pointer(struct snd_pcm_substream
+                                                *substream)
+{
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct snd_soc_dai *cpu_dai = soc_runtime->dai->cpu_dai;
+        struct kirkwood_dma_data *priv;
+        snd_pcm_uframes_t count;
+
+        priv = snd_soc_dai_get_dma_data(cpu_dai, substream);
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                count = bytes_to_frames(substream->runtime,
+                        readl(priv->io + KIRKWOOD_PLAY_BYTE_COUNT));
+        else
+                count = bytes_to_frames(substream->runtime,
+                        readl(priv->io + KIRKWOOD_REC_BYTE_COUNT));
+
+        return count;
+}
+
+struct snd_pcm_ops kirkwood_dma_ops = {
+        .open =         kirkwood_dma_open,
+        .close =        kirkwood_dma_close,
+        .ioctl =        snd_pcm_lib_ioctl,
+        .hw_params =    kirkwood_dma_hw_params,
+        .hw_free =      kirkwood_dma_hw_free,
+        .prepare =      kirkwood_dma_prepare,
+        .pointer =      kirkwood_dma_pointer,
+};
+
+static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm,
+                int stream)
+{
+        struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+        struct snd_dma_buffer *buf = &substream->dma_buffer;
+        size_t size = kirkwood_dma_snd_hw.buffer_bytes_max;
+
+        buf->dev.type = SNDRV_DMA_TYPE_DEV;
+        buf->dev.dev = pcm->card->dev;
+        buf->area = dma_alloc_coherent(pcm->card->dev, size,
+                        &buf->addr, GFP_KERNEL);
+        if (!buf->area)
+                return -ENOMEM;
+        buf->bytes = size;
+        buf->private_data = NULL;
+
+        return 0;
+}
+
+static int kirkwood_dma_new(struct snd_card *card,
+                struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+        int ret;
+
+        if (!card->dev->dma_mask)
+                card->dev->dma_mask = &kirkwood_dma_dmamask;
+        if (!card->dev->coherent_dma_mask)
+                card->dev->coherent_dma_mask = 0xffffffff;
+
+        if (dai->playback.channels_min) {
+                ret = kirkwood_dma_preallocate_dma_buffer(pcm,
+                                SNDRV_PCM_STREAM_PLAYBACK);
+                if (ret)
+                        return ret;
+        }
+
+        if (dai->capture.channels_min) {
+                ret = kirkwood_dma_preallocate_dma_buffer(pcm,
+                                SNDRV_PCM_STREAM_CAPTURE);
+                if (ret)
+                        return ret;
+        }
+
+        return 0;
+}
+
+static void kirkwood_dma_free_dma_buffers(struct snd_pcm *pcm)
+{
+        struct snd_pcm_substream *substream;
+        struct snd_dma_buffer *buf;
+        int stream;
+
+        for (stream = 0; stream < 2; stream++) {
+                substream = pcm->streams[stream].substream;
+                if (!substream)
+                        continue;
+                buf = &substream->dma_buffer;
+                if (!buf->area)
+                        continue;
+
+                dma_free_coherent(pcm->card->dev, buf->bytes,
+                                buf->area, buf->addr);
+                buf->area = NULL;
+        }
+}
+
+struct snd_soc_platform kirkwood_soc_platform = {
+        .name           = "kirkwood-dma",
+        .pcm_ops        = &kirkwood_dma_ops,
+        .pcm_new        = kirkwood_dma_new,
+        .pcm_free       = kirkwood_dma_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(kirkwood_soc_platform);
+
+static int __init kirkwood_soc_platform_init(void)
+{
+        return snd_soc_register_platform(&kirkwood_soc_platform);
+}
+module_init(kirkwood_soc_platform_init);
+
+static void __exit kirkwood_soc_platform_exit(void)
+{
+        snd_soc_unregister_platform(&kirkwood_soc_platform);
+}
+module_exit(kirkwood_soc_platform_exit);
+
+MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>");
+MODULE_DESCRIPTION("Marvell Kirkwood Audio DMA module");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/kirkwood/kirkwood-dma.h b/sound/soc/kirkwood/kirkwood-dma.h
new file mode 100644
index 000000000000..ba4454cd34f1
--- /dev/null
+++ b/sound/soc/kirkwood/kirkwood-dma.h
@@ -0,0 +1,17 @@
+/*
+ * kirkwood-dma.h
+ *
+ * (c) 2010 Arnaud Patard <apatard@mandriva.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 _KIRKWOOD_DMA_H
+#define _KIRKWOOD_DMA_H
+
+extern struct snd_soc_platform kirkwood_soc_platform;
+
+#endif
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
new file mode 100644
index 000000000000..981ffc2a13c8
--- /dev/null
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -0,0 +1,495 @@
+/*
+ * kirkwood-i2s.c
+ *
+ * (c) 2010 Arnaud Patard <apatard@mandriva.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+#include <linux/mbus.h>
+#include <linux/delay.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <plat/audio.h>
+#include "kirkwood-i2s.h"
+#include "kirkwood.h"
+
+#define DRV_NAME        "kirkwood-i2s"
+
+#define KIRKWOOD_I2S_RATES \
+        (SNDRV_PCM_RATE_44100 | \
+         SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+#define KIRKWOOD_I2S_FORMATS \
+        (SNDRV_PCM_FMTBIT_S16_LE | \
+         SNDRV_PCM_FMTBIT_S24_LE | \
+         SNDRV_PCM_FMTBIT_S32_LE)
+
+
+struct snd_soc_dai kirkwood_i2s_dai;
+static struct kirkwood_dma_data *priv;
+
+static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+                unsigned int fmt)
+{
+        unsigned long mask;
+        unsigned long value;
+
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_RIGHT_J:
+                mask = KIRKWOOD_I2S_CTL_RJ;
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                mask = KIRKWOOD_I2S_CTL_LJ;
+                break;
+        case SND_SOC_DAIFMT_I2S:
+                mask = KIRKWOOD_I2S_CTL_I2S;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /*
+         * Set same format for playback and record
+         * This avoids some troubles.
+         */
+        value = readl(priv->io+KIRKWOOD_I2S_PLAYCTL);
+        value &= ~KIRKWOOD_I2S_CTL_JUST_MASK;
+        value |= mask;
+        writel(value, priv->io+KIRKWOOD_I2S_PLAYCTL);
+
+        value = readl(priv->io+KIRKWOOD_I2S_RECCTL);
+        value &= ~KIRKWOOD_I2S_CTL_JUST_MASK;
+        value |= mask;
+        writel(value, priv->io+KIRKWOOD_I2S_RECCTL);
+
+        return 0;
+}
+
+static inline void kirkwood_set_dco(void __iomem *io, unsigned long rate)
+{
+        unsigned long value;
+
+        value = KIRKWOOD_DCO_CTL_OFFSET_0;
+        switch (rate) {
+        default:
+        case 44100:
+                value |= KIRKWOOD_DCO_CTL_FREQ_11;
+                break;
+        case 48000:
+                value |= KIRKWOOD_DCO_CTL_FREQ_12;
+                break;
+        case 96000:
+                value |= KIRKWOOD_DCO_CTL_FREQ_24;
+                break;
+        }
+        writel(value, io + KIRKWOOD_DCO_CTL);
+
+        /* wait for dco locked */
+        do {
+                cpu_relax();
+                value = readl(io + KIRKWOOD_DCO_SPCR_STATUS);
+                value &= KIRKWOOD_DCO_SPCR_STATUS;
+        } while (value == 0);
+}
+
+static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream,
+                                 struct snd_pcm_hw_params *params,
+                                 struct snd_soc_dai *dai)
+{
+        unsigned int i2s_reg, reg;
+        unsigned long i2s_value, value;
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                i2s_reg = KIRKWOOD_I2S_PLAYCTL;
+                reg = KIRKWOOD_PLAYCTL;
+        } else {
+                i2s_reg = KIRKWOOD_I2S_RECCTL;
+                reg = KIRKWOOD_RECCTL;
+        }
+
+        /* set dco conf */
+        kirkwood_set_dco(priv->io, params_rate(params));
+
+        i2s_value = readl(priv->io+i2s_reg);
+        i2s_value &= ~KIRKWOOD_I2S_CTL_SIZE_MASK;
+
+        value = readl(priv->io+reg);
+        value &= ~KIRKWOOD_PLAYCTL_SIZE_MASK;
+
+        /*
+         * Size settings in play/rec i2s control regs and play/rec control
+         * regs must be the same.
+         */
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16;
+                value |= KIRKWOOD_PLAYCTL_SIZE_16_C;
+                break;
+        /*
+         * doesn't work... S20_3LE != kirkwood 20bit format ?
+         *
+        case SNDRV_PCM_FORMAT_S20_3LE:
+                i2s_value |= KIRKWOOD_I2S_CTL_SIZE_20;
+                value |= KIRKWOOD_PLAYCTL_SIZE_20;
+                break;
+        */
+        case SNDRV_PCM_FORMAT_S24_LE:
+                i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24;
+                value |= KIRKWOOD_PLAYCTL_SIZE_24;
+                break;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32;
+                value |= KIRKWOOD_PLAYCTL_SIZE_32;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                value &= ~KIRKWOOD_PLAYCTL_MONO_MASK;
+                if (params_channels(params) == 1)
+                        value |= KIRKWOOD_PLAYCTL_MONO_BOTH;
+                else
+                        value |= KIRKWOOD_PLAYCTL_MONO_OFF;
+        }
+
+        writel(i2s_value, priv->io+i2s_reg);
+        writel(value, priv->io+reg);
+
+        return 0;
+}
+
+static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream,
+                                int cmd, struct snd_soc_dai *dai)
+{
+        unsigned long value;
+
+        /*
+         * specs says KIRKWOOD_PLAYCTL must be read 2 times before
+         * changing it. So read 1 time here and 1 later.
+         */
+        value = readl(priv->io + KIRKWOOD_PLAYCTL);
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+                /* stop audio, enable interrupts */
+                value = readl(priv->io + KIRKWOOD_PLAYCTL);
+                value |= KIRKWOOD_PLAYCTL_PAUSE;
+                writel(value, priv->io + KIRKWOOD_PLAYCTL);
+
+                value = readl(priv->io + KIRKWOOD_INT_MASK);
+                value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES;
+                writel(value, priv->io + KIRKWOOD_INT_MASK);
+
+                /* configure audio & enable i2s playback */
+                value = readl(priv->io + KIRKWOOD_PLAYCTL);
+                value &= ~KIRKWOOD_PLAYCTL_BURST_MASK;
+                value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE
+                                | KIRKWOOD_PLAYCTL_SPDIF_EN);
+
+                if (priv->burst == 32)
+                        value |= KIRKWOOD_PLAYCTL_BURST_32;
+                else
+                        value |= KIRKWOOD_PLAYCTL_BURST_128;
+                value |= KIRKWOOD_PLAYCTL_I2S_EN;
+                writel(value, priv->io + KIRKWOOD_PLAYCTL);
+                break;
+
+        case SNDRV_PCM_TRIGGER_STOP:
+                /* stop audio, disable interrupts */
+                value = readl(priv->io + KIRKWOOD_PLAYCTL);
+                value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE;
+                writel(value, priv->io + KIRKWOOD_PLAYCTL);
+
+                value = readl(priv->io + KIRKWOOD_INT_MASK);
+                value &= ~KIRKWOOD_INT_CAUSE_PLAY_BYTES;
+                writel(value, priv->io + KIRKWOOD_INT_MASK);
+
+                /* disable all playbacks */
+                value = readl(priv->io + KIRKWOOD_PLAYCTL);
+                value &= ~(KIRKWOOD_PLAYCTL_I2S_EN | KIRKWOOD_PLAYCTL_SPDIF_EN);
+                writel(value, priv->io + KIRKWOOD_PLAYCTL);
+                break;
+
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+                value = readl(priv->io + KIRKWOOD_PLAYCTL);
+                value |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE;
+                writel(value, priv->io + KIRKWOOD_PLAYCTL);
+                break;
+
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                value = readl(priv->io + KIRKWOOD_PLAYCTL);
+                value &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE);
+                writel(value, priv->io + KIRKWOOD_PLAYCTL);
+                break;
+
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream,
+                                int cmd, struct snd_soc_dai *dai)
+{
+        unsigned long value;
+
+        value = readl(priv->io + KIRKWOOD_RECCTL);
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+                /* stop audio, enable interrupts */
+                value = readl(priv->io + KIRKWOOD_RECCTL);
+                value |= KIRKWOOD_RECCTL_PAUSE;
+                writel(value, priv->io + KIRKWOOD_RECCTL);
+
+                value = readl(priv->io + KIRKWOOD_INT_MASK);
+                value |= KIRKWOOD_INT_CAUSE_REC_BYTES;
+                writel(value, priv->io + KIRKWOOD_INT_MASK);
+
+                /* configure audio & enable i2s record */
+                value = readl(priv->io + KIRKWOOD_RECCTL);
+                value &= ~KIRKWOOD_RECCTL_BURST_MASK;
+                value &= ~KIRKWOOD_RECCTL_MONO;
+                value &= ~(KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE
+                        | KIRKWOOD_RECCTL_SPDIF_EN);
+
+                if (priv->burst == 32)
+                        value |= KIRKWOOD_RECCTL_BURST_32;
+                else
+                        value |= KIRKWOOD_RECCTL_BURST_128;
+                value |= KIRKWOOD_RECCTL_I2S_EN;
+
+                writel(value, priv->io + KIRKWOOD_RECCTL);
+                break;
+
+        case SNDRV_PCM_TRIGGER_STOP:
+                /* stop audio, disable interrupts */
+                value = readl(priv->io + KIRKWOOD_RECCTL);
+                value |= KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE;
+                writel(value, priv->io + KIRKWOOD_RECCTL);
+
+                value = readl(priv->io + KIRKWOOD_INT_MASK);
+                value &= ~KIRKWOOD_INT_CAUSE_REC_BYTES;
+                writel(value, priv->io + KIRKWOOD_INT_MASK);
+
+                /* disable all records */
+                value = readl(priv->io + KIRKWOOD_RECCTL);
+                value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+                writel(value, priv->io + KIRKWOOD_RECCTL);
+                break;
+
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+                value = readl(priv->io + KIRKWOOD_RECCTL);
+                value |= KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE;
+                writel(value, priv->io + KIRKWOOD_RECCTL);
+                break;
+
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                value = readl(priv->io + KIRKWOOD_RECCTL);
+                value &= ~(KIRKWOOD_RECCTL_PAUSE | KIRKWOOD_RECCTL_MUTE);
+                writel(value, priv->io + KIRKWOOD_RECCTL);
+                break;
+
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+                               struct snd_soc_dai *dai)
+{
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                return kirkwood_i2s_play_trigger(substream, cmd, dai);
+        else
+                return kirkwood_i2s_rec_trigger(substream, cmd, dai);
+
+        return 0;
+}
+
+static int kirkwood_i2s_probe(struct platform_device *pdev,
+                             struct snd_soc_dai *dai)
+{
+        unsigned long value;
+        unsigned int reg_data;
+
+        /* put system in a "safe" state : */
+        /* disable audio interrupts */
+        writel(0xffffffff, priv->io + KIRKWOOD_INT_CAUSE);
+        writel(0, priv->io + KIRKWOOD_INT_MASK);
+
+        reg_data = readl(priv->io + 0x1200);
+        reg_data &= (~(0x333FF8));
+        reg_data |= 0x111D18;
+        writel(reg_data, priv->io + 0x1200);
+
+        msleep(500);
+
+        reg_data = readl(priv->io + 0x1200);
+        reg_data &= (~(0x333FF8));
+        reg_data |= 0x111D18;
+        writel(reg_data, priv->io + 0x1200);
+
+        /* disable playback/record */
+        value = readl(priv->io + KIRKWOOD_PLAYCTL);
+        value &= ~(KIRKWOOD_PLAYCTL_I2S_EN|KIRKWOOD_PLAYCTL_SPDIF_EN);
+        writel(value, priv->io + KIRKWOOD_PLAYCTL);
+
+        value = readl(priv->io + KIRKWOOD_RECCTL);
+        value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN);
+        writel(value, priv->io + KIRKWOOD_RECCTL);
+
+        return 0;
+
+}
+
+static void kirkwood_i2s_remove(struct platform_device *pdev,
+                                struct snd_soc_dai *dai)
+{
+}
+
+static struct snd_soc_dai_ops kirkwood_i2s_dai_ops = {
+        .trigger        = kirkwood_i2s_trigger,
+        .hw_params      = kirkwood_i2s_hw_params,
+        .set_fmt        = kirkwood_i2s_set_fmt,
+};
+
+
+struct snd_soc_dai kirkwood_i2s_dai = {
+        .name = DRV_NAME,
+        .id = 0,
+        .probe = kirkwood_i2s_probe,
+        .remove = kirkwood_i2s_remove,
+        .playback = {
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = KIRKWOOD_I2S_RATES,
+                .formats = KIRKWOOD_I2S_FORMATS,},
+        .capture = {
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = KIRKWOOD_I2S_RATES,
+                .formats = KIRKWOOD_I2S_FORMATS,},
+        .ops = &kirkwood_i2s_dai_ops,
+};
+EXPORT_SYMBOL_GPL(kirkwood_i2s_dai);
+
+static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev)
+{
+        struct resource *mem;
+        struct kirkwood_asoc_platform_data *data =
+                pdev->dev.platform_data;
+        int err;
+
+        priv = kzalloc(sizeof(struct kirkwood_dma_data), GFP_KERNEL);
+        if (!priv) {
+                dev_err(&pdev->dev, "allocation failed\n");
+                err = -ENOMEM;
+                goto error;
+        }
+
+        mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!mem) {
+                dev_err(&pdev->dev, "platform_get_resource failed\n");
+                err = -ENXIO;
+                goto err_alloc;
+        }
+
+        priv->mem = request_mem_region(mem->start, SZ_16K, DRV_NAME);
+        if (!priv->mem) {
+                dev_err(&pdev->dev, "request_mem_region failed\n");
+                err = -EBUSY;
+                goto error;
+        }
+
+        priv->io = ioremap(priv->mem->start, SZ_16K);
+        if (!priv->io) {
+                dev_err(&pdev->dev, "ioremap failed\n");
+                err = -ENOMEM;
+                goto err_iomem;
+        }
+
+        priv->irq = platform_get_irq(pdev, 0);
+        if (priv->irq <= 0) {
+                dev_err(&pdev->dev, "platform_get_irq failed\n");
+                err = -ENXIO;
+                goto err_ioremap;
+        }
+
+        if (!data || !data->dram) {
+                dev_err(&pdev->dev, "no platform data ?!\n");
+                err = -EINVAL;
+                goto err_ioremap;
+        }
+
+        priv->dram = data->dram;
+        priv->burst = data->burst;
+
+        kirkwood_i2s_dai.capture.dma_data = priv;
+        kirkwood_i2s_dai.playback.dma_data = priv;
+
+        return snd_soc_register_dai(&kirkwood_i2s_dai);
+
+err_ioremap:
+        iounmap(priv->io);
+err_iomem:
+        release_mem_region(priv->mem->start, SZ_16K);
+err_alloc:
+        kfree(priv);
+error:
+        return err;
+}
+
+static __devexit int kirkwood_i2s_dev_remove(struct platform_device *pdev)
+{
+        if (priv) {
+                iounmap(priv->io);
+                release_mem_region(priv->mem->start, SZ_16K);
+                kfree(priv);
+        }
+        snd_soc_unregister_dai(&kirkwood_i2s_dai);
+        return 0;
+}
+
+static struct platform_driver kirkwood_i2s_driver = {
+        .probe  = kirkwood_i2s_dev_probe,
+        .remove = kirkwood_i2s_dev_remove,
+        .driver = {
+                .name = DRV_NAME,
+                .owner = THIS_MODULE,
+        },
+};
+
+static int __init kirkwood_i2s_init(void)
+{
+        return platform_driver_register(&kirkwood_i2s_driver);
+}
+module_init(kirkwood_i2s_init);
+
+static void __exit kirkwood_i2s_exit(void)
+{
+        platform_driver_unregister(&kirkwood_i2s_driver);
+}
+module_exit(kirkwood_i2s_exit);
+
+/* Module information */
+MODULE_AUTHOR("Arnaud Patard, <apatard@mandriva.com>");
+MODULE_DESCRIPTION("Kirkwood I2S SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:kirkwood-i2s");
diff --git a/sound/soc/kirkwood/kirkwood-i2s.h b/sound/soc/kirkwood/kirkwood-i2s.h
new file mode 100644
index 000000000000..c5595c616d7a
--- /dev/null
+++ b/sound/soc/kirkwood/kirkwood-i2s.h
@@ -0,0 +1,17 @@
+/*
+ * kirkwood-i2s.h
+ *
+ * (c) 2010 Arnaud Patard <apatard@mandriva.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 _KIRKWOOD_I2S_H
+#define _KIRKWOOD_I2S_H
+
+extern struct snd_soc_dai kirkwood_i2s_dai;
+
+#endif
diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c
new file mode 100644
index 000000000000..0353d06bc41a
--- /dev/null
+++ b/sound/soc/kirkwood/kirkwood-openrd.c
@@ -0,0 +1,126 @@
+/*
+ * kirkwood-openrd.c
+ *
+ * (c) 2010 Arnaud Patard <apatard@mandriva.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/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <mach/kirkwood.h>
+#include <plat/audio.h>
+#include <asm/mach-types.h>
+#include "kirkwood-i2s.h"
+#include "kirkwood-dma.h"
+#include "../codecs/cs42l51.h"
+
+static int openrd_client_hw_params(struct snd_pcm_substream *substream,
+                struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        int ret;
+        unsigned int freq, fmt;
+
+        fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
+        ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+        if (ret < 0)
+                return ret;
+
+        ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+        if (ret < 0)
+                return ret;
+
+        switch (params_rate(params)) {
+        default:
+        case 44100:
+                freq = 11289600;
+                break;
+        case 48000:
+                freq = 12288000;
+                break;
+        case 96000:
+                freq = 24576000;
+                break;
+        }
+
+        return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN);
+
+}
+
+static struct snd_soc_ops openrd_client_ops = {
+        .hw_params = openrd_client_hw_params,
+};
+
+
+static struct snd_soc_dai_link openrd_client_dai[] = {
+{
+        .name = "CS42L51",
+        .stream_name = "CS42L51 HiFi",
+        .cpu_dai = &kirkwood_i2s_dai,
+        .codec_dai = &cs42l51_dai,
+        .ops = &openrd_client_ops,
+},
+};
+
+
+static struct snd_soc_card openrd_client = {
+        .name = "OpenRD Client",
+        .platform = &kirkwood_soc_platform,
+        .dai_link = openrd_client_dai,
+        .num_links = ARRAY_SIZE(openrd_client_dai),
+};
+
+static struct snd_soc_device openrd_client_snd_devdata = {
+        .card = &openrd_client,
+        .codec_dev = &soc_codec_device_cs42l51,
+};
+
+static struct platform_device *openrd_client_snd_device;
+
+static int __init openrd_client_init(void)
+{
+        int ret;
+
+        if (!machine_is_openrd_client())
+                return 0;
+
+        openrd_client_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!openrd_client_snd_device)
+                return -ENOMEM;
+
+        platform_set_drvdata(openrd_client_snd_device,
+                        &openrd_client_snd_devdata);
+        openrd_client_snd_devdata.dev = &openrd_client_snd_device->dev;
+
+        ret = platform_device_add(openrd_client_snd_device);
+        if (ret) {
+                printk(KERN_ERR "%s: platform_device_add failed\n", __func__);
+                platform_device_put(openrd_client_snd_device);
+        }
+
+        return ret;
+}
+
+static void __exit openrd_client_exit(void)
+{
+        platform_device_unregister(openrd_client_snd_device);
+}
+
+module_init(openrd_client_init);
+module_exit(openrd_client_exit);
+
+/* Module information */
+MODULE_AUTHOR("Arnaud Patard <apatard@mandriva.com>");
+MODULE_DESCRIPTION("ALSA SoC OpenRD Client");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:soc-audio");
diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h
new file mode 100644
index 000000000000..bb6e6a5648c9
--- /dev/null
+++ b/sound/soc/kirkwood/kirkwood.h
@@ -0,0 +1,129 @@
+/*
+ * kirkwood.h
+ *
+ * (c) 2010 Arnaud Patard <apatard@mandriva.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 _KIRKWOOD_AUDIO_H
+#define _KIRKWOOD_AUDIO_H
+
+#define KIRKWOOD_RECORD_WIN                     0
+#define KIRKWOOD_PLAYBACK_WIN                   1
+#define KIRKWOOD_MAX_AUDIO_WIN                  2
+
+#define KIRKWOOD_AUDIO_WIN_BASE_REG(win)        (0xA00 + ((win)<<3))
+#define KIRKWOOD_AUDIO_WIN_CTRL_REG(win)        (0xA04 + ((win)<<3))
+
+
+#define KIRKWOOD_RECCTL                 0x1000
+#define KIRKWOOD_RECCTL_SPDIF_EN                (1<<11)
+#define KIRKWOOD_RECCTL_I2S_EN                  (1<<10)
+#define KIRKWOOD_RECCTL_PAUSE                   (1<<9)
+#define KIRKWOOD_RECCTL_MUTE                    (1<<8)
+#define KIRKWOOD_RECCTL_BURST_MASK              (3<<5)
+#define KIRKWOOD_RECCTL_BURST_128               (2<<5)
+#define KIRKWOOD_RECCTL_BURST_32                (1<<5)
+#define KIRKWOOD_RECCTL_MONO                    (1<<4)
+#define KIRKWOOD_RECCTL_MONO_CHAN_RIGHT (1<<3)
+#define KIRKWOOD_RECCTL_MONO_CHAN_LEFT          (0<<3)
+#define KIRKWOOD_RECCTL_SIZE_MASK               (7<<0)
+#define KIRKWOOD_RECCTL_SIZE_16         (7<<0)
+#define KIRKWOOD_RECCTL_SIZE_16_C               (3<<0)
+#define KIRKWOOD_RECCTL_SIZE_20         (2<<0)
+#define KIRKWOOD_RECCTL_SIZE_24         (1<<0)
+#define KIRKWOOD_RECCTL_SIZE_32         (0<<0)
+
+#define KIRKWOOD_REC_BUF_ADDR                   0x1004
+#define KIRKWOOD_REC_BUF_SIZE                   0x1008
+#define KIRKWOOD_REC_BYTE_COUNT                 0x100C
+
+#define KIRKWOOD_PLAYCTL                        0x1100
+#define KIRKWOOD_PLAYCTL_PLAY_BUSY              (1<<16)
+#define KIRKWOOD_PLAYCTL_BURST_MASK             (3<<11)
+#define KIRKWOOD_PLAYCTL_BURST_128              (2<<11)
+#define KIRKWOOD_PLAYCTL_BURST_32               (1<<11)
+#define KIRKWOOD_PLAYCTL_PAUSE                  (1<<9)
+#define KIRKWOOD_PLAYCTL_SPDIF_MUTE             (1<<8)
+#define KIRKWOOD_PLAYCTL_MONO_MASK              (3<<5)
+#define KIRKWOOD_PLAYCTL_MONO_BOTH              (3<<5)
+#define KIRKWOOD_PLAYCTL_MONO_OFF               (0<<5)
+#define KIRKWOOD_PLAYCTL_I2S_MUTE               (1<<7)
+#define KIRKWOOD_PLAYCTL_SPDIF_EN               (1<<4)
+#define KIRKWOOD_PLAYCTL_I2S_EN         (1<<3)
+#define KIRKWOOD_PLAYCTL_SIZE_MASK              (7<<0)
+#define KIRKWOOD_PLAYCTL_SIZE_16                (7<<0)
+#define KIRKWOOD_PLAYCTL_SIZE_16_C              (3<<0)
+#define KIRKWOOD_PLAYCTL_SIZE_20                (2<<0)
+#define KIRKWOOD_PLAYCTL_SIZE_24                (1<<0)
+#define KIRKWOOD_PLAYCTL_SIZE_32                (0<<0)
+
+#define KIRKWOOD_PLAY_BUF_ADDR                  0x1104
+#define KIRKWOOD_PLAY_BUF_SIZE                  0x1108
+#define KIRKWOOD_PLAY_BYTE_COUNT                0x110C
+
+#define KIRKWOOD_DCO_CTL                        0x1204
+#define KIRKWOOD_DCO_CTL_OFFSET_MASK            (0xFFF<<2)
+#define KIRKWOOD_DCO_CTL_OFFSET_0               (0x800<<2)
+#define KIRKWOOD_DCO_CTL_FREQ_MASK              (3<<0)
+#define KIRKWOOD_DCO_CTL_FREQ_11                (0<<0)
+#define KIRKWOOD_DCO_CTL_FREQ_12                (1<<0)
+#define KIRKWOOD_DCO_CTL_FREQ_24                (2<<0)
+
+#define KIRKWOOD_DCO_SPCR_STATUS                0x120c
+#define KIRKWOOD_DCO_SPCR_STATUS_DCO_LOCK       (1<<16)
+
+#define KIRKWOOD_ERR_CAUSE                      0x1300
+#define KIRKWOOD_ERR_MASK                       0x1304
+
+#define KIRKWOOD_INT_CAUSE                      0x1308
+#define KIRKWOOD_INT_MASK                       0x130C
+#define KIRKWOOD_INT_CAUSE_PLAY_BYTES           (1<<14)
+#define KIRKWOOD_INT_CAUSE_REC_BYTES            (1<<13)
+#define KIRKWOOD_INT_CAUSE_DMA_PLAY_END (1<<7)
+#define KIRKWOOD_INT_CAUSE_DMA_PLAY_3Q          (1<<6)
+#define KIRKWOOD_INT_CAUSE_DMA_PLAY_HALF        (1<<5)
+#define KIRKWOOD_INT_CAUSE_DMA_PLAY_1Q          (1<<4)
+#define KIRKWOOD_INT_CAUSE_DMA_REC_END          (1<<3)
+#define KIRKWOOD_INT_CAUSE_DMA_REC_3Q           (1<<2)
+#define KIRKWOOD_INT_CAUSE_DMA_REC_HALF (1<<1)
+#define KIRKWOOD_INT_CAUSE_DMA_REC_1Q           (1<<0)
+
+#define KIRKWOOD_REC_BYTE_INT_COUNT             0x1310
+#define KIRKWOOD_PLAY_BYTE_INT_COUNT            0x1314
+#define KIRKWOOD_BYTE_INT_COUNT_MASK            0xffffff
+
+#define KIRKWOOD_I2S_PLAYCTL                    0x2508
+#define KIRKWOOD_I2S_RECCTL                     0x2408
+#define KIRKWOOD_I2S_CTL_JUST_MASK              (0xf<<26)
+#define KIRKWOOD_I2S_CTL_LJ                     (0<<26)
+#define KIRKWOOD_I2S_CTL_I2S                    (5<<26)
+#define KIRKWOOD_I2S_CTL_RJ                     (8<<26)
+#define KIRKWOOD_I2S_CTL_SIZE_MASK              (3<<30)
+#define KIRKWOOD_I2S_CTL_SIZE_16                (3<<30)
+#define KIRKWOOD_I2S_CTL_SIZE_20                (2<<30)
+#define KIRKWOOD_I2S_CTL_SIZE_24                (1<<30)
+#define KIRKWOOD_I2S_CTL_SIZE_32                (0<<30)
+
+#define KIRKWOOD_AUDIO_BUF_MAX                  (16*1024*1024)
+
+/* Theses values come from the marvell alsa driver */
+/* need to find where they come from               */
+#define KIRKWOOD_SND_MIN_PERIODS                8
+#define KIRKWOOD_SND_MAX_PERIODS                16
+#define KIRKWOOD_SND_MIN_PERIOD_BYTES           0x4000
+#define KIRKWOOD_SND_MAX_PERIOD_BYTES           0x4000
+
+struct kirkwood_dma_data {
+        struct resource *mem;
+        void __iomem *io;
+        int irq;
+        int burst;
+        struct mbus_dram_target_info *dram;
+};
+
+#endif
diff --git a/sound/soc/nuc900/Kconfig b/sound/soc/nuc900/Kconfig
new file mode 100644
index 000000000000..a0ed1c618f60
--- /dev/null
+++ b/sound/soc/nuc900/Kconfig
@@ -0,0 +1,27 @@
+##
+## NUC900 series AC97 API
+##
+config SND_SOC_NUC900
+        tristate "SoC Audio for NUC900 series"
+        depends on ARCH_W90X900
+        help
+          This option enables support for AC97 mode on the NUC900 SoC.
+
+config SND_SOC_NUC900_AC97
+        tristate
+        select AC97_BUS
+        select SND_AC97_CODEC
+        select SND_SOC_AC97_BUS
+
+
+##
+## Boards
+##
+config SND_SOC_NUC900EVB
+        tristate "NUC900 AC97 support for demo board"
+        depends on SND_SOC_NUC900
+        select SND_SOC_NUC900_AC97
+        select SND_SOC_AC97_CODEC
+        help
+          Select this option to enable audio (AC97) on the
+          NUC900 demoboard.
diff --git a/sound/soc/nuc900/Makefile b/sound/soc/nuc900/Makefile
new file mode 100644
index 000000000000..7e46c7150316
--- /dev/null
+++ b/sound/soc/nuc900/Makefile
@@ -0,0 +1,11 @@
+# NUC900 series audio
+snd-soc-nuc900-pcm-objs := nuc900-pcm.o
+snd-soc-nuc900-ac97-objs := nuc900-ac97.o
+
+obj-$(CONFIG_SND_SOC_NUC900) += snd-soc-nuc900-pcm.o
+obj-$(CONFIG_SND_SOC_NUC900_AC97) += snd-soc-nuc900-ac97.o
+
+# Boards
+snd-soc-nuc900-audio-objs := nuc900-audio.o
+
+obj-$(CONFIG_SND_SOC_NUC900EVB) += snd-soc-nuc900-audio.o
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
new file mode 100644
index 000000000000..caa7c901bc2e
--- /dev/null
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -0,0 +1,430 @@
+/*
+ * Copyright (c) 2009-2010 Nuvoton technology corporation.
+ *
+ * Wan ZongShun <mcuos.com@gmail.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;version 2 of the License.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/mutex.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+
+#include <mach/mfp.h>
+
+#include "nuc900-audio.h"
+
+static DEFINE_MUTEX(ac97_mutex);
+struct nuc900_audio *nuc900_ac97_data;
+
+static int nuc900_checkready(void)
+{
+        struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
+
+        if (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS0) & CODEC_READY))
+                return -EPERM;
+
+        return 0;
+}
+
+/* AC97 controller reads codec register */
+static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97,
+                                        unsigned short reg)
+{
+        struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
+        unsigned long timeout = 0x10000, val;
+
+        mutex_lock(&ac97_mutex);
+
+        val = nuc900_checkready();
+        if (!!val) {
+                dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
+                goto out;
+        }
+
+        /* set the R_WB bit and write register index */
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, R_WB | reg);
+
+        /* set the valid frame bit and valid slots */
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
+        val |= (VALID_FRAME | SLOT1_VALID);
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val);
+
+        udelay(100);
+
+        /* polling the AC_R_FINISH */
+        while (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_R_FINISH)
+                                                                && timeout--)
+                mdelay(1);
+
+        if (!timeout) {
+                dev_err(nuc900_audio->dev, "AC97 read register time out !\n");
+                val = -EPERM;
+                goto out;
+        }
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ;
+        val &= ~SLOT1_VALID;
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val);
+
+        if (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS1) >> 2 != reg) {
+                dev_err(nuc900_audio->dev,
+                                "R_INDEX of REG_ACTL_ACIS1 not match!\n");
+        }
+
+        udelay(100);
+        val = (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS2) & 0xFFFF);
+
+out:
+        mutex_unlock(&ac97_mutex);
+        return val;
+}
+
+/* AC97 controller writes to codec register */
+static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+                                unsigned short val)
+{
+        struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
+        unsigned long tmp, timeout = 0x10000;
+
+        mutex_lock(&ac97_mutex);
+
+        tmp = nuc900_checkready();
+        if (!!tmp)
+                dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
+
+        /* clear the R_WB bit and write register index */
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, reg);
+
+        /* write register value */
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS2, val);
+
+        /* set the valid frame bit and valid slots */
+        tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
+        tmp |= SLOT1_VALID | SLOT2_VALID | VALID_FRAME;
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
+
+        udelay(100);
+
+        /* polling the AC_W_FINISH */
+        while ((AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_W_FINISH)
+                                                                && timeout--)
+                mdelay(1);
+
+        if (!timeout)
+                dev_err(nuc900_audio->dev, "AC97 write register time out !\n");
+
+        tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
+        tmp &= ~(SLOT1_VALID | SLOT2_VALID);
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
+
+        mutex_unlock(&ac97_mutex);
+
+}
+
+static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+        struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
+        unsigned long val;
+
+        mutex_lock(&ac97_mutex);
+
+        /* warm reset AC 97 */
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
+        val |= AC_W_RES;
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
+
+        udelay(100);
+
+        val = nuc900_checkready();
+        if (!!val)
+                dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
+
+        mutex_unlock(&ac97_mutex);
+}
+
+static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+        struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
+        unsigned long val;
+
+        mutex_lock(&ac97_mutex);
+
+        /* reset Audio Controller */
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
+        val |= ACTL_RESET_BIT;
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
+        val &= (~ACTL_RESET_BIT);
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
+
+        /* reset AC-link interface */
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
+        val |= AC_RESET;
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
+        val &= ~AC_RESET;
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
+
+        /* cold reset AC 97 */
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
+        val |= AC_C_RES;
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON);
+        val &= (~AC_C_RES);
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val);
+
+        udelay(100);
+
+        mutex_unlock(&ac97_mutex);
+
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+        .read           = nuc900_ac97_read,
+        .write          = nuc900_ac97_write,
+        .reset          = nuc900_ac97_cold_reset,
+        .warm_reset     = nuc900_ac97_warm_reset,
+}
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int nuc900_ac97_trigger(struct snd_pcm_substream *substream,
+                                int cmd, struct snd_soc_dai *dai)
+{
+        struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
+        int ret;
+        unsigned long val, tmp;
+
+        ret = 0;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+                val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
+                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                        tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
+                        tmp |= (SLOT3_VALID | SLOT4_VALID | VALID_FRAME);
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
+
+                        tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR);
+                        tmp |= (P_DMA_END_IRQ | P_DMA_MIDDLE_IRQ);
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, tmp);
+                        val |= AC_PLAY;
+                } else {
+                        tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR);
+                        tmp |= (R_DMA_END_IRQ | R_DMA_MIDDLE_IRQ);
+
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, tmp);
+                        val |= AC_RECORD;
+                }
+
+                AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
+
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+                val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
+                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                        tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0);
+                        tmp &= ~(SLOT3_VALID | SLOT4_VALID);
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp);
+
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, RESET_PRSR);
+                        val &= ~AC_PLAY;
+                } else {
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, RESET_PRSR);
+                        val &= ~AC_RECORD;
+                }
+
+                AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
+
+                break;
+        default:
+                ret = -EINVAL;
+        }
+
+        return ret;
+}
+
+static int nuc900_ac97_probe(struct platform_device *pdev,
+                                        struct snd_soc_dai *dai)
+{
+        struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
+        unsigned long val;
+
+        mutex_lock(&ac97_mutex);
+
+        /* enable unit clock */
+        clk_enable(nuc900_audio->clk);
+
+        /* enable audio controller and AC-link interface */
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
+        val |= (IIS_AC_PIN_SEL | ACLINK_EN);
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
+
+        mutex_unlock(&ac97_mutex);
+
+        return 0;
+}
+
+static void nuc900_ac97_remove(struct platform_device *pdev,
+                                                struct snd_soc_dai *dai)
+{
+        struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
+
+        clk_disable(nuc900_audio->clk);
+}
+
+static struct snd_soc_dai_ops nuc900_ac97_dai_ops = {
+        .trigger        = nuc900_ac97_trigger,
+};
+
+struct snd_soc_dai nuc900_ac97_dai = {
+        .name                   = "nuc900-ac97",
+        .probe                  = nuc900_ac97_probe,
+        .remove                 = nuc900_ac97_remove,
+        .ac97_control           = 1,
+        .playback = {
+                .rates          = SNDRV_PCM_RATE_8000_48000,
+                .formats        = SNDRV_PCM_FMTBIT_S16_LE,
+                .channels_min   = 1,
+                .channels_max   = 2,
+        },
+        .capture = {
+                .rates          = SNDRV_PCM_RATE_8000_48000,
+                .formats        = SNDRV_PCM_FMTBIT_S16_LE,
+                .channels_min   = 1,
+                .channels_max   = 2,
+        },
+        .ops = &nuc900_ac97_dai_ops,
+}
+EXPORT_SYMBOL_GPL(nuc900_ac97_dai);
+
+static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev)
+{
+        struct nuc900_audio *nuc900_audio;
+        int ret;
+
+        if (nuc900_ac97_data)
+                return -EBUSY;
+
+        nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL);
+        if (!nuc900_audio)
+                return -ENOMEM;
+
+        spin_lock_init(&nuc900_audio->lock);
+
+        nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!nuc900_audio->res) {
+                ret = -ENODEV;
+                goto out0;
+        }
+
+        if (!request_mem_region(nuc900_audio->res->start,
+                        resource_size(nuc900_audio->res), pdev->name)) {
+                ret = -EBUSY;
+                goto out0;
+        }
+
+        nuc900_audio->mmio = ioremap(nuc900_audio->res->start,
+                                        resource_size(nuc900_audio->res));
+        if (!nuc900_audio->mmio) {
+                ret = -ENOMEM;
+                goto out1;
+        }
+
+        nuc900_audio->clk = clk_get(&pdev->dev, NULL);
+        if (IS_ERR(nuc900_audio->clk)) {
+                ret = PTR_ERR(nuc900_audio->clk);
+                goto out2;
+        }
+
+        nuc900_audio->irq_num = platform_get_irq(pdev, 0);
+        if (!nuc900_audio->irq_num) {
+                ret = -EBUSY;
+                goto out2;
+        }
+
+        nuc900_ac97_data = nuc900_audio;
+
+        nuc900_audio->dev = nuc900_ac97_dai.dev =  &pdev->dev;
+
+        ret = snd_soc_register_dai(&nuc900_ac97_dai);
+        if (ret)
+                goto out3;
+
+        mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/
+
+        return 0;
+
+out3:
+        clk_put(nuc900_audio->clk);
+out2:
+        iounmap(nuc900_audio->mmio);
+out1:
+        release_mem_region(nuc900_audio->res->start,
+                                        resource_size(nuc900_audio->res));
+out0:
+        kfree(nuc900_audio);
+        return ret;
+}
+
+static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev)
+{
+
+        snd_soc_unregister_dai(&nuc900_ac97_dai);
+
+        clk_put(nuc900_ac97_data->clk);
+        iounmap(nuc900_ac97_data->mmio);
+        release_mem_region(nuc900_ac97_data->res->start,
+                                resource_size(nuc900_ac97_data->res));
+
+        nuc900_ac97_data = NULL;
+
+        return 0;
+}
+
+static struct platform_driver nuc900_ac97_driver = {
+        .driver = {
+                .name   = "nuc900-audio",
+                .owner  = THIS_MODULE,
+        },
+        .probe          = nuc900_ac97_drvprobe,
+        .remove         = __devexit_p(nuc900_ac97_drvremove),
+};
+
+static int __init nuc900_ac97_init(void)
+{
+        return platform_driver_register(&nuc900_ac97_driver);
+}
+
+static void __exit nuc900_ac97_exit(void)
+{
+        platform_driver_unregister(&nuc900_ac97_driver);
+}
+
+module_init(nuc900_ac97_init);
+module_exit(nuc900_ac97_exit);
+
+MODULE_AUTHOR("Wan ZongShun <mcuos.com@gmail.com>");
+MODULE_DESCRIPTION("NUC900 AC97 SoC driver!");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:nuc900-ac97");
diff --git a/sound/soc/nuc900/nuc900-audio.c b/sound/soc/nuc900/nuc900-audio.c
new file mode 100644
index 000000000000..72e6f518f7b2
--- /dev/null
+++ b/sound/soc/nuc900/nuc900-audio.c
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2010 Nuvoton technology corporation.
+ *
+ * Wan ZongShun <mcuos.com@gmail.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;version 2 of the License.
+ *
+ */
+
+#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 "../codecs/ac97.h"
+#include "nuc900-audio.h"
+
+static struct snd_soc_dai_link nuc900evb_ac97_dai = {
+        .name           = "AC97",
+        .stream_name    = "AC97 HiFi",
+        .cpu_dai        = &nuc900_ac97_dai,
+        .codec_dai      = &ac97_dai,
+};
+
+static struct snd_soc_card nuc900evb_audio_machine = {
+        .name           = "NUC900EVB_AC97",
+        .dai_link       = &nuc900evb_ac97_dai,
+        .num_links      = 1,
+        .platform       = &nuc900_soc_platform,
+};
+
+static struct snd_soc_device nuc900evb_ac97_devdata = {
+        .card           = &nuc900evb_audio_machine,
+        .codec_dev      = &soc_codec_dev_ac97,
+};
+
+static struct platform_device *nuc900evb_asoc_dev;
+
+static int __init nuc900evb_audio_init(void)
+{
+        int ret;
+
+        ret = -ENOMEM;
+        nuc900evb_asoc_dev = platform_device_alloc("soc-audio", -1);
+        if (!nuc900evb_asoc_dev)
+                goto out;
+
+        /* nuc900 board audio device */
+        platform_set_drvdata(nuc900evb_asoc_dev, &nuc900evb_ac97_devdata);
+
+        nuc900evb_ac97_devdata.dev = &nuc900evb_asoc_dev->dev;
+        ret = platform_device_add(nuc900evb_asoc_dev);
+
+        if (ret) {
+                platform_device_put(nuc900evb_asoc_dev);
+                nuc900evb_asoc_dev = NULL;
+        }
+
+out:
+        return ret;
+}
+
+static void __exit nuc900evb_audio_exit(void)
+{
+        platform_device_unregister(nuc900evb_asoc_dev);
+}
+
+module_init(nuc900evb_audio_init);
+module_exit(nuc900evb_audio_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("NUC900 Series ASoC audio support");
+MODULE_AUTHOR("Wan ZongShun");
diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h
new file mode 100644
index 000000000000..3038f519729f
--- /dev/null
+++ b/sound/soc/nuc900/nuc900-audio.h
@@ -0,0 +1,117 @@
+/*
+ * Copyright (c) 2010 Nuvoton technology corporation.
+ *
+ * Wan ZongShun <mcuos.com@gmail.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;version 2 of the License.
+ *
+ */
+
+#ifndef _NUC900_AUDIO_H
+#define _NUC900_AUDIO_H
+
+#include <linux/io.h>
+
+/* Audio Control Registers */
+#define ACTL_CON                0x00
+#define ACTL_RESET              0x04
+#define ACTL_RDSTB              0x08
+#define ACTL_RDST_LENGTH        0x0C
+#define ACTL_RDSTC              0x10
+#define ACTL_RSR                0x14
+#define ACTL_PDSTB              0x18
+#define ACTL_PDST_LENGTH        0x1C
+#define ACTL_PDSTC              0x20
+#define ACTL_PSR                0x24
+#define ACTL_IISCON             0x28
+#define ACTL_ACCON              0x2C
+#define ACTL_ACOS0              0x30
+#define ACTL_ACOS1              0x34
+#define ACTL_ACOS2              0x38
+#define ACTL_ACIS0              0x3C
+#define ACTL_ACIS1              0x40
+#define ACTL_ACIS2              0x44
+#define ACTL_COUNTER            0x48
+
+/* bit definition of REG_ACTL_CON register */
+#define R_DMA_IRQ               0x1000
+#define T_DMA_IRQ               0x0800
+#define IIS_AC_PIN_SEL          0x0100
+#define FIFO_TH                 0x0080
+#define ADC_EN                  0x0010
+#define M80_EN                  0x0008
+#define ACLINK_EN               0x0004
+#define IIS_EN                  0x0002
+
+/* bit definition of REG_ACTL_RESET register */
+#define W5691_PLAY              0x20000
+#define ACTL_RESET_BIT          0x10000
+#define RECORD_RIGHT_CHNNEL     0x08000
+#define RECORD_LEFT_CHNNEL      0x04000
+#define PLAY_RIGHT_CHNNEL       0x02000
+#define PLAY_LEFT_CHNNEL        0x01000
+#define DAC_PLAY                0x00800
+#define ADC_RECORD              0x00400
+#define M80_PLAY                0x00200
+#define AC_RECORD               0x00100
+#define AC_PLAY                 0x00080
+#define IIS_RECORD              0x00040
+#define IIS_PLAY                0x00020
+#define DAC_RESET               0x00010
+#define ADC_RESET               0x00008
+#define M80_RESET               0x00004
+#define AC_RESET                0x00002
+#define IIS_RESET               0x00001
+
+/* bit definition of REG_ACTL_ACCON register */
+#define AC_BCLK_PU_EN           0x20
+#define AC_R_FINISH             0x10
+#define AC_W_FINISH             0x08
+#define AC_W_RES                0x04
+#define AC_C_RES                0x02
+
+/* bit definition of ACTL_RSR register */
+#define R_FIFO_EMPTY            0x04
+#define R_DMA_END_IRQ           0x02
+#define R_DMA_MIDDLE_IRQ        0x01
+
+/* bit definition of ACTL_PSR register */
+#define P_FIFO_EMPTY            0x04
+#define P_DMA_END_IRQ           0x02
+#define P_DMA_MIDDLE_IRQ        0x01
+
+/* bit definition of ACTL_ACOS0 register */
+#define SLOT1_VALID             0x01
+#define SLOT2_VALID             0x02
+#define SLOT3_VALID             0x04
+#define SLOT4_VALID             0x08
+#define VALID_FRAME             0x10
+
+/* bit definition of ACTL_ACOS1 register */
+#define R_WB                    0x80
+
+#define CODEC_READY             0x10
+#define RESET_PRSR              0x00
+#define AUDIO_WRITE(addr, val)  __raw_writel(val, addr)
+#define AUDIO_READ(addr)        __raw_readl(addr)
+
+struct nuc900_audio {
+        void __iomem *mmio;
+        spinlock_t lock;
+        dma_addr_t dma_addr[2];
+        unsigned long buffersize[2];
+        unsigned long irq_num;
+        struct snd_pcm_substream *substream;
+        struct resource *res;
+        struct clk *clk;
+        struct device *dev;
+
+};
+
+extern struct nuc900_audio *nuc900_ac97_data;
+extern struct snd_soc_dai nuc900_ac97_dai;
+extern struct snd_soc_platform nuc900_soc_platform;
+
+#endif /*end _NUC900_AUDIO_H */
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
new file mode 100644
index 000000000000..e81e803b3a63
--- /dev/null
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -0,0 +1,354 @@
+/*
+ * Copyright (c) 2010 Nuvoton technology corporation.
+ *
+ * Wan ZongShun <mcuos.com@gmail.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;version 2 of the License.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/io.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 <mach/hardware.h>
+
+#include "nuc900-audio.h"
+
+static const struct snd_pcm_hardware nuc900_pcm_hardware = {
+        .info                   = SNDRV_PCM_INFO_INTERLEAVED |
+                                        SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                                        SNDRV_PCM_INFO_MMAP |
+                                        SNDRV_PCM_INFO_MMAP_VALID |
+                                        SNDRV_PCM_INFO_PAUSE |
+                                        SNDRV_PCM_INFO_RESUME,
+        .formats                = SNDRV_PCM_FMTBIT_S16_LE,
+        .channels_min           = 1,
+        .channels_max           = 2,
+        .buffer_bytes_max       = 4*1024,
+        .period_bytes_min       = 1*1024,
+        .period_bytes_max       = 4*1024,
+        .periods_min            = 1,
+        .periods_max            = 1024,
+};
+
+static int nuc900_dma_hw_params(struct snd_pcm_substream *substream,
+        struct snd_pcm_hw_params *params)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct nuc900_audio *nuc900_audio = runtime->private_data;
+        unsigned long flags;
+        int ret = 0;
+
+        spin_lock_irqsave(&nuc900_audio->lock, flags);
+
+        ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+        if (ret < 0)
+                return ret;
+
+        nuc900_audio->substream = substream;
+        nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr;
+        nuc900_audio->buffersize[substream->stream] =
+                                                params_buffer_bytes(params);
+
+        spin_unlock_irqrestore(&nuc900_audio->lock, flags);
+
+        return ret;
+}
+
+static void nuc900_update_dma_register(struct snd_pcm_substream *substream,
+                                dma_addr_t dma_addr, size_t count)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct nuc900_audio *nuc900_audio = runtime->private_data;
+        void __iomem *mmio_addr, *mmio_len;
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                mmio_addr = nuc900_audio->mmio + ACTL_PDSTB;
+                mmio_len = nuc900_audio->mmio + ACTL_PDST_LENGTH;
+        } else {
+                mmio_addr = nuc900_audio->mmio + ACTL_RDSTB;
+                mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH;
+        }
+
+        AUDIO_WRITE(mmio_addr, dma_addr);
+        AUDIO_WRITE(mmio_len, count);
+}
+
+static void nuc900_dma_start(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct nuc900_audio *nuc900_audio = runtime->private_data;
+        unsigned long val;
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
+        val |= (T_DMA_IRQ | R_DMA_IRQ);
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
+}
+
+static void nuc900_dma_stop(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct nuc900_audio *nuc900_audio = runtime->private_data;
+        unsigned long val;
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
+        val &= ~(T_DMA_IRQ | R_DMA_IRQ);
+        AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val);
+}
+
+static irqreturn_t nuc900_dma_interrupt(int irq, void *dev_id)
+{
+        struct snd_pcm_substream *substream = dev_id;
+        struct nuc900_audio *nuc900_audio = substream->runtime->private_data;
+        unsigned long val;
+
+        spin_lock(&nuc900_audio->lock);
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON);
+
+        if (val & R_DMA_IRQ) {
+                AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | R_DMA_IRQ);
+
+                val = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR);
+
+                if (val & R_DMA_MIDDLE_IRQ) {
+                        val |= R_DMA_MIDDLE_IRQ;
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val);
+                }
+
+                if (val & R_DMA_END_IRQ) {
+                        val |= R_DMA_END_IRQ;
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val);
+                }
+        } else if (val & T_DMA_IRQ) {
+                AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | T_DMA_IRQ);
+
+                val = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR);
+
+                if (val & P_DMA_MIDDLE_IRQ) {
+                        val |= P_DMA_MIDDLE_IRQ;
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val);
+                }
+
+                if (val & P_DMA_END_IRQ) {
+                        val |= P_DMA_END_IRQ;
+                        AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val);
+                }
+        } else {
+                dev_err(nuc900_audio->dev, "Wrong DMA interrupt status!\n");
+                spin_unlock(&nuc900_audio->lock);
+                return IRQ_HANDLED;
+        }
+
+        spin_unlock(&nuc900_audio->lock);
+
+        snd_pcm_period_elapsed(substream);
+
+        return IRQ_HANDLED;
+}
+
+static int nuc900_dma_hw_free(struct snd_pcm_substream *substream)
+{
+        snd_pcm_lib_free_pages(substream);
+        return 0;
+}
+
+static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct nuc900_audio *nuc900_audio = runtime->private_data;
+        unsigned long flags, val;
+
+        spin_lock_irqsave(&nuc900_audio->lock, flags);
+
+        nuc900_update_dma_register(substream,
+                                nuc900_audio->dma_addr[substream->stream],
+                                nuc900_audio->buffersize[substream->stream]);
+
+        val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET);
+
+        switch (runtime->channels) {
+        case 1:
+                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                        val &= ~(PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL);
+                        val |= PLAY_RIGHT_CHNNEL;
+                } else {
+                        val &= ~(RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL);
+                        val |= RECORD_RIGHT_CHNNEL;
+                }
+                AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
+                break;
+        case 2:
+                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                        val |= (PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL);
+                else
+                        val |= (RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL);
+                AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
+                break;
+        default:
+                return -EINVAL;
+        }
+        spin_unlock_irqrestore(&nuc900_audio->lock, flags);
+        return 0;
+}
+
+static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        int ret = 0;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+                nuc900_dma_start(substream);
+                break;
+
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+                nuc900_dma_stop(substream);
+                break;
+
+        default:
+                ret = -EINVAL;
+                break;
+        }
+
+        return ret;
+}
+
+int nuc900_dma_getposition(struct snd_pcm_substream *substream,
+                                        dma_addr_t *src, dma_addr_t *dst)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct nuc900_audio *nuc900_audio = runtime->private_data;
+
+        if (src != NULL)
+                *src = AUDIO_READ(nuc900_audio->mmio + ACTL_PDSTC);
+
+        if (dst != NULL)
+                *dst = AUDIO_READ(nuc900_audio->mmio + ACTL_RDSTC);
+
+        return 0;
+}
+
+static snd_pcm_uframes_t nuc900_dma_pointer(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        dma_addr_t src, dst;
+        unsigned long res;
+
+        nuc900_dma_getposition(substream, &src, &dst);
+
+        if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+                res = dst - runtime->dma_addr;
+        else
+                res = src - runtime->dma_addr;
+
+        return bytes_to_frames(substream->runtime, res);
+}
+
+static int nuc900_dma_open(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct nuc900_audio *nuc900_audio;
+
+        snd_soc_set_runtime_hwparams(substream, &nuc900_pcm_hardware);
+
+        nuc900_audio = nuc900_ac97_data;
+
+        if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt,
+                        IRQF_DISABLED, "nuc900-dma", substream))
+                return -EBUSY;
+
+        runtime->private_data = nuc900_audio;
+
+        return 0;
+}
+
+static int nuc900_dma_close(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct nuc900_audio *nuc900_audio = runtime->private_data;
+
+        free_irq(nuc900_audio->irq_num, substream);
+
+        return 0;
+}
+
+static int nuc900_dma_mmap(struct snd_pcm_substream *substream,
+        struct vm_area_struct *vma)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+
+        return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+                                        runtime->dma_area,
+                                        runtime->dma_addr,
+                                        runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops nuc900_dma_ops = {
+        .open           = nuc900_dma_open,
+        .close          = nuc900_dma_close,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .hw_params      = nuc900_dma_hw_params,
+        .hw_free        = nuc900_dma_hw_free,
+        .prepare        = nuc900_dma_prepare,
+        .trigger        = nuc900_dma_trigger,
+        .pointer        = nuc900_dma_pointer,
+        .mmap           = nuc900_dma_mmap,
+};
+
+static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm)
+{
+        snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32);
+static int nuc900_dma_new(struct snd_card *card,
+        struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+        if (!card->dev->dma_mask)
+                card->dev->dma_mask = &nuc900_pcm_dmamask;
+        if (!card->dev->coherent_dma_mask)
+                card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+        snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+                card->dev, 4 * 1024, (4 * 1024) - 1);
+
+        return 0;
+}
+
+struct snd_soc_platform nuc900_soc_platform = {
+        .name           = "nuc900-dma",
+        .pcm_ops        = &nuc900_dma_ops,
+        .pcm_new        = nuc900_dma_new,
+        .pcm_free       = nuc900_dma_free_dma_buffers,
+}
+EXPORT_SYMBOL_GPL(nuc900_soc_platform);
+
+static int __init nuc900_soc_platform_init(void)
+{
+        return snd_soc_register_platform(&nuc900_soc_platform);
+}
+
+static void __exit nuc900_soc_platform_exit(void)
+{
+        snd_soc_unregister_platform(&nuc900_soc_platform);
+}
+
+module_init(nuc900_soc_platform_init);
+module_exit(nuc900_soc_platform_exit);
+
+MODULE_AUTHOR("Wan ZongShun, <mcuos.com@gmail.com>");
+MODULE_DESCRIPTION("nuc900 Audio DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6f44cb4d30b8..86f213905e2c 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -59,6 +59,7 @@ struct omap_mcbsp_data {
         int                             configured;
         unsigned int                    in_freq;
         int                             clk_div;
+        int                             wlen;
 };
 
 #define to_mcbsp(priv)  container_of((priv), struct omap_mcbsp_data, bus_id)
@@ -154,20 +155,51 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
         struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+        struct omap_pcm_dma_data *dma_data;
         int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
-        int samples;
+        int words;
+
+        dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
         /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
         if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
-                samples = snd_pcm_lib_period_bytes(substream) >> 1;
+                /*
+                 * Configure McBSP threshold based on either:
+                 * packet_size, when the sDMA is in packet mode, or
+                 * based on the period size.
+                 */
+                if (dma_data->packet_size)
+                        words = dma_data->packet_size;
+                else
+                        words = snd_pcm_lib_period_bytes(substream) /
+                                                        (mcbsp_data->wlen / 8);
         else
-                samples = 1;
+                words = 1;
 
         /* Configure McBSP internal buffer usage */
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-                omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1);
+                omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, words);
         else
-                omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1);
+                omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, words);
+}
+
+static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params,
+                                    struct snd_pcm_hw_rule *rule)
+{
+        struct snd_interval *buffer_size = hw_param_interval(params,
+                                        SNDRV_PCM_HW_PARAM_BUFFER_SIZE);
+        struct snd_interval *channels = hw_param_interval(params,
+                                        SNDRV_PCM_HW_PARAM_CHANNELS);
+        struct omap_mcbsp_data *mcbsp_data = rule->private;
+        struct snd_interval frames;
+        int size;
+
+        snd_interval_any(&frames);
+        size = omap_mcbsp_get_fifo_size(mcbsp_data->bus_id);
+
+        frames.min = size / channels->min;
+        frames.integer = 1;
+        return snd_interval_refine(buffer_size, &frames);
 }
 
 static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
@@ -182,33 +214,35 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
         if (!cpu_dai->active)
                 err = omap_mcbsp_request(bus_id);
 
+        /*
+         * OMAP3 McBSP FIFO is word structured.
+         * McBSP2 has 1024 + 256 = 1280 word long buffer,
+         * McBSP1,3,4,5 has 128 word long buffer
+         * This means that the size of the FIFO depends on the sample format.
+         * For example on McBSP3:
+         * 16bit samples: size is 128 * 2 = 256 bytes
+         * 32bit samples: size is 128 * 4 = 512 bytes
+         * It is simpler to place constraint for buffer and period based on
+         * channels.
+         * McBSP3 as example again (16 or 32 bit samples):
+         * 1 channel (mono): size is 128 frames (128 words)
+         * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
+         * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
+         */
         if (cpu_is_omap343x()) {
-                int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id);
-                int max_period;
-
                 /*
-                 * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
-                 * Set constraint for minimum buffer size to the same than FIFO
-                 * size in order to avoid underruns in playback startup because
-                 * HW is keeping the DMA request active until FIFO is filled.
-                 */
-                if (bus_id == 1)
-                        snd_pcm_hw_constraint_minmax(substream->runtime,
-                                        SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
-                                        4096, UINT_MAX);
-
-                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-                        max_period = omap_mcbsp_get_max_tx_threshold(bus_id);
-                else
-                        max_period = omap_mcbsp_get_max_rx_threshold(bus_id);
-
-                max_period++;
-                max_period <<= 1;
-
-                if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
-                        snd_pcm_hw_constraint_minmax(substream->runtime,
-                                                SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
-                                                32, max_period);
+                * Rule for the buffer size. We should not allow
+                * smaller buffer than the FIFO size to avoid underruns
+                */
+                snd_pcm_hw_rule_add(substream->runtime, 0,
+                                    SNDRV_PCM_HW_PARAM_CHANNELS,
+                                    omap_mcbsp_hwrule_min_buffersize,
+                                    mcbsp_data,
+                                    SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1);
+
+                /* Make sure, that the period size is always even */
+                snd_pcm_hw_constraint_step(substream->runtime, 0,
+                                           SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
         }
 
         return err;
@@ -289,11 +323,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
         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;
+        struct omap_pcm_dma_data *dma_data;
+        int dma, bus_id = mcbsp_data->bus_id;
         int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
+        int pkt_size = 0;
         unsigned long port;
         unsigned int format, div, framesize, master;
 
+        dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream];
         if (cpu_class_is_omap1()) {
                 dma = omap1_dma_reqs[bus_id][substream->stream];
                 port = omap1_mcbsp_port[bus_id][substream->stream];
@@ -306,35 +343,74 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
         } else if (cpu_is_omap343x()) {
                 dma = omap24xx_dma_reqs[bus_id][substream->stream];
                 port = omap34xx_mcbsp_port[bus_id][substream->stream];
-                omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold =
-                                                omap_mcbsp_set_threshold;
-                /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
-                if (omap_mcbsp_get_dma_op_mode(bus_id) ==
-                                                MCBSP_DMA_MODE_THRESHOLD)
-                        sync_mode = OMAP_DMA_SYNC_FRAME;
         } else {
                 return -ENODEV;
         }
-        omap_mcbsp_dai_dma_params[id][substream->stream].name =
-                substream->stream ? "Audio Capture" : "Audio Playback";
-        omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
-        omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
-        omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
         switch (params_format(params)) {
         case SNDRV_PCM_FORMAT_S16_LE:
-                omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
-                         OMAP_DMA_DATA_TYPE_S16;
+                dma_data->data_type = OMAP_DMA_DATA_TYPE_S16;
+                wlen = 16;
                 break;
         case SNDRV_PCM_FORMAT_S32_LE:
-                omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
-                         OMAP_DMA_DATA_TYPE_S32;
+                dma_data->data_type = OMAP_DMA_DATA_TYPE_S32;
+                wlen = 32;
                 break;
         default:
                 return -EINVAL;
         }
+        if (cpu_is_omap343x()) {
+                dma_data->set_threshold = omap_mcbsp_set_threshold;
+                /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+                if (omap_mcbsp_get_dma_op_mode(bus_id) ==
+                                                MCBSP_DMA_MODE_THRESHOLD) {
+                        int period_words, max_thrsh;
+
+                        period_words = params_period_bytes(params) / (wlen / 8);
+                        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                                max_thrsh = omap_mcbsp_get_max_tx_threshold(
+                                                            mcbsp_data->bus_id);
+                        else
+                                max_thrsh = omap_mcbsp_get_max_rx_threshold(
+                                                            mcbsp_data->bus_id);
+                        /*
+                         * If the period contains less or equal number of words,
+                         * we are using the original threshold mode setup:
+                         * McBSP threshold = sDMA frame size = period_size
+                         * Otherwise we switch to sDMA packet mode:
+                         * McBSP threshold = sDMA packet size
+                         * sDMA frame size = period size
+                         */
+                        if (period_words > max_thrsh) {
+                                int divider = 0;
+
+                                /*
+                                 * Look for the biggest threshold value, which
+                                 * divides the period size evenly.
+                                 */
+                                divider = period_words / max_thrsh;
+                                if (period_words % max_thrsh)
+                                        divider++;
+                                while (period_words % divider &&
+                                        divider < period_words)
+                                        divider++;
+                                if (divider == period_words)
+                                        return -EINVAL;
+
+                                pkt_size = period_words / divider;
+                                sync_mode = OMAP_DMA_SYNC_PACKET;
+                        } else {
+                                sync_mode = OMAP_DMA_SYNC_FRAME;
+                        }
+                }
+        }
 
-        snd_soc_dai_set_dma_data(cpu_dai, substream,
-                &omap_mcbsp_dai_dma_params[id][substream->stream]);
+        dma_data->name = substream->stream ? "Audio Capture" : "Audio Playback";
+        dma_data->dma_req = dma;
+        dma_data->port_addr = port;
+        dma_data->sync_mode = sync_mode;
+        dma_data->packet_size = pkt_size;
+
+        snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
         if (mcbsp_data->configured) {
                 /* McBSP already configured by another stream */
@@ -360,7 +436,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
         switch (params_format(params)) {
         case SNDRV_PCM_FORMAT_S16_LE:
                 /* Set word lengths */
-                wlen = 16;
                 regs->rcr2      |= RWDLEN2(OMAP_MCBSP_WORD_16);
                 regs->rcr1      |= RWDLEN1(OMAP_MCBSP_WORD_16);
                 regs->xcr2      |= XWDLEN2(OMAP_MCBSP_WORD_16);
@@ -368,7 +443,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
                 break;
         case SNDRV_PCM_FORMAT_S32_LE:
                 /* Set word lengths */
-                wlen = 32;
                 regs->rcr2      |= RWDLEN2(OMAP_MCBSP_WORD_32);
                 regs->rcr1      |= RWDLEN1(OMAP_MCBSP_WORD_32);
                 regs->xcr2      |= XWDLEN2(OMAP_MCBSP_WORD_32);
@@ -409,6 +483,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
         }
 
         omap_mcbsp_config(bus_id, &mcbsp_data->regs);
+        mcbsp_data->wlen = wlen;
         mcbsp_data->configured = 1;
 
         return 0;
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 87ce842fa2e8..9eecac135bbb 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -43,12 +43,14 @@
 
 static struct regulator *omap3pandora_dac_reg;
 
-static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
-        struct snd_pcm_hw_params *params, unsigned int fmt)
+static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
+        struct snd_pcm_hw_params *params)
 {
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
         struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+                  SND_SOC_DAIFMT_CBS_CFS;
         int ret;
 
         /* Set codec DAI configuration */
@@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
         return 0;
 }
 
-static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream,
-        struct snd_pcm_hw_params *params)
-{
-        return omap3pandora_cmn_hw_params(substream, params,
-                                          SND_SOC_DAIFMT_I2S |
-                                          SND_SOC_DAIFMT_IB_NF |
-                                          SND_SOC_DAIFMT_CBS_CFS);
-}
-
-static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream,
-        struct snd_pcm_hw_params *params)
-{
-        return omap3pandora_cmn_hw_params(substream, params,
-                                          SND_SOC_DAIFMT_I2S |
-                                          SND_SOC_DAIFMT_NB_NF |
-                                          SND_SOC_DAIFMT_CBS_CFS);
-}
-
 static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
         struct snd_kcontrol *k, int event)
 {
@@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec)
         return snd_soc_dapm_sync(codec);
 }
 
-static struct snd_soc_ops omap3pandora_out_ops = {
-        .hw_params = omap3pandora_out_hw_params,
-};
-
-static struct snd_soc_ops omap3pandora_in_ops = {
-        .hw_params = omap3pandora_in_hw_params,
+static struct snd_soc_ops omap3pandora_ops = {
+        .hw_params = omap3pandora_hw_params,
 };
 
 /* Digital audio interface glue - connects codec <--> CPU */
@@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
                 .stream_name = "HiFi Out",
                 .cpu_dai = &omap_mcbsp_dai[0],
                 .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
-                .ops = &omap3pandora_out_ops,
+                .ops = &omap3pandora_ops,
                 .init = omap3pandora_out_init,
         }, {
                 .name = "TWL4030",
                 .stream_name = "Line/Mic In",
                 .cpu_dai = &omap_mcbsp_dai[1],
                 .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
-                .ops = &omap3pandora_in_ops,
+                .ops = &omap3pandora_ops,
                 .init = omap3pandora_in_init,
         }
 };
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 47d831ef2dbb..88052d29617f 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -27,6 +27,7 @@
 #include <linux/gpio.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
+#include <sound/jack.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
@@ -37,14 +38,22 @@
 #include "omap-pcm.h"
 #include "../codecs/tlv320aic3x.h"
 
+#define RX51_TVOUT_SEL_GPIO             40
+#define RX51_JACK_DETECT_GPIO           177
 /*
  * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
  * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
  */
 #define RX51_SPEAKER_AMP_TWL_GPIO       (192 + 7)
 
+enum {
+        RX51_JACK_DISABLED,
+        RX51_JACK_TVOUT,                /* tv-out */
+};
+
 static int rx51_spk_func;
 static int rx51_dmic_func;
+static int rx51_jack_func;
 
 static void rx51_ext_control(struct snd_soc_codec *codec)
 {
@@ -57,6 +66,9 @@ static void rx51_ext_control(struct snd_soc_codec *codec)
         else
                 snd_soc_dapm_disable_pin(codec, "DMic");
 
+        gpio_set_value(RX51_TVOUT_SEL_GPIO,
+                       rx51_jack_func == RX51_JACK_TVOUT);
+
         snd_soc_dapm_sync(codec);
 }
 
@@ -162,6 +174,40 @@ static int rx51_set_input(struct snd_kcontrol *kcontrol,
         return 1;
 }
 
+static int rx51_get_jack(struct snd_kcontrol *kcontrol,
+                         struct snd_ctl_elem_value *ucontrol)
+{
+        ucontrol->value.integer.value[0] = rx51_jack_func;
+
+        return 0;
+}
+
+static int rx51_set_jack(struct snd_kcontrol *kcontrol,
+                         struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+        if (rx51_jack_func == ucontrol->value.integer.value[0])
+                return 0;
+
+        rx51_jack_func = ucontrol->value.integer.value[0];
+        rx51_ext_control(codec);
+
+        return 1;
+}
+
+static struct snd_soc_jack rx51_av_jack;
+
+static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
+        {
+                .gpio = RX51_JACK_DETECT_GPIO,
+                .name = "avdet-gpio",
+                .report = SND_JACK_VIDEOOUT,
+                .invert = 1,
+                .debounce_time = 200,
+        },
+};
+
 static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
         SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
         SND_SOC_DAPM_MIC("DMic", NULL),
@@ -177,10 +223,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 static const char *spk_function[] = {"Off", "On"};
 static const char *input_function[] = {"ADC", "Digital Mic"};
+static const char *jack_function[] = {"Off", "TV-OUT"};
 
 static const struct soc_enum rx51_enum[] = {
         SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
         SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
+        SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
 };
 
 static const struct snd_kcontrol_new aic34_rx51_controls[] = {
@@ -188,10 +236,13 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = {
                      rx51_get_spk, rx51_set_spk),
         SOC_ENUM_EXT("Input Select",  rx51_enum[1],
                      rx51_get_input, rx51_set_input),
+        SOC_ENUM_EXT("Jack Function", rx51_enum[2],
+                     rx51_get_jack, rx51_set_jack),
 };
 
 static int rx51_aic34_init(struct snd_soc_codec *codec)
 {
+        struct snd_soc_card *card = codec->socdev->card;
         int err;
 
         /* Set up NC codec pins */
@@ -214,7 +265,16 @@ static int rx51_aic34_init(struct snd_soc_codec *codec)
 
         snd_soc_dapm_sync(codec);
 
-        return 0;
+        /* AV jack detection */
+        err = snd_soc_jack_new(card, "AV Jack",
+                               SND_JACK_VIDEOOUT, &rx51_av_jack);
+        if (err)
+                return err;
+        err = snd_soc_jack_add_gpios(&rx51_av_jack,
+                                     ARRAY_SIZE(rx51_av_jack_gpios),
+                                     rx51_av_jack_gpios);
+
+        return err;
 }
 
 /* Digital audio interface glue - connects codec <--> CPU */
@@ -259,6 +319,11 @@ static int __init rx51_soc_init(void)
         if (!machine_is_nokia_rx51())
                 return -ENODEV;
 
+        err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel");
+        if (err)
+                goto err_gpio_tvout_sel;
+        gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0);
+
         rx51_snd_device = platform_device_alloc("soc-audio", -1);
         if (!rx51_snd_device) {
                 err = -ENOMEM;
@@ -277,13 +342,19 @@ static int __init rx51_soc_init(void)
 err2:
         platform_device_put(rx51_snd_device);
 err1:
+        gpio_free(RX51_TVOUT_SEL_GPIO);
+err_gpio_tvout_sel:
 
         return err;
 }
 
 static void __exit rx51_soc_exit(void)
 {
+        snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
+                                rx51_av_jack_gpios);
+
         platform_device_unregister(rx51_snd_device);
+        gpio_free(RX51_TVOUT_SEL_GPIO);
 }
 
 module_init(rx51_soc_init);
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 2a7cc222d098..213963ac3c28 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,6 +1,6 @@
 config SND_S3C24XX_SOC
         tristate "SoC Audio for the Samsung S3CXXXX chips"
-        depends on ARCH_S3C2410 || ARCH_S3C64XX
+        depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210
         select S3C64XX_DMA if ARCH_S3C64XX
         help
           Say Y or M if you want to add support for codecs attached to
@@ -120,8 +120,14 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES
 
 config SND_SOC_SMDK_WM9713
         tristate "SoC AC97 Audio support for SMDK with WM9713"
-        depends on SND_S3C24XX_SOC && MACH_SMDK6410
+        depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110)
         select SND_SOC_WM9713
         select SND_S3C_SOC_AC97
         help
           Sat Y if you want to add support for SoC audio on the SMDK.
+
+config SND_S3C64XX_SOC_SMARTQ
+        tristate "SoC I2S Audio support for SmartQ board"
+        depends on SND_S3C24XX_SOC && MACH_SMARTQ
+        select SND_S3C64XX_SOC_I2S
+        select SND_SOC_WM8750
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 81d8dc503f87..50172c385d90 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -29,6 +29,7 @@ snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
 snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
 snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o
 snd-soc-smdk-wm9713-objs := smdk_wm9713.o
+snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -41,3 +42,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o
 obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o
+obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o
diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c
index ecf4fd04ae96..31f6d45b6384 100644
--- a/sound/soc/s3c24xx/s3c-ac97.c
+++ b/sound/soc/s3c24xx/s3c-ac97.c
@@ -31,7 +31,6 @@
 #define AC_CMD_DATA(x) (x & 0xffff)
 
 struct s3c_ac97_info {
-        unsigned           state;
         struct clk         *ac97_clk;
         void __iomem       *regs;
         struct mutex       lock;
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 13311c8cf965..64376b2aac73 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -32,7 +32,8 @@
 
 #undef S3C_IIS_V2_SUPPORTED
 
-#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) \
+        || defined(CONFIG_CPU_S5PV210)
 #define S3C_IIS_V2_SUPPORTED
 #endif
 
diff --git a/sound/soc/s3c24xx/smartq_wm8987.c b/sound/soc/s3c24xx/smartq_wm8987.c
new file mode 100644
index 000000000000..b480348140b0
--- /dev/null
+++ b/sound/soc/s3c24xx/smartq_wm8987.c
@@ -0,0 +1,295 @@
+/* sound/soc/s3c24xx/smartq_wm8987.c
+ *
+ * Copyright 2010 Maurus Cuelenaere <mcuelenaere@gmail.com>
+ *
+ * Based on smdk6410_wm8987.c
+ *     Copyright 2007 Wolfson Microelectronics PLC. - linux@wolfsonmicro.com
+ *     Graeme Gregory - graeme.gregory@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/platform_device.h>
+#include <linux/gpio.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "s3c-dma.h"
+#include "s3c64xx-i2s.h"
+
+#include "../codecs/wm8750.h"
+
+/*
+ * WM8987 is register compatible with WM8750, so using that as base driver.
+ */
+
+static struct snd_soc_card snd_soc_smartq;
+
+static int smartq_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_dai *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        struct s3c_i2sv2_rate_calc div;
+        unsigned int clk = 0;
+        int ret;
+
+        s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
+                                s3c_i2sv2_get_clock(cpu_dai));
+
+        switch (params_rate(params)) {
+        case 8000:
+        case 16000:
+        case 32000:
+        case 48000:
+        case 96000:
+                clk = 12288000;
+                break;
+        case 11025:
+        case 22050:
+        case 44100:
+        case 88200:
+                clk = 11289600;
+                break;
+        }
+
+        /* set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+                                             SND_SOC_DAIFMT_NB_NF |
+                                             SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        /* set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+                                           SND_SOC_DAIFMT_NB_NF |
+                                           SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        /* set the codec system clock for DAC and ADC */
+        ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+                                     SND_SOC_CLOCK_IN);
+        if (ret < 0)
+                return ret;
+
+        /* set MCLK division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, div.fs_div);
+        if (ret < 0)
+                return ret;
+
+        /* set prescaler division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_PRESCALER,
+                                     div.clk_div - 1);
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+/*
+ * SmartQ WM8987 HiFi DAI operations.
+ */
+static struct snd_soc_ops smartq_hifi_ops = {
+        .hw_params = smartq_hifi_hw_params,
+};
+
+static struct snd_soc_jack smartq_jack;
+
+static struct snd_soc_jack_pin smartq_jack_pins[] = {
+        /* Disable speaker when headphone is plugged in */
+        {
+                .pin    = "Internal Speaker",
+                .mask   = SND_JACK_HEADPHONE,
+        },
+};
+
+static struct snd_soc_jack_gpio smartq_jack_gpios[] = {
+        {
+                .gpio           = S3C64XX_GPL(12),
+                .name           = "headphone detect",
+                .report         = SND_JACK_HEADPHONE,
+                .debounce_time  = 200,
+        },
+};
+
+static const struct snd_kcontrol_new wm8987_smartq_controls[] = {
+        SOC_DAPM_PIN_SWITCH("Internal Speaker"),
+        SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+        SOC_DAPM_PIN_SWITCH("Internal Mic"),
+};
+
+static int smartq_speaker_event(struct snd_soc_dapm_widget *w,
+                                struct snd_kcontrol *k,
+                                int event)
+{
+        gpio_set_value(S3C64XX_GPK(12), SND_SOC_DAPM_EVENT_OFF(event));
+
+        return 0;
+}
+
+static const struct snd_soc_dapm_widget wm8987_dapm_widgets[] = {
+        SND_SOC_DAPM_SPK("Internal Speaker", smartq_speaker_event),
+        SND_SOC_DAPM_HP("Headphone Jack", NULL),
+        SND_SOC_DAPM_MIC("Internal Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+        {"Headphone Jack", NULL, "LOUT2"},
+        {"Headphone Jack", NULL, "ROUT2"},
+
+        {"Internal Speaker", NULL, "LOUT2"},
+        {"Internal Speaker", NULL, "ROUT2"},
+
+        {"Mic Bias", NULL, "Internal Mic"},
+        {"LINPUT2", NULL, "Mic Bias"},
+};
+
+static int smartq_wm8987_init(struct snd_soc_codec *codec)
+{
+        int err = 0;
+
+        /* Add SmartQ specific widgets */
+        snd_soc_dapm_new_controls(codec, wm8987_dapm_widgets,
+                                  ARRAY_SIZE(wm8987_dapm_widgets));
+
+        /* add SmartQ specific controls */
+        err = snd_soc_add_controls(codec, wm8987_smartq_controls,
+                                   ARRAY_SIZE(wm8987_smartq_controls));
+
+        if (err < 0)
+                return err;
+
+        /* setup SmartQ specific audio path */
+        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+        /* set endpoints to not connected */
+        snd_soc_dapm_nc_pin(codec, "LINPUT1");
+        snd_soc_dapm_nc_pin(codec, "RINPUT1");
+        snd_soc_dapm_nc_pin(codec, "OUT3");
+        snd_soc_dapm_nc_pin(codec, "ROUT1");
+
+        /* set endpoints to default off mode */
+        snd_soc_dapm_enable_pin(codec, "Internal Speaker");
+        snd_soc_dapm_enable_pin(codec, "Internal Mic");
+        snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+        err = snd_soc_dapm_sync(codec);
+        if (err)
+                return err;
+
+        /* Headphone jack detection */
+        err = snd_soc_jack_new(&snd_soc_smartq, "Headphone Jack",
+                               SND_JACK_HEADPHONE, &smartq_jack);
+        if (err)
+                return err;
+
+        err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins),
+                                    smartq_jack_pins);
+        if (err)
+                return err;
+
+        err = snd_soc_jack_add_gpios(&smartq_jack,
+                                     ARRAY_SIZE(smartq_jack_gpios),
+                                     smartq_jack_gpios);
+
+        return err;
+}
+
+static struct snd_soc_dai_link smartq_dai[] = {
+        {
+                .name           = "wm8987",
+                .stream_name    = "SmartQ Hi-Fi",
+                .cpu_dai        = &s3c64xx_i2s_dai[0],
+                .codec_dai      = &wm8750_dai,
+                .init           = smartq_wm8987_init,
+                .ops            = &smartq_hifi_ops,
+        },
+};
+
+static struct snd_soc_card snd_soc_smartq = {
+        .name = "SmartQ",
+        .platform = &s3c24xx_soc_platform,
+        .dai_link = smartq_dai,
+        .num_links = ARRAY_SIZE(smartq_dai),
+};
+
+static struct snd_soc_device smartq_snd_devdata = {
+        .card = &snd_soc_smartq,
+        .codec_dev = &soc_codec_dev_wm8750,
+};
+
+static struct platform_device *smartq_snd_device;
+
+static int __init smartq_init(void)
+{
+        int ret;
+
+        if (!machine_is_smartq7() && !machine_is_smartq5()) {
+                pr_info("Only SmartQ is supported by this ASoC driver\n");
+                return -ENODEV;
+        }
+
+        smartq_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!smartq_snd_device)
+                return -ENOMEM;
+
+        platform_set_drvdata(smartq_snd_device, &smartq_snd_devdata);
+        smartq_snd_devdata.dev = &smartq_snd_device->dev;
+
+        ret = platform_device_add(smartq_snd_device);
+        if (ret) {
+                platform_device_put(smartq_snd_device);
+                return ret;
+        }
+
+        /* Initialise GPIOs used by amplifiers */
+        ret = gpio_request(S3C64XX_GPK(12), "amplifiers shutdown");
+        if (ret) {
+                dev_err(&smartq_snd_device->dev, "Failed to register GPK12\n");
+                goto err_unregister_device;
+        }
+
+        /* Disable amplifiers */
+        ret = gpio_direction_output(S3C64XX_GPK(12), 1);
+        if (ret) {
+                dev_err(&smartq_snd_device->dev, "Failed to configure GPK12\n");
+                goto err_free_gpio_amp_shut;
+        }
+
+        return 0;
+
+err_free_gpio_amp_shut:
+        gpio_free(S3C64XX_GPK(12));
+err_unregister_device:
+        platform_device_unregister(smartq_snd_device);
+
+        return ret;
+}
+
+static void __exit smartq_exit(void)
+{
+        snd_soc_jack_free_gpios(&smartq_jack, ARRAY_SIZE(smartq_jack_gpios),
+                                smartq_jack_gpios);
+
+        platform_device_unregister(smartq_snd_device);
+}
+
+module_init(smartq_init);
+module_exit(smartq_exit);
+
+/* Module information */
+MODULE_AUTHOR("Maurus Cuelenaere <mcuelenaere@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC SmartQ WM8987");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c
index 24fd39f38ccb..5527b9e88c98 100644
--- a/sound/soc/s3c24xx/smdk_wm9713.c
+++ b/sound/soc/s3c24xx/smdk_wm9713.c
@@ -25,6 +25,9 @@ static struct snd_soc_card smdk;
  * Default CFG switch settings to use this driver:
  *
  *   SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off
+ *   SMDKC100: Set CFG6 1-3 On, CFG7 1   On
+ *   SMDKC110: Set CFGB10 1-2 Off, CFGB12 1-3 On
+ *   SMDKV210: Set CFGB10 1-2 Off, CFGB12 1-3 On
  */
 
 /*
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 5b9ac1759bd2..59e3fa7bcb05 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -451,16 +451,15 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
                 goto err_release_none;
         }
 
-        region = request_mem_region(scbmem->start,
-                                    scbmem->end - scbmem->start + 1,
-                                    pdev->name);
+        region = request_mem_region(scbmem->start, resource_size(scbmem),
+                                                                pdev->name);
         if (!region) {
                 dev_err(&pdev->dev, "I2S SCB region already claimed\n");
                 ret = -EBUSY;
                 goto err_release_none;
         }
 
-        mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1);
+        mmio = ioremap(scbmem->start, resource_size(scbmem));
         if (!mmio) {
                 dev_err(&pdev->dev, "can't ioremap SCB region\n");
                 ret = -ENOMEM;
@@ -474,9 +473,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
                 goto err_release_map;
         }
 
-        region = request_mem_region(sifmem->start,
-                                    sifmem->end - sifmem->start + 1,
-                                    pdev->name);
+        region = request_mem_region(sifmem->start, resource_size(sifmem),
+                                                                pdev->name);
         if (!region) {
                 dev_err(&pdev->dev, "I2S SIF region already claimed\n");
                 ret = -EBUSY;
@@ -490,8 +488,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
                 goto err_release_sif;
         }
 
-        region = request_mem_region(dma1->start, dma1->end - dma1->start + 1,
-                                    pdev->name);
+        region = request_mem_region(dma1->start, resource_size(dma1),
+                                                                pdev->name);
         if (!region) {
                 dev_err(&pdev->dev, "I2S DMA region already claimed\n");
                 ret = -EBUSY;
@@ -500,9 +498,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev)
 
         dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1);
         if (dma2) {
-                region = request_mem_region(dma2->start,
-                                            dma2->end - dma2->start + 1,
-                                            pdev->name);
+                region = request_mem_region(dma2->start, resource_size(dma2),
+                                                                pdev->name);
                 if (!region) {
                         dev_err(&pdev->dev,
                                 "I2S DMA region already claimed\n");
@@ -561,15 +558,15 @@ err_release_dev:
         kfree(dev);
 err_release_dma2:
         if (dma2)
-                release_mem_region(dma2->start, dma2->end - dma2->start + 1);
+                release_mem_region(dma2->start, resource_size(dma2));
 err_release_dma1:
-        release_mem_region(dma1->start, dma1->end - dma1->start + 1);
+        release_mem_region(dma1->start, resource_size(dma1));
 err_release_sif:
-        release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1);
+        release_mem_region(sifmem->start, resource_size(sifmem));
 err_release_map:
         iounmap(mmio);
 err_release_scb:
-        release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1);
+        release_mem_region(scbmem->start, resource_size(scbmem));
 err_release_none:
         return ret;
 }
@@ -590,19 +587,18 @@ static void __devexit s6000_i2s_remove(struct platform_device *pdev)
         kfree(dev);
 
         region = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-        release_mem_region(region->start, region->end - region->start + 1);
+        release_mem_region(region->start, resource_size(region));
 
         region = platform_get_resource(pdev, IORESOURCE_DMA, 1);
         if (region)
-                release_mem_region(region->start,
-                                   region->end - region->start + 1);
+                release_mem_region(region->start, resource_size(region));
 
         region = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-        release_mem_region(region->start, (region->end - region->start) + 1);
+        release_mem_region(region->start, resource_size(region));
 
         iounmap(mmio);
         region = platform_get_resource(pdev, IORESOURCE_IO, 0);
-        release_mem_region(region->start, (region->end - region->start) + 1);
+        release_mem_region(region->start, resource_size(region));
 }
 
 static struct platform_driver s6000_i2s_driver = {
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index a1d14bc3c76f..52d7e8ed9c1f 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -48,7 +48,7 @@ config SND_SH7760_AC97
 
 config SND_FSI_AK4642
         bool "FSI-AK4642 sound support"
-        depends on SND_SOC_SH4_FSI
+        depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE
         select SND_SOC_AK4642
         help
           This option enables generic sound support for the
@@ -56,7 +56,7 @@ config SND_FSI_AK4642
 
 config SND_FSI_DA7210
         bool "FSI-DA7210 sound support"
-        depends on SND_SOC_SH4_FSI
+        depends on SND_SOC_SH4_FSI && I2C_SH_MOBILE
         select SND_SOC_DA7210
         help
           This option enables generic sound support for the
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index be018542314e..dad575a22622 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -9,16 +9,7 @@
  * for more details.
  */
 
-#include <linux/module.h>
-#include <linux/moduleparam.h>
 #include <linux/platform_device.h>
-#include <linux/i2c.h>
-#include <linux/io.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-
 #include <sound/sh_fsi.h>
 #include <../sound/soc/codecs/ak4642.h>
 
@@ -38,7 +29,7 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec)
 static struct snd_soc_dai_link fsi_dai_link = {
         .name           = "AK4642",
         .stream_name    = "AK4642",
-        .cpu_dai        = &fsi_soc_dai[0], /* fsi */
+        .cpu_dai        = &fsi_soc_dai[FSI_PORT_A],
         .codec_dai      = &ak4642_dai,
         .init           = fsi_ak4642_dai_init,
         .ops            = NULL,
@@ -62,7 +53,7 @@ static int __init fsi_ak4642_init(void)
 {
         int ret = -ENOMEM;
 
-        fsi_snd_device = platform_device_alloc("soc-audio", -1);
+        fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A);
         if (!fsi_snd_device)
                 goto out;
 
diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c
index 33b4d177f466..121bbb07bb03 100644
--- a/sound/soc/sh/fsi-da7210.c
+++ b/sound/soc/sh/fsi-da7210.c
@@ -10,16 +10,7 @@
  *  option) any later version.
  */
 
-#include <linux/interrupt.h>
 #include <linux/platform_device.h>
-#include <linux/io.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/soc-dapm.h>
-
 #include <sound/sh_fsi.h>
 #include "../codecs/da7210.h"
 
@@ -33,7 +24,7 @@ static int fsi_da7210_init(struct snd_soc_codec *codec)
 static struct snd_soc_dai_link fsi_da7210_dai = {
         .name           = "DA7210",
         .stream_name    = "DA7210",
-        .cpu_dai        = &fsi_soc_dai[1], /* FSI B */
+        .cpu_dai        = &fsi_soc_dai[FSI_PORT_B],
         .codec_dai      = &da7210_dai,
         .init           = fsi_da7210_init,
 };
@@ -56,7 +47,7 @@ static int __init fsi_da7210_sound_init(void)
 {
         int ret;
 
-        fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1);
+        fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B);
         if (!fsi_da7210_snd_device)
                 return -ENOMEM;
 
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index ec4acac49ebd..58c6bec642de 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -12,21 +12,12 @@
  * published by the Free Software Foundation.
  */
 
-#include <linux/init.h>
-#include <linux/module.h>
-#include <linux/platform_device.h>
 #include <linux/delay.h>
-#include <linux/list.h>
 #include <linux/pm_runtime.h>
 #include <linux/io.h>
 #include <linux/slab.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/initval.h>
 #include <sound/soc.h>
-#include <sound/pcm_params.h>
 #include <sound/sh_fsi.h>
-#include <asm/atomic.h>
 
 #define DO_FMT          0x0000
 #define DOFF_CTL        0x0004
@@ -39,9 +30,11 @@
 #define DIDT            0x0020
 #define DODT            0x0024
 #define MUTE_ST         0x0028
-#define REG_END         MUTE_ST
-
+#define OUT_SEL         0x0030
+#define REG_END         OUT_SEL
 
+#define A_MST_CTLR      0x0180
+#define B_MST_CTLR      0x01A0
 #define CPU_INT_ST      0x01F4
 #define CPU_IEMSK       0x01F8
 #define CPU_IMSK        0x01FC
@@ -52,18 +45,18 @@
 #define CLK_RST         0x0210
 #define SOFT_RST        0x0214
 #define FIFO_SZ         0x0218
-#define MREG_START      CPU_INT_ST
+#define MREG_START      A_MST_CTLR
 #define MREG_END        FIFO_SZ
 
 /* DO_FMT */
 /* DI_FMT */
-#define CR_FMT(param) ((param) << 4)
-# define CR_MONO        0x0
-# define CR_MONO_D      0x1
-# define CR_PCM         0x2
-# define CR_I2S         0x3
-# define CR_TDM         0x4
-# define CR_TDM_D       0x5
+#define CR_MONO         (0x0 << 4)
+#define CR_MONO_D       (0x1 << 4)
+#define CR_PCM          (0x2 << 4)
+#define CR_I2S          (0x3 << 4)
+#define CR_TDM          (0x4 << 4)
+#define CR_TDM_D        (0x5 << 4)
+#define CR_SPDIF        0x00100120
 
 /* DOFF_CTL */
 /* DIFF_CTL */
@@ -75,6 +68,14 @@
 #define ERR_UNDER       0x00000001
 #define ST_ERR          (ERR_OVER | ERR_UNDER)
 
+/* CKG1 */
+#define ACKMD_MASK      0x00007000
+#define BPFMD_MASK      0x00000700
+
+/* A/B MST_CTLR */
+#define BP      (1 << 4)        /* Fix the signal of Biphase output */
+#define SE      (1 << 0)        /* Fix the master clock */
+
 /* CLK_RST */
 #define B_CLK           0x00000010
 #define A_CLK           0x00000001
@@ -119,9 +120,13 @@ struct fsi_priv {
         int period_len;
         int buffer_len;
         int periods;
+
+        u32 mst_ctrl;
 };
 
-struct fsi_regs {
+struct fsi_core {
+        int ver;
+
         u32 int_st;
         u32 iemsk;
         u32 imsk;
@@ -132,7 +137,7 @@ struct fsi_master {
         int irq;
         struct fsi_priv fsia;
         struct fsi_priv fsib;
-        struct fsi_regs *regs;
+        struct fsi_core *core;
         struct sh_fsi_platform_info *info;
         spinlock_t lock;
 };
@@ -169,24 +174,30 @@ static void __fsi_reg_mask_set(u32 reg, u32 mask, u32 data)
 
 static void fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data)
 {
-        if (reg > REG_END)
+        if (reg > REG_END) {
+                pr_err("fsi: register access err (%s)\n", __func__);
                 return;
+        }
 
         __fsi_reg_write((u32)(fsi->base + reg), data);
 }
 
 static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg)
 {
-        if (reg > REG_END)
+        if (reg > REG_END) {
+                pr_err("fsi: register access err (%s)\n", __func__);
                 return 0;
+        }
 
         return __fsi_reg_read((u32)(fsi->base + reg));
 }
 
 static void fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data)
 {
-        if (reg > REG_END)
+        if (reg > REG_END) {
+                pr_err("fsi: register access err (%s)\n", __func__);
                 return;
+        }
 
         __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data);
 }
@@ -196,8 +207,10 @@ static void fsi_master_write(struct fsi_master *master, u32 reg, u32 data)
         unsigned long flags;
 
         if ((reg < MREG_START) ||
-            (reg > MREG_END))
+            (reg > MREG_END)) {
+                pr_err("fsi: register access err (%s)\n", __func__);
                 return;
+        }
 
         spin_lock_irqsave(&master->lock, flags);
         __fsi_reg_write((u32)(master->base + reg), data);
@@ -210,8 +223,10 @@ static u32 fsi_master_read(struct fsi_master *master, u32 reg)
         unsigned long flags;
 
         if ((reg < MREG_START) ||
-            (reg > MREG_END))
+            (reg > MREG_END)) {
+                pr_err("fsi: register access err (%s)\n", __func__);
                 return 0;
+        }
 
         spin_lock_irqsave(&master->lock, flags);
         ret = __fsi_reg_read((u32)(master->base + reg));
@@ -226,8 +241,10 @@ static void fsi_master_mask_set(struct fsi_master *master,
         unsigned long flags;
 
         if ((reg < MREG_START) ||
-            (reg > MREG_END))
+            (reg > MREG_END)) {
+                pr_err("fsi: register access err (%s)\n", __func__);
                 return;
+        }
 
         spin_lock_irqsave(&master->lock, flags);
         __fsi_reg_mask_set((u32)(master->base + reg), mask, data);
@@ -349,8 +366,8 @@ static void fsi_irq_enable(struct fsi_priv *fsi, int is_play)
         u32 data = fsi_port_ab_io_bit(fsi, is_play);
         struct fsi_master *master = fsi_get_master(fsi);
 
-        fsi_master_mask_set(master, master->regs->imsk,  data, data);
-        fsi_master_mask_set(master, master->regs->iemsk, data, data);
+        fsi_master_mask_set(master, master->core->imsk,  data, data);
+        fsi_master_mask_set(master, master->core->iemsk, data, data);
 }
 
 static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
@@ -358,18 +375,18 @@ static void fsi_irq_disable(struct fsi_priv *fsi, int is_play)
         u32 data = fsi_port_ab_io_bit(fsi, is_play);
         struct fsi_master *master = fsi_get_master(fsi);
 
-        fsi_master_mask_set(master, master->regs->imsk,  data, 0);
-        fsi_master_mask_set(master, master->regs->iemsk, data, 0);
+        fsi_master_mask_set(master, master->core->imsk,  data, 0);
+        fsi_master_mask_set(master, master->core->iemsk, data, 0);
 }
 
 static u32 fsi_irq_get_status(struct fsi_master *master)
 {
-        return fsi_master_read(master, master->regs->int_st);
+        return fsi_master_read(master, master->core->int_st);
 }
 
 static void fsi_irq_clear_all_status(struct fsi_master *master)
 {
-        fsi_master_write(master, master->regs->int_st, 0x0000000);
+        fsi_master_write(master, master->core->int_st, 0);
 }
 
 static void fsi_irq_clear_status(struct fsi_priv *fsi)
@@ -381,7 +398,30 @@ static void fsi_irq_clear_status(struct fsi_priv *fsi)
         data |= fsi_port_ab_io_bit(fsi, 1);
 
         /* clear interrupt factor */
-        fsi_master_mask_set(master, master->regs->int_st, data, 0);
+        fsi_master_mask_set(master, master->core->int_st, data, 0);
+}
+
+/************************************************************************
+
+
+                SPDIF master clock function
+
+These functions are used later FSI2
+************************************************************************/
+static void fsi_spdif_clk_ctrl(struct fsi_priv *fsi, int enable)
+{
+        struct fsi_master *master = fsi_get_master(fsi);
+        u32 val = BP | SE;
+
+        if (master->core->ver < 2) {
+                pr_err("fsi: register access err (%s)\n", __func__);
+                return;
+        }
+
+        if (enable)
+                fsi_master_mask_set(master, fsi->mst_ctrl, val, val);
+        else
+                fsi_master_mask_set(master, fsi->mst_ctrl, val, 0);
 }
 
 /************************************************************************
@@ -662,8 +702,8 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
                            struct snd_soc_dai *dai)
 {
         struct fsi_priv *fsi = fsi_get_priv(substream);
-        const char *msg;
         u32 flags = fsi_get_info_flags(fsi);
+        struct fsi_master *master = fsi_get_master(fsi);
         u32 fmt;
         u32 reg;
         u32 data;
@@ -700,36 +740,40 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
         fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags);
         switch (fmt) {
         case SH_FSI_FMT_MONO:
-                msg = "MONO";
-                data = CR_FMT(CR_MONO);
+                data = CR_MONO;
                 fsi->chan = 1;
                 break;
         case SH_FSI_FMT_MONO_DELAY:
-                msg = "MONO Delay";
-                data = CR_FMT(CR_MONO_D);
+                data = CR_MONO_D;
                 fsi->chan = 1;
                 break;
         case SH_FSI_FMT_PCM:
-                msg = "PCM";
-                data = CR_FMT(CR_PCM);
+                data = CR_PCM;
                 fsi->chan = 2;
                 break;
         case SH_FSI_FMT_I2S:
-                msg = "I2S";
-                data = CR_FMT(CR_I2S);
+                data = CR_I2S;
                 fsi->chan = 2;
                 break;
         case SH_FSI_FMT_TDM:
-                msg = "TDM";
                 fsi->chan = is_play ?
                         SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
-                data = CR_FMT(CR_TDM) | (fsi->chan - 1);
+                data = CR_TDM | (fsi->chan - 1);
                 break;
         case SH_FSI_FMT_TDM_DELAY:
-                msg = "TDM Delay";
                 fsi->chan = is_play ?
                         SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
-                data = CR_FMT(CR_TDM_D) | (fsi->chan - 1);
+                data = CR_TDM_D | (fsi->chan - 1);
+                break;
+        case SH_FSI_FMT_SPDIF:
+                if (master->core->ver < 2) {
+                        dev_err(dai->dev, "This FSI can not use SPDIF\n");
+                        return -EINVAL;
+                }
+                data = CR_SPDIF;
+                fsi->chan = 2;
+                fsi_spdif_clk_ctrl(fsi, 1);
+                fsi_reg_mask_set(fsi, OUT_SEL, 0x0010, 0x0010);
                 break;
         default:
                 dev_err(dai->dev, "unknown format.\n");
@@ -737,12 +781,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
         }
         fsi_reg_write(fsi, reg, data);
 
-        /*
-         * clear clk reset if master mode
-         */
-        if (is_master)
-                fsi_clk_ctrl(fsi, 1);
-
         /* irq clear */
         fsi_irq_disable(fsi, is_play);
         fsi_irq_clear_status(fsi);
@@ -789,10 +827,93 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
         return ret;
 }
 
+static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
+                             struct snd_pcm_hw_params *params,
+                             struct snd_soc_dai *dai)
+{
+        struct fsi_priv *fsi = fsi_get_priv(substream);
+        struct fsi_master *master = fsi_get_master(fsi);
+        int (*set_rate)(int is_porta, int rate) = master->info->set_rate;
+        int fsi_ver = master->core->ver;
+        int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+        int ret;
+
+        /* if slave mode, set_rate is not needed */
+        if (!fsi_is_master_mode(fsi, is_play))
+                return 0;
+
+        /* it is error if no set_rate */
+        if (!set_rate)
+                return -EIO;
+
+        ret = set_rate(fsi_is_port_a(fsi), params_rate(params));
+        if (ret > 0) {
+                u32 data = 0;
+
+                switch (ret & SH_FSI_ACKMD_MASK) {
+                default:
+                        /* FALL THROUGH */
+                case SH_FSI_ACKMD_512:
+                        data |= (0x0 << 12);
+                        break;
+                case SH_FSI_ACKMD_256:
+                        data |= (0x1 << 12);
+                        break;
+                case SH_FSI_ACKMD_128:
+                        data |= (0x2 << 12);
+                        break;
+                case SH_FSI_ACKMD_64:
+                        data |= (0x3 << 12);
+                        break;
+                case SH_FSI_ACKMD_32:
+                        if (fsi_ver < 2)
+                                dev_err(dai->dev, "unsupported ACKMD\n");
+                        else
+                                data |= (0x4 << 12);
+                        break;
+                }
+
+                switch (ret & SH_FSI_BPFMD_MASK) {
+                default:
+                        /* FALL THROUGH */
+                case SH_FSI_BPFMD_32:
+                        data |= (0x0 << 8);
+                        break;
+                case SH_FSI_BPFMD_64:
+                        data |= (0x1 << 8);
+                        break;
+                case SH_FSI_BPFMD_128:
+                        data |= (0x2 << 8);
+                        break;
+                case SH_FSI_BPFMD_256:
+                        data |= (0x3 << 8);
+                        break;
+                case SH_FSI_BPFMD_512:
+                        data |= (0x4 << 8);
+                        break;
+                case SH_FSI_BPFMD_16:
+                        if (fsi_ver < 2)
+                                dev_err(dai->dev, "unsupported ACKMD\n");
+                        else
+                                data |= (0x7 << 8);
+                        break;
+                }
+
+                fsi_reg_mask_set(fsi, CKG1, (ACKMD_MASK | BPFMD_MASK) , data);
+                udelay(10);
+                fsi_clk_ctrl(fsi, 1);
+                ret = 0;
+        }
+
+        return ret;
+
+}
+
 static struct snd_soc_dai_ops fsi_dai_ops = {
         .startup        = fsi_dai_startup,
         .shutdown       = fsi_dai_shutdown,
         .trigger        = fsi_dai_trigger,
+        .hw_params      = fsi_dai_hw_params,
 };
 
 /************************************************************************
@@ -965,11 +1086,6 @@ static int fsi_probe(struct platform_device *pdev)
         unsigned int irq;
         int ret;
 
-        if (0 != pdev->id) {
-                dev_err(&pdev->dev, "current fsi support id 0 only now\n");
-                return -ENODEV;
-        }
-
         id_entry = pdev->id_entry;
         if (!id_entry) {
                 dev_err(&pdev->dev, "unknown fsi device\n");
@@ -998,14 +1114,21 @@ static int fsi_probe(struct platform_device *pdev)
                 goto exit_kfree;
         }
 
+        /* master setting */
         master->irq             = irq;
         master->info            = pdev->dev.platform_data;
+        master->core            = (struct fsi_core *)id_entry->driver_data;
+        spin_lock_init(&master->lock);
+
+        /* FSI A setting */
         master->fsia.base       = master->base;
         master->fsia.master     = master;
+        master->fsia.mst_ctrl   = A_MST_CTLR;
+
+        /* FSI B setting */
         master->fsib.base       = master->base + 0x40;
         master->fsib.master     = master;
-        master->regs            = (struct fsi_regs *)id_entry->driver_data;
-        spin_lock_init(&master->lock);
+        master->fsib.mst_ctrl   = B_MST_CTLR;
 
         pm_runtime_enable(&pdev->dev);
         pm_runtime_resume(&pdev->dev);
@@ -1085,21 +1208,27 @@ static struct dev_pm_ops fsi_pm_ops = {
         .runtime_resume         = fsi_runtime_nop,
 };
 
-static struct fsi_regs fsi_regs = {
+static struct fsi_core fsi1_core = {
+        .ver    = 1,
+
+        /* Interrupt */
         .int_st = INT_ST,
         .iemsk  = IEMSK,
         .imsk   = IMSK,
 };
 
-static struct fsi_regs fsi2_regs = {
+static struct fsi_core fsi2_core = {
+        .ver    = 2,
+
+        /* Interrupt */
         .int_st = CPU_INT_ST,
         .iemsk  = CPU_IEMSK,
         .imsk   = CPU_IMSK,
 };
 
 static struct platform_device_id fsi_id_table[] = {
-        { "sh_fsi",     (kernel_ulong_t)&fsi_regs },
-        { "sh_fsi2",    (kernel_ulong_t)&fsi2_regs },
+        { "sh_fsi",     (kernel_ulong_t)&fsi1_core },
+        { "sh_fsi2",    (kernel_ulong_t)&fsi2_core },
 };
 
 static struct platform_driver fsi_driver = {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 998569d60330..5299932db0b6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -84,7 +84,7 @@ static int run_delayed_work(struct delayed_work *dwork)
 /* codec register dump */
 static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
 {
-        int i, step = 1, count = 0;
+        int ret, i, step = 1, count = 0;
 
         if (!codec->reg_cache_size)
                 return 0;
@@ -101,12 +101,24 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
                 if (count >= PAGE_SIZE - 1)
                         break;
 
-                if (codec->display_register)
+                if (codec->display_register) {
                         count += codec->display_register(codec, buf + count,
                                                          PAGE_SIZE - count, i);
-                else
-                        count += snprintf(buf + count, PAGE_SIZE - count,
-                                          "%4x", codec->read(codec, i));
+                } else {
+                        /* If the read fails it's almost certainly due to
+                         * the register being volatile and the device being
+                         * powered off.
+                         */
+                        ret = codec->read(codec, i);
+                        if (ret >= 0)
+                                count += snprintf(buf + count,
+                                                  PAGE_SIZE - count,
+                                                  "%4x", ret);
+                        else
+                                count += snprintf(buf + count,
+                                                  PAGE_SIZE - count,
+                                                  "<no data: %d>", ret);
+                }
 
                 if (count >= PAGE_SIZE - 1)
                         break;
@@ -2353,6 +2365,99 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec,
 EXPORT_SYMBOL_GPL(snd_soc_limit_volume);
 
 /**
+ * snd_soc_info_volsw_2r_sx - double with tlv and variable data size
+ *  mixer info callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol,
+                        struct snd_ctl_elem_info *uinfo)
+{
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
+        int max = mc->max;
+        int min = mc->min;
+
+        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_2r_sx);
+
+/**
+ * snd_soc_get_volsw_2r_sx - double with tlv and variable data size
+ *  mixer get callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol,
+                        struct snd_ctl_elem_value *ucontrol)
+{
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        unsigned int mask = (1<<mc->shift)-1;
+        int min = mc->min;
+        int val = snd_soc_read(codec, mc->reg) & mask;
+        int valr = snd_soc_read(codec, mc->rreg) & mask;
+
+        ucontrol->value.integer.value[0] = ((val & 0xff)-min) & mask;
+        ucontrol->value.integer.value[1] = ((valr & 0xff)-min) & mask;
+        return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx);
+
+/**
+ * snd_soc_put_volsw_2r_sx - double with tlv and variable data size
+ *  mixer put callback
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
+                        struct snd_ctl_elem_value *ucontrol)
+{
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        unsigned int mask = (1<<mc->shift)-1;
+        int min = mc->min;
+        int ret;
+        unsigned int val, valr, oval, ovalr;
+
+        val = ((ucontrol->value.integer.value[0]+min) & 0xff);
+        val &= mask;
+        valr = ((ucontrol->value.integer.value[1]+min) & 0xff);
+        valr &= mask;
+
+        oval = snd_soc_read(codec, mc->reg) & mask;
+        ovalr = snd_soc_read(codec, mc->rreg) & mask;
+
+        ret = 0;
+        if (oval != val) {
+                ret = snd_soc_write(codec, mc->reg, val);
+                if (ret < 0)
+                        return ret;
+        }
+        if (ovalr != valr) {
+                ret = snd_soc_write(codec, mc->rreg, valr);
+                if (ret < 0)
+                        return ret;
+        }
+
+        return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx);
+
+/**
  * snd_soc_dai_set_sysclk - configure DAI system or master clock.
  * @dai: DAI
  * @clk_id: DAI specific clock ID