104 files changed, 12625 insertions, 1734 deletions

diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 3d2bb6fc6dcc..d3e786a9a0a7 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -32,7 +32,9 @@ source "sound/soc/fsl/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
+source "sound/soc/s6000/Kconfig"
 source "sound/soc/sh/Kconfig"
+source "sound/soc/txx9/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 0237879fd412..6f1e28de23cf 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -10,4 +10,6 @@ obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)   += omap/
 obj-$(CONFIG_SND_SOC)   += pxa/
 obj-$(CONFIG_SND_SOC)   += s3c24xx/
+obj-$(CONFIG_SND_SOC)   += s6000/
 obj-$(CONFIG_SND_SOC)   += sh/
+obj-$(CONFIG_SND_SOC)   += txx9/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index a608d7009dbd..e720d5e6f04c 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE
           and FRAME signals on the PlayPaq.  Unless you want to play
           with the AT32 as the SSC master, you probably want to say N here,
           as this will give you better sound quality.
+
+config SND_AT91_SOC_AFEB9260
+        tristate "SoC Audio support for AFEB9260 board"
+        depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
+        select SND_ATMEL_SOC_SSC
+        select SND_SOC_TLV320AIC23
+        help
+          Say Y here to support sound on AFEB9260 board.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index f54a7cc68e66..e7ea56bd5f82 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
+obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 70657534e6b1..9eb610c2ba91 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -117,7 +117,7 @@ static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
          * Find actual rate, compare to requested rate
          */
         actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
-        pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n",
+        pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
                  rate, actual_rate);
 
 
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
new file mode 100644
index 000000000000..23349de27313
--- /dev/null
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -0,0 +1,203 @@
+/*
+ * afeb9260.c  --  SoC audio for AFEB9260
+ *
+ * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+#define CODEC_CLOCK     12000000
+
+static int afeb9260_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;
+
+        /* Set codec DAI configuration */
+        err = snd_soc_dai_set_fmt(codec_dai,
+                                  SND_SOC_DAIFMT_I2S|
+                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_CBM_CFM);
+        if (err < 0) {
+                printk(KERN_ERR "can't set codec DAI configuration\n");
+                return err;
+        }
+
+        /* Set cpu DAI configuration */
+        err = snd_soc_dai_set_fmt(cpu_dai,
+                                  SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_CBM_CFM);
+        if (err < 0) {
+                printk(KERN_ERR "can't set cpu DAI configuration\n");
+                return err;
+        }
+
+        /* Set the codec system clock for DAC and ADC */
+        err =
+            snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+        if (err < 0) {
+                printk(KERN_ERR "can't set codec system clock\n");
+                return err;
+        }
+
+        return err;
+}
+
+static struct snd_soc_ops afeb9260_ops = {
+        .hw_params = afeb9260_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 afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+        /* Add afeb9260 specific widgets */
+        snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+                                  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+        /* Set up afeb9260 specific audio path audio_map */
+        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+        snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+        snd_soc_dapm_enable_pin(codec, "Line In");
+        snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+        snd_soc_dapm_sync(codec);
+
+        return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link afeb9260_dai = {
+        .name = "TLV320AIC23",
+        .stream_name = "AIC23",
+        .cpu_dai = &atmel_ssc_dai[0],
+        .codec_dai = &tlv320aic23_dai,
+        .init = afeb9260_tlv320aic23_init,
+        .ops = &afeb9260_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_machine_afeb9260 = {
+        .name = "AFEB9260",
+        .platform = &atmel_soc_platform,
+        .dai_link = &afeb9260_dai,
+        .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device afeb9260_snd_devdata = {
+        .card = &snd_soc_machine_afeb9260,
+        .codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *afeb9260_snd_device;
+
+static int __init afeb9260_soc_init(void)
+{
+        int err;
+        struct device *dev;
+        struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
+        struct ssc_device *ssc = NULL;
+
+        if (!(machine_is_afeb9260()))
+                return -ENODEV;
+
+        ssc = ssc_request(0);
+        if (IS_ERR(ssc)) {
+                printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
+                err = PTR_ERR(ssc);
+                ssc = NULL;
+                goto err_ssc;
+        }
+        ssc_p->ssc = ssc;
+
+        afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!afeb9260_snd_device) {
+                printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+                return -ENOMEM;
+        }
+
+        platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata);
+        afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
+        err = platform_device_add(afeb9260_snd_device);
+        if (err)
+                goto err1;
+
+        dev = &afeb9260_snd_device->dev;
+
+        return 0;
+err1:
+        platform_device_del(afeb9260_snd_device);
+        platform_device_put(afeb9260_snd_device);
+err_ssc:
+        return err;
+
+}
+
+static void __exit afeb9260_soc_exit(void)
+{
+        platform_device_unregister(afeb9260_snd_device);
+}
+
+module_init(afeb9260_soc_init);
+module_exit(afeb9260_soc_exit);
+
+MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
+MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a259148..594c6c5b7838 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
 /* PCM hardware DMA capabilities - platform specific */
 static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
         .info             = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-                            SNDRV_PCM_INFO_INTERLEAVED,
+                            SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
         .formats          = AU1XPSC_PCM_FMTS,
         .period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
         .period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 8a935f2d1767..b1ed423fabd5 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -31,6 +31,15 @@
 #include "bf5xx-sport.h"
 #include "bf5xx-ac97.h"
 
+/* Anomaly notes:
+ *  05000250 -  AD1980 is running in TDM mode and RFS/TFS are generated by SPORT
+ *              contrtoller. But, RFSDIV and TFSDIV are always set to 16*16-1,
+ *              while the max AC97 data size is 13*16. The DIV is always larger
+ *              than data size. AD73311 and ad2602 are not running in TDM mode.
+ *              AD1836 and AD73322 depend on external RFS/TFS only. So, this
+ *              anomaly does not affect blackfin sound drivers.
+*/
+
 static int *cmd_count;
 static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
 
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
index b7953c8cf838..469ce7fab20c 100644
--- a/sound/soc/blackfin/bf5xx-sport.c
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -190,7 +190,7 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport)
         desc = get_dma_next_desc_ptr(sport->dma_rx_chan);
         /* Copy the descriptor which will be damaged to backup */
         temp_desc = *desc;
-        desc->x_count = 0xa;
+        desc->x_count = sport->dummy_count / 2;
         desc->y_count = 0;
         desc->next_desc_addr = sport->dummy_rx_desc;
         local_irq_restore(flags);
@@ -309,7 +309,7 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport)
         desc = get_dma_next_desc_ptr(sport->dma_tx_chan);
         /* Store the descriptor which will be damaged */
         temp_desc = *desc;
-        desc->x_count = 0xa;
+        desc->x_count = sport->dummy_count / 2;
         desc->y_count = 0;
         desc->next_desc_addr = sport->dummy_tx_desc;
         local_irq_restore(flags);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index b6c7f7a01cb0..bbc97fd76648 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -18,7 +18,9 @@ config SND_SOC_ALL_CODECS
         select SND_SOC_AK4535 if I2C
         select SND_SOC_CS4270 if I2C
         select SND_SOC_PCM3008
+        select SND_SOC_SPDIF
         select SND_SOC_SSM2602 if I2C
+        select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
         select SND_SOC_TLV320AIC23 if I2C
         select SND_SOC_TLV320AIC26 if SPI_MASTER
         select SND_SOC_TLV320AIC3X if I2C
@@ -35,8 +37,12 @@ config SND_SOC_ALL_CODECS
         select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
         select SND_SOC_WM8900 if I2C
         select SND_SOC_WM8903 if I2C
+        select SND_SOC_WM8940 if I2C
+        select SND_SOC_WM8960 if I2C
         select SND_SOC_WM8971 if I2C
+        select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
         select SND_SOC_WM8990 if I2C
+        select SND_SOC_WM9081 if I2C
         select SND_SOC_WM9705 if SND_SOC_AC97_BUS
         select SND_SOC_WM9712 if SND_SOC_AC97_BUS
         select SND_SOC_WM9713 if SND_SOC_AC97_BUS
@@ -86,9 +92,15 @@ config SND_SOC_L3
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_SPDIF
+        tristate
+
 config SND_SOC_SSM2602
         tristate
 
+config SND_SOC_STAC9766
+        tristate
+
 config SND_SOC_TLV320AIC23
         tristate
 
@@ -138,12 +150,24 @@ config SND_SOC_WM8900
 config SND_SOC_WM8903
         tristate
 
+config SND_SOC_WM8940
+        tristate
+
+config SND_SOC_WM8960
+        tristate
+
 config SND_SOC_WM8971
         tristate
 
+config SND_SOC_WM8988
+        tristate
+
 config SND_SOC_WM8990
         tristate
 
+config SND_SOC_WM9081
+        tristate
+
 config SND_SOC_WM9705
         tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 030d2454725f..8b7530546f4d 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -6,7 +6,9 @@ snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-l3-objs := l3.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -23,8 +25,12 @@ snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
+snd-soc-wm8940-objs := wm8940.o
+snd-soc-wm8960-objs := wm8960.o
 snd-soc-wm8971-objs := wm8971.o
+snd-soc-wm8988-objs := wm8988.o
 snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9081-objs := wm9081.o
 snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
@@ -37,7 +43,9 @@ obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)    += snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_L3)        += snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_PCM3008)   += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_SPDIF)     += snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)   += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STAC9766)  += snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)       += snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)       += snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)       += snd-soc-tlv320aic3x.o
@@ -55,8 +63,11 @@ obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8900)    += snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)    += snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)    += snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8940)    += snd-soc-wm8940.o
+obj-$(CONFIG_SND_SOC_WM8960)    += snd-soc-wm8960.o
+obj-$(CONFIG_SND_SOC_WM8988)    += snd-soc-wm8988.o
 obj-$(CONFIG_SND_SOC_WM8990)    += snd-soc-wm8990.o
-obj-$(CONFIG_SND_SOC_WM8991)    += snd-soc-wm8991.o
+obj-$(CONFIG_SND_SOC_WM9081)    += snd-soc-wm9081.o
 obj-$(CONFIG_SND_SOC_WM9705)    += snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)    += snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)    += snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index b0d4af145b87..932299bb5d1e 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -53,13 +53,13 @@ struct snd_soc_dai ac97_dai = {
                 .channels_min = 1,
                 .channels_max = 2,
                 .rates = STD_AC97_RATES,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+                .formats = SND_SOC_STD_AC97_FMTS,},
         .capture = {
                 .stream_name = "AC97 Capture",
                 .channels_min = 1,
                 .channels_max = 2,
                 .rates = STD_AC97_RATES,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+                .formats = SND_SOC_STD_AC97_FMTS,},
         .ops = &ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ac97_dai);
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index ddb3b08ac23c..d7440a982d22 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -137,13 +137,13 @@ struct snd_soc_dai ad1980_dai = {
                 .channels_min = 2,
                 .channels_max = 6,
                 .rates = SNDRV_PCM_RATE_48000,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+                .formats = SND_SOC_STD_AC97_FMTS, },
         .capture = {
                 .stream_name = "Capture",
                 .channels_min = 2,
                 .channels_max = 2,
                 .rates = SNDRV_PCM_RATE_48000,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+                .formats = SND_SOC_STD_AC97_FMTS, },
 };
 EXPORT_SYMBOL_GPL(ad1980_dai);
 
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 7fa09a387622..a32b8226c8a4 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -18,7 +18,7 @@
  * - The machine driver's 'startup' function must call
  *   cs4270_set_dai_sysclk() with the value of MCLK.
  * - Only I2S and left-justified modes are supported
- * - Power management is not supported
+ * - Power management is supported
  */
 
 #include <linux/module.h>
@@ -27,6 +27,7 @@
 #include <sound/soc.h>
 #include <sound/initval.h>
 #include <linux/i2c.h>
+#include <linux/delay.h>
 
 #include "cs4270.h"
 
@@ -56,6 +57,7 @@
 #define CS4270_FIRSTREG 0x01
 #define CS4270_LASTREG  0x08
 #define CS4270_NUMREGS  (CS4270_LASTREG - CS4270_FIRSTREG + 1)
+#define CS4270_I2C_INCR 0x80
 
 /* Bit masks for the CS4270 registers */
 #define CS4270_CHIPID_ID        0xF0
@@ -64,6 +66,8 @@
 #define CS4270_PWRCTL_PDN_ADC   0x20
 #define CS4270_PWRCTL_PDN_DAC   0x02
 #define CS4270_PWRCTL_PDN       0x01
+#define CS4270_PWRCTL_PDN_ALL   \
+        (CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN)
 #define CS4270_MODE_SPEED_MASK  0x30
 #define CS4270_MODE_1X          0x00
 #define CS4270_MODE_2X          0x10
@@ -109,6 +113,7 @@ struct cs4270_private {
         unsigned int mclk; /* Input frequency of the MCLK pin */
         unsigned int mode; /* The mode (I2S or left-justified) */
         unsigned int slave_mode;
+        unsigned int manual_mute;
 };
 
 /**
@@ -295,7 +300,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
         s32 length;
 
         length = i2c_smbus_read_i2c_block_data(i2c_client,
-                CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
+                CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache);
 
         if (length != CS4270_NUMREGS) {
                 dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
@@ -453,7 +458,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 }
 
 /**
- * cs4270_mute - enable/disable the CS4270 external mute
+ * cs4270_dai_mute - enable/disable the CS4270 external mute
  * @dai: the SOC DAI
  * @mute: 0 = disable mute, 1 = enable mute
  *
@@ -462,21 +467,52 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
  * board does not have the MUTEA or MUTEB pins connected to such circuitry,
  * then this function will do nothing.
  */
-static int cs4270_mute(struct snd_soc_dai *dai, int mute)
+static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute)
 {
         struct snd_soc_codec *codec = dai->codec;
+        struct cs4270_private *cs4270 = codec->private_data;
         int reg6;
 
         reg6 = snd_soc_read(codec, CS4270_MUTE);
 
         if (mute)
                 reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
-        else
+        else {
                 reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+                reg6 |= cs4270->manual_mute;
+        }
 
         return snd_soc_write(codec, CS4270_MUTE, reg6);
 }
 
+/**
+ * cs4270_soc_put_mute - put callback for the 'Master Playback switch'
+ *                       alsa control.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * This function basically passes the arguments on to the generic
+ * snd_soc_put_volsw() function and saves the mute information in
+ * our private data structure. This is because we want to prevent
+ * cs4270_dai_mute() neglecting the user's decision to manually
+ * mute the codec's output.
+ *
+ * Returns 0 for success.
+ */
+static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct cs4270_private *cs4270 = codec->private_data;
+        int left = !ucontrol->value.integer.value[0];
+        int right = !ucontrol->value.integer.value[1];
+
+        cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) |
+                              (right ? CS4270_MUTE_DAC_B : 0);
+
+        return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
         SOC_DOUBLE_R("Master Playback Volume",
@@ -486,7 +522,9 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
         SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
         SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
         SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
-        SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
+        SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1),
+        SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1,
+                snd_soc_get_volsw, cs4270_soc_put_mute),
 };
 
 /*
@@ -506,7 +544,7 @@ static struct snd_soc_dai_ops cs4270_dai_ops = {
         .hw_params      = cs4270_hw_params,
         .set_sysclk     = cs4270_set_dai_sysclk,
         .set_fmt        = cs4270_set_dai_fmt,
-        .digital_mute   = cs4270_mute,
+        .digital_mute   = cs4270_dai_mute,
 };
 
 struct snd_soc_dai cs4270_dai = {
@@ -753,6 +791,57 @@ static struct i2c_device_id cs4270_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, cs4270_id);
 
+#ifdef CONFIG_PM
+
+/* This suspend/resume implementation can handle both - a simple standby
+ * where the codec remains powered, and a full suspend, where the voltage
+ * domain the codec is connected to is teared down and/or any other hardware
+ * reset condition is asserted.
+ *
+ * The codec's own power saving features are enabled in the suspend callback,
+ * and all registers are written back to the hardware when resuming.
+ */
+
+static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg)
+{
+        struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+        struct snd_soc_codec *codec = &cs4270->codec;
+        int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
+
+        return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+
+static int cs4270_i2c_resume(struct i2c_client *client)
+{
+        struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+        struct snd_soc_codec *codec = &cs4270->codec;
+        int reg;
+
+        /* In case the device was put to hard reset during sleep, we need to
+         * wait 500ns here before any I2C communication. */
+        ndelay(500);
+
+        /* first restore the entire register cache ... */
+        for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
+                u8 val = snd_soc_read(codec, reg);
+
+                if (i2c_smbus_write_byte_data(client, reg, val)) {
+                        dev_err(codec->dev, "i2c write failed\n");
+                        return -EIO;
+                }
+        }
+
+        /* ... then disable the power-down bits */
+        reg = snd_soc_read(codec, CS4270_PWRCTL);
+        reg &= ~CS4270_PWRCTL_PDN_ALL;
+
+        return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+#else
+#define cs4270_i2c_suspend      NULL
+#define cs4270_i2c_resume       NULL
+#endif /* CONFIG_PM */
+
 /*
  * cs4270_i2c_driver - I2C device identification
  *
@@ -767,6 +856,8 @@ static struct i2c_driver cs4270_i2c_driver = {
         .id_table = cs4270_id,
         .probe = cs4270_i2c_probe,
         .remove = cs4270_i2c_remove,
+        .suspend = cs4270_i2c_suspend,
+        .resume = cs4270_i2c_resume,
 };
 
 /*
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
new file mode 100644
index 000000000000..218b33adad90
--- /dev/null
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -0,0 +1,71 @@
+/*
+ * ALSA SoC SPDIF DIT driver
+ *
+ *  This driver is used by controllers which can operate in DIT (SPDI/F) where
+ *  no codec is needed.  This file provides stub codec that can be used
+ *  in these configurations. TI DaVinci Audio controller uses this driver.
+ *
+ * Author:      Steve Chen,  <schen@mvista.com>
+ * Copyright:   (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright:   (C) 2009  Texas Instruments, India
+ *
+ * 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 <sound/soc.h>
+#include <sound/pcm.h>
+
+#include "spdif_transciever.h"
+
+#define STUB_RATES      SNDRV_PCM_RATE_8000_96000
+#define STUB_FORMATS    SNDRV_PCM_FMTBIT_S16_LE
+
+struct snd_soc_dai dit_stub_dai = {
+        .name           = "DIT",
+        .playback       = {
+                .stream_name    = "Playback",
+                .channels_min   = 1,
+                .channels_max   = 384,
+                .rates          = STUB_RATES,
+                .formats        = STUB_FORMATS,
+        },
+};
+
+static int spdif_dit_probe(struct platform_device *pdev)
+{
+        dit_stub_dai.dev = &pdev->dev;
+        return snd_soc_register_dai(&dit_stub_dai);
+}
+
+static int spdif_dit_remove(struct platform_device *pdev)
+{
+        snd_soc_unregister_dai(&dit_stub_dai);
+        return 0;
+}
+
+static struct platform_driver spdif_dit_driver = {
+        .probe          = spdif_dit_probe,
+        .remove         = spdif_dit_remove,
+        .driver         = {
+                .name   = "spdif-dit",
+                .owner  = THIS_MODULE,
+        },
+};
+
+static int __init dit_modinit(void)
+{
+        return platform_driver_register(&spdif_dit_driver);
+}
+
+static void __exit dit_exit(void)
+{
+        platform_driver_unregister(&spdif_dit_driver);
+}
+
+module_init(dit_modinit);
+module_exit(dit_exit);
+
diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h
new file mode 100644
index 000000000000..296f2eb6c4ef
--- /dev/null
+++ b/sound/soc/codecs/spdif_transciever.h
@@ -0,0 +1,17 @@
+/*
+ * ALSA SoC DIT/DIR driver header
+ *
+ * Author:      Steve Chen,  <schen@mvista.com>
+ * Copyright:   (C) 2008 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef CODEC_STUBS_H
+#define CODEC_STUBS_H
+
+extern struct snd_soc_dai dit_stub_dai;
+
+#endif /* CODEC_STUBS_H */
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 87f606c76822..c550750c79c0 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -336,15 +336,17 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
                         master_runtime->sample_bits,
                         master_runtime->rate);
 
-                snd_pcm_hw_constraint_minmax(substream->runtime,
-                                             SNDRV_PCM_HW_PARAM_RATE,
-                                             master_runtime->rate,
-                                             master_runtime->rate);
-
-                snd_pcm_hw_constraint_minmax(substream->runtime,
-                                             SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-                                             master_runtime->sample_bits,
-                                             master_runtime->sample_bits);
+                if (master_runtime->rate != 0)
+                        snd_pcm_hw_constraint_minmax(substream->runtime,
+                                                     SNDRV_PCM_HW_PARAM_RATE,
+                                                     master_runtime->rate,
+                                                     master_runtime->rate);
+
+                if (master_runtime->sample_bits != 0)
+                        snd_pcm_hw_constraint_minmax(substream->runtime,
+                                                     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+                                                     master_runtime->sample_bits,
+                                                     master_runtime->sample_bits);
 
                 ssm2602->slave_substream = substream;
         } else
@@ -372,6 +374,7 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
         struct snd_soc_device *socdev = rtd->socdev;
         struct snd_soc_codec *codec = socdev->card->codec;
         struct ssm2602_priv *ssm2602 = codec->private_data;
+
         /* deactivate */
         if (!codec->active)
                 ssm2602_write(codec, SSM2602_ACTIVE, 0);
@@ -497,11 +500,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
         return 0;
 }
 
-#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-                SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
-                SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
-                SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
-                SNDRV_PCM_RATE_96000)
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\
+                SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+                SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
                 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
new file mode 100644
index 000000000000..8ad4b7b3e3ba
--- /dev/null
+++ b/sound/soc/codecs/stac9766.c
@@ -0,0 +1,463 @@
+/*
+ * stac9766.c  --  ALSA SoC STAC9766 codec support
+ *
+ * Copyright 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@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;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ *  Features:-
+ *
+ *   o Support for AC97 Codec, S/PDIF
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/soc-of-simple.h>
+
+#include "stac9766.h"
+
+#define STAC9766_VERSION "0.10"
+
+/*
+ * STAC9766 register cache
+ */
+static const u16 stac9766_reg[] = {
+        0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
+        0x0000, 0x0000, 0x8008, 0x8008, /* e */
+        0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
+        0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
+        0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+        0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
+        0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+        0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
+        0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
+        0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
+        0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
+        0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
+        0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
+        0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+        0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+        0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
+};
+
+static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
+                        "Line", "Stereo Mix", "Mono Mix", "Phone"};
+static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
+static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
+static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
+static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
+static const char *stac9766_record_all_mux[] = {"All analog",
+        "Analog plus DAC"};
+static const char *stac9766_boost1[] = {"0dB", "10dB"};
+static const char *stac9766_boost2[] = {"0dB", "20dB"};
+static const char *stac9766_stereo_mic[] = {"Off", "On"};
+
+static const struct soc_enum stac9766_record_enum =
+        SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
+static const struct soc_enum stac9766_mono_enum =
+        SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
+static const struct soc_enum stac9766_mic_enum =
+        SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
+static const struct soc_enum stac9766_SPDIF_enum =
+        SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
+static const struct soc_enum stac9766_popbypass_enum =
+        SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
+static const struct soc_enum stac9766_record_all_enum =
+        SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
+                        stac9766_record_all_mux);
+static const struct soc_enum stac9766_boost1_enum =
+        SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
+static const struct soc_enum stac9766_boost2_enum =
+        SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
+static const struct soc_enum stac9766_stereo_mic_enum =
+        SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
+
+static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
+static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
+static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
+static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
+
+static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
+        SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
+        SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
+        SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
+                       master_tlv),
+        SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
+        SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
+                       master_tlv),
+        SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+        SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
+        SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
+
+
+        SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
+        SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
+        SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
+        SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
+        SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
+
+        SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
+        SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
+        SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
+        SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+        SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
+
+        SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
+        SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
+        SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
+        SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
+        SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
+        SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
+        SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
+        SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
+
+        SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
+        SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
+        SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
+        SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
+        SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
+
+        SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
+        SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
+        SOC_ENUM("Record All Mux", stac9766_record_all_enum),
+        SOC_ENUM("Record Mux", stac9766_record_enum),
+        SOC_ENUM("Mono Mux", stac9766_mono_enum),
+        SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
+};
+
+static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+                               unsigned int val)
+{
+        u16 *cache = codec->reg_cache;
+
+        if (reg > AC97_STAC_PAGE0) {
+                stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+                soc_ac97_ops.write(codec->ac97, reg, val);
+                stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+                return 0;
+        }
+        if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+                return -EIO;
+
+        soc_ac97_ops.write(codec->ac97, reg, val);
+        cache[reg / 2] = val;
+        return 0;
+}
+
+static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
+                                       unsigned int reg)
+{
+        u16 val = 0, *cache = codec->reg_cache;
+
+        if (reg > AC97_STAC_PAGE0) {
+                stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+                val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
+                stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+                return val;
+        }
+        if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+                return -EIO;
+
+        if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+                reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
+                reg == AC97_VENDOR_ID2) {
+
+                val = soc_ac97_ops.read(codec->ac97, reg);
+                return val;
+        }
+        return cache[reg / 2];
+}
+
+static int ac97_analog_prepare(struct snd_pcm_substream *substream,
+                               struct snd_soc_dai *dai)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        unsigned short reg, vra;
+
+        vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+
+        vra |= 0x1; /* enable variable rate audio */
+
+        stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                reg = AC97_PCM_FRONT_DAC_RATE;
+        else
+                reg = AC97_PCM_LR_ADC_RATE;
+
+        return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_prepare(struct snd_pcm_substream *substream,
+                                struct snd_soc_dai *dai)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        unsigned short reg, vra;
+
+        stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
+
+        vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+        vra |= 0x5; /* Enable VRA and SPDIF out */
+
+        stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+        reg = AC97_PCM_FRONT_DAC_RATE;
+
+        return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_trigger(struct snd_pcm_substream *substream,
+                                int cmd, struct snd_soc_dai *dai)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        unsigned short vra;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_STOP:
+                vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+                vra &= !0x04;
+                stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+                break;
+        }
+        return 0;
+}
+
+static int stac9766_set_bias_level(struct snd_soc_codec *codec,
+                                   enum snd_soc_bias_level level)
+{
+        switch (level) {
+        case SND_SOC_BIAS_ON: /* full On */
+        case SND_SOC_BIAS_PREPARE: /* partial On */
+        case SND_SOC_BIAS_STANDBY: /* Off, with power */
+                stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
+                break;
+        case SND_SOC_BIAS_OFF: /* Off, without power */
+                /* disable everything including AC link */
+                stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
+                break;
+        }
+        codec->bias_level = level;
+        return 0;
+}
+
+static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
+{
+        if (try_warm && soc_ac97_ops.warm_reset) {
+                soc_ac97_ops.warm_reset(codec->ac97);
+                if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
+                        return 1;
+        }
+
+        soc_ac97_ops.reset(codec->ac97);
+        if (soc_ac97_ops.warm_reset)
+                soc_ac97_ops.warm_reset(codec->ac97);
+        if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
+                return -EIO;
+        return 0;
+}
+
+static int stac9766_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;
+
+        stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
+        return 0;
+}
+
+static int stac9766_codec_resume(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+        u16 id, reset;
+
+        reset = 0;
+        /* give the codec an AC97 warm reset to start the link */
+reset:
+        if (reset > 5) {
+                printk(KERN_ERR "stac9766 failed to resume");
+                return -EIO;
+        }
+        codec->ac97->bus->ops->warm_reset(codec->ac97);
+        id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
+        if (id != 0x4c13) {
+                stac9766_reset(codec, 0);
+                reset++;
+                goto reset;
+        }
+        stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+        if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+                stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+        return 0;
+}
+
+static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
+        .prepare = ac97_analog_prepare,
+};
+
+static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
+        .prepare = ac97_digital_prepare,
+        .trigger = ac97_digital_trigger,
+};
+
+struct snd_soc_dai stac9766_dai[] = {
+{
+        .name = "stac9766 analog",
+        .id = 0,
+        .ac97_control = 1,
+
+        /* stream cababilities */
+        .playback = {
+                .stream_name = "stac9766 analog",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000_48000,
+                .formats = SND_SOC_STD_AC97_FMTS,
+        },
+        .capture = {
+                .stream_name = "stac9766 analog",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000_48000,
+                .formats = SND_SOC_STD_AC97_FMTS,
+        },
+        /* alsa ops */
+        .ops = &stac9766_dai_ops_analog,
+},
+{
+        .name = "stac9766 IEC958",
+        .id = 1,
+        .ac97_control = 1,
+
+        /* stream cababilities */
+        .playback = {
+                .stream_name = "stac9766 IEC958",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_32000 | \
+                        SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+                .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
+        },
+        /* alsa ops */
+        .ops = &stac9766_dai_ops_digital,
+}
+};
+EXPORT_SYMBOL_GPL(stac9766_dai);
+
+static int stac9766_codec_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec;
+        int ret = 0;
+
+        printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
+
+        socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+        if (socdev->card->codec == NULL)
+                return -ENOMEM;
+        codec = socdev->card->codec;
+        mutex_init(&codec->mutex);
+
+        codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg),
+                                   GFP_KERNEL);
+        if (codec->reg_cache == NULL) {
+                ret = -ENOMEM;
+                goto cache_err;
+        }
+        codec->reg_cache_size = sizeof(stac9766_reg);
+        codec->reg_cache_step = 2;
+
+        codec->name = "STAC9766";
+        codec->owner = THIS_MODULE;
+        codec->dai = stac9766_dai;
+        codec->num_dai = ARRAY_SIZE(stac9766_dai);
+        codec->write = stac9766_ac97_write;
+        codec->read = stac9766_ac97_read;
+        codec->set_bias_level = stac9766_set_bias_level;
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+        if (ret < 0)
+                goto codec_err;
+
+        /* register pcms */
+        ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+        if (ret < 0)
+                goto pcm_err;
+
+        /* do a cold reset for the controller and then try
+         * a warm reset followed by an optional cold reset for codec */
+        stac9766_reset(codec, 0);
+        ret = stac9766_reset(codec, 1);
+        if (ret < 0) {
+                printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
+                goto reset_err;
+        }
+
+        stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+        snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
+                             ARRAY_SIZE(stac9766_snd_ac97_controls));
+
+        ret = snd_soc_init_card(socdev);
+        if (ret < 0)
+                goto reset_err;
+        return 0;
+
+reset_err:
+        snd_soc_free_pcms(socdev);
+pcm_err:
+        snd_soc_free_ac97_codec(codec);
+codec_err:
+        kfree(codec->private_data);
+cache_err:
+        kfree(socdev->card->codec);
+        socdev->card->codec = NULL;
+        return ret;
+}
+
+static int stac9766_codec_remove(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+
+        if (codec == NULL)
+                return 0;
+
+        snd_soc_free_pcms(socdev);
+        snd_soc_free_ac97_codec(codec);
+        kfree(codec->reg_cache);
+        kfree(codec);
+        return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_stac9766 = {
+        .probe = stac9766_codec_probe,
+        .remove = stac9766_codec_remove,
+        .suspend = stac9766_codec_suspend,
+        .resume = stac9766_codec_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);
+
+MODULE_DESCRIPTION("ASoC stac9766 driver");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h
new file mode 100644
index 000000000000..65642eb8393e
--- /dev/null
+++ b/sound/soc/codecs/stac9766.h
@@ -0,0 +1,21 @@
+/*
+ * stac9766.h  --  STAC9766 Soc Audio driver
+ */
+
+#ifndef _STAC9766_H
+#define _STAC9766_H
+
+#define AC97_STAC_PAGE0 0x1000
+#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
+#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
+#define AC97_STAC_STEREO_MIC 0x78
+
+/* STAC9766 DAI ID's */
+#define STAC9766_DAI_AC97_ANALOG                0
+#define STAC9766_DAI_AC97_DIGITAL               1
+
+extern struct snd_soc_dai stac9766_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_stac9766;
+
+
+#endif
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index c3f4afb5d017..0b8dcb5cd729 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -86,7 +86,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
          */
 
         if ((reg < 0 || reg > 9) && (reg != 15)) {
-                printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+                printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
                 return -1;
         }
 
@@ -98,7 +98,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
         if (codec->hw_write(codec->control_data, data, 2) == 2)
                 return 0;
 
-        printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
+        printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__,
                value, reg);
 
         return -EIO;
@@ -273,14 +273,14 @@ static const unsigned short sr_valid_mask[] = {
  * Every divisor is a factor of 11*12
  */
 #define SR_MULT (11*12)
-#define A(x) (x) ? (SR_MULT/x) : 0
+#define A(x) (SR_MULT/x)
 static const unsigned char sr_adc_mult_table[] = {
-        A(2), A(2), A(12), A(12),  A(0), A(0), A(3), A(1),
-        A(2), A(2), A(11), A(11),  A(0), A(0), A(0), A(1)
+        A(2), A(2), A(12), A(12),  0, 0, A(3), A(1),
+        A(2), A(2), A(11), A(11),  0, 0, 0, A(1)
 };
 static const unsigned char sr_dac_mult_table[] = {
-        A(2), A(12), A(2), A(12),  A(0), A(0), A(3), A(1),
-        A(2), A(11), A(2), A(11),  A(0), A(0), A(0), A(1)
+        A(2), A(12), A(2), A(12),  0, 0, A(3), A(1),
+        A(2), A(11), A(2), A(11),  0, 0, 0, A(1)
 };
 
 static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
@@ -523,6 +523,8 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
         case SND_SOC_DAIFMT_I2S:
                 iface_reg |= TLV320AIC23_FOR_I2S;
                 break;
+        case SND_SOC_DAIFMT_DSP_A:
+                iface_reg |= TLV320AIC23_LRP_ON;
         case SND_SOC_DAIFMT_DSP_B:
                 iface_reg |= TLV320AIC23_FOR_DSP;
                 break;
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28a..4dbb853eef5a 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -115,6 +115,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
         0x00, /* REG_VIBRA_PWM_SET      (0x47)  */
         0x00, /* REG_ANAMIC_GAIN        (0x48)  */
         0x00, /* REG_MISC_SET_2         (0x49)  */
+        0x00, /* REG_SW_SHADOW          (0x4A)  - Shadow, non HW register */
 };
 
 /* codec private data */
@@ -125,6 +126,17 @@ struct twl4030_priv {
 
         struct snd_pcm_substream *master_substream;
         struct snd_pcm_substream *slave_substream;
+
+        unsigned int configured;
+        unsigned int rate;
+        unsigned int sample_bits;
+        unsigned int channels;
+
+        unsigned int sysclk;
+
+        /* Headset output state handling */
+        unsigned int hsl_enabled;
+        unsigned int hsr_enabled;
 };
 
 /*
@@ -161,7 +173,11 @@ static int twl4030_write(struct snd_soc_codec *codec,
                         unsigned int reg, unsigned int value)
 {
         twl4030_write_reg_cache(codec, reg, value);
-        return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+        if (likely(reg < TWL4030_REG_SW_SHADOW))
+                return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
+                                            reg);
+        else
+                return 0;
 }
 
 static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
@@ -188,6 +204,7 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 
 static void twl4030_init_chip(struct snd_soc_codec *codec)
 {
+        u8 *cache = codec->reg_cache;
         int i;
 
         /* clear CODECPDZ prior to setting register defaults */
@@ -195,7 +212,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
         /* set all audio section registers to reasonable defaults */
         for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
-                twl4030_write(codec, i, twl4030_reg[i]);
+                twl4030_write(codec, i, cache[i]);
 
 }
 
@@ -232,7 +249,7 @@ static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute)
                                         TWL4030_REG_PRECKL_CTL);
                 reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL);
                 twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
-                                        reg_val & (~TWL4030_PRECKL_GAIN),
+                                        reg_val & (~TWL4030_PRECKR_GAIN),
                                         TWL4030_REG_PRECKR_CTL);
 
                 /* Disable PLL */
@@ -316,104 +333,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec)
 }
 
 /* Earpiece */
-static const char *twl4030_earpiece_texts[] =
-                {"Off", "DACL1", "DACL2", "DACR1"};
-
-static const unsigned int twl4030_earpiece_values[] =
-                {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_earpiece_enum =
-        SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7,
-                        ARRAY_SIZE(twl4030_earpiece_texts),
-                        twl4030_earpiece_texts,
-                        twl4030_earpiece_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum);
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = {
+        SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0),
+        SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0),
+        SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0),
+        SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0),
+};
 
 /* PreDrive Left */
-static const char *twl4030_predrivel_texts[] =
-                {"Off", "DACL1", "DACL2", "DACR2"};
-
-static const unsigned int twl4030_predrivel_values[] =
-                {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_predrivel_enum =
-        SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7,
-                        ARRAY_SIZE(twl4030_predrivel_texts),
-                        twl4030_predrivel_texts,
-                        twl4030_predrivel_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum);
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = {
+        SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0),
+        SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0),
+        SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0),
+        SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0),
+};
 
 /* PreDrive Right */
-static const char *twl4030_predriver_texts[] =
-                {"Off", "DACR1", "DACR2", "DACL2"};
-
-static const unsigned int twl4030_predriver_values[] =
-                {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_predriver_enum =
-        SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7,
-                        ARRAY_SIZE(twl4030_predriver_texts),
-                        twl4030_predriver_texts,
-                        twl4030_predriver_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum);
+static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = {
+        SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0),
+        SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0),
+        SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0),
+        SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0),
+};
 
 /* Headset Left */
-static const char *twl4030_hsol_texts[] =
-                {"Off", "DACL1", "DACL2"};
-
-static const struct soc_enum twl4030_hsol_enum =
-        SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
-                        ARRAY_SIZE(twl4030_hsol_texts),
-                        twl4030_hsol_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
-SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = {
+        SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0),
+        SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0),
+        SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0),
+};
 
 /* Headset Right */
-static const char *twl4030_hsor_texts[] =
-                {"Off", "DACR1", "DACR2"};
-
-static const struct soc_enum twl4030_hsor_enum =
-        SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
-                        ARRAY_SIZE(twl4030_hsor_texts),
-                        twl4030_hsor_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
-SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = {
+        SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0),
+        SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0),
+        SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0),
+};
 
 /* Carkit Left */
-static const char *twl4030_carkitl_texts[] =
-                {"Off", "DACL1", "DACL2"};
-
-static const struct soc_enum twl4030_carkitl_enum =
-        SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
-                        ARRAY_SIZE(twl4030_carkitl_texts),
-                        twl4030_carkitl_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
-SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = {
+        SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0),
+        SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0),
+        SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0),
+};
 
 /* Carkit Right */
-static const char *twl4030_carkitr_texts[] =
-                {"Off", "DACR1", "DACR2"};
-
-static const struct soc_enum twl4030_carkitr_enum =
-        SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
-                        ARRAY_SIZE(twl4030_carkitr_texts),
-                        twl4030_carkitr_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
-SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = {
+        SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0),
+        SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0),
+        SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0),
+};
 
 /* Handsfree Left */
 static const char *twl4030_handsfreel_texts[] =
-                {"Voice", "DACL1", "DACL2", "DACR2"};
+                {"Voice", "AudioL1", "AudioL2", "AudioR2"};
 
 static const struct soc_enum twl4030_handsfreel_enum =
         SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
@@ -423,9 +396,13 @@ static const struct soc_enum twl4030_handsfreel_enum =
 static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
 SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
 
+/* Handsfree Left virtual mute */
+static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control =
+        SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0);
+
 /* Handsfree Right */
 static const char *twl4030_handsfreer_texts[] =
-                {"Voice", "DACR1", "DACR2", "DACL2"};
+                {"Voice", "AudioR1", "AudioR2", "AudioL2"};
 
 static const struct soc_enum twl4030_handsfreer_enum =
         SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
@@ -435,37 +412,48 @@ static const struct soc_enum twl4030_handsfreer_enum =
 static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
 SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
 
-/* Left analog microphone selection */
-static const char *twl4030_analoglmic_texts[] =
-                {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};
+/* Handsfree Right virtual mute */
+static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control =
+        SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0);
 
-static const unsigned int twl4030_analoglmic_values[] =
-                {0x0, 0x1, 0x2, 0x4, 0x8};
+/* Vibra */
+/* Vibra audio path selection */
+static const char *twl4030_vibra_texts[] =
+                {"AudioL1", "AudioR1", "AudioL2", "AudioR2"};
 
-static const struct soc_enum twl4030_analoglmic_enum =
-        SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf,
-                        ARRAY_SIZE(twl4030_analoglmic_texts),
-                        twl4030_analoglmic_texts,
-                        twl4030_analoglmic_values);
+static const struct soc_enum twl4030_vibra_enum =
+        SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2,
+                        ARRAY_SIZE(twl4030_vibra_texts),
+                        twl4030_vibra_texts);
 
-static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum);
+static const struct snd_kcontrol_new twl4030_dapm_vibra_control =
+SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
 
-/* Right analog microphone selection */
-static const char *twl4030_analogrmic_texts[] =
-                {"Off", "Sub mic", "AUXR"};
+/* Vibra path selection: local vibrator (PWM) or audio driven */
+static const char *twl4030_vibrapath_texts[] =
+                {"Local vibrator", "Audio"};
+
+static const struct soc_enum twl4030_vibrapath_enum =
+        SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4,
+                        ARRAY_SIZE(twl4030_vibrapath_texts),
+                        twl4030_vibrapath_texts);
 
-static const unsigned int twl4030_analogrmic_values[] =
-                {0x0, 0x1, 0x4};
+static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
+SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
 
-static const struct soc_enum twl4030_analogrmic_enum =
-        SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5,
-                        ARRAY_SIZE(twl4030_analogrmic_texts),
-                        twl4030_analogrmic_texts,
-                        twl4030_analogrmic_values);
+/* Left analog microphone selection */
+static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
+        SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0),
+        SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0),
+        SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0),
+        SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0),
+};
 
-static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum);
+/* Right analog microphone selection */
+static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
+        SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0),
+        SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 2, 1, 0),
+};
 
 /* TX1 L/R Analog/Digital microphone selection */
 static const char *twl4030_micpathtx1_texts[] =
@@ -507,6 +495,10 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control =
 static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
         SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
 
+/* Analog bypass for Voice */
+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 */
 static const unsigned int twl4030_dapm_dbypass_tlv[] = {
         TLV_DB_RANGE_HEAD(2),
@@ -526,6 +518,18 @@ static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control =
                         TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
                         twl4030_dapm_dbypass_tlv);
 
+/*
+ * Voice Sidetone GAIN volume control:
+ * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB)
+ */
+static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1);
+
+/* Digital bypass voice: sidetone (VUL -> VDL)*/
+static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
+        SOC_DAPM_SINGLE_TLV("Volume",
+                        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)
 {
@@ -556,63 +560,143 @@ static int micpath_event(struct snd_soc_dapm_widget *w,
         return 0;
 }
 
-static int handsfree_event(struct snd_soc_dapm_widget *w,
-                struct snd_kcontrol *kcontrol, int event)
+static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
 {
-        struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
         unsigned char hs_ctl;
 
-        hs_ctl = twl4030_read_reg_cache(w->codec, e->reg);
+        hs_ctl = twl4030_read_reg_cache(codec, reg);
 
-        if (hs_ctl & TWL4030_HF_CTL_REF_EN) {
+        if (ramp) {
+                /* HF ramp-up */
+                hs_ctl |= TWL4030_HF_CTL_REF_EN;
+                twl4030_write(codec, reg, hs_ctl);
+                udelay(10);
                 hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
-                twl4030_write(w->codec, e->reg, hs_ctl);
+                twl4030_write(codec, reg, hs_ctl);
+                udelay(40);
                 hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
-                twl4030_write(w->codec, e->reg, hs_ctl);
                 hs_ctl |= TWL4030_HF_CTL_HB_EN;
-                twl4030_write(w->codec, e->reg, hs_ctl);
+                twl4030_write(codec, reg, hs_ctl);
         } else {
-                hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN
-                                | TWL4030_HF_CTL_HB_EN);
-                twl4030_write(w->codec, e->reg, hs_ctl);
+                /* HF ramp-down */
+                hs_ctl &= ~TWL4030_HF_CTL_LOOP_EN;
+                hs_ctl &= ~TWL4030_HF_CTL_HB_EN;
+                twl4030_write(codec, reg, hs_ctl);
+                hs_ctl &= ~TWL4030_HF_CTL_RAMP_EN;
+                twl4030_write(codec, reg, hs_ctl);
+                udelay(40);
+                hs_ctl &= ~TWL4030_HF_CTL_REF_EN;
+                twl4030_write(codec, reg, hs_ctl);
         }
+}
 
+static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
+                struct snd_kcontrol *kcontrol, int event)
+{
+        switch (event) {
+        case SND_SOC_DAPM_POST_PMU:
+                handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1);
+                break;
+        case SND_SOC_DAPM_POST_PMD:
+                handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0);
+                break;
+        }
         return 0;
 }
 
-static int headsetl_event(struct snd_soc_dapm_widget *w,
+static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
                 struct snd_kcontrol *kcontrol, int event)
 {
+        switch (event) {
+        case SND_SOC_DAPM_POST_PMU:
+                handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1);
+                break;
+        case SND_SOC_DAPM_POST_PMD:
+                handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0);
+                break;
+        }
+        return 0;
+}
+
+static void headset_ramp(struct snd_soc_codec *codec, int ramp)
+{
         unsigned char hs_gain, hs_pop;
+        struct twl4030_priv *twl4030 = codec->private_data;
+        /* Base values for ramp delay calculation: 2^19 - 2^26 */
+        unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304,
+                                    8388608, 16777216, 33554432, 67108864};
 
-        /* Save the current volume */
-        hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
-        hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET);
+        hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET);
+        hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
 
-        switch (event) {
-        case SND_SOC_DAPM_POST_PMU:
-                /* Do the anti-pop/bias ramp enable according to the TRM */
+        if (ramp) {
+                /* Headset ramp-up according to the TRM */
                 hs_pop |= TWL4030_VMID_EN;
-                twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-                /* Is this needed? Can we just use whatever gain here? */
-                twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET,
-                                (hs_gain & (~0x0f)) | 0x0a);
+                twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+                twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
                 hs_pop |= TWL4030_RAMP_EN;
-                twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-
-                /* Restore the original volume */
-                twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
-                break;
-        case SND_SOC_DAPM_POST_PMD:
-                /* Do the anti-pop/bias ramp disable according to the TRM */
+                twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+        } else {
+                /* Headset ramp-down _not_ according to
+                 * the TRM, but in a way that it is working */
                 hs_pop &= ~TWL4030_RAMP_EN;
-                twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+                twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+                /* Wait ramp delay time + 1, so the VMID can settle */
+                mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
+                        twl4030->sysclk) + 1);
                 /* Bypass the reg_cache to mute the headset */
                 twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
                                         hs_gain & (~0x0f),
                                         TWL4030_REG_HS_GAIN_SET);
+
                 hs_pop &= ~TWL4030_VMID_EN;
-                twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+                twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+        }
+}
+
+static int headsetlpga_event(struct snd_soc_dapm_widget *w,
+                struct snd_kcontrol *kcontrol, int event)
+{
+        struct twl4030_priv *twl4030 = w->codec->private_data;
+
+        switch (event) {
+        case SND_SOC_DAPM_POST_PMU:
+                /* Do the ramp-up only once */
+                if (!twl4030->hsr_enabled)
+                        headset_ramp(w->codec, 1);
+
+                twl4030->hsl_enabled = 1;
+                break;
+        case SND_SOC_DAPM_POST_PMD:
+                /* Do the ramp-down only if both headsetL/R is disabled */
+                if (!twl4030->hsr_enabled)
+                        headset_ramp(w->codec, 0);
+
+                twl4030->hsl_enabled = 0;
+                break;
+        }
+        return 0;
+}
+
+static int headsetrpga_event(struct snd_soc_dapm_widget *w,
+                struct snd_kcontrol *kcontrol, int event)
+{
+        struct twl4030_priv *twl4030 = w->codec->private_data;
+
+        switch (event) {
+        case SND_SOC_DAPM_POST_PMU:
+                /* Do the ramp-up only once */
+                if (!twl4030->hsl_enabled)
+                        headset_ramp(w->codec, 1);
+
+                twl4030->hsr_enabled = 1;
+                break;
+        case SND_SOC_DAPM_POST_PMD:
+                /* Do the ramp-down only if both headsetL/R is disabled */
+                if (!twl4030->hsl_enabled)
+                        headset_ramp(w->codec, 0);
+
+                twl4030->hsr_enabled = 0;
                 break;
         }
         return 0;
@@ -624,7 +708,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
         struct soc_mixer_control *m =
                 (struct soc_mixer_control *)w->kcontrols->private_value;
         struct twl4030_priv *twl4030 = w->codec->private_data;
-        unsigned char reg;
+        unsigned char reg, misc;
 
         reg = twl4030_read_reg_cache(w->codec, m->reg);
 
@@ -636,14 +720,34 @@ static int bypass_event(struct snd_soc_dapm_widget *w,
                 else
                         twl4030->bypass_state &=
                                 ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+        } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) {
+                /* Analog voice bypass */
+                if (reg & (1 << m->shift))
+                        twl4030->bypass_state |= (1 << 4);
+                else
+                        twl4030->bypass_state &= ~(1 << 4);
+        } else if (m->reg == TWL4030_REG_VSTPGA) {
+                /* Voice digital bypass */
+                if (reg)
+                        twl4030->bypass_state |= (1 << 5);
+                else
+                        twl4030->bypass_state &= ~(1 << 5);
         } else {
                 /* Digital bypass */
                 if (reg & (0x7 << m->shift))
-                        twl4030->bypass_state |= (1 << (m->shift ? 5 : 4));
+                        twl4030->bypass_state |= (1 << (m->shift ? 7 : 6));
                 else
-                        twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4));
+                        twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6));
         }
 
+        /* Enable master analog loopback mode if any analog switch is enabled*/
+        misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1);
+        if (twl4030->bypass_state & 0x1F)
+                misc |= TWL4030_FMLOOP_EN;
+        else
+                misc &= ~TWL4030_FMLOOP_EN;
+        twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc);
+
         if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
                 if (twl4030->bypass_state)
                         twl4030_codec_mute(w->codec, 0);
@@ -810,6 +914,48 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol,
         return err;
 }
 
+/* Codec operation modes */
+static const char *twl4030_op_modes_texts[] = {
+        "Option 2 (voice/audio)", "Option 1 (audio)"
+};
+
+static const struct soc_enum twl4030_op_modes_enum =
+        SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0,
+                        ARRAY_SIZE(twl4030_op_modes_texts),
+                        twl4030_op_modes_texts);
+
+int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct twl4030_priv *twl4030 = codec->private_data;
+        struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+        unsigned short val;
+        unsigned short mask, bitmask;
+
+        if (twl4030->configured) {
+                printk(KERN_ERR "twl4030 operation mode cannot be "
+                        "changed on-the-fly\n");
+                return -EBUSY;
+        }
+
+        for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+                ;
+        if (ucontrol->value.enumerated.item[0] > e->max - 1)
+                return -EINVAL;
+
+        val = ucontrol->value.enumerated.item[0] << e->shift_l;
+        mask = (bitmask - 1) << e->shift_l;
+        if (e->shift_l != e->shift_r) {
+                if (ucontrol->value.enumerated.item[1] > e->max - 1)
+                        return -EINVAL;
+                val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+                mask |= (bitmask - 1) << e->shift_r;
+        }
+
+        return snd_soc_update_bits(codec, e->reg, mask, val);
+}
+
 /*
  * FGAIN volume control:
  * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
@@ -824,6 +970,12 @@ static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1);
 static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
 
 /*
+ * Voice Downlink GAIN volume control:
+ * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB)
+ */
+static DECLARE_TLV_DB_SCALE(digital_voice_downlink_tlv, -3700, 100, 1);
+
+/*
  * Analog playback gain
  * -24 dB to 12 dB in 2 dB steps
  */
@@ -836,6 +988,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
 static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
 
 /*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
  * Capture gain after the ADCs
  * from 0 dB to 31 dB in 1 dB steps
  */
@@ -858,7 +1016,32 @@ static const struct soc_enum twl4030_rampdelay_enum =
                         ARRAY_SIZE(twl4030_rampdelay_texts),
                         twl4030_rampdelay_texts);
 
+/* Vibra H-bridge direction mode */
+static const char *twl4030_vibradirmode_texts[] = {
+        "Vibra H-bridge direction", "Audio data MSB",
+};
+
+static const struct soc_enum twl4030_vibradirmode_enum =
+        SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5,
+                        ARRAY_SIZE(twl4030_vibradirmode_texts),
+                        twl4030_vibradirmode_texts);
+
+/* Vibra H-bridge direction */
+static const char *twl4030_vibradir_texts[] = {
+        "Positive polarity", "Negative polarity",
+};
+
+static const struct soc_enum twl4030_vibradir_enum =
+        SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1,
+                        ARRAY_SIZE(twl4030_vibradir_texts),
+                        twl4030_vibradir_texts);
+
 static const struct snd_kcontrol_new twl4030_snd_controls[] = {
+        /* Codec operation mode control */
+        SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum,
+                snd_soc_get_enum_double,
+                snd_soc_put_twl4030_opmode_enum_double),
+
         /* Common playback gain controls */
         SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
                 TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
@@ -887,6 +1070,16 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
                 TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
                 1, 1, 0),
 
+        /* Common voice downlink gain controls */
+        SOC_SINGLE_TLV("DAC Voice Digital Downlink Volume",
+                TWL4030_REG_VRXPGA, 0, 0x31, 0, digital_voice_downlink_tlv),
+
+        SOC_SINGLE_TLV("DAC Voice Analog Downlink Volume",
+                TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv),
+
+        SOC_SINGLE("DAC Voice Analog Downlink Switch",
+                TWL4030_REG_VDL_APGA_CTL, 1, 1, 0),
+
         /* Separate output gain controls */
         SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
                 TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
@@ -900,7 +1093,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
                 4, 3, 0, output_tvl),
 
         SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
-                TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+                TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
 
         /* Common capture gain controls */
         SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
@@ -914,6 +1107,9 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
                 0, 3, 5, 0, input_gain_tlv),
 
         SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
+
+        SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
+        SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum),
 };
 
 static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
@@ -941,26 +1137,19 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
         SND_SOC_DAPM_OUTPUT("CARKITR"),
         SND_SOC_DAPM_OUTPUT("HFL"),
         SND_SOC_DAPM_OUTPUT("HFR"),
+        SND_SOC_DAPM_OUTPUT("VIBRA"),
 
         /* DACs */
-        SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
+        SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback",
                         SND_SOC_NOPM, 0, 0),
-        SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
+        SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback",
                         SND_SOC_NOPM, 0, 0),
-        SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
+        SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback",
                         SND_SOC_NOPM, 0, 0),
-        SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
+        SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback",
+                        SND_SOC_NOPM, 0, 0),
+        SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback",
                         SND_SOC_NOPM, 0, 0),
-
-        /* Analog PGAs */
-        SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
-                        0, 0, NULL, 0),
-        SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
-                        0, 0, NULL, 0),
-        SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
-                        0, 0, NULL, 0),
-        SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
-                        0, 0, NULL, 0),
 
         /* Analog bypasses */
         SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
@@ -975,6 +1164,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
         SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
                         &twl4030_dapm_abypassl2_control,
                         bypass_event, SND_SOC_DAPM_POST_REG),
+        SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_abypassv_control,
+                        bypass_event, SND_SOC_DAPM_POST_REG),
 
         /* Digital bypasses */
         SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
@@ -983,43 +1175,88 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
         SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
                         &twl4030_dapm_dbypassr_control, bypass_event,
                         SND_SOC_DAPM_POST_REG),
+        SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_dbypassv_control, bypass_event,
+                        SND_SOC_DAPM_POST_REG),
 
-        SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
-                        0, 0, NULL, 0),
-        SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
-                        1, 0, NULL, 0),
-        SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
-                        2, 0, NULL, 0),
-        SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
-                        3, 0, NULL, 0),
-
-        /* Output MUX controls */
+        /* Digital mixers, power control for the physical DACs */
+        SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer",
+                        TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0),
+        SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer",
+                        TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0),
+        SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer",
+                        TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0),
+        SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer",
+                        TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0),
+        SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer",
+                        TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0),
+
+        /* Analog mixers, power control for the physical PGAs */
+        SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer",
+                        TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0),
+        SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer",
+                        TWL4030_REG_ARXL1_APGA_CTL, 0, 0, NULL, 0),
+        SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer",
+                        TWL4030_REG_ARXR2_APGA_CTL, 0, 0, NULL, 0),
+        SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer",
+                        TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0),
+        SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer",
+                        TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0),
+
+        /* Output MIXER controls */
         /* Earpiece */
-        SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_earpiece_control),
+        SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_earpiece_controls[0],
+                        ARRAY_SIZE(twl4030_dapm_earpiece_controls)),
         /* PreDrivL/R */
-        SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_predrivel_control),
-        SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_predriver_control),
+        SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_predrivel_controls[0],
+                        ARRAY_SIZE(twl4030_dapm_predrivel_controls)),
+        SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_predriver_controls[0],
+                        ARRAY_SIZE(twl4030_dapm_predriver_controls)),
         /* HeadsetL/R */
-        SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_hsol_control, headsetl_event,
-                SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
-        SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_hsor_control),
+        SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_hsol_controls[0],
+                        ARRAY_SIZE(twl4030_dapm_hsol_controls)),
+        SND_SOC_DAPM_PGA_E("HeadsetL PGA", SND_SOC_NOPM,
+                        0, 0, NULL, 0, headsetlpga_event,
+                        SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+        SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_hsor_controls[0],
+                        ARRAY_SIZE(twl4030_dapm_hsor_controls)),
+        SND_SOC_DAPM_PGA_E("HeadsetR PGA", SND_SOC_NOPM,
+                        0, 0, NULL, 0, headsetrpga_event,
+                        SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
         /* CarkitL/R */
-        SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_carkitl_control),
-        SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
-                &twl4030_dapm_carkitr_control),
+        SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_carkitl_controls[0],
+                        ARRAY_SIZE(twl4030_dapm_carkitl_controls)),
+        SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_carkitr_controls[0],
+                        ARRAY_SIZE(twl4030_dapm_carkitr_controls)),
+
+        /* Output MUX controls */
         /* HandsfreeL/R */
-        SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
-                &twl4030_dapm_handsfreel_control, handsfree_event,
-                SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
-        SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
-                &twl4030_dapm_handsfreer_control, handsfree_event,
-                SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+        SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0,
+                &twl4030_dapm_handsfreel_control),
+        SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_handsfreelmute_control),
+        SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM,
+                        0, 0, NULL, 0, handsfreelpga_event,
+                        SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+        SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0,
+                &twl4030_dapm_handsfreer_control),
+        SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0,
+                        &twl4030_dapm_handsfreermute_control),
+        SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM,
+                        0, 0, NULL, 0, handsfreerpga_event,
+                        SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+        /* Vibra */
+        SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0,
+                &twl4030_dapm_vibra_control),
+        SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0,
+                &twl4030_dapm_vibrapath_control),
 
         /* Introducing four virtual ADC, since TWL4030 have four channel for
            capture */
@@ -1044,11 +1281,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
                 SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
                 SND_SOC_DAPM_POST_REG),
 
-        /* Analog input muxes with switch for the capture amplifiers */
-        SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
-                TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
-        SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
-                TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
+        /* Analog input mixers for the capture amplifiers */
+        SND_SOC_DAPM_MIXER("Analog Left Capture Route",
+                TWL4030_REG_ANAMICL, 4, 0,
+                &twl4030_dapm_analoglmic_controls[0],
+                ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
+        SND_SOC_DAPM_MIXER("Analog Right Capture Route",
+                TWL4030_REG_ANAMICR, 4, 0,
+                &twl4030_dapm_analogrmic_controls[0],
+                ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),
 
         SND_SOC_DAPM_PGA("ADC Physical Left",
                 TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
@@ -1067,62 +1308,86 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
-        {"Analog L1 Playback Mixer", NULL, "DAC Left1"},
-        {"Analog R1 Playback Mixer", NULL, "DAC Right1"},
-        {"Analog L2 Playback Mixer", NULL, "DAC Left2"},
-        {"Analog R2 Playback Mixer", NULL, "DAC Right2"},
-
-        {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"},
-        {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"},
-        {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"},
-        {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"},
+        {"Digital L1 Playback Mixer", NULL, "DAC Left1"},
+        {"Digital R1 Playback Mixer", NULL, "DAC Right1"},
+        {"Digital L2 Playback Mixer", NULL, "DAC Left2"},
+        {"Digital R2 Playback Mixer", NULL, "DAC Right2"},
+        {"Digital Voice Playback Mixer", NULL, "DAC Voice"},
+
+        {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"},
+        {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"},
+        {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"},
+        {"Analog R2 Playback Mixer", NULL, "Digital R2 Playback Mixer"},
+        {"Analog Voice Playback Mixer", NULL, "Digital Voice Playback Mixer"},
 
         /* Internal playback routings */
         /* Earpiece */
-        {"Earpiece Mux", "DACL1", "ARXL1_APGA"},
-        {"Earpiece Mux", "DACL2", "ARXL2_APGA"},
-        {"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+        {"Earpiece Mixer", "Voice", "Analog Voice Playback Mixer"},
+        {"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+        {"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+        {"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"},
         /* PreDrivL */
-        {"PredriveL Mux", "DACL1", "ARXL1_APGA"},
-        {"PredriveL Mux", "DACL2", "ARXL2_APGA"},
-        {"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+        {"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"},
+        {"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+        {"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+        {"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"},
         /* PreDrivR */
-        {"PredriveR Mux", "DACR1", "ARXR1_APGA"},
-        {"PredriveR Mux", "DACR2", "ARXR2_APGA"},
-        {"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+        {"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"},
+        {"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+        {"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+        {"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"},
         /* HeadsetL */
-        {"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
-        {"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+        {"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"},
+        {"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+        {"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+        {"HeadsetL PGA", NULL, "HeadsetL Mixer"},
         /* HeadsetR */
-        {"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
-        {"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+        {"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"},
+        {"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+        {"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+        {"HeadsetR PGA", NULL, "HeadsetR Mixer"},
         /* CarkitL */
-        {"CarkitL Mux", "DACL1", "ARXL1_APGA"},
-        {"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+        {"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"},
+        {"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+        {"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
         /* CarkitR */
-        {"CarkitR Mux", "DACR1", "ARXR1_APGA"},
-        {"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+        {"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"},
+        {"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+        {"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
         /* HandsfreeL */
-        {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
-        {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
-        {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+        {"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"},
+        {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"},
+        {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"},
+        {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"},
+        {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"},
+        {"HandsfreeL PGA", NULL, "HandsfreeL Switch"},
         /* HandsfreeR */
-        {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
-        {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
-        {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
+        {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"},
+        {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"},
+        {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"},
+        {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"},
+        {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"},
+        {"HandsfreeR PGA", NULL, "HandsfreeR Switch"},
+        /* Vibra */
+        {"Vibra Mux", "AudioL1", "DAC Left1"},
+        {"Vibra Mux", "AudioR1", "DAC Right1"},
+        {"Vibra Mux", "AudioL2", "DAC Left2"},
+        {"Vibra Mux", "AudioR2", "DAC Right2"},
 
         /* outputs */
-        {"OUTL", NULL, "ARXL2_APGA"},
-        {"OUTR", NULL, "ARXR2_APGA"},
-        {"EARPIECE", NULL, "Earpiece Mux"},
-        {"PREDRIVEL", NULL, "PredriveL Mux"},
-        {"PREDRIVER", NULL, "PredriveR Mux"},
-        {"HSOL", NULL, "HeadsetL Mux"},
-        {"HSOR", NULL, "HeadsetR Mux"},
-        {"CARKITL", NULL, "CarkitL Mux"},
-        {"CARKITR", NULL, "CarkitR Mux"},
-        {"HFL", NULL, "HandsfreeL Mux"},
-        {"HFR", NULL, "HandsfreeR Mux"},
+        {"OUTL", NULL, "Analog L2 Playback Mixer"},
+        {"OUTR", NULL, "Analog R2 Playback Mixer"},
+        {"EARPIECE", NULL, "Earpiece Mixer"},
+        {"PREDRIVEL", NULL, "PredriveL Mixer"},
+        {"PREDRIVER", NULL, "PredriveR Mixer"},
+        {"HSOL", NULL, "HeadsetL PGA"},
+        {"HSOR", NULL, "HeadsetR PGA"},
+        {"CARKITL", NULL, "CarkitL Mixer"},
+        {"CARKITR", NULL, "CarkitR Mixer"},
+        {"HFL", NULL, "HandsfreeL PGA"},
+        {"HFR", NULL, "HandsfreeR PGA"},
+        {"Vibra Route", "Audio", "Vibra Mux"},
+        {"VIBRA", NULL, "Vibra Route"},
 
         /* Capture path */
         {"Analog Left Capture Route", "Main mic", "MAINMIC"},
@@ -1162,18 +1427,22 @@ static const struct snd_soc_dapm_route intercon[] = {
         {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
         {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
         {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
+        {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"},
 
         {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
         {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
         {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
         {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
+        {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"},
 
         /* Digital bypass routes */
         {"Right Digital Loopback", "Volume", "TX1 Capture Route"},
         {"Left Digital Loopback", "Volume", "TX1 Capture Route"},
+        {"Voice Digital Loopback", "Volume", "TX2 Capture Route"},
 
-        {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"},
-        {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"},
+        {"Digital R2 Playback Mixer", NULL, "Right Digital Loopback"},
+        {"Digital L2 Playback Mixer", NULL, "Left Digital Loopback"},
+        {"Digital Voice Playback Mixer", NULL, "Voice Digital Loopback"},
 
 };
 
@@ -1220,6 +1489,58 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
         return 0;
 }
 
+static void twl4030_constraints(struct twl4030_priv *twl4030,
+                                struct snd_pcm_substream *mst_substream)
+{
+        struct snd_pcm_substream *slv_substream;
+
+        /* Pick the stream, which need to be constrained */
+        if (mst_substream == twl4030->master_substream)
+                slv_substream = twl4030->slave_substream;
+        else if (mst_substream == twl4030->slave_substream)
+                slv_substream = twl4030->master_substream;
+        else /* This should not happen.. */
+                return;
+
+        /* Set the constraints according to the already configured stream */
+        snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+                                SNDRV_PCM_HW_PARAM_RATE,
+                                twl4030->rate,
+                                twl4030->rate);
+
+        snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+                                SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+                                twl4030->sample_bits,
+                                twl4030->sample_bits);
+
+        snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+                                SNDRV_PCM_HW_PARAM_CHANNELS,
+                                twl4030->channels,
+                                twl4030->channels);
+}
+
+/* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for
+ * capture has to be enabled/disabled. */
+static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
+                                int enable)
+{
+        u8 reg, mask;
+
+        reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+
+        if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+                mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN;
+        else
+                mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
+
+        if (enable)
+                reg |= mask;
+        else
+                reg &= ~mask;
+
+        twl4030_write(codec, TWL4030_REG_OPTION, reg);
+}
+
 static int twl4030_startup(struct snd_pcm_substream *substream,
                            struct snd_soc_dai *dai)
 {
@@ -1228,26 +1549,25 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
         struct snd_soc_codec *codec = socdev->card->codec;
         struct twl4030_priv *twl4030 = codec->private_data;
 
-        /* If we already have a playback or capture going then constrain
-         * this substream to match it.
-         */
         if (twl4030->master_substream) {
-                struct snd_pcm_runtime *master_runtime;
-                master_runtime = twl4030->master_substream->runtime;
-
-                snd_pcm_hw_constraint_minmax(substream->runtime,
-                                             SNDRV_PCM_HW_PARAM_RATE,
-                                             master_runtime->rate,
-                                             master_runtime->rate);
-
-                snd_pcm_hw_constraint_minmax(substream->runtime,
-                                             SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-                                             master_runtime->sample_bits,
-                                             master_runtime->sample_bits);
-
                 twl4030->slave_substream = substream;
-        } else
+                /* The DAI has one configuration for playback and capture, so
+                 * if the DAI has been already configured then constrain this
+                 * substream to match it. */
+                if (twl4030->configured)
+                        twl4030_constraints(twl4030, twl4030->master_substream);
+        } else {
+                if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) &
+                        TWL4030_OPTION_1)) {
+                        /* In option2 4 channel is not supported, set the
+                         * constraint for the first stream for channels, the
+                         * second stream will 'inherit' this cosntraint */
+                        snd_pcm_hw_constraint_minmax(substream->runtime,
+                                                SNDRV_PCM_HW_PARAM_CHANNELS,
+                                                2, 2);
+                }
                 twl4030->master_substream = substream;
+        }
 
         return 0;
 }
@@ -1264,6 +1584,17 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream,
                 twl4030->master_substream = twl4030->slave_substream;
 
         twl4030->slave_substream = NULL;
+
+        /* If all streams are closed, or the remaining stream has not yet
+         * been configured than set the DAI as not configured. */
+        if (!twl4030->master_substream)
+                twl4030->configured = 0;
+         else if (!twl4030->master_substream->runtime->channels)
+                twl4030->configured = 0;
+
+         /* If the closing substream had 4 channel, do the necessary cleanup */
+        if (substream->runtime->channels == 4)
+                twl4030_tdm_enable(codec, substream->stream, 0);
 }
 
 static int twl4030_hw_params(struct snd_pcm_substream *substream,
@@ -1276,8 +1607,24 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
         struct twl4030_priv *twl4030 = codec->private_data;
         u8 mode, old_mode, format, old_format;
 
-        if (substream == twl4030->slave_substream)
-                /* Ignoring hw_params for slave substream */
+         /* If the substream has 4 channel, do the necessary setup */
+        if (params_channels(params) == 4) {
+                u8 format, mode;
+
+                format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+                mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+
+                /* Safety check: are we in the correct operating mode and
+                 * the interface is in TDM mode? */
+                if ((mode & TWL4030_OPTION_1) &&
+                    ((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM))
+                        twl4030_tdm_enable(codec, substream->stream, 1);
+                else
+                        return -EINVAL;
+        }
+
+        if (twl4030->configured)
+                /* Ignoring hw_params for already configured DAI */
                 return 0;
 
         /* bit rate */
@@ -1357,6 +1704,21 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
                 /* set CODECPDZ afterwards */
                 twl4030_codec_enable(codec, 1);
         }
+
+        /* Store the important parameters for the DAI configuration and set
+         * the DAI as configured */
+        twl4030->configured = 1;
+        twl4030->rate = params_rate(params);
+        twl4030->sample_bits = hw_param_interval(params,
+                                        SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
+        twl4030->channels = params_channels(params);
+
+        /* If both playback and capture streams are open, and one of them
+         * is setting the hw parameters right now (since we are here), set
+         * constraints to the other stream to match the current one. */
+        if (twl4030->slave_substream)
+                twl4030_constraints(twl4030, substream);
+
         return 0;
 }
 
@@ -1364,17 +1726,21 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
                 int clk_id, unsigned int freq, int dir)
 {
         struct snd_soc_codec *codec = codec_dai->codec;
+        struct twl4030_priv *twl4030 = codec->private_data;
         u8 infreq;
 
         switch (freq) {
         case 19200000:
                 infreq = TWL4030_APLL_INFREQ_19200KHZ;
+                twl4030->sysclk = 19200;
                 break;
         case 26000000:
                 infreq = TWL4030_APLL_INFREQ_26000KHZ;
+                twl4030->sysclk = 26000;
                 break;
         case 38400000:
                 infreq = TWL4030_APLL_INFREQ_38400KHZ;
+                twl4030->sysclk = 38400;
                 break;
         default:
                 printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n",
@@ -1418,6 +1784,9 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
         case SND_SOC_DAIFMT_I2S:
                 format |= TWL4030_AIF_FORMAT_CODEC;
                 break;
+        case SND_SOC_DAIFMT_DSP_A:
+                format |= TWL4030_AIF_FORMAT_TDM;
+                break;
         default:
                 return -EINVAL;
         }
@@ -1437,6 +1806,180 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
         return 0;
 }
 
+/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R
+ * (VTXL, VTXR) for uplink has to be enabled/disabled. */
+static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
+                                int enable)
+{
+        u8 reg, mask;
+
+        reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+
+        if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+                mask = TWL4030_ARXL1_VRX_EN;
+        else
+                mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
+
+        if (enable)
+                reg |= mask;
+        else
+                reg &= ~mask;
+
+        twl4030_write(codec, TWL4030_REG_OPTION, reg);
+}
+
+static int twl4030_voice_startup(struct snd_pcm_substream *substream,
+                struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_soc_codec *codec = socdev->card->codec;
+        u8 infreq;
+        u8 mode;
+
+        /* If the system master clock is not 26MHz, the voice PCM interface is
+         * not avilable.
+         */
+        infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL)
+                & TWL4030_APLL_INFREQ;
+
+        if (infreq != TWL4030_APLL_INFREQ_26000KHZ) {
+                printk(KERN_ERR "TWL4030 voice startup: "
+                        "MCLK is not 26MHz, call set_sysclk() on init\n");
+                return -EINVAL;
+        }
+
+        /* If the codec mode is not option2, the voice PCM interface is not
+         * avilable.
+         */
+        mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+                & TWL4030_OPT_MODE;
+
+        if (mode != TWL4030_OPTION_2) {
+                printk(KERN_ERR "TWL4030 voice startup: "
+                        "the codec mode is not option2\n");
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
+                                struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_soc_codec *codec = socdev->card->codec;
+
+        /* Enable voice digital filters */
+        twl4030_voice_enable(codec, substream->stream, 0);
+}
+
+static int twl4030_voice_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;
+        u8 old_mode, mode;
+
+        /* Enable voice digital filters */
+        twl4030_voice_enable(codec, substream->stream, 1);
+
+        /* bit rate */
+        old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+                & ~(TWL4030_CODECPDZ);
+        mode = old_mode;
+
+        switch (params_rate(params)) {
+        case 8000:
+                mode &= ~(TWL4030_SEL_16K);
+                break;
+        case 16000:
+                mode |= TWL4030_SEL_16K;
+                break;
+        default:
+                printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n",
+                        params_rate(params));
+                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);
+        }
+
+        return 0;
+}
+
+static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+                int clk_id, unsigned int freq, int dir)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u8 infreq;
+
+        switch (freq) {
+        case 26000000:
+                infreq = TWL4030_APLL_INFREQ_26000KHZ;
+                break;
+        default:
+                printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n",
+                        freq);
+                return -EINVAL;
+        }
+
+        infreq |= TWL4030_APLL_EN;
+        twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+
+        return 0;
+}
+
+static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+                unsigned int fmt)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u8 old_format, format;
+
+        /* get format */
+        old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
+        format = old_format;
+
+        /* set master/slave audio interface */
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBS_CFM:
+                format &= ~(TWL4030_VIF_SLAVE_EN);
+                break;
+        case SND_SOC_DAIFMT_CBS_CFS:
+                format |= TWL4030_VIF_SLAVE_EN;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /* clock inversion */
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_IB_NF:
+                format &= ~(TWL4030_VIF_FORMAT);
+                break;
+        case SND_SOC_DAIFMT_NB_IF:
+                format |= TWL4030_VIF_FORMAT;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        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);
+        }
+
+        return 0;
+}
+
 #define TWL4030_RATES    (SNDRV_PCM_RATE_8000_48000)
 #define TWL4030_FORMATS  (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
 
@@ -1448,21 +1991,47 @@ static struct snd_soc_dai_ops twl4030_dai_ops = {
         .set_fmt        = twl4030_set_dai_fmt,
 };
 
-struct snd_soc_dai twl4030_dai = {
+static struct snd_soc_dai_ops twl4030_dai_voice_ops = {
+        .startup        = twl4030_voice_startup,
+        .shutdown       = twl4030_voice_shutdown,
+        .hw_params      = twl4030_voice_hw_params,
+        .set_sysclk     = twl4030_voice_set_dai_sysclk,
+        .set_fmt        = twl4030_voice_set_dai_fmt,
+};
+
+struct snd_soc_dai twl4030_dai[] = {
+{
         .name = "twl4030",
         .playback = {
-                .stream_name = "Playback",
+                .stream_name = "HiFi Playback",
                 .channels_min = 2,
-                .channels_max = 2,
+                .channels_max = 4,
                 .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
                 .formats = TWL4030_FORMATS,},
         .capture = {
                 .stream_name = "Capture",
                 .channels_min = 2,
-                .channels_max = 2,
+                .channels_max = 4,
                 .rates = TWL4030_RATES,
                 .formats = TWL4030_FORMATS,},
         .ops = &twl4030_dai_ops,
+},
+{
+        .name = "twl4030 Voice",
+        .playback = {
+                .stream_name = "Voice Playback",
+                .channels_min = 1,
+                .channels_max = 1,
+                .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+        .capture = {
+                .stream_name = "Capture",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+        .ops = &twl4030_dai_voice_ops,
+},
 };
 EXPORT_SYMBOL_GPL(twl4030_dai);
 
@@ -1494,6 +2063,8 @@ static int twl4030_resume(struct platform_device *pdev)
 static int twl4030_init(struct snd_soc_device *socdev)
 {
         struct snd_soc_codec *codec = socdev->card->codec;
+        struct twl4030_setup_data *setup = socdev->codec_data;
+        struct twl4030_priv *twl4030 = codec->private_data;
         int ret = 0;
 
         printk(KERN_INFO "TWL4030 Audio Codec init \n");
@@ -1503,14 +2074,31 @@ static int twl4030_init(struct snd_soc_device *socdev)
         codec->read = twl4030_read_reg_cache;
         codec->write = twl4030_write;
         codec->set_bias_level = twl4030_set_bias_level;
-        codec->dai = &twl4030_dai;
-        codec->num_dai = 1;
+        codec->dai = twl4030_dai;
+        codec->num_dai = ARRAY_SIZE(twl4030_dai),
         codec->reg_cache_size = sizeof(twl4030_reg);
         codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
                                         GFP_KERNEL);
         if (codec->reg_cache == NULL)
                 return -ENOMEM;
 
+        /* Configuration for headset ramp delay from setup data */
+        if (setup) {
+                unsigned char hs_pop;
+
+                if (setup->sysclk)
+                        twl4030->sysclk = setup->sysclk;
+                else
+                        twl4030->sysclk = 26000;
+
+                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);
+        } else {
+                twl4030->sysclk = 26000;
+        }
+
         /* register pcms */
         ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
         if (ret < 0) {
@@ -1598,13 +2186,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
 
 static int __init twl4030_modinit(void)
 {
-        return snd_soc_register_dai(&twl4030_dai);
+        return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
 }
 module_init(twl4030_modinit);
 
 static void __exit twl4030_exit(void)
 {
-        snd_soc_unregister_dai(&twl4030_dai);
+        snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
 }
 module_exit(twl4030_exit);
 
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index cb63765db1df..fe5f395d9e4f 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -92,8 +92,9 @@
 #define TWL4030_REG_VIBRA_PWM_SET       0x47
 #define TWL4030_REG_ANAMIC_GAIN         0x48
 #define TWL4030_REG_MISC_SET_2          0x49
+#define TWL4030_REG_SW_SHADOW           0x4A
 
-#define TWL4030_CACHEREGNUM     (TWL4030_REG_MISC_SET_2 + 1)
+#define TWL4030_CACHEREGNUM     (TWL4030_REG_SW_SHADOW + 1)
 
 /* Bitfield Definitions */
 
@@ -110,9 +111,22 @@
 #define TWL4030_APLL_RATE_44100         0x90
 #define TWL4030_APLL_RATE_48000         0xA0
 #define TWL4030_APLL_RATE_96000         0xE0
-#define TWL4030_SEL_16K                 0x04
+#define TWL4030_SEL_16K                 0x08
 #define TWL4030_CODECPDZ                0x02
 #define TWL4030_OPT_MODE                0x01
+#define TWL4030_OPTION_1                (1 << 0)
+#define TWL4030_OPTION_2                (0 << 0)
+
+/* TWL4030_OPTION (0x02) Fields */
+
+#define TWL4030_ATXL1_EN                (1 << 0)
+#define TWL4030_ATXR1_EN                (1 << 1)
+#define TWL4030_ATXL2_VTXL_EN           (1 << 2)
+#define TWL4030_ATXR2_VTXR_EN           (1 << 3)
+#define TWL4030_ARXL1_VRX_EN            (1 << 4)
+#define TWL4030_ARXR1_EN                (1 << 5)
+#define TWL4030_ARXL2_EN                (1 << 6)
+#define TWL4030_ARXR2_EN                (1 << 7)
 
 /* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
 
@@ -171,6 +185,17 @@
 #define TWL4030_CLK256FS_EN             0x02
 #define TWL4030_AIF_EN                  0x01
 
+/* VOICE_IF (0x0F) Fields */
+
+#define TWL4030_VIF_SLAVE_EN            0x80
+#define TWL4030_VIF_DIN_EN              0x40
+#define TWL4030_VIF_DOUT_EN             0x20
+#define TWL4030_VIF_SWAP                0x10
+#define TWL4030_VIF_FORMAT              0x08
+#define TWL4030_VIF_TRI_EN              0x04
+#define TWL4030_VIF_SUB_EN              0x02
+#define TWL4030_VIF_EN                  0x01
+
 /* EAR_CTL (0x21) */
 #define TWL4030_EAR_GAIN                0x30
 
@@ -236,7 +261,19 @@
 #define TWL4030_SMOOTH_ANAVOL_EN        0x02
 #define TWL4030_DIGMIC_LR_SWAP_EN       0x01
 
-extern struct snd_soc_dai twl4030_dai;
+/* TWL4030_REG_SW_SHADOW (0x4A) Fields */
+#define TWL4030_HFL_EN                  0x01
+#define TWL4030_HFR_EN                  0x02
+
+#define TWL4030_DAI_HIFI                0
+#define TWL4030_DAI_VOICE               1
+
+extern struct snd_soc_dai twl4030_dai[2];
 extern struct snd_soc_codec_device soc_codec_dev_twl4030;
 
+struct twl4030_setup_data {
+        unsigned int ramp_delay_value;
+        unsigned int sysclk;
+};
+
 #endif  /* End of __TWL4030_AUDIO_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index ddefb8f80145..269b108e1de6 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
         pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
 
         if (reg >= UDA134X_REGS_NUM) {
-                printk(KERN_ERR "%s unkown register: reg: %d",
+                printk(KERN_ERR "%s unkown register: reg: %u",
                        __func__, reg);
                 return -EINVAL;
         }
@@ -296,7 +296,7 @@ static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
         struct snd_soc_codec *codec = codec_dai->codec;
         struct uda134x_priv *uda134x = codec->private_data;
 
-        pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+        pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__,
                  clk_id, freq, dir);
 
         /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed9..e7348d341b76 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
          * required for LRC in master mode. The DACs or ADCs need a
          * valid audio path i.e. pin -> ADC or DAC -> pin before
          * the LRC will be enabled in master mode. */
-        if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+        if (!master || cmd != SNDRV_PCM_TRIGGER_START)
                 return 0;
 
         if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -1108,7 +1108,7 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
         if (ret < 0)
                 return ret;
         dev_dbg(wm8350->dev,
-                "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+                "FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d",
                 freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
                 fll_div.ratio);
 
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
index d11bd9288cf9..d088eb4b88bb 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -13,6 +13,7 @@
 #define _WM8350_H
 
 #include <sound/soc.h>
+#include <linux/mfd/wm8350/audio.h>
 
 extern struct snd_soc_dai wm8350_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8350;
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 510efa604008..502eefac1ecd 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -954,7 +954,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
                 factors->outdiv *= 2;
                 if (factors->outdiv > 32) {
                         dev_err(wm8400->wm8400->dev,
-                                "Unsupported FLL output frequency %dHz\n",
+                                "Unsupported FLL output frequency %uHz\n",
                                 Fout);
                         return -EINVAL;
                 }
@@ -1003,7 +1003,7 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
         factors->k = K / 10;
 
         dev_dbg(wm8400->wm8400->dev,
-                "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
+                "FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
                 Fref, Fout,
                 factors->n, factors->k, factors->fratio, factors->outdiv);
 
@@ -1473,8 +1473,8 @@ static int wm8400_codec_probe(struct platform_device *dev)
 
         codec = &priv->codec;
         codec->private_data = priv;
-        codec->control_data = dev->dev.driver_data;
-        priv->wm8400 = dev->dev.driver_data;
+        codec->control_data = dev_get_drvdata(&dev->dev);
+        priv->wm8400 = dev_get_drvdata(&dev->dev);
 
         ret = regulator_bulk_get(priv->wm8400->dev,
                                  ARRAY_SIZE(power), &power[0]);
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 6a4cea09c45d..c8b8dba85890 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -298,7 +298,7 @@ static void pll_factors(unsigned int target, unsigned int source)
 
         if ((Ndiv < 6) || (Ndiv > 12))
                 printk(KERN_WARNING
-                        "WM8510 N value %d outwith recommended range!d\n",
+                        "WM8510 N value %u outwith recommended range!d\n",
                         Ndiv);
 
         pll_div.n = Ndiv;
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 442ea6f160fc..86c4b24db817 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static int wm8580_out_vu(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);
-        int reg = kcontrol->private_value & 0xff;
-        int reg2 = (kcontrol->private_value >> 24) & 0xff;
+        unsigned int reg = mc->reg;
+        unsigned int reg2 = mc->rreg;
         int ret;
         u16 val;
 
@@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
         return 0;
 }
 
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+                                    xinvert, tlv_array)                 \
 {       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
         .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
                 SNDRV_CTL_ELEM_ACCESS_READWRITE,  \
         .tlv.p = (tlv_array), \
         .info = snd_soc_info_volsw_2r, \
         .get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
-        .private_value = (reg_left) | ((shift) << 8)  |         \
-                ((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+        .private_value = (unsigned long)&(struct soc_mixer_control) \
+                {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+                .max = xmax, .invert = xinvert} }
 
 static const struct snd_kcontrol_new wm8580_snd_controls[] = {
 SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
@@ -411,7 +415,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target,
         unsigned int K, Ndiv, Nmod;
         int i;
 
-        pr_debug("wm8580: PLL %dHz->%dHz\n", source, target);
+        pr_debug("wm8580: PLL %uHz->%uHz\n", source, target);
 
         /* Scale the output frequency up; the PLL should run in the
          * region of 90-100MHz.
@@ -443,7 +447,7 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target,
 
         if ((Ndiv < 5) || (Ndiv > 13)) {
                 printk(KERN_ERR
-                        "WM8580 N=%d outside supported range\n", Ndiv);
+                        "WM8580 N=%u outside supported range\n", Ndiv);
                 return -EINVAL;
         }
 
@@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
         reg = wm8580_read(codec, WM8580_PLLA4 + offset);
         reg &= ~0x3f;
         reg |= pll_div.prescale | pll_div.postscale << 1 |
-                pll_div.freqmode << 4;
+                pll_div.freqmode << 3;
 
         wm8580_write(codec, WM8580_PLLA4 + offset, reg);
 
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index e043e3f60008..7a205876ef4f 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -666,14 +666,14 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi)
         codec->hw_write = (hw_write_t)wm8731_spi_write;
         codec->dev = &spi->dev;
 
-        spi->dev.driver_data = wm8731;
+        dev_set_drvdata(&spi->dev, wm8731);
 
         return wm8731_register(wm8731);
 }
 
 static int __devexit wm8731_spi_remove(struct spi_device *spi)
 {
-        struct wm8731_priv *wm8731 = spi->dev.driver_data;
+        struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev);
 
         wm8731_unregister(wm8731);
 
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a6e8f3f7f052..d28eeaceb857 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -703,7 +703,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 
         if ((Ndiv < 6) || (Ndiv > 12))
                 printk(KERN_WARNING
-                        "wm8753: unsupported N = %d\n", Ndiv);
+                        "wm8753: unsupported N = %u\n", Ndiv);
 
         pll_div->n = Ndiv;
         Nmod = target % source;
@@ -1822,14 +1822,14 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi)
         codec->hw_write = (hw_write_t)wm8753_spi_write;
         codec->dev = &spi->dev;
 
-        spi->dev.driver_data = wm8753;
+        dev_set_drvdata(&spi->dev, wm8753);
 
         return wm8753_register(wm8753);
 }
 
 static int __devexit wm8753_spi_remove(struct spi_device *spi)
 {
-        struct wm8753_priv *wm8753 = spi->dev.driver_data;
+        struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev);
         wm8753_unregister(wm8753);
         return 0;
 }
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 46c5ea1ff921..3c78945244b8 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -778,11 +778,11 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
         }
 
         if (target > 100000000)
-                printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d"
-                       " Fout=%d\n", target, Fref, Fout);
+                printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u"
+                       " Fout=%u\n", target, Fref, Fout);
         if (div > 32) {
                 printk(KERN_ERR "wm8900: Invalid FLL division rate %u, "
-                       "Fref=%d, Fout=%d, target=%d\n",
+                       "Fref=%u, Fout=%u, target=%u\n",
                        div, Fref, Fout, target);
                 return -EINVAL;
         }
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 8cf571f1a803..e8d2e3e14c45 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -217,7 +217,6 @@ struct wm8903_priv {
         int sysclk;
 
         /* Reference counts */
-        int charge_pump_users;
         int class_w_users;
         int playback_active;
         int capture_active;
@@ -373,6 +372,15 @@ static void wm8903_reset(struct snd_soc_codec *codec)
 #define WM8903_OUTPUT_INT   0x2
 #define WM8903_OUTPUT_IN    0x1
 
+static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
+                           struct snd_kcontrol *kcontrol, int event)
+{
+        WARN_ON(event != SND_SOC_DAPM_POST_PMU);
+        mdelay(4);
+
+        return 0;
+}
+
 /*
  * Event for headphone and line out amplifier power changes.  Special
  * power up/down sequences are required in order to maximise pop/click
@@ -382,19 +390,20 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
                                struct snd_kcontrol *kcontrol, int event)
 {
         struct snd_soc_codec *codec = w->codec;
-        struct wm8903_priv *wm8903 = codec->private_data;
-        struct i2c_client *i2c = codec->control_data;
         u16 val;
         u16 reg;
+        u16 dcs_reg;
+        u16 dcs_bit;
         int shift;
-        u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0);
 
         switch (w->reg) {
         case WM8903_POWER_MANAGEMENT_2:
                 reg = WM8903_ANALOGUE_HP_0;
+                dcs_bit = 0 + w->shift;
                 break;
         case WM8903_POWER_MANAGEMENT_3:
                 reg = WM8903_ANALOGUE_LINEOUT_0;
+                dcs_bit = 2 + w->shift;
                 break;
         default:
                 BUG();
@@ -419,18 +428,6 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
                 /* Short the output */
                 val &= ~(WM8903_OUTPUT_SHORT << shift);
                 wm8903_write(codec, reg, val);
-
-                wm8903->charge_pump_users++;
-
-                dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
-                        wm8903->charge_pump_users);
-
-                if (wm8903->charge_pump_users == 1) {
-                        dev_dbg(&i2c->dev, "Enabling charge pump\n");
-                        wm8903_write(codec, WM8903_CHARGE_PUMP_0,
-                                     cp_reg | WM8903_CP_ENA);
-                        mdelay(4);
-                }
         }
 
         if (event & SND_SOC_DAPM_POST_PMU) {
@@ -446,6 +443,11 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
                 val |= (WM8903_OUTPUT_OUT << shift);
                 wm8903_write(codec, reg, val);
 
+                /* Enable the DC servo */
+                dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+                dcs_reg |= dcs_bit;
+                wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+
                 /* Remove the short */
                 val |= (WM8903_OUTPUT_SHORT << shift);
                 wm8903_write(codec, reg, val);
@@ -458,25 +460,17 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w,
                 val &= ~(WM8903_OUTPUT_SHORT << shift);
                 wm8903_write(codec, reg, val);
 
+                /* Disable the DC servo */
+                dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+                dcs_reg &= ~dcs_bit;
+                wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+
                 /* Then disable the intermediate and output stages */
                 val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
                           WM8903_OUTPUT_IN) << shift);
                 wm8903_write(codec, reg, val);
         }
 
-        if (event & SND_SOC_DAPM_POST_PMD) {
-                wm8903->charge_pump_users--;
-
-                dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
-                        wm8903->charge_pump_users);
-
-                if (wm8903->charge_pump_users == 0) {
-                        dev_dbg(&i2c->dev, "Disabling charge pump\n");
-                        wm8903_write(codec, WM8903_CHARGE_PUMP_0,
-                                     cp_reg & ~WM8903_CP_ENA);
-                }
-        }
-
         return 0;
 }
 
@@ -539,6 +533,7 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
 /* ALSA can only do steps of .01dB */
 static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
 
+static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
 
 static const DECLARE_TLV_DB_SCALE(drc_tlv_thresh, 0, 75, 0);
@@ -657,6 +652,16 @@ static const struct soc_enum rinput_inv_enum =
         SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text);
 
 
+static const char *sidetone_text[] = {
+        "None", "Left", "Right"
+};
+
+static const struct soc_enum lsidetone_enum =
+        SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text);
+
+static const struct soc_enum rsidetone_enum =
+        SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
+
 static const struct snd_kcontrol_new wm8903_snd_controls[] = {
 
 /* Input PGAs - No TLV since the scale depends on PGA mode */
@@ -700,6 +705,9 @@ SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
 SOC_ENUM("ADC Companding Mode", adc_companding),
 SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
 
+SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8,
+               12, 0, digital_sidetone_tlv),
+
 /* DAC */
 SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT,
                  WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
@@ -762,6 +770,12 @@ static const struct snd_kcontrol_new rinput_mux =
 static const struct snd_kcontrol_new rinput_inv_mux =
         SOC_DAPM_ENUM("Right Inverting Input Mux", rinput_inv_enum);
 
+static const struct snd_kcontrol_new lsidetone_mux =
+        SOC_DAPM_ENUM("DACL Sidetone Mux", lsidetone_enum);
+
+static const struct snd_kcontrol_new rsidetone_mux =
+        SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum);
+
 static const struct snd_kcontrol_new left_output_mixer[] = {
 SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
@@ -828,6 +842,9 @@ SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0),
 SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
 SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
 
+SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux),
+SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux),
+
 SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
 SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
 
@@ -844,26 +861,29 @@ SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
 SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
                    1, 0, NULL, 0, wm8903_output_event,
                    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-                   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+                   SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
                    0, 0, NULL, 0, wm8903_output_event,
                    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-                   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+                   SND_SOC_DAPM_PRE_PMD),
 
 SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
                    NULL, 0, wm8903_output_event,
                    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-                   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+                   SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
                    NULL, 0, wm8903_output_event,
                    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-                   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+                   SND_SOC_DAPM_PRE_PMD),
 
 SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
                  NULL, 0),
 SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0,
                  NULL, 0),
 
+SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0,
+                    wm8903_cp_event, SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0),
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
@@ -909,7 +929,19 @@ static const struct snd_soc_dapm_route intercon[] = {
         { "Right Input PGA", NULL, "Right Input Mode Mux" },
 
         { "ADCL", NULL, "Left Input PGA" },
+        { "ADCL", NULL, "CLK_DSP" },
         { "ADCR", NULL, "Right Input PGA" },
+        { "ADCR", NULL, "CLK_DSP" },
+
+        { "DACL Sidetone", "Left", "ADCL" },
+        { "DACL Sidetone", "Right", "ADCR" },
+        { "DACR Sidetone", "Left", "ADCL" },
+        { "DACR Sidetone", "Right", "ADCR" },
+
+        { "DACL", NULL, "DACL Sidetone" },
+        { "DACL", NULL, "CLK_DSP" },
+        { "DACR", NULL, "DACR Sidetone" },
+        { "DACR", NULL, "CLK_DSP" },
 
         { "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" },
         { "Left Output Mixer", "Right Bypass Switch", "Right Input PGA" },
@@ -951,6 +983,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 
         { "ROP", NULL, "Right Speaker PGA" },
         { "RON", NULL, "Right Speaker PGA" },
+
+        { "Left Headphone Output PGA", NULL, "Charge Pump" },
+        { "Right Headphone Output PGA", NULL, "Charge Pump" },
+        { "Left Line Output PGA", NULL, "Charge Pump" },
+        { "Right Line Output PGA", NULL, "Charge Pump" },
 };
 
 static int wm8903_add_widgets(struct snd_soc_codec *codec)
@@ -985,6 +1022,11 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec,
                         wm8903_write(codec, WM8903_CLOCK_RATES_2,
                                      WM8903_CLK_SYS_ENA);
 
+                        /* Change DC servo dither level in startup sequence */
+                        wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
+                        wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
+                        wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
+
                         wm8903_run_sequence(codec, 0);
                         wm8903_sync_reg_cache(codec, codec->reg_cache);
 
@@ -1215,22 +1257,18 @@ static struct {
         int div;
 } bclk_divs[] = {
         {  10,  0 },
-        {  15,  1 },
         {  20,  2 },
         {  30,  3 },
         {  40,  4 },
         {  50,  5 },
-        {  55,  6 },
         {  60,  7 },
         {  80,  8 },
         { 100,  9 },
-        { 110, 10 },
         { 120, 11 },
         { 160, 12 },
         { 200, 13 },
         { 220, 14 },
         { 240, 15 },
-        { 250, 16 },
         { 300, 17 },
         { 320, 18 },
         { 440, 19 },
@@ -1277,14 +1315,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream,
         if (wm8903->master_substream) {
                 master_runtime = wm8903->master_substream->runtime;
 
-                dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
-                        master_runtime->sample_bits,
-                        master_runtime->rate);
-
-                snd_pcm_hw_constraint_minmax(substream->runtime,
-                                             SNDRV_PCM_HW_PARAM_RATE,
-                                             master_runtime->rate,
-                                             master_runtime->rate);
+                dev_dbg(&i2c->dev, "Constraining to %d bits\n",
+                        master_runtime->sample_bits);
 
                 snd_pcm_hw_constraint_minmax(substream->runtime,
                                              SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
@@ -1523,6 +1555,7 @@ struct snd_soc_dai wm8903_dai = {
                  .formats = WM8903_FORMATS,
          },
         .ops = &wm8903_dai_ops,
+        .symmetric_rates = 1,
 };
 EXPORT_SYMBOL_GPL(wm8903_dai);
 
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
new file mode 100644
index 000000000000..b8e17d6bc1f7
--- /dev/null
+++ b/sound/soc/codecs/wm8940.c
@@ -0,0 +1,955 @@
+/*
+ * wm8940.c  --  WM8940 ALSA Soc Audio driver
+ *
+ * Author: Jonathan Cameron <jic23@cam.ac.uk>
+ *
+ * Based on wm8510.c
+ *    Copyright  2006 Wolfson Microelectronics PLC.
+ *    Author:  Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * Not currently handled:
+ * Notch filter control
+ * AUXMode (inverting vs mixer)
+ * No means to obtain current gain if alc enabled.
+ * No use made of gpio
+ * Fast VMID discharge for power down
+ * Soft Start
+ * DLR and ALR Swaps not enabled
+ * Digital Sidetone not supported
+ */
+#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/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8940.h"
+
+struct wm8940_priv {
+        unsigned int sysclk;
+        u16 reg_cache[WM8940_CACHEREGNUM];
+        struct snd_soc_codec codec;
+};
+
+static u16 wm8940_reg_defaults[] = {
+        0x8940, /* Soft Reset */
+        0x0000, /* Power 1 */
+        0x0000, /* Power 2 */
+        0x0000, /* Power 3 */
+        0x0010, /* Interface Control */
+        0x0000, /* Companding Control */
+        0x0140, /* Clock Control */
+        0x0000, /* Additional Controls */
+        0x0000, /* GPIO Control */
+        0x0002, /* Auto Increment Control */
+        0x0000, /* DAC Control */
+        0x00FF, /* DAC Volume */
+        0,
+        0,
+        0x0100, /* ADC Control */
+        0x00FF, /* ADC Volume */
+        0x0000, /* Notch Filter 1 Control 1 */
+        0x0000, /* Notch Filter 1 Control 2 */
+        0x0000, /* Notch Filter 2 Control 1 */
+        0x0000, /* Notch Filter 2 Control 2 */
+        0x0000, /* Notch Filter 3 Control 1 */
+        0x0000, /* Notch Filter 3 Control 2 */
+        0x0000, /* Notch Filter 4 Control 1 */
+        0x0000, /* Notch Filter 4 Control 2 */
+        0x0032, /* DAC Limit Control 1 */
+        0x0000, /* DAC Limit Control 2 */
+        0,
+        0,
+        0,
+        0,
+        0,
+        0,
+        0x0038, /* ALC Control 1 */
+        0x000B, /* ALC Control 2 */
+        0x0032, /* ALC Control 3 */
+        0x0000, /* Noise Gate */
+        0x0041, /* PLLN */
+        0x000C, /* PLLK1 */
+        0x0093, /* PLLK2 */
+        0x00E9, /* PLLK3 */
+        0,
+        0,
+        0x0030, /* ALC Control 4 */
+        0,
+        0x0002, /* Input Control */
+        0x0050, /* PGA Gain */
+        0,
+        0x0002, /* ADC Boost Control */
+        0,
+        0x0002, /* Output Control */
+        0x0000, /* Speaker Mixer Control */
+        0,
+        0,
+        0,
+        0x0079, /* Speaker Volume */
+        0,
+        0x0000, /* Mono Mixer Control */
+};
+
+static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec,
+                                                 unsigned int reg)
+{
+        u16 *cache = codec->reg_cache;
+
+        if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+                return -1;
+
+        return cache[reg];
+}
+
+static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec,
+                                          u16 reg, unsigned int value)
+{
+        u16 *cache = codec->reg_cache;
+
+        if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+                return -1;
+
+        cache[reg] = value;
+
+        return 0;
+}
+
+static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg,
+                        unsigned int value)
+{
+        int ret;
+        u8 data[3] = { reg,
+                       (value & 0xff00) >> 8,
+                       (value & 0x00ff)
+        };
+
+        wm8940_write_reg_cache(codec, reg, value);
+
+        ret = codec->hw_write(codec->control_data, data, 3);
+
+        if (ret < 0)
+                return ret;
+        else if (ret != 3)
+                return -EIO;
+        return 0;
+}
+
+static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
+static const struct soc_enum wm8940_adc_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding);
+static const struct soc_enum wm8940_dac_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding);
+
+static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"};
+static const struct soc_enum wm8940_alc_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text);
+
+static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"};
+static const struct soc_enum wm8940_mic_bias_level_enum
+= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text);
+
+static const char *wm8940_filter_mode_text[] = {"Audio", "Application"};
+static const struct soc_enum wm8940_filter_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text);
+
+static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0);
+
+static const struct snd_kcontrol_new wm8940_snd_controls[] = {
+        SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL,
+                   6, 1, 0),
+        SOC_ENUM("DAC Companding", wm8940_dac_companding_enum),
+        SOC_ENUM("ADC Companding", wm8940_adc_companding_enum),
+
+        SOC_ENUM("ALC Mode", wm8940_alc_mode_enum),
+        SOC_SINGLE("ALC Switch", WM8940_ALC1, 8, 1, 0),
+        SOC_SINGLE_TLV("ALC Capture Max Gain", WM8940_ALC1,
+                       3, 7, 1, wm8940_alc_max_tlv),
+        SOC_SINGLE_TLV("ALC Capture Min Gain", WM8940_ALC1,
+                       0, 7, 0, wm8940_alc_min_tlv),
+        SOC_SINGLE_TLV("ALC Capture Target", WM8940_ALC2,
+                       0, 14, 0, wm8940_alc_tar_tlv),
+        SOC_SINGLE("ALC Capture Hold", WM8940_ALC2, 4, 10, 0),
+        SOC_SINGLE("ALC Capture Decay", WM8940_ALC3, 4, 10, 0),
+        SOC_SINGLE("ALC Capture Attach", WM8940_ALC3, 0, 10, 0),
+        SOC_SINGLE("ALC ZC Switch", WM8940_ALC4, 1, 1, 0),
+        SOC_SINGLE("ALC Capture Noise Gate Switch", WM8940_NOISEGATE,
+                   3, 1, 0),
+        SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8940_NOISEGATE,
+                   0, 7, 0),
+
+        SOC_SINGLE("DAC Playback Limiter Switch", WM8940_DACLIM1, 8, 1, 0),
+        SOC_SINGLE("DAC Playback Limiter Attack", WM8940_DACLIM1, 0, 9, 0),
+        SOC_SINGLE("DAC Playback Limiter Decay", WM8940_DACLIM1, 4, 11, 0),
+        SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8940_DACLIM2,
+                       4, 9, 1, wm8940_lim_thresh_tlv),
+        SOC_SINGLE_TLV("DAC Playback Limiter Boost", WM8940_DACLIM2,
+                       0, 12, 0, wm8940_lim_boost_tlv),
+
+        SOC_SINGLE("Capture PGA ZC Switch", WM8940_PGAGAIN, 7, 1, 0),
+        SOC_SINGLE_TLV("Capture PGA Volume", WM8940_PGAGAIN,
+                       0, 63, 0, wm8940_pga_vol_tlv),
+        SOC_SINGLE_TLV("Digital Playback Volume", WM8940_DACVOL,
+                       0, 255, 0, wm8940_adc_tlv),
+        SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL,
+                       0, 255, 0, wm8940_adc_tlv),
+        SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum),
+        SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST,
+                       8, 1, 0, wm8940_capture_boost_vol_tlv),
+        SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL,
+                       0, 63, 0, wm8940_spk_vol_tlv),
+        SOC_SINGLE("Speaker Playback Switch", WM8940_SPKVOL,  6, 1, 1),
+
+        SOC_SINGLE_TLV("Speaker Mixer Line Bypass Volume", WM8940_SPKVOL,
+                       8, 1, 1, wm8940_att_tlv),
+        SOC_SINGLE("Speaker Playback ZC Switch", WM8940_SPKVOL, 7, 1, 0),
+
+        SOC_SINGLE("Mono Out Switch", WM8940_MONOMIX, 6, 1, 1),
+        SOC_SINGLE_TLV("Mono Mixer Line Bypass Volume", WM8940_MONOMIX,
+                       7, 1, 1, wm8940_att_tlv),
+
+        SOC_SINGLE("High Pass Filter Switch", WM8940_ADC, 8, 1, 0),
+        SOC_ENUM("High Pass Filter Mode", wm8940_filter_mode_enum),
+        SOC_SINGLE("High Pass Filter Cut Off", WM8940_ADC, 4, 7, 0),
+        SOC_SINGLE("ADC Inversion Switch", WM8940_ADC, 0, 1, 0),
+        SOC_SINGLE("DAC Inversion Switch", WM8940_DAC, 0, 1, 0),
+        SOC_SINGLE("DAC Auto Mute Switch", WM8940_DAC, 2, 1, 0),
+        SOC_SINGLE("ZC Timeout Clock Switch", WM8940_ADDCNTRL, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_speaker_mixer_controls[] = {
+        SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_SPKMIX, 1, 1, 0),
+        SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_SPKMIX, 5, 1, 0),
+        SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_SPKMIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = {
+        SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_MONOMIX, 1, 1, 0),
+        SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_MONOMIX, 2, 1, 0),
+        SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1);
+static const struct snd_kcontrol_new wm8940_input_boost_controls[] = {
+        SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1),
+        SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST,
+                            0, 7, 0, wm8940_boost_vol_tlv),
+        SOC_DAPM_SINGLE_TLV("Mic Volume", WM8940_ADCBOOST,
+                            4, 7, 0, wm8940_boost_vol_tlv),
+};
+
+static const struct snd_kcontrol_new wm8940_micpga_controls[] = {
+        SOC_DAPM_SINGLE("AUX Switch", WM8940_INPUTCTL, 2, 1, 0),
+        SOC_DAPM_SINGLE("MICP Switch", WM8940_INPUTCTL, 0, 1, 0),
+        SOC_DAPM_SINGLE("MICN Switch", WM8940_INPUTCTL, 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = {
+        SND_SOC_DAPM_MIXER("Speaker Mixer", WM8940_POWER3, 2, 0,
+                           &wm8940_speaker_mixer_controls[0],
+                           ARRAY_SIZE(wm8940_speaker_mixer_controls)),
+        SND_SOC_DAPM_MIXER("Mono Mixer", WM8940_POWER3, 3, 0,
+                           &wm8940_mono_mixer_controls[0],
+                           ARRAY_SIZE(wm8940_mono_mixer_controls)),
+        SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8940_POWER3, 0, 0),
+
+        SND_SOC_DAPM_PGA("SpkN Out", WM8940_POWER3, 5, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("SpkP Out", WM8940_POWER3, 6, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("Mono Out", WM8940_POWER3, 7, 0, NULL, 0),
+        SND_SOC_DAPM_OUTPUT("MONOOUT"),
+        SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+        SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+
+        SND_SOC_DAPM_PGA("Aux Input", WM8940_POWER1, 6, 0, NULL, 0),
+        SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8940_POWER2, 0, 0),
+        SND_SOC_DAPM_MIXER("Mic PGA", WM8940_POWER2, 2, 0,
+                           &wm8940_micpga_controls[0],
+                           ARRAY_SIZE(wm8940_micpga_controls)),
+        SND_SOC_DAPM_MIXER("Boost Mixer", WM8940_POWER2, 4, 0,
+                           &wm8940_input_boost_controls[0],
+                           ARRAY_SIZE(wm8940_input_boost_controls)),
+        SND_SOC_DAPM_MICBIAS("Mic Bias", WM8940_POWER1, 4, 0),
+
+        SND_SOC_DAPM_INPUT("MICN"),
+        SND_SOC_DAPM_INPUT("MICP"),
+        SND_SOC_DAPM_INPUT("AUX"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+        /* Mono output mixer */
+        {"Mono Mixer", "PCM Playback Switch", "DAC"},
+        {"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+        {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+        /* Speaker output mixer */
+        {"Speaker Mixer", "PCM Playback Switch", "DAC"},
+        {"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+        {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+        /* Outputs */
+        {"Mono Out", NULL, "Mono Mixer"},
+        {"MONOOUT", NULL, "Mono Out"},
+        {"SpkN Out", NULL, "Speaker Mixer"},
+        {"SpkP Out", NULL, "Speaker Mixer"},
+        {"SPKOUTN", NULL, "SpkN Out"},
+        {"SPKOUTP", NULL, "SpkP Out"},
+
+        /*  Microphone PGA */
+        {"Mic PGA", "MICN Switch", "MICN"},
+        {"Mic PGA", "MICP Switch", "MICP"},
+        {"Mic PGA", "AUX Switch", "AUX"},
+
+        /* Boost Mixer */
+        {"Boost Mixer", "Mic PGA Switch", "Mic PGA"},
+        {"Boost Mixer", "Mic Volume",  "MICP"},
+        {"Boost Mixer", "Aux Volume", "Aux Input"},
+
+        {"ADC", NULL, "Boost Mixer"},
+};
+
+static int wm8940_add_widgets(struct snd_soc_codec *codec)
+{
+        int ret;
+
+        ret = snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets,
+                                        ARRAY_SIZE(wm8940_dapm_widgets));
+        if (ret)
+                goto error_ret;
+        ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+        if (ret)
+                goto error_ret;
+        ret = snd_soc_dapm_new_widgets(codec);
+
+error_ret:
+        return ret;
+}
+
+#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0);
+
+static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
+                              unsigned int fmt)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67;
+        u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe;
+
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBM_CFM:
+                clk |= 1;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFS:
+                break;
+        default:
+                return -EINVAL;
+        }
+        wm8940_write(codec, WM8940_CLOCK, clk);
+
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_I2S:
+                iface |= (2 << 3);
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                iface |= (1 << 3);
+                break;
+        case SND_SOC_DAIFMT_RIGHT_J:
+                break;
+        case SND_SOC_DAIFMT_DSP_A:
+                iface |= (3 << 3);
+                break;
+        case SND_SOC_DAIFMT_DSP_B:
+                iface |= (3 << 3) | (1 << 7);
+                break;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                break;
+        case SND_SOC_DAIFMT_NB_IF:
+                iface |= (1 << 7);
+                break;
+        case SND_SOC_DAIFMT_IB_NF:
+                iface |= (1 << 8);
+                break;
+        case SND_SOC_DAIFMT_IB_IF:
+                iface |= (1 << 8) | (1 << 7);
+                break;
+        }
+
+        wm8940_write(codec, WM8940_IFACE, iface);
+
+        return 0;
+}
+
+static int wm8940_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_device *socdev = rtd->socdev;
+        struct snd_soc_codec *codec = socdev->card->codec;
+        u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F;
+        u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1;
+        u16 companding =  wm8940_read_reg_cache(codec,
+                                                WM8940_COMPANDINGCTL) & 0xFFDF;
+        int ret;
+
+        /* LoutR control */
+        if (substream->stream == SNDRV_PCM_STREAM_CAPTURE
+            && params_channels(params) == 2)
+                iface |= (1 << 9);
+
+        switch (params_rate(params)) {
+        case SNDRV_PCM_RATE_8000:
+                addcntrl |= (0x5 << 1);
+                break;
+        case SNDRV_PCM_RATE_11025:
+                addcntrl |= (0x4 << 1);
+                break;
+        case SNDRV_PCM_RATE_16000:
+                addcntrl |= (0x3 << 1);
+                break;
+        case SNDRV_PCM_RATE_22050:
+                addcntrl |= (0x2 << 1);
+                break;
+        case SNDRV_PCM_RATE_32000:
+                addcntrl |= (0x1 << 1);
+                break;
+        case SNDRV_PCM_RATE_44100:
+        case SNDRV_PCM_RATE_48000:
+                break;
+        }
+        ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl);
+        if (ret)
+                goto error_ret;
+
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S8:
+                companding = companding | (1 << 5);
+                break;
+        case SNDRV_PCM_FORMAT_S16_LE:
+                break;
+        case SNDRV_PCM_FORMAT_S20_3LE:
+                iface |= (1 << 5);
+                break;
+        case SNDRV_PCM_FORMAT_S24_LE:
+                iface |= (2 << 5);
+                break;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                iface |= (3 << 5);
+                break;
+        }
+        ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding);
+        if (ret)
+                goto error_ret;
+        ret = wm8940_write(codec, WM8940_IFACE, iface);
+
+error_ret:
+        return ret;
+}
+
+static int wm8940_mute(struct snd_soc_dai *dai, int mute)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf;
+
+        if (mute)
+                mute_reg |= 0x40;
+
+        return wm8940_write(codec, WM8940_DAC, mute_reg);
+}
+
+static int wm8940_set_bias_level(struct snd_soc_codec *codec,
+                                 enum snd_soc_bias_level level)
+{
+        u16 val;
+        u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0;
+        int ret = 0;
+
+        switch (level) {
+        case SND_SOC_BIAS_ON:
+                /* ensure bufioen and biasen */
+                pwr_reg |= (1 << 2) | (1 << 3);
+                /* Enable thermal shutdown */
+                val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+                ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2);
+                if (ret)
+                        break;
+                /* set vmid to 75k */
+                ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+                break;
+        case SND_SOC_BIAS_PREPARE:
+                /* ensure bufioen and biasen */
+                pwr_reg |= (1 << 2) | (1 << 3);
+                ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+                break;
+        case SND_SOC_BIAS_STANDBY:
+                /* ensure bufioen and biasen */
+                pwr_reg |= (1 << 2) | (1 << 3);
+                /* set vmid to 300k for standby */
+                ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2);
+                break;
+        case SND_SOC_BIAS_OFF:
+                ret = wm8940_write(codec, WM8940_POWER1, pwr_reg);
+                break;
+        }
+
+        return ret;
+}
+
+struct pll_ {
+        unsigned int pre_scale:2;
+        unsigned int n:4;
+        unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+static void pll_factors(unsigned int target, unsigned int source)
+{
+        unsigned long long Kpart;
+        unsigned int K, Ndiv, Nmod;
+        /* The left shift ist to avoid accuracy loss when right shifting */
+        Ndiv = target / source;
+
+        if (Ndiv > 12) {
+                source <<= 1;
+                /* Multiply by 2 */
+                pll_div.pre_scale = 0;
+                Ndiv = target / source;
+        } else if (Ndiv < 3) {
+                source >>= 2;
+                /* Divide by 4 */
+                pll_div.pre_scale = 3;
+                Ndiv = target / source;
+        } else if (Ndiv < 6) {
+                source >>= 1;
+                /* divide by 2 */
+                pll_div.pre_scale = 2;
+                Ndiv = target / source;
+        } else
+                pll_div.pre_scale = 1;
+
+        if ((Ndiv < 6) || (Ndiv > 12))
+                printk(KERN_WARNING
+                        "WM8940 N value %d outwith recommended range!d\n",
+                        Ndiv);
+
+        pll_div.n = Ndiv;
+        Nmod = target % source;
+        Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+        do_div(Kpart, source);
+
+        K = Kpart & 0xFFFFFFFF;
+
+        /* Check if we need to round */
+        if ((K % 10) >= 5)
+                K += 5;
+
+        /* Move down to proper range now rounding is done */
+        K /= 10;
+
+        pll_div.k = K;
+}
+
+/* Untested at the moment */
+static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai,
+                int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u16 reg;
+
+        /* Turn off PLL */
+        reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+        wm8940_write(codec, WM8940_POWER1, reg & 0x1df);
+
+        if (freq_in == 0 || freq_out == 0) {
+                /* Clock CODEC directly from MCLK */
+                reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+                wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff);
+                /* Pll power down */
+                wm8940_write(codec, WM8940_PLLN, (1 << 7));
+                return 0;
+        }
+
+        /* Pll is followed by a frequency divide by 4 */
+        pll_factors(freq_out*4, freq_in);
+        if (pll_div.k)
+                wm8940_write(codec, WM8940_PLLN,
+                             (pll_div.pre_scale << 4) | pll_div.n | (1 << 6));
+        else /* No factional component */
+                wm8940_write(codec, WM8940_PLLN,
+                             (pll_div.pre_scale << 4) | pll_div.n);
+        wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18);
+        wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff);
+        wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff);
+        /* Enable the PLL */
+        reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+        wm8940_write(codec, WM8940_POWER1, reg | 0x020);
+
+        /* Run CODEC from PLL instead of MCLK */
+        reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+        wm8940_write(codec, WM8940_CLOCK, reg | 0x100);
+
+        return 0;
+}
+
+static int wm8940_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 wm8940_priv *wm8940 = codec->private_data;
+
+        switch (freq) {
+        case 11289600:
+        case 12000000:
+        case 12288000:
+        case 16934400:
+        case 18432000:
+                wm8940->sysclk = freq;
+                return 0;
+        }
+        return -EINVAL;
+}
+
+static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+                                 int div_id, int div)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u16 reg;
+        int ret = 0;
+
+        switch (div_id) {
+        case WM8940_BCLKDIV:
+                reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3;
+                ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2));
+                break;
+        case WM8940_MCLKDIV:
+                reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F;
+                ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5));
+                break;
+        case WM8940_OPCLKDIV:
+                reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF;
+                ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4));
+                break;
+        }
+        return ret;
+}
+
+#define WM8940_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S8 |                           \
+                        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 wm8940_dai_ops = {
+        .hw_params = wm8940_i2s_hw_params,
+        .set_sysclk = wm8940_set_dai_sysclk,
+        .digital_mute = wm8940_mute,
+        .set_fmt = wm8940_set_dai_fmt,
+        .set_clkdiv = wm8940_set_dai_clkdiv,
+        .set_pll = wm8940_set_dai_pll,
+};
+
+struct snd_soc_dai wm8940_dai = {
+        .name = "WM8940",
+        .playback = {
+                .stream_name = "Playback",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = WM8940_RATES,
+                .formats = WM8940_FORMATS,
+        },
+        .capture = {
+                .stream_name = "Capture",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = WM8940_RATES,
+                .formats = WM8940_FORMATS,
+        },
+        .ops = &wm8940_dai_ops,
+        .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8940_dai);
+
+static int wm8940_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 wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8940_resume(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+        int i;
+        int ret;
+        u8 data[3];
+        u16 *cache = codec->reg_cache;
+
+        /* Sync reg_cache with the hardware
+         * Could use auto incremented writes to speed this up
+         */
+        for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) {
+                data[0] = i;
+                data[1] = (cache[i] & 0xFF00) >> 8;
+                data[2] = cache[i] & 0x00FF;
+                ret = codec->hw_write(codec->control_data, data, 3);
+                if (ret < 0)
+                        goto error_ret;
+                else if (ret != 3) {
+                        ret = -EIO;
+                        goto error_ret;
+                }
+        }
+        ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+        if (ret)
+                goto error_ret;
+        ret = wm8940_set_bias_level(codec, codec->suspend_bias_level);
+
+error_ret:
+        return ret;
+}
+
+static struct snd_soc_codec *wm8940_codec;
+
+static int wm8940_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec;
+
+        int ret = 0;
+
+        if (wm8940_codec == NULL) {
+                dev_err(&pdev->dev, "Codec device not registered\n");
+                return -ENODEV;
+        }
+
+        socdev->card->codec = wm8940_codec;
+        codec = wm8940_codec;
+
+        mutex_init(&codec->mutex);
+        /* 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;
+        }
+
+        ret = snd_soc_add_controls(codec, wm8940_snd_controls,
+                             ARRAY_SIZE(wm8940_snd_controls));
+        if (ret)
+                goto error_free_pcms;
+        ret = wm8940_add_widgets(codec);
+        if (ret)
+                goto error_free_pcms;
+
+        ret = snd_soc_init_card(socdev);
+        if (ret < 0) {
+                dev_err(codec->dev, "failed to register card: %d\n", ret);
+                goto error_free_pcms;
+        }
+
+        return ret;
+
+error_free_pcms:
+        snd_soc_free_pcms(socdev);
+        snd_soc_dapm_free(socdev);
+pcm_err:
+        return ret;
+}
+
+static int wm8940_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_wm8940 = {
+        .probe = wm8940_probe,
+        .remove = wm8940_remove,
+        .suspend = wm8940_suspend,
+        .resume = wm8940_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940);
+
+static int wm8940_register(struct wm8940_priv *wm8940)
+{
+        struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data;
+        struct snd_soc_codec *codec = &wm8940->codec;
+        int ret;
+        u16 reg;
+        if (wm8940_codec) {
+                dev_err(codec->dev, "Another WM8940 is registered\n");
+                return -EINVAL;
+        }
+
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        codec->private_data = wm8940;
+        codec->name = "WM8940";
+        codec->owner = THIS_MODULE;
+        codec->read = wm8940_read_reg_cache;
+        codec->write = wm8940_write;
+        codec->bias_level = SND_SOC_BIAS_OFF;
+        codec->set_bias_level = wm8940_set_bias_level;
+        codec->dai = &wm8940_dai;
+        codec->num_dai = 1;
+        codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults);
+        codec->reg_cache = &wm8940->reg_cache;
+
+        memcpy(codec->reg_cache, wm8940_reg_defaults,
+               sizeof(wm8940_reg_defaults));
+
+        ret = wm8940_reset(codec);
+        if (ret < 0) {
+                dev_err(codec->dev, "Failed to issue reset\n");
+                return ret;
+        }
+
+        wm8940_dai.dev = codec->dev;
+
+        wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+        ret = wm8940_write(codec, WM8940_POWER1, 0x180);
+        if (ret < 0)
+                return ret;
+
+        if (!pdata)
+                dev_warn(codec->dev, "No platform data supplied\n");
+        else {
+                reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+                ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi);
+                if (ret < 0)
+                        return ret;
+        }
+
+
+        wm8940_codec = codec;
+
+        ret = snd_soc_register_codec(codec);
+        if (ret) {
+                dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+                return ret;
+        }
+
+        ret = snd_soc_register_dai(&wm8940_dai);
+        if (ret) {
+                dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+                snd_soc_unregister_codec(codec);
+                return ret;
+        }
+
+        return 0;
+}
+
+static void wm8940_unregister(struct wm8940_priv *wm8940)
+{
+        wm8940_set_bias_level(&wm8940->codec, SND_SOC_BIAS_OFF);
+        snd_soc_unregister_dai(&wm8940_dai);
+        snd_soc_unregister_codec(&wm8940->codec);
+        kfree(wm8940);
+        wm8940_codec = NULL;
+}
+
+static int wm8940_i2c_probe(struct i2c_client *i2c,
+                            const struct i2c_device_id *id)
+{
+        struct wm8940_priv *wm8940;
+        struct snd_soc_codec *codec;
+
+        wm8940 = kzalloc(sizeof *wm8940, GFP_KERNEL);
+        if (wm8940 == NULL)
+                return -ENOMEM;
+
+        codec = &wm8940->codec;
+        codec->hw_write = (hw_write_t)i2c_master_send;
+        i2c_set_clientdata(i2c, wm8940);
+        codec->control_data = i2c;
+        codec->dev = &i2c->dev;
+
+        return wm8940_register(wm8940);
+}
+
+static int __devexit wm8940_i2c_remove(struct i2c_client *client)
+{
+        struct wm8940_priv *wm8940 = i2c_get_clientdata(client);
+
+        wm8940_unregister(wm8940);
+
+        return 0;
+}
+
+static const struct i2c_device_id wm8940_i2c_id[] = {
+        { "wm8940", 0 },
+        { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id);
+
+static struct i2c_driver wm8940_i2c_driver = {
+        .driver = {
+                .name = "WM8940 I2C Codec",
+                .owner = THIS_MODULE,
+        },
+        .probe = wm8940_i2c_probe,
+        .remove = __devexit_p(wm8940_i2c_remove),
+        .id_table = wm8940_i2c_id,
+};
+
+static int __init wm8940_modinit(void)
+{
+        int ret;
+
+        ret = i2c_add_driver(&wm8940_i2c_driver);
+        if (ret)
+                printk(KERN_ERR "Failed to register WM8940 I2C driver: %d\n",
+                       ret);
+        return ret;
+}
+module_init(wm8940_modinit);
+
+static void __exit wm8940_exit(void)
+{
+        i2c_del_driver(&wm8940_i2c_driver);
+}
+module_exit(wm8940_exit);
+
+MODULE_DESCRIPTION("ASoC WM8940 driver");
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8940.h b/sound/soc/codecs/wm8940.h
new file mode 100644
index 000000000000..8410eed3ef84
--- /dev/null
+++ b/sound/soc/codecs/wm8940.h
@@ -0,0 +1,104 @@
+/*
+ * wm8940.h -- WM8940 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8940_H
+#define _WM8940_H
+
+struct wm8940_setup_data {
+        /* Vref to analogue output resistance */
+#define WM8940_VROI_1K 0
+#define WM8940_VROI_30K 1
+        unsigned int vroi:1;
+};
+extern struct snd_soc_dai wm8940_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8940;
+
+/* WM8940 register space */
+#define WM8940_SOFTRESET        0x00
+#define WM8940_POWER1           0x01
+#define WM8940_POWER2           0x02
+#define WM8940_POWER3           0x03
+#define WM8940_IFACE            0x04
+#define WM8940_COMPANDINGCTL    0x05
+#define WM8940_CLOCK            0x06
+#define WM8940_ADDCNTRL         0x07
+#define WM8940_GPIO             0x08
+#define WM8940_CTLINT           0x09
+#define WM8940_DAC              0x0A
+#define WM8940_DACVOL           0x0B
+
+#define WM8940_ADC              0x0E
+#define WM8940_ADCVOL           0x0F
+#define WM8940_NOTCH1           0x10
+#define WM8940_NOTCH2           0x11
+#define WM8940_NOTCH3           0x12
+#define WM8940_NOTCH4           0x13
+#define WM8940_NOTCH5           0x14
+#define WM8940_NOTCH6           0x15
+#define WM8940_NOTCH7           0x16
+#define WM8940_NOTCH8           0x17
+#define WM8940_DACLIM1          0x18
+#define WM8940_DACLIM2          0x19
+
+#define WM8940_ALC1             0x20
+#define WM8940_ALC2             0x21
+#define WM8940_ALC3             0x22
+#define WM8940_NOISEGATE        0x23
+#define WM8940_PLLN             0x24
+#define WM8940_PLLK1            0x25
+#define WM8940_PLLK2            0x26
+#define WM8940_PLLK3            0x27
+
+#define WM8940_ALC4             0x2A
+
+#define WM8940_INPUTCTL         0x2C
+#define WM8940_PGAGAIN          0x2D
+
+#define WM8940_ADCBOOST         0x2F
+
+#define WM8940_OUTPUTCTL        0x31
+#define WM8940_SPKMIX           0x32
+
+#define WM8940_SPKVOL           0x36
+
+#define WM8940_MONOMIX          0x38
+
+#define WM8940_CACHEREGNUM  0x57
+
+
+/* Clock divider Id's */
+#define WM8940_BCLKDIV 0
+#define WM8940_MCLKDIV 1
+#define WM8940_OPCLKDIV 2
+
+/* MCLK clock dividers */
+#define WM8940_MCLKDIV_1        0
+#define WM8940_MCLKDIV_1_5      1
+#define WM8940_MCLKDIV_2        2
+#define WM8940_MCLKDIV_3        3
+#define WM8940_MCLKDIV_4        4
+#define WM8940_MCLKDIV_6        5
+#define WM8940_MCLKDIV_8        6
+#define WM8940_MCLKDIV_12       7
+
+/* BCLK clock dividers */
+#define WM8940_BCLKDIV_1 0
+#define WM8940_BCLKDIV_2 1
+#define WM8940_BCLKDIV_4 2
+#define WM8940_BCLKDIV_8 3
+#define WM8940_BCLKDIV_16 4
+#define WM8940_BCLKDIV_32 5
+
+/* PLL Out Dividers */
+#define WM8940_OPCLKDIV_1 0
+#define WM8940_OPCLKDIV_2 1
+#define WM8940_OPCLKDIV_3 2
+#define WM8940_OPCLKDIV_4 3
+
+#endif /* _WM8940_H */
+
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
new file mode 100644
index 000000000000..e224d8add170
--- /dev/null
+++ b/sound/soc/codecs/wm8960.c
@@ -0,0 +1,969 @@
+/*
+ * wm8960.c  --  WM8960 ALSA SoC Audio driver
+ *
+ * Author: Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8960.h"
+
+#define AUDIO_NAME "wm8960"
+
+struct snd_soc_codec_device soc_codec_dev_wm8960;
+
+/* R25 - Power 1 */
+#define WM8960_VREF      0x40
+
+/* R28 - Anti-pop 1 */
+#define WM8960_POBCTRL   0x80
+#define WM8960_BUFDCOPEN 0x10
+#define WM8960_BUFIOEN   0x08
+#define WM8960_SOFT_ST   0x04
+#define WM8960_HPSTBY    0x01
+
+/* R29 - Anti-pop 2 */
+#define WM8960_DISOP     0x40
+
+/*
+ * wm8960 register cache
+ * We can't read the WM8960 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8960_reg[WM8960_CACHEREGNUM] = {
+        0x0097, 0x0097, 0x0000, 0x0000,
+        0x0000, 0x0008, 0x0000, 0x000a,
+        0x01c0, 0x0000, 0x00ff, 0x00ff,
+        0x0000, 0x0000, 0x0000, 0x0000,
+        0x0000, 0x007b, 0x0100, 0x0032,
+        0x0000, 0x00c3, 0x00c3, 0x01c0,
+        0x0000, 0x0000, 0x0000, 0x0000,
+        0x0000, 0x0000, 0x0000, 0x0000,
+        0x0100, 0x0100, 0x0050, 0x0050,
+        0x0050, 0x0050, 0x0000, 0x0000,
+        0x0000, 0x0000, 0x0040, 0x0000,
+        0x0000, 0x0050, 0x0050, 0x0000,
+        0x0002, 0x0037, 0x004d, 0x0080,
+        0x0008, 0x0031, 0x0026, 0x00e9,
+};
+
+struct wm8960_priv {
+        u16 reg_cache[WM8960_CACHEREGNUM];
+        struct snd_soc_codec codec;
+};
+
+/*
+ * read wm8960 register cache
+ */
+static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec,
+        unsigned int reg)
+{
+        u16 *cache = codec->reg_cache;
+        if (reg == WM8960_RESET)
+                return 0;
+        if (reg >= WM8960_CACHEREGNUM)
+                return -1;
+        return cache[reg];
+}
+
+/*
+ * write wm8960 register cache
+ */
+static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec,
+        u16 reg, unsigned int value)
+{
+        u16 *cache = codec->reg_cache;
+        if (reg >= WM8960_CACHEREGNUM)
+                return;
+        cache[reg] = value;
+}
+
+static inline unsigned int wm8960_read(struct snd_soc_codec *codec,
+        unsigned int reg)
+{
+        return wm8960_read_reg_cache(codec, reg);
+}
+
+/*
+ * write to the WM8960 register space
+ */
+static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg,
+        unsigned int value)
+{
+        u8 data[2];
+
+        /* data is
+         *   D15..D9 WM8960 register offset
+         *   D8...D0 register data
+         */
+        data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+        data[1] = value & 0x00ff;
+
+        wm8960_write_reg_cache(codec, reg, value);
+        if (codec->hw_write(codec->control_data, data, 2) == 2)
+                return 0;
+        else
+                return -EIO;
+}
+
+#define wm8960_reset(c) wm8960_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"};
+static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"};
+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),
+        SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff),
+        SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc),
+        SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
+};
+
+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);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+
+static const struct snd_kcontrol_new wm8960_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
+                 0, 63, 0, adc_tlv),
+SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
+        6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
+        7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
+                 0, 255, 0, dac_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1,
+                 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1,
+        7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8960_LOUT2, WM8960_ROUT2,
+                 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8960_LOUT2, WM8960_ROUT2,
+        7, 1, 0),
+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_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
+
+SOC_ENUM("DAC Polarity", wm8960_enum[2]),
+
+SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
+SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
+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_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_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
+
+SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0),
+SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0),
+
+SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH,
+        0, 127, 0),
+
+SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume",
+               WM8960_BYPASS1, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Left Output Mixer LINPUT3 Volume",
+               WM8960_LOUTMIX, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume",
+               WM8960_BYPASS2, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume",
+               WM8960_ROUTMIX, 4, 7, 1, bypass_tlv),
+};
+
+static const struct snd_kcontrol_new wm8960_lin_boost[] = {
+SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0),
+SOC_DAPM_SINGLE("LINPUT1 Switch", WM8960_LINPATH, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_lin[] = {
+SOC_DAPM_SINGLE("Boost Switch", WM8960_LINPATH, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_rin_boost[] = {
+SOC_DAPM_SINGLE("RINPUT2 Switch", WM8960_RINPATH, 6, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_RINPATH, 7, 1, 0),
+SOC_DAPM_SINGLE("RINPUT1 Switch", WM8960_RINPATH, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_rin[] = {
+SOC_DAPM_SINGLE("Boost Switch", WM8960_RINPATH, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_loutput_mixer[] = {
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_LOUTMIX, 8, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LOUTMIX, 7, 1, 0),
+SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS1, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_routput_mixer[] = {
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_ROUTMIX, 8, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_ROUTMIX, 7, 1, 0),
+SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS2, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_mono_out[] = {
+SOC_DAPM_SINGLE("Left Switch", WM8960_MONOMIX1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Switch", WM8960_MONOMIX2, 7, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("LINPUT1"),
+SND_SOC_DAPM_INPUT("RINPUT1"),
+SND_SOC_DAPM_INPUT("LINPUT2"),
+SND_SOC_DAPM_INPUT("RINPUT2"),
+SND_SOC_DAPM_INPUT("LINPUT3"),
+SND_SOC_DAPM_INPUT("RINPUT3"),
+
+SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0),
+
+SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0,
+                   wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)),
+SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8960_POWER1, 4, 0,
+                   wm8960_rin_boost, ARRAY_SIZE(wm8960_rin_boost)),
+
+SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0,
+                   wm8960_lin, ARRAY_SIZE(wm8960_lin)),
+SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0,
+                   wm8960_rin, ARRAY_SIZE(wm8960_rin)),
+
+SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0),
+
+SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0),
+
+SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0,
+        &wm8960_loutput_mixer[0],
+        ARRAY_SIZE(wm8960_loutput_mixer)),
+SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0,
+        &wm8960_routput_mixer[0],
+        ARRAY_SIZE(wm8960_routput_mixer)),
+
+SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0,
+        &wm8960_mono_out[0],
+        ARRAY_SIZE(wm8960_mono_out)),
+
+SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Left Speaker PGA", WM8960_POWER2, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Speaker PGA", WM8960_POWER2, 3, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Right Speaker Output", WM8960_CLASSD1, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left Speaker Output", WM8960_CLASSD1, 6, 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("SPK_LP"),
+SND_SOC_DAPM_OUTPUT("SPK_LN"),
+SND_SOC_DAPM_OUTPUT("HP_L"),
+SND_SOC_DAPM_OUTPUT("HP_R"),
+SND_SOC_DAPM_OUTPUT("SPK_RP"),
+SND_SOC_DAPM_OUTPUT("SPK_RN"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+};
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+        { "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" },
+        { "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" },
+        { "Left Boost Mixer", "LINPUT3 Switch", "LINPUT3" },
+
+        { "Left Input Mixer", "Boost Switch", "Left Boost Mixer", },
+        { "Left Input Mixer", NULL, "LINPUT1", },  /* Really Boost Switch */
+        { "Left Input Mixer", NULL, "LINPUT2" },
+        { "Left Input Mixer", NULL, "LINPUT3" },
+
+        { "Right Boost Mixer", "RINPUT1 Switch", "RINPUT1" },
+        { "Right Boost Mixer", "RINPUT2 Switch", "RINPUT2" },
+        { "Right Boost Mixer", "RINPUT3 Switch", "RINPUT3" },
+
+        { "Right Input Mixer", "Boost Switch", "Right Boost Mixer", },
+        { "Right Input Mixer", NULL, "RINPUT1", },  /* Really Boost Switch */
+        { "Right Input Mixer", NULL, "RINPUT2" },
+        { "Right Input Mixer", NULL, "LINPUT3" },
+
+        { "Left ADC", NULL, "Left Input Mixer" },
+        { "Right ADC", NULL, "Right Input Mixer" },
+
+        { "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" },
+        { "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} ,
+        { "Left Output Mixer", "PCM Playback Switch", "Left DAC" },
+
+        { "Right Output Mixer", "RINPUT3 Switch", "RINPUT3" },
+        { "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } ,
+        { "Right Output Mixer", "PCM Playback Switch", "Right DAC" },
+
+        { "Mono Output Mixer", "Left Switch", "Left Output Mixer" },
+        { "Mono Output Mixer", "Right Switch", "Right Output Mixer" },
+
+        { "LOUT1 PGA", NULL, "Left Output Mixer" },
+        { "ROUT1 PGA", NULL, "Right Output Mixer" },
+
+        { "HP_L", NULL, "LOUT1 PGA" },
+        { "HP_R", NULL, "ROUT1 PGA" },
+
+        { "Left Speaker PGA", NULL, "Left Output Mixer" },
+        { "Right Speaker PGA", NULL, "Right Output Mixer" },
+
+        { "Left Speaker Output", NULL, "Left Speaker PGA" },
+        { "Right Speaker Output", NULL, "Right Speaker PGA" },
+
+        { "SPK_LN", NULL, "Left Speaker Output" },
+        { "SPK_LP", NULL, "Left Speaker Output" },
+        { "SPK_RN", NULL, "Right Speaker Output" },
+        { "SPK_RP", NULL, "Right Speaker Output" },
+
+        { "OUT3", NULL, "Mono Output Mixer", }
+};
+
+static int wm8960_add_widgets(struct snd_soc_codec *codec)
+{
+        snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
+                                  ARRAY_SIZE(wm8960_dapm_widgets));
+
+        snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+
+        snd_soc_dapm_new_widgets(codec);
+        return 0;
+}
+
+static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
+                unsigned int fmt)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u16 iface = 0;
+
+        /* set master/slave audio interface */
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBM_CFM:
+                iface |= 0x0040;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFS:
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /* interface format */
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_I2S:
+                iface |= 0x0002;
+                break;
+        case SND_SOC_DAIFMT_RIGHT_J:
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                iface |= 0x0001;
+                break;
+        case SND_SOC_DAIFMT_DSP_A:
+                iface |= 0x0003;
+                break;
+        case SND_SOC_DAIFMT_DSP_B:
+                iface |= 0x0013;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /* clock inversion */
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                break;
+        case SND_SOC_DAIFMT_IB_IF:
+                iface |= 0x0090;
+                break;
+        case SND_SOC_DAIFMT_IB_NF:
+                iface |= 0x0080;
+                break;
+        case SND_SOC_DAIFMT_NB_IF:
+                iface |= 0x0010;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /* set iface */
+        wm8960_write(codec, WM8960_IFACE1, iface);
+        return 0;
+}
+
+static int wm8960_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;
+        u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3;
+
+        /* bit size */
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                break;
+        case SNDRV_PCM_FORMAT_S20_3LE:
+                iface |= 0x0004;
+                break;
+        case SNDRV_PCM_FORMAT_S24_LE:
+                iface |= 0x0008;
+                break;
+        }
+
+        /* set iface */
+        wm8960_write(codec, WM8960_IFACE1, iface);
+        return 0;
+}
+
+static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7;
+
+        if (mute)
+                wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8);
+        else
+                wm8960_write(codec, WM8960_DACCTL1, mute_reg);
+        return 0;
+}
+
+static int wm8960_set_bias_level(struct snd_soc_codec *codec,
+                                 enum snd_soc_bias_level level)
+{
+        struct wm8960_data *pdata = codec->dev->platform_data;
+        u16 reg;
+
+        switch (level) {
+        case SND_SOC_BIAS_ON:
+                break;
+
+        case SND_SOC_BIAS_PREPARE:
+                /* Set VMID to 2x50k */
+                reg = wm8960_read(codec, WM8960_POWER1);
+                reg &= ~0x180;
+                reg |= 0x80;
+                wm8960_write(codec, WM8960_POWER1, reg);
+                break;
+
+        case SND_SOC_BIAS_STANDBY:
+                if (codec->bias_level == SND_SOC_BIAS_OFF) {
+                        /* Enable anti-pop features */
+                        wm8960_write(codec, WM8960_APOP1,
+                                     WM8960_POBCTRL | WM8960_SOFT_ST |
+                                     WM8960_BUFDCOPEN | WM8960_BUFIOEN);
+
+                        /* Discharge HP output */
+                        reg = WM8960_DISOP;
+                        if (pdata)
+                                reg |= pdata->dres << 4;
+                        wm8960_write(codec, WM8960_APOP2, reg);
+
+                        msleep(400);
+
+                        wm8960_write(codec, WM8960_APOP2, 0);
+
+                        /* Enable & ramp VMID at 2x50k */
+                        reg = wm8960_read(codec, WM8960_POWER1);
+                        reg |= 0x80;
+                        wm8960_write(codec, WM8960_POWER1, reg);
+                        msleep(100);
+
+                        /* Enable VREF */
+                        wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF);
+
+                        /* Disable anti-pop features */
+                        wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN);
+                }
+
+                /* Set VMID to 2x250k */
+                reg = wm8960_read(codec, WM8960_POWER1);
+                reg &= ~0x180;
+                reg |= 0x100;
+                wm8960_write(codec, WM8960_POWER1, reg);
+                break;
+
+        case SND_SOC_BIAS_OFF:
+                /* Enable anti-pop features */
+                wm8960_write(codec, WM8960_APOP1,
+                             WM8960_POBCTRL | WM8960_SOFT_ST |
+                             WM8960_BUFDCOPEN | WM8960_BUFIOEN);
+
+                /* Disable VMID and VREF, let them discharge */
+                wm8960_write(codec, WM8960_POWER1, 0);
+                msleep(600);
+
+                wm8960_write(codec, WM8960_APOP1, 0);
+                break;
+        }
+
+        codec->bias_level = level;
+
+        return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+        u32 pre_div:1;
+        u32 n:4;
+        u32 k:24;
+};
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+
+static int pll_factors(unsigned int source, unsigned int target,
+                       struct _pll_div *pll_div)
+{
+        unsigned long long Kpart;
+        unsigned int K, Ndiv, Nmod;
+
+        pr_debug("WM8960 PLL: setting %dHz->%dHz\n", source, target);
+
+        /* Scale up target to PLL operating frequency */
+        target *= 4;
+
+        Ndiv = target / source;
+        if (Ndiv < 6) {
+                source >>= 1;
+                pll_div->pre_div = 1;
+                Ndiv = target / source;
+        } else
+                pll_div->pre_div = 0;
+
+        if ((Ndiv < 6) || (Ndiv > 12)) {
+                pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv);
+                return -EINVAL;
+        }
+
+        pll_div->n = Ndiv;
+        Nmod = target % source;
+        Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
+        do_div(Kpart, source);
+
+        K = Kpart & 0xFFFFFFFF;
+
+        /* Check if we need to round */
+        if ((K % 10) >= 5)
+                K += 5;
+
+        /* Move down to proper range now rounding is done */
+        K /= 10;
+
+        pll_div->k = K;
+
+        pr_debug("WM8960 PLL: N=%x K=%x pre_div=%d\n",
+                 pll_div->n, pll_div->k, pll_div->pre_div);
+
+        return 0;
+}
+
+static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai,
+                int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u16 reg;
+        static struct _pll_div pll_div;
+        int ret;
+
+        if (freq_in && freq_out) {
+                ret = pll_factors(freq_in, freq_out, &pll_div);
+                if (ret != 0)
+                        return ret;
+        }
+
+        /* Disable the PLL: even if we are changing the frequency the
+         * PLL needs to be disabled while we do so. */
+        wm8960_write(codec, WM8960_CLOCK1,
+                     wm8960_read(codec, WM8960_CLOCK1) & ~1);
+        wm8960_write(codec, WM8960_POWER2,
+                     wm8960_read(codec, WM8960_POWER2) & ~1);
+
+        if (!freq_in || !freq_out)
+                return 0;
+
+        reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f;
+        reg |= pll_div.pre_div << 4;
+        reg |= pll_div.n;
+
+        if (pll_div.k) {
+                reg |= 0x20;
+
+                wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
+                wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
+                wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+        }
+        wm8960_write(codec, WM8960_PLL1, reg);
+
+        /* Turn it on */
+        wm8960_write(codec, WM8960_POWER2,
+                     wm8960_read(codec, WM8960_POWER2) | 1);
+        msleep(250);
+        wm8960_write(codec, WM8960_CLOCK1,
+                     wm8960_read(codec, WM8960_CLOCK1) | 1);
+
+        return 0;
+}
+
+static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+                int div_id, int div)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u16 reg;
+
+        switch (div_id) {
+        case WM8960_SYSCLKSEL:
+                reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe;
+                wm8960_write(codec, WM8960_CLOCK1, reg | div);
+                break;
+        case WM8960_SYSCLKDIV:
+                reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9;
+                wm8960_write(codec, WM8960_CLOCK1, reg | div);
+                break;
+        case WM8960_DACDIV:
+                reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7;
+                wm8960_write(codec, WM8960_CLOCK1, reg | div);
+                break;
+        case WM8960_OPCLKDIV:
+                reg = wm8960_read(codec, WM8960_PLL1) & 0x03f;
+                wm8960_write(codec, WM8960_PLL1, reg | div);
+                break;
+        case WM8960_DCLKDIV:
+                reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f;
+                wm8960_write(codec, WM8960_CLOCK2, reg | div);
+                break;
+        case WM8960_TOCLKSEL:
+                reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd;
+                wm8960_write(codec, WM8960_ADDCTL1, reg | div);
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+#define WM8960_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8960_FORMATS \
+        (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+        SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8960_dai_ops = {
+        .hw_params = wm8960_hw_params,
+        .digital_mute = wm8960_mute,
+        .set_fmt = wm8960_set_dai_fmt,
+        .set_clkdiv = wm8960_set_dai_clkdiv,
+        .set_pll = wm8960_set_dai_pll,
+};
+
+struct snd_soc_dai wm8960_dai = {
+        .name = "WM8960",
+        .playback = {
+                .stream_name = "Playback",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = WM8960_RATES,
+                .formats = WM8960_FORMATS,},
+        .capture = {
+                .stream_name = "Capture",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = WM8960_RATES,
+                .formats = WM8960_FORMATS,},
+        .ops = &wm8960_dai_ops,
+        .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8960_dai);
+
+static int wm8960_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;
+
+        wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF);
+        return 0;
+}
+
+static int wm8960_resume(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+        int i;
+        u8 data[2];
+        u16 *cache = codec->reg_cache;
+
+        /* Sync reg_cache with the hardware */
+        for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) {
+                data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+                data[1] = cache[i] & 0x00ff;
+                codec->hw_write(codec->control_data, data, 2);
+        }
+
+        wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+        wm8960_set_bias_level(codec, codec->suspend_bias_level);
+        return 0;
+}
+
+static struct snd_soc_codec *wm8960_codec;
+
+static int wm8960_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec;
+        int ret = 0;
+
+        if (wm8960_codec == NULL) {
+                dev_err(&pdev->dev, "Codec device not registered\n");
+                return -ENODEV;
+        }
+
+        socdev->card->codec = wm8960_codec;
+        codec = wm8960_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, wm8960_snd_controls,
+                             ARRAY_SIZE(wm8960_snd_controls));
+        wm8960_add_widgets(codec);
+        ret = snd_soc_init_card(socdev);
+        if (ret < 0) {
+                dev_err(codec->dev, "failed to register card: %d\n", ret);
+                goto card_err;
+        }
+
+        return ret;
+
+card_err:
+        snd_soc_free_pcms(socdev);
+        snd_soc_dapm_free(socdev);
+pcm_err:
+        return ret;
+}
+
+/* power down chip */
+static int wm8960_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_wm8960 = {
+        .probe =        wm8960_probe,
+        .remove =       wm8960_remove,
+        .suspend =      wm8960_suspend,
+        .resume =       wm8960_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960);
+
+static int wm8960_register(struct wm8960_priv *wm8960)
+{
+        struct wm8960_data *pdata = wm8960->codec.dev->platform_data;
+        struct snd_soc_codec *codec = &wm8960->codec;
+        int ret;
+        u16 reg;
+
+        if (wm8960_codec) {
+                dev_err(codec->dev, "Another WM8960 is registered\n");
+                return -EINVAL;
+        }
+
+        if (!pdata) {
+                dev_warn(codec->dev, "No platform data supplied\n");
+        } else {
+                if (pdata->dres > WM8960_DRES_MAX) {
+                        dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres);
+                        pdata->dres = 0;
+                }
+        }
+
+        mutex_init(&codec->mutex);
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        codec->private_data = wm8960;
+        codec->name = "WM8960";
+        codec->owner = THIS_MODULE;
+        codec->read = wm8960_read_reg_cache;
+        codec->write = wm8960_write;
+        codec->bias_level = SND_SOC_BIAS_OFF;
+        codec->set_bias_level = wm8960_set_bias_level;
+        codec->dai = &wm8960_dai;
+        codec->num_dai = 1;
+        codec->reg_cache_size = WM8960_CACHEREGNUM;
+        codec->reg_cache = &wm8960->reg_cache;
+
+        memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg));
+
+        ret = wm8960_reset(codec);
+        if (ret < 0) {
+                dev_err(codec->dev, "Failed to issue reset\n");
+                return ret;
+        }
+
+        wm8960_dai.dev = codec->dev;
+
+        wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+        /* Latch the update bits */
+        reg = wm8960_read(codec, WM8960_LINVOL);
+        wm8960_write(codec, WM8960_LINVOL, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_RINVOL);
+        wm8960_write(codec, WM8960_RINVOL, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_LADC);
+        wm8960_write(codec, WM8960_LADC, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_RADC);
+        wm8960_write(codec, WM8960_RADC, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_LDAC);
+        wm8960_write(codec, WM8960_LDAC, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_RDAC);
+        wm8960_write(codec, WM8960_RDAC, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_LOUT1);
+        wm8960_write(codec, WM8960_LOUT1, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_ROUT1);
+        wm8960_write(codec, WM8960_ROUT1, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_LOUT2);
+        wm8960_write(codec, WM8960_LOUT2, reg | 0x100);
+        reg = wm8960_read(codec, WM8960_ROUT2);
+        wm8960_write(codec, WM8960_ROUT2, reg | 0x100);
+
+        wm8960_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(&wm8960_dai);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+                snd_soc_unregister_codec(codec);
+                return ret;
+        }
+
+        return 0;
+}
+
+static void wm8960_unregister(struct wm8960_priv *wm8960)
+{
+        wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF);
+        snd_soc_unregister_dai(&wm8960_dai);
+        snd_soc_unregister_codec(&wm8960->codec);
+        kfree(wm8960);
+        wm8960_codec = NULL;
+}
+
+static __devinit int wm8960_i2c_probe(struct i2c_client *i2c,
+                                      const struct i2c_device_id *id)
+{
+        struct wm8960_priv *wm8960;
+        struct snd_soc_codec *codec;
+
+        wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL);
+        if (wm8960 == NULL)
+                return -ENOMEM;
+
+        codec = &wm8960->codec;
+        codec->hw_write = (hw_write_t)i2c_master_send;
+
+        i2c_set_clientdata(i2c, wm8960);
+        codec->control_data = i2c;
+
+        codec->dev = &i2c->dev;
+
+        return wm8960_register(wm8960);
+}
+
+static __devexit int wm8960_i2c_remove(struct i2c_client *client)
+{
+        struct wm8960_priv *wm8960 = i2c_get_clientdata(client);
+        wm8960_unregister(wm8960);
+        return 0;
+}
+
+static const struct i2c_device_id wm8960_i2c_id[] = {
+        { "wm8960", 0 },
+        { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
+
+static struct i2c_driver wm8960_i2c_driver = {
+        .driver = {
+                .name = "WM8960 I2C Codec",
+                .owner = THIS_MODULE,
+        },
+        .probe =    wm8960_i2c_probe,
+        .remove =   __devexit_p(wm8960_i2c_remove),
+        .id_table = wm8960_i2c_id,
+};
+
+static int __init wm8960_modinit(void)
+{
+        int ret;
+
+        ret = i2c_add_driver(&wm8960_i2c_driver);
+        if (ret != 0) {
+                printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n",
+                       ret);
+        }
+
+        return ret;
+}
+module_init(wm8960_modinit);
+
+static void __exit wm8960_exit(void)
+{
+        i2c_del_driver(&wm8960_i2c_driver);
+}
+module_exit(wm8960_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8960 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h
new file mode 100644
index 000000000000..c9af56c9d9d4
--- /dev/null
+++ b/sound/soc/codecs/wm8960.h
@@ -0,0 +1,127 @@
+/*
+ * wm8960.h  --  WM8960 Soc Audio driver
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM8960_H
+#define _WM8960_H
+
+/* WM8960 register space */
+
+
+#define WM8960_CACHEREGNUM      56
+
+#define WM8960_LINVOL           0x0
+#define WM8960_RINVOL           0x1
+#define WM8960_LOUT1            0x2
+#define WM8960_ROUT1            0x3
+#define WM8960_CLOCK1           0x4
+#define WM8960_DACCTL1          0x5
+#define WM8960_DACCTL2          0x6
+#define WM8960_IFACE1           0x7
+#define WM8960_CLOCK2           0x8
+#define WM8960_IFACE2           0x9
+#define WM8960_LDAC             0xa
+#define WM8960_RDAC             0xb
+
+#define WM8960_RESET            0xf
+#define WM8960_3D               0x10
+#define WM8960_ALC1             0x11
+#define WM8960_ALC2             0x12
+#define WM8960_ALC3             0x13
+#define WM8960_NOISEG           0x14
+#define WM8960_LADC             0x15
+#define WM8960_RADC             0x16
+#define WM8960_ADDCTL1          0x17
+#define WM8960_ADDCTL2          0x18
+#define WM8960_POWER1           0x19
+#define WM8960_POWER2           0x1a
+#define WM8960_ADDCTL3          0x1b
+#define WM8960_APOP1            0x1c
+#define WM8960_APOP2            0x1d
+
+#define WM8960_LINPATH          0x20
+#define WM8960_RINPATH          0x21
+#define WM8960_LOUTMIX          0x22
+
+#define WM8960_ROUTMIX          0x25
+#define WM8960_MONOMIX1         0x26
+#define WM8960_MONOMIX2         0x27
+#define WM8960_LOUT2            0x28
+#define WM8960_ROUT2            0x29
+#define WM8960_MONO             0x2a
+#define WM8960_INBMIX1          0x2b
+#define WM8960_INBMIX2          0x2c
+#define WM8960_BYPASS1          0x2d
+#define WM8960_BYPASS2          0x2e
+#define WM8960_POWER3           0x2f
+#define WM8960_ADDCTL4          0x30
+#define WM8960_CLASSD1          0x31
+
+#define WM8960_CLASSD3          0x33
+#define WM8960_PLL1             0x34
+#define WM8960_PLL2             0x35
+#define WM8960_PLL3             0x36
+#define WM8960_PLL4             0x37
+
+
+/*
+ * WM8960 Clock dividers
+ */
+#define WM8960_SYSCLKDIV                0
+#define WM8960_DACDIV                   1
+#define WM8960_OPCLKDIV                 2
+#define WM8960_DCLKDIV                  3
+#define WM8960_TOCLKSEL                 4
+#define WM8960_SYSCLKSEL                5
+
+#define WM8960_SYSCLK_DIV_1             (0 << 1)
+#define WM8960_SYSCLK_DIV_2             (2 << 1)
+
+#define WM8960_SYSCLK_MCLK              (0 << 0)
+#define WM8960_SYSCLK_PLL               (1 << 0)
+
+#define WM8960_DAC_DIV_1                (0 << 3)
+#define WM8960_DAC_DIV_1_5              (1 << 3)
+#define WM8960_DAC_DIV_2                (2 << 3)
+#define WM8960_DAC_DIV_3                (3 << 3)
+#define WM8960_DAC_DIV_4                (4 << 3)
+#define WM8960_DAC_DIV_5_5              (5 << 3)
+#define WM8960_DAC_DIV_6                (6 << 3)
+
+#define WM8960_DCLK_DIV_1_5             (0 << 6)
+#define WM8960_DCLK_DIV_2               (1 << 6)
+#define WM8960_DCLK_DIV_3               (2 << 6)
+#define WM8960_DCLK_DIV_4               (3 << 6)
+#define WM8960_DCLK_DIV_6               (4 << 6)
+#define WM8960_DCLK_DIV_8               (5 << 6)
+#define WM8960_DCLK_DIV_12              (6 << 6)
+#define WM8960_DCLK_DIV_16              (7 << 6)
+
+#define WM8960_TOCLK_F19                (0 << 1)
+#define WM8960_TOCLK_F21                (1 << 1)
+
+#define WM8960_OPCLK_DIV_1              (0 << 0)
+#define WM8960_OPCLK_DIV_2              (1 << 0)
+#define WM8960_OPCLK_DIV_3              (2 << 0)
+#define WM8960_OPCLK_DIV_4              (3 << 0)
+#define WM8960_OPCLK_DIV_5_5            (4 << 0)
+#define WM8960_OPCLK_DIV_6              (5 << 0)
+
+extern struct snd_soc_dai wm8960_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8960;
+
+#define WM8960_DRES_400R 0
+#define WM8960_DRES_200R 1
+#define WM8960_DRES_600R 2
+#define WM8960_DRES_150R 3
+#define WM8960_DRES_MAX  3
+
+struct wm8960_data {
+        int dres;
+};
+
+#endif
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
new file mode 100644
index 000000000000..c05f71803aa8
--- /dev/null
+++ b/sound/soc/codecs/wm8988.c
@@ -0,0 +1,1097 @@
+/*
+ * wm8988.c -- WM8988 ALSA SoC audio driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8988.h"
+
+/*
+ * wm8988 register cache
+ * We can't read the WM8988 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8988_reg[] = {
+        0x0097, 0x0097, 0x0079, 0x0079,  /*  0 */
+        0x0000, 0x0008, 0x0000, 0x000a,  /*  4 */
+        0x0000, 0x0000, 0x00ff, 0x00ff,  /*  8 */
+        0x000f, 0x000f, 0x0000, 0x0000,  /* 12 */
+        0x0000, 0x007b, 0x0000, 0x0032,  /* 16 */
+        0x0000, 0x00c3, 0x00c3, 0x00c0,  /* 20 */
+        0x0000, 0x0000, 0x0000, 0x0000,  /* 24 */
+        0x0000, 0x0000, 0x0000, 0x0000,  /* 28 */
+        0x0000, 0x0000, 0x0050, 0x0050,  /* 32 */
+        0x0050, 0x0050, 0x0050, 0x0050,  /* 36 */
+        0x0079, 0x0079, 0x0079,          /* 40 */
+};
+
+/* codec private data */
+struct wm8988_priv {
+        unsigned int sysclk;
+        struct snd_soc_codec codec;
+        struct snd_pcm_hw_constraint_list *sysclk_constraints;
+        u16 reg_cache[WM8988_NUM_REG];
+};
+
+
+/*
+ * read wm8988 register cache
+ */
+static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec,
+        unsigned int reg)
+{
+        u16 *cache = codec->reg_cache;
+        if (reg > WM8988_NUM_REG)
+                return -1;
+        return cache[reg];
+}
+
+/*
+ * write wm8988 register cache
+ */
+static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec,
+        unsigned int reg, unsigned int value)
+{
+        u16 *cache = codec->reg_cache;
+        if (reg > WM8988_NUM_REG)
+                return;
+        cache[reg] = value;
+}
+
+static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg,
+        unsigned int value)
+{
+        u8 data[2];
+
+        /* data is
+         *   D15..D9 WM8753 register offset
+         *   D8...D0 register data
+         */
+        data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+        data[1] = value & 0x00ff;
+
+        wm8988_write_reg_cache(codec, reg, value);
+        if (codec->hw_write(codec->control_data, data, 2) == 2)
+                return 0;
+        else
+                return -EIO;
+}
+
+#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0)
+
+/*
+ * WM8988 Controls
+ */
+
+static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"};
+static const struct soc_enum bass_boost =
+        SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt);
+
+static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
+static const struct soc_enum bass_filter =
+        SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt);
+
+static const char *treble_txt[] = {"8kHz", "4kHz"};
+static const struct soc_enum treble =
+        SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt);
+
+static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"};
+static const struct soc_enum stereo_3d_lc =
+        SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt);
+
+static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"};
+static const struct soc_enum stereo_3d_uc =
+        SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt);
+
+static const char *stereo_3d_func_txt[] = {"Capture", "Playback"};
+static const struct soc_enum stereo_3d_func =
+        SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt);
+
+static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"};
+static const struct soc_enum alc_func =
+        SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt);
+
+static const char *ng_type_txt[] = {"Constant PGA Gain",
+                                    "Mute ADC Output"};
+static const struct soc_enum ng_type =
+        SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt);
+
+static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const struct soc_enum deemph =
+        SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt);
+
+static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert",
+                                   "L + R Invert"};
+static const struct soc_enum adcpol =
+        SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt);
+
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
+
+static const struct snd_kcontrol_new wm8988_snd_controls[] = {
+
+SOC_ENUM("Bass Boost", bass_boost),
+SOC_ENUM("Bass Filter", bass_filter),
+SOC_SINGLE("Bass Volume", WM8988_BASS, 0, 15, 1),
+
+SOC_SINGLE("Treble Volume", WM8988_TREBLE, 0, 15, 0),
+SOC_ENUM("Treble Cut-off", treble),
+
+SOC_SINGLE("3D Switch", WM8988_3D, 0, 1, 0),
+SOC_SINGLE("3D Volume", WM8988_3D, 1, 15, 0),
+SOC_ENUM("3D Lower Cut-off", stereo_3d_lc),
+SOC_ENUM("3D Upper Cut-off", stereo_3d_uc),
+SOC_ENUM("3D Mode", stereo_3d_func),
+
+SOC_SINGLE("ALC Capture Target Volume", WM8988_ALC1, 0, 7, 0),
+SOC_SINGLE("ALC Capture Max Volume", WM8988_ALC1, 4, 7, 0),
+SOC_ENUM("ALC Capture Function", alc_func),
+SOC_SINGLE("ALC Capture ZC Switch", WM8988_ALC2, 7, 1, 0),
+SOC_SINGLE("ALC Capture Hold Time", WM8988_ALC2, 0, 15, 0),
+SOC_SINGLE("ALC Capture Decay Time", WM8988_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack Time", WM8988_ALC3, 0, 15, 0),
+SOC_SINGLE("ALC Capture NG Threshold", WM8988_NGATE, 3, 31, 0),
+SOC_ENUM("ALC Capture NG Type", ng_type),
+SOC_SINGLE("ALC Capture NG Switch", WM8988_NGATE, 0, 1, 0),
+
+SOC_SINGLE("ZC Timeout Switch", WM8988_ADCTL1, 0, 1, 0),
+
+SOC_DOUBLE_R_TLV("Capture Digital Volume", WM8988_LADC, WM8988_RADC,
+                 0, 255, 0, adc_tlv),
+SOC_DOUBLE_R_TLV("Capture Volume", WM8988_LINVOL, WM8988_RINVOL,
+                 0, 63, 0, pga_tlv),
+SOC_DOUBLE_R("Capture ZC Switch", WM8988_LINVOL, WM8988_RINVOL, 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8988_LINVOL, WM8988_RINVOL, 7, 1, 1),
+
+SOC_ENUM("Playback De-emphasis", deemph),
+
+SOC_ENUM("Capture Polarity", adcpol),
+SOC_SINGLE("Playback 6dB Attenuate", WM8988_ADCDAC, 7, 1, 0),
+SOC_SINGLE("Capture 6dB Attenuate", WM8988_ADCDAC, 8, 1, 0),
+
+SOC_DOUBLE_R_TLV("PCM Volume", WM8988_LDAC, WM8988_RDAC, 0, 255, 0, dac_tlv),
+
+SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", WM8988_LOUTM1, 4, 7, 1,
+               bypass_tlv),
+SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", WM8988_LOUTM2, 4, 7, 1,
+               bypass_tlv),
+SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", WM8988_ROUTM1, 4, 7, 1,
+               bypass_tlv),
+SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", WM8988_ROUTM2, 4, 7, 1,
+               bypass_tlv),
+
+SOC_DOUBLE_R("Output 1 Playback ZC Switch", WM8988_LOUT1V,
+             WM8988_ROUT1V, 7, 1, 0),
+SOC_DOUBLE_R_TLV("Output 1 Playback Volume", WM8988_LOUT1V, WM8988_ROUT1V,
+                 0, 127, 0, out_tlv),
+
+SOC_DOUBLE_R("Output 2 Playback ZC Switch", WM8988_LOUT2V,
+             WM8988_ROUT2V, 7, 1, 0),
+SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V,
+                 0, 127, 0, out_tlv),
+
+};
+
+/*
+ * DAPM Controls
+ */
+
+static int wm8988_lrc_control(struct snd_soc_dapm_widget *w,
+                              struct snd_kcontrol *kcontrol, int event)
+{
+        struct snd_soc_codec *codec = w->codec;
+        u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2);
+
+        /* Use the DAC to gate LRC if active, otherwise use ADC */
+        if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180)
+                adctl2 &= ~0x4;
+        else
+                adctl2 |= 0x4;
+
+        return wm8988_write(codec, WM8988_ADCTL2, adctl2);
+}
+
+static const char *wm8988_line_texts[] = {
+        "Line 1", "Line 2", "PGA", "Differential"};
+
+static const unsigned int wm8988_line_values[] = {
+        0, 1, 3, 4};
+
+static const struct soc_enum wm8988_lline_enum =
+        SOC_VALUE_ENUM_SINGLE(WM8988_LOUTM1, 0, 7,
+                              ARRAY_SIZE(wm8988_line_texts),
+                              wm8988_line_texts,
+                              wm8988_line_values);
+static const struct snd_kcontrol_new wm8988_left_line_controls =
+        SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum);
+
+static const struct soc_enum wm8988_rline_enum =
+        SOC_VALUE_ENUM_SINGLE(WM8988_ROUTM1, 0, 7,
+                              ARRAY_SIZE(wm8988_line_texts),
+                              wm8988_line_texts,
+                              wm8988_line_values);
+static const struct snd_kcontrol_new wm8988_right_line_controls =
+        SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum);
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = {
+        SOC_DAPM_SINGLE("Playback Switch", WM8988_LOUTM1, 8, 1, 0),
+        SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_LOUTM1, 7, 1, 0),
+        SOC_DAPM_SINGLE("Right Playback Switch", WM8988_LOUTM2, 8, 1, 0),
+        SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_LOUTM2, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new wm8988_right_mixer_controls[] = {
+        SOC_DAPM_SINGLE("Left Playback Switch", WM8988_ROUTM1, 8, 1, 0),
+        SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_ROUTM1, 7, 1, 0),
+        SOC_DAPM_SINGLE("Playback Switch", WM8988_ROUTM2, 8, 1, 0),
+        SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_ROUTM2, 7, 1, 0),
+};
+
+static const char *wm8988_pga_sel[] = {"Line 1", "Line 2", "Differential"};
+static const unsigned int wm8988_pga_val[] = { 0, 1, 3 };
+
+/* Left PGA Mux */
+static const struct soc_enum wm8988_lpga_enum =
+        SOC_VALUE_ENUM_SINGLE(WM8988_LADCIN, 6, 3,
+                              ARRAY_SIZE(wm8988_pga_sel),
+                              wm8988_pga_sel,
+                              wm8988_pga_val);
+static const struct snd_kcontrol_new wm8988_left_pga_controls =
+        SOC_DAPM_VALUE_ENUM("Route", wm8988_lpga_enum);
+
+/* Right PGA Mux */
+static const struct soc_enum wm8988_rpga_enum =
+        SOC_VALUE_ENUM_SINGLE(WM8988_RADCIN, 6, 3,
+                              ARRAY_SIZE(wm8988_pga_sel),
+                              wm8988_pga_sel,
+                              wm8988_pga_val);
+static const struct snd_kcontrol_new wm8988_right_pga_controls =
+        SOC_DAPM_VALUE_ENUM("Route", wm8988_rpga_enum);
+
+/* Differential Mux */
+static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"};
+static const struct soc_enum diffmux =
+        SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel);
+static const struct snd_kcontrol_new wm8988_diffmux_controls =
+        SOC_DAPM_ENUM("Route", diffmux);
+
+/* Mono ADC Mux */
+static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)",
+        "Mono (Right)", "Digital Mono"};
+static const struct soc_enum monomux =
+        SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux);
+static const struct snd_kcontrol_new wm8988_monomux_controls =
+        SOC_DAPM_ENUM("Route", monomux);
+
+static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = {
+        SND_SOC_DAPM_MICBIAS("Mic Bias", WM8988_PWR1, 1, 0),
+
+        SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+                &wm8988_diffmux_controls),
+        SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+                &wm8988_monomux_controls),
+        SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+                &wm8988_monomux_controls),
+
+        SND_SOC_DAPM_MUX("Left PGA Mux", WM8988_PWR1, 5, 0,
+                &wm8988_left_pga_controls),
+        SND_SOC_DAPM_MUX("Right PGA Mux", WM8988_PWR1, 4, 0,
+                &wm8988_right_pga_controls),
+
+        SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+                &wm8988_left_line_controls),
+        SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+                &wm8988_right_line_controls),
+
+        SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8988_PWR1, 2, 0),
+        SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8988_PWR1, 3, 0),
+
+        SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8988_PWR2, 7, 0),
+        SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8988_PWR2, 8, 0),
+
+        SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+                &wm8988_left_mixer_controls[0],
+                ARRAY_SIZE(wm8988_left_mixer_controls)),
+        SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+                &wm8988_right_mixer_controls[0],
+                ARRAY_SIZE(wm8988_right_mixer_controls)),
+
+        SND_SOC_DAPM_PGA("Right Out 2", WM8988_PWR2, 3, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("Left Out 2", WM8988_PWR2, 4, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("Right Out 1", WM8988_PWR2, 5, 0, NULL, 0),
+        SND_SOC_DAPM_PGA("Left Out 1", WM8988_PWR2, 6, 0, NULL, 0),
+
+        SND_SOC_DAPM_POST("LRC control", wm8988_lrc_control),
+
+        SND_SOC_DAPM_OUTPUT("LOUT1"),
+        SND_SOC_DAPM_OUTPUT("ROUT1"),
+        SND_SOC_DAPM_OUTPUT("LOUT2"),
+        SND_SOC_DAPM_OUTPUT("ROUT2"),
+        SND_SOC_DAPM_OUTPUT("VREF"),
+
+        SND_SOC_DAPM_INPUT("LINPUT1"),
+        SND_SOC_DAPM_INPUT("LINPUT2"),
+        SND_SOC_DAPM_INPUT("RINPUT1"),
+        SND_SOC_DAPM_INPUT("RINPUT2"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+        { "Left Line Mux", "Line 1", "LINPUT1" },
+        { "Left Line Mux", "Line 2", "LINPUT2" },
+        { "Left Line Mux", "PGA", "Left PGA Mux" },
+        { "Left Line Mux", "Differential", "Differential Mux" },
+
+        { "Right Line Mux", "Line 1", "RINPUT1" },
+        { "Right Line Mux", "Line 2", "RINPUT2" },
+        { "Right Line Mux", "PGA", "Right PGA Mux" },
+        { "Right Line Mux", "Differential", "Differential Mux" },
+
+        { "Left PGA Mux", "Line 1", "LINPUT1" },
+        { "Left PGA Mux", "Line 2", "LINPUT2" },
+        { "Left PGA Mux", "Differential", "Differential Mux" },
+
+        { "Right PGA Mux", "Line 1", "RINPUT1" },
+        { "Right PGA Mux", "Line 2", "RINPUT2" },
+        { "Right PGA Mux", "Differential", "Differential Mux" },
+
+        { "Differential Mux", "Line 1", "LINPUT1" },
+        { "Differential Mux", "Line 1", "RINPUT1" },
+        { "Differential Mux", "Line 2", "LINPUT2" },
+        { "Differential Mux", "Line 2", "RINPUT2" },
+
+        { "Left ADC Mux", "Stereo", "Left PGA Mux" },
+        { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" },
+        { "Left ADC Mux", "Digital Mono", "Left PGA Mux" },
+
+        { "Right ADC Mux", "Stereo", "Right PGA Mux" },
+        { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" },
+        { "Right ADC Mux", "Digital Mono", "Right PGA Mux" },
+
+        { "Left ADC", NULL, "Left ADC Mux" },
+        { "Right ADC", NULL, "Right ADC Mux" },
+
+        { "Left Line Mux", "Line 1", "LINPUT1" },
+        { "Left Line Mux", "Line 2", "LINPUT2" },
+        { "Left Line Mux", "PGA", "Left PGA Mux" },
+        { "Left Line Mux", "Differential", "Differential Mux" },
+
+        { "Right Line Mux", "Line 1", "RINPUT1" },
+        { "Right Line Mux", "Line 2", "RINPUT2" },
+        { "Right Line Mux", "PGA", "Right PGA Mux" },
+        { "Right Line Mux", "Differential", "Differential Mux" },
+
+        { "Left Mixer", "Playback Switch", "Left DAC" },
+        { "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
+        { "Left Mixer", "Right Playback Switch", "Right DAC" },
+        { "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+        { "Right Mixer", "Left Playback Switch", "Left DAC" },
+        { "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
+        { "Right Mixer", "Playback Switch", "Right DAC" },
+        { "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+        { "Left Out 1", NULL, "Left Mixer" },
+        { "LOUT1", NULL, "Left Out 1" },
+        { "Right Out 1", NULL, "Right Mixer" },
+        { "ROUT1", NULL, "Right Out 1" },
+
+        { "Left Out 2", NULL, "Left Mixer" },
+        { "LOUT2", NULL, "Left Out 2" },
+        { "Right Out 2", NULL, "Right Mixer" },
+        { "ROUT2", NULL, "Right Out 2" },
+};
+
+struct _coeff_div {
+        u32 mclk;
+        u32 rate;
+        u16 fs;
+        u8 sr:5;
+        u8 usb:1;
+};
+
+/* codec hifi mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+        /* 8k */
+        {12288000, 8000, 1536, 0x6, 0x0},
+        {11289600, 8000, 1408, 0x16, 0x0},
+        {18432000, 8000, 2304, 0x7, 0x0},
+        {16934400, 8000, 2112, 0x17, 0x0},
+        {12000000, 8000, 1500, 0x6, 0x1},
+
+        /* 11.025k */
+        {11289600, 11025, 1024, 0x18, 0x0},
+        {16934400, 11025, 1536, 0x19, 0x0},
+        {12000000, 11025, 1088, 0x19, 0x1},
+
+        /* 16k */
+        {12288000, 16000, 768, 0xa, 0x0},
+        {18432000, 16000, 1152, 0xb, 0x0},
+        {12000000, 16000, 750, 0xa, 0x1},
+
+        /* 22.05k */
+        {11289600, 22050, 512, 0x1a, 0x0},
+        {16934400, 22050, 768, 0x1b, 0x0},
+        {12000000, 22050, 544, 0x1b, 0x1},
+
+        /* 32k */
+        {12288000, 32000, 384, 0xc, 0x0},
+        {18432000, 32000, 576, 0xd, 0x0},
+        {12000000, 32000, 375, 0xa, 0x1},
+
+        /* 44.1k */
+        {11289600, 44100, 256, 0x10, 0x0},
+        {16934400, 44100, 384, 0x11, 0x0},
+        {12000000, 44100, 272, 0x11, 0x1},
+
+        /* 48k */
+        {12288000, 48000, 256, 0x0, 0x0},
+        {18432000, 48000, 384, 0x1, 0x0},
+        {12000000, 48000, 250, 0x0, 0x1},
+
+        /* 88.2k */
+        {11289600, 88200, 128, 0x1e, 0x0},
+        {16934400, 88200, 192, 0x1f, 0x0},
+        {12000000, 88200, 136, 0x1f, 0x1},
+
+        /* 96k */
+        {12288000, 96000, 128, 0xe, 0x0},
+        {18432000, 96000, 192, 0xf, 0x0},
+        {12000000, 96000, 125, 0xe, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+        int i;
+
+        for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+                if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+                        return i;
+        }
+
+        return -EINVAL;
+}
+
+/* The set of rates we can generate from the above for each SYSCLK */
+
+static unsigned int rates_12288[] = {
+        8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_12288 = {
+        .count  = ARRAY_SIZE(rates_12288),
+        .list   = rates_12288,
+};
+
+static unsigned int rates_112896[] = {
+        8000, 11025, 22050, 44100,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_112896 = {
+        .count  = ARRAY_SIZE(rates_112896),
+        .list   = rates_112896,
+};
+
+static unsigned int rates_12[] = {
+        8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
+        48000, 88235, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_12 = {
+        .count  = ARRAY_SIZE(rates_12),
+        .list   = rates_12,
+};
+
+/*
+ * Note that this should be called from init rather than from hw_params.
+ */
+static int wm8988_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 wm8988_priv *wm8988 = codec->private_data;
+
+        switch (freq) {
+        case 11289600:
+        case 18432000:
+        case 22579200:
+        case 36864000:
+                wm8988->sysclk_constraints = &constraints_112896;
+                wm8988->sysclk = freq;
+                return 0;
+
+        case 12288000:
+        case 16934400:
+        case 24576000:
+        case 33868800:
+                wm8988->sysclk_constraints = &constraints_12288;
+                wm8988->sysclk = freq;
+                return 0;
+
+        case 12000000:
+        case 24000000:
+                wm8988->sysclk_constraints = &constraints_12;
+                wm8988->sysclk = freq;
+                return 0;
+        }
+        return -EINVAL;
+}
+
+static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai,
+                unsigned int fmt)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        u16 iface = 0;
+
+        /* set master/slave audio interface */
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBM_CFM:
+                iface = 0x0040;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFS:
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /* interface format */
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_I2S:
+                iface |= 0x0002;
+                break;
+        case SND_SOC_DAIFMT_RIGHT_J:
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                iface |= 0x0001;
+                break;
+        case SND_SOC_DAIFMT_DSP_A:
+                iface |= 0x0003;
+                break;
+        case SND_SOC_DAIFMT_DSP_B:
+                iface |= 0x0013;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        /* clock inversion */
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                break;
+        case SND_SOC_DAIFMT_IB_IF:
+                iface |= 0x0090;
+                break;
+        case SND_SOC_DAIFMT_IB_NF:
+                iface |= 0x0080;
+                break;
+        case SND_SOC_DAIFMT_NB_IF:
+                iface |= 0x0010;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        wm8988_write(codec, WM8988_IFACE, iface);
+        return 0;
+}
+
+static int wm8988_pcm_startup(struct snd_pcm_substream *substream,
+                              struct snd_soc_dai *dai)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        struct wm8988_priv *wm8988 = codec->private_data;
+
+        /* The set of sample rates that can be supported depends on the
+         * MCLK supplied to the CODEC - enforce this.
+         */
+        if (!wm8988->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,
+                                   wm8988->sysclk_constraints);
+
+        return 0;
+}
+
+static int wm8988_pcm_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 wm8988_priv *wm8988 = codec->private_data;
+        u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3;
+        u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180;
+        int coeff;
+
+        coeff = get_coeff(wm8988->sysclk, params_rate(params));
+        if (coeff < 0) {
+                coeff = get_coeff(wm8988->sysclk / 2, params_rate(params));
+                srate |= 0x40;
+        }
+        if (coeff < 0) {
+                dev_err(codec->dev,
+                        "Unable to configure sample rate %dHz with %dHz MCLK\n",
+                        params_rate(params), wm8988->sysclk);
+                return coeff;
+        }
+
+        /* bit size */
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                break;
+        case SNDRV_PCM_FORMAT_S20_3LE:
+                iface |= 0x0004;
+                break;
+        case SNDRV_PCM_FORMAT_S24_LE:
+                iface |= 0x0008;
+                break;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                iface |= 0x000c;
+                break;
+        }
+
+        /* set iface & srate */
+        wm8988_write(codec, WM8988_IFACE, iface);
+        if (coeff >= 0)
+                wm8988_write(codec, WM8988_SRATE, srate |
+                        (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
+
+        return 0;
+}
+
+static int wm8988_mute(struct snd_soc_dai *dai, int mute)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7;
+
+        if (mute)
+                wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8);
+        else
+                wm8988_write(codec, WM8988_ADCDAC, mute_reg);
+        return 0;
+}
+
+static int wm8988_set_bias_level(struct snd_soc_codec *codec,
+                                 enum snd_soc_bias_level level)
+{
+        u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1;
+
+        switch (level) {
+        case SND_SOC_BIAS_ON:
+                break;
+
+        case SND_SOC_BIAS_PREPARE:
+                /* VREF, VMID=2x50k, digital enabled */
+                wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0);
+                break;
+
+        case SND_SOC_BIAS_STANDBY:
+                if (codec->bias_level == SND_SOC_BIAS_OFF) {
+                        /* VREF, VMID=2x5k */
+                        wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
+
+                        /* Charge caps */
+                        msleep(100);
+                }
+
+                /* VREF, VMID=2*500k, digital stopped */
+                wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141);
+                break;
+
+        case SND_SOC_BIAS_OFF:
+                wm8988_write(codec, WM8988_PWR1, 0x0000);
+                break;
+        }
+        codec->bias_level = level;
+        return 0;
+}
+
+#define WM8988_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+        SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8988_ops = {
+        .startup = wm8988_pcm_startup,
+        .hw_params = wm8988_pcm_hw_params,
+        .set_fmt = wm8988_set_dai_fmt,
+        .set_sysclk = wm8988_set_dai_sysclk,
+        .digital_mute = wm8988_mute,
+};
+
+struct snd_soc_dai wm8988_dai = {
+        .name = "WM8988",
+        .playback = {
+                .stream_name = "Playback",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = WM8988_RATES,
+                .formats = WM8988_FORMATS,
+        },
+        .capture = {
+                .stream_name = "Capture",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = WM8988_RATES,
+                .formats = WM8988_FORMATS,
+         },
+        .ops = &wm8988_ops,
+        .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8988_dai);
+
+static int wm8988_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;
+
+        wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF);
+        return 0;
+}
+
+static int wm8988_resume(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+        int i;
+        u8 data[2];
+        u16 *cache = codec->reg_cache;
+
+        /* Sync reg_cache with the hardware */
+        for (i = 0; i < WM8988_NUM_REG; i++) {
+                if (i == WM8988_RESET)
+                        continue;
+                data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+                data[1] = cache[i] & 0x00ff;
+                codec->hw_write(codec->control_data, data, 2);
+        }
+
+        wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+        return 0;
+}
+
+static struct snd_soc_codec *wm8988_codec;
+
+static int wm8988_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec;
+        int ret = 0;
+
+        if (wm8988_codec == NULL) {
+                dev_err(&pdev->dev, "Codec device not registered\n");
+                return -ENODEV;
+        }
+
+        socdev->card->codec = wm8988_codec;
+        codec = wm8988_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, wm8988_snd_controls,
+                                ARRAY_SIZE(wm8988_snd_controls));
+        snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets,
+                                  ARRAY_SIZE(wm8988_dapm_widgets));
+        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+        snd_soc_dapm_new_widgets(codec);
+
+        ret = snd_soc_init_card(socdev);
+        if (ret < 0) {
+                dev_err(codec->dev, "failed to register card: %d\n", ret);
+                goto card_err;
+        }
+
+        return ret;
+
+card_err:
+        snd_soc_free_pcms(socdev);
+        snd_soc_dapm_free(socdev);
+pcm_err:
+        return ret;
+}
+
+static int wm8988_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_wm8988 = {
+        .probe =        wm8988_probe,
+        .remove =       wm8988_remove,
+        .suspend =      wm8988_suspend,
+        .resume =       wm8988_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988);
+
+static int wm8988_register(struct wm8988_priv *wm8988)
+{
+        struct snd_soc_codec *codec = &wm8988->codec;
+        int ret;
+        u16 reg;
+
+        if (wm8988_codec) {
+                dev_err(codec->dev, "Another WM8988 is registered\n");
+                ret = -EINVAL;
+                goto err;
+        }
+
+        mutex_init(&codec->mutex);
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        codec->private_data = wm8988;
+        codec->name = "WM8988";
+        codec->owner = THIS_MODULE;
+        codec->read = wm8988_read_reg_cache;
+        codec->write = wm8988_write;
+        codec->dai = &wm8988_dai;
+        codec->num_dai = 1;
+        codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache);
+        codec->reg_cache = &wm8988->reg_cache;
+        codec->bias_level = SND_SOC_BIAS_OFF;
+        codec->set_bias_level = wm8988_set_bias_level;
+
+        memcpy(codec->reg_cache, wm8988_reg,
+               sizeof(wm8988_reg));
+
+        ret = wm8988_reset(codec);
+        if (ret < 0) {
+                dev_err(codec->dev, "Failed to issue reset\n");
+                return ret;
+        }
+
+        /* set the update bits (we always update left then right) */
+        reg = wm8988_read_reg_cache(codec, WM8988_RADC);
+        wm8988_write(codec, WM8988_RADC, reg | 0x100);
+        reg = wm8988_read_reg_cache(codec, WM8988_RDAC);
+        wm8988_write(codec, WM8988_RDAC, reg | 0x0100);
+        reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V);
+        wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100);
+        reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V);
+        wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100);
+        reg = wm8988_read_reg_cache(codec, WM8988_RINVOL);
+        wm8988_write(codec, WM8988_RINVOL, reg | 0x0100);
+
+        wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY);
+
+        wm8988_dai.dev = codec->dev;
+
+        wm8988_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(&wm8988_dai);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+                snd_soc_unregister_codec(codec);
+                return ret;
+        }
+
+        return 0;
+
+err:
+        kfree(wm8988);
+        return ret;
+}
+
+static void wm8988_unregister(struct wm8988_priv *wm8988)
+{
+        wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_OFF);
+        snd_soc_unregister_dai(&wm8988_dai);
+        snd_soc_unregister_codec(&wm8988->codec);
+        kfree(wm8988);
+        wm8988_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static int wm8988_i2c_probe(struct i2c_client *i2c,
+                            const struct i2c_device_id *id)
+{
+        struct wm8988_priv *wm8988;
+        struct snd_soc_codec *codec;
+
+        wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL);
+        if (wm8988 == NULL)
+                return -ENOMEM;
+
+        codec = &wm8988->codec;
+        codec->hw_write = (hw_write_t)i2c_master_send;
+
+        i2c_set_clientdata(i2c, wm8988);
+        codec->control_data = i2c;
+
+        codec->dev = &i2c->dev;
+
+        return wm8988_register(wm8988);
+}
+
+static int wm8988_i2c_remove(struct i2c_client *client)
+{
+        struct wm8988_priv *wm8988 = i2c_get_clientdata(client);
+        wm8988_unregister(wm8988);
+        return 0;
+}
+
+static const struct i2c_device_id wm8988_i2c_id[] = {
+        { "wm8988", 0 },
+        { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id);
+
+static struct i2c_driver wm8988_i2c_driver = {
+        .driver = {
+                .name = "WM8988",
+                .owner = THIS_MODULE,
+        },
+        .probe = wm8988_i2c_probe,
+        .remove = wm8988_i2c_remove,
+        .id_table = wm8988_i2c_id,
+};
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int wm8988_spi_write(struct spi_device *spi, const char *data, int len)
+{
+        struct spi_transfer t;
+        struct spi_message m;
+        u8 msg[2];
+
+        if (len <= 0)
+                return 0;
+
+        msg[0] = data[0];
+        msg[1] = data[1];
+
+        spi_message_init(&m);
+        memset(&t, 0, (sizeof t));
+
+        t.tx_buf = &msg[0];
+        t.len = len;
+
+        spi_message_add_tail(&t, &m);
+        spi_sync(spi, &m);
+
+        return len;
+}
+
+static int __devinit wm8988_spi_probe(struct spi_device *spi)
+{
+        struct wm8988_priv *wm8988;
+        struct snd_soc_codec *codec;
+
+        wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL);
+        if (wm8988 == NULL)
+                return -ENOMEM;
+
+        codec = &wm8988->codec;
+        codec->hw_write = (hw_write_t)wm8988_spi_write;
+        codec->control_data = spi;
+        codec->dev = &spi->dev;
+
+        spi->dev.driver_data = wm8988;
+
+        return wm8988_register(wm8988);
+}
+
+static int __devexit wm8988_spi_remove(struct spi_device *spi)
+{
+        struct wm8988_priv *wm8988 = spi->dev.driver_data;
+
+        wm8988_unregister(wm8988);
+
+        return 0;
+}
+
+static struct spi_driver wm8988_spi_driver = {
+        .driver = {
+                .name   = "wm8988",
+                .bus    = &spi_bus_type,
+                .owner  = THIS_MODULE,
+        },
+        .probe          = wm8988_spi_probe,
+        .remove         = __devexit_p(wm8988_spi_remove),
+};
+#endif
+
+static int __init wm8988_modinit(void)
+{
+        int ret;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+        ret = i2c_add_driver(&wm8988_i2c_driver);
+        if (ret != 0)
+                pr_err("WM8988: Unable to register I2C driver: %d\n", ret);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+        ret = spi_register_driver(&wm8988_spi_driver);
+        if (ret != 0)
+                pr_err("WM8988: Unable to register SPI driver: %d\n", ret);
+#endif
+        return ret;
+}
+module_init(wm8988_modinit);
+
+static void __exit wm8988_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+        i2c_del_driver(&wm8988_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+        spi_unregister_driver(&wm8988_spi_driver);
+#endif
+}
+module_exit(wm8988_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8988 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8988.h b/sound/soc/codecs/wm8988.h
new file mode 100644
index 000000000000..4552d37fdd41
--- /dev/null
+++ b/sound/soc/codecs/wm8988.h
@@ -0,0 +1,60 @@
+/*
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#ifndef _WM8988_H
+#define _WM8988_H
+
+/* WM8988 register space */
+
+#define WM8988_LINVOL    0x00
+#define WM8988_RINVOL    0x01
+#define WM8988_LOUT1V    0x02
+#define WM8988_ROUT1V    0x03
+#define WM8988_ADCDAC    0x05
+#define WM8988_IFACE     0x07
+#define WM8988_SRATE     0x08
+#define WM8988_LDAC      0x0a
+#define WM8988_RDAC      0x0b
+#define WM8988_BASS      0x0c
+#define WM8988_TREBLE    0x0d
+#define WM8988_RESET     0x0f
+#define WM8988_3D        0x10
+#define WM8988_ALC1      0x11
+#define WM8988_ALC2      0x12
+#define WM8988_ALC3      0x13
+#define WM8988_NGATE     0x14
+#define WM8988_LADC      0x15
+#define WM8988_RADC      0x16
+#define WM8988_ADCTL1    0x17
+#define WM8988_ADCTL2    0x18
+#define WM8988_PWR1      0x19
+#define WM8988_PWR2      0x1a
+#define WM8988_ADCTL3    0x1b
+#define WM8988_ADCIN     0x1f
+#define WM8988_LADCIN    0x20
+#define WM8988_RADCIN    0x21
+#define WM8988_LOUTM1    0x22
+#define WM8988_LOUTM2    0x23
+#define WM8988_ROUTM1    0x24
+#define WM8988_ROUTM2    0x25
+#define WM8988_LOUT2V    0x28
+#define WM8988_ROUT2V    0x29
+#define WM8988_LPPB      0x43
+#define WM8988_NUM_REG   0x44
+
+#define WM8988_SYSCLK   0
+
+extern struct snd_soc_dai wm8988_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8988;
+
+#endif
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index c518c3e5aa3f..d029818350e9 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_PWR_BIT, 0,
         inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINLMUX */
-SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0,
         &wm8990_dapm_ainlmux_controls, inmixer_event,
         SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_PWR_BIT, 0,
         inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
 /* AINRMUX */
-SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
+SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0,
         &wm8990_dapm_ainrmux_controls, inmixer_event,
         SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
@@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = {
         {"LIN12 PGA", "LIN2 Switch", "LIN2"},
         /* LIN34 PGA */
         {"LIN34 PGA", "LIN3 Switch", "LIN3"},
-        {"LIN34 PGA", "LIN4 Switch", "LIN4"},
+        {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
         /* INMIXL */
         {"INMIXL", "Record Left Volume", "LOMIX"},
         {"INMIXL", "LIN2 Volume", "LIN2"},
         {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
         {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
-        /* AILNMUX */
-        {"AILNMUX", "INMIXL Mix", "INMIXL"},
-        {"AILNMUX", "DIFFINL Mix", "LIN12PGA"},
-        {"AILNMUX", "DIFFINL Mix", "LIN34PGA"},
-        {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
-        {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+        /* AINLMUX */
+        {"AINLMUX", "INMIXL Mix", "INMIXL"},
+        {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"},
+        {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"},
+        {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"},
+        {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"},
         /* ADC */
-        {"Left ADC", NULL, "AILNMUX"},
+        {"Left ADC", NULL, "AINLMUX"},
 
         /* RIN12 PGA */
         {"RIN12 PGA", "RIN1 Switch", "RIN1"},
         {"RIN12 PGA", "RIN2 Switch", "RIN2"},
         /* RIN34 PGA */
         {"RIN34 PGA", "RIN3 Switch", "RIN3"},
-        {"RIN34 PGA", "RIN4 Switch", "RIN4"},
+        {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
         /* INMIXL */
         {"INMIXR", "Record Right Volume", "ROMIX"},
         {"INMIXR", "RIN2 Volume", "RIN2"},
         {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
         {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
-        /* AIRNMUX */
-        {"AIRNMUX", "INMIXR Mix", "INMIXR"},
-        {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"},
-        {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"},
-        {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"},
-        {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+        /* AINRMUX */
+        {"AINRMUX", "INMIXR Mix", "INMIXR"},
+        {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"},
+        {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"},
+        {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"},
+        {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"},
         /* ADC */
-        {"Right ADC", NULL, "AIRNMUX"},
+        {"Right ADC", NULL, "AINRMUX"},
 
         /* LOMIX */
         {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
@@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
         {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
 
         /* OUT3MIX */
-        {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"},
+        {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
         {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
 
         /* OUT4MIX */
@@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
         /* Output Pins */
         {"LON", NULL, "LONMIX"},
         {"LOP", NULL, "LOPMIX"},
-        {"OUT", NULL, "OUT3MIX"},
+        {"OUT3", NULL, "OUT3MIX"},
         {"LOUT", NULL, "LOUT PGA"},
         {"SPKN", NULL, "SPKMIX"},
         {"ROUT", NULL, "ROUT PGA"},
@@ -998,7 +998,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 
         if ((Ndiv < 6) || (Ndiv > 12))
                 printk(KERN_WARNING
-                "WM8990 N value outwith recommended range! N = %d\n", Ndiv);
+                "WM8990 N value outwith recommended range! N = %u\n", Ndiv);
 
         pll_div->n = Ndiv;
         Nmod = target % source;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
new file mode 100644
index 000000000000..86fc57e25f97
--- /dev/null
+++ b/sound/soc/codecs/wm9081.c
@@ -0,0 +1,1534 @@
+/*
+ * wm9081.c  --  WM9081 ALSA SoC Audio driver
+ *
+ * Author: Mark Brown
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <sound/wm9081.h>
+#include "wm9081.h"
+
+static u16 wm9081_reg_defaults[] = {
+        0x0000,     /* R0  - Software Reset */
+        0x0000,     /* R1 */
+        0x00B9,     /* R2  - Analogue Lineout */
+        0x00B9,     /* R3  - Analogue Speaker PGA */
+        0x0001,     /* R4  - VMID Control */
+        0x0068,     /* R5  - Bias Control 1 */
+        0x0000,     /* R6 */
+        0x0000,     /* R7  - Analogue Mixer */
+        0x0000,     /* R8  - Anti Pop Control */
+        0x01DB,     /* R9  - Analogue Speaker 1 */
+        0x0018,     /* R10 - Analogue Speaker 2 */
+        0x0180,     /* R11 - Power Management */
+        0x0000,     /* R12 - Clock Control 1 */
+        0x0038,     /* R13 - Clock Control 2 */
+        0x4000,     /* R14 - Clock Control 3 */
+        0x0000,     /* R15 */
+        0x0000,     /* R16 - FLL Control 1 */
+        0x0200,     /* R17 - FLL Control 2 */
+        0x0000,     /* R18 - FLL Control 3 */
+        0x0204,     /* R19 - FLL Control 4 */
+        0x0000,     /* R20 - FLL Control 5 */
+        0x0000,     /* R21 */
+        0x0000,     /* R22 - Audio Interface 1 */
+        0x0002,     /* R23 - Audio Interface 2 */
+        0x0008,     /* R24 - Audio Interface 3 */
+        0x0022,     /* R25 - Audio Interface 4 */
+        0x0000,     /* R26 - Interrupt Status */
+        0x0006,     /* R27 - Interrupt Status Mask */
+        0x0000,     /* R28 - Interrupt Polarity */
+        0x0000,     /* R29 - Interrupt Control */
+        0x00C0,     /* R30 - DAC Digital 1 */
+        0x0008,     /* R31 - DAC Digital 2 */
+        0x09AF,     /* R32 - DRC 1 */
+        0x4201,     /* R33 - DRC 2 */
+        0x0000,     /* R34 - DRC 3 */
+        0x0000,     /* R35 - DRC 4 */
+        0x0000,     /* R36 */
+        0x0000,     /* R37 */
+        0x0000,     /* R38 - Write Sequencer 1 */
+        0x0000,     /* R39 - Write Sequencer 2 */
+        0x0002,     /* R40 - MW Slave 1 */
+        0x0000,     /* R41 */
+        0x0000,     /* R42 - EQ 1 */
+        0x0000,     /* R43 - EQ 2 */
+        0x0FCA,     /* R44 - EQ 3 */
+        0x0400,     /* R45 - EQ 4 */
+        0x00B8,     /* R46 - EQ 5 */
+        0x1EB5,     /* R47 - EQ 6 */
+        0xF145,     /* R48 - EQ 7 */
+        0x0B75,     /* R49 - EQ 8 */
+        0x01C5,     /* R50 - EQ 9 */
+        0x169E,     /* R51 - EQ 10 */
+        0xF829,     /* R52 - EQ 11 */
+        0x07AD,     /* R53 - EQ 12 */
+        0x1103,     /* R54 - EQ 13 */
+        0x1C58,     /* R55 - EQ 14 */
+        0xF373,     /* R56 - EQ 15 */
+        0x0A54,     /* R57 - EQ 16 */
+        0x0558,     /* R58 - EQ 17 */
+        0x0564,     /* R59 - EQ 18 */
+        0x0559,     /* R60 - EQ 19 */
+        0x4000,     /* R61 - EQ 20 */
+};
+
+static struct {
+        int ratio;
+        int clk_sys_rate;
+} clk_sys_rates[] = {
+        { 64,   0 },
+        { 128,  1 },
+        { 192,  2 },
+        { 256,  3 },
+        { 384,  4 },
+        { 512,  5 },
+        { 768,  6 },
+        { 1024, 7 },
+        { 1408, 8 },
+        { 1536, 9 },
+};
+
+static struct {
+        int rate;
+        int sample_rate;
+} sample_rates[] = {
+        { 8000,  0  },
+        { 11025, 1  },
+        { 12000, 2  },
+        { 16000, 3  },
+        { 22050, 4  },
+        { 24000, 5  },
+        { 32000, 6  },
+        { 44100, 7  },
+        { 48000, 8  },
+        { 88200, 9  },
+        { 96000, 10 },
+};
+
+static struct {
+        int div; /* *10 due to .5s */
+        int bclk_div;
+} bclk_divs[] = {
+        { 10,  0  },
+        { 15,  1  },
+        { 20,  2  },
+        { 30,  3  },
+        { 40,  4  },
+        { 50,  5  },
+        { 55,  6  },
+        { 60,  7  },
+        { 80,  8  },
+        { 100, 9  },
+        { 110, 10 },
+        { 120, 11 },
+        { 160, 12 },
+        { 200, 13 },
+        { 220, 14 },
+        { 240, 15 },
+        { 250, 16 },
+        { 300, 17 },
+        { 320, 18 },
+        { 440, 19 },
+        { 480, 20 },
+};
+
+struct wm9081_priv {
+        struct snd_soc_codec codec;
+        u16 reg_cache[WM9081_MAX_REGISTER + 1];
+        int sysclk_source;
+        int mclk_rate;
+        int sysclk_rate;
+        int fs;
+        int bclk;
+        int master;
+        int fll_fref;
+        int fll_fout;
+        struct wm9081_retune_mobile_config *retune;
+};
+
+static int wm9081_reg_is_volatile(int reg)
+{
+        switch (reg) {
+        default:
+                return 0;
+        }
+}
+
+static unsigned int wm9081_read_reg_cache(struct snd_soc_codec *codec,
+                                          unsigned int reg)
+{
+        u16 *cache = codec->reg_cache;
+        BUG_ON(reg > WM9081_MAX_REGISTER);
+        return cache[reg];
+}
+
+static unsigned int wm9081_read_hw(struct snd_soc_codec *codec, u8 reg)
+{
+        struct i2c_msg xfer[2];
+        u16 data;
+        int ret;
+        struct i2c_client *client = codec->control_data;
+
+        BUG_ON(reg > WM9081_MAX_REGISTER);
+
+        /* Write register */
+        xfer[0].addr = client->addr;
+        xfer[0].flags = 0;
+        xfer[0].len = 1;
+        xfer[0].buf = &reg;
+
+        /* Read data */
+        xfer[1].addr = client->addr;
+        xfer[1].flags = I2C_M_RD;
+        xfer[1].len = 2;
+        xfer[1].buf = (u8 *)&data;
+
+        ret = i2c_transfer(client->adapter, xfer, 2);
+        if (ret != 2) {
+                dev_err(&client->dev, "i2c_transfer() returned %d\n", ret);
+                return 0;
+        }
+
+        return (data >> 8) | ((data & 0xff) << 8);
+}
+
+static unsigned int wm9081_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+        if (wm9081_reg_is_volatile(reg))
+                return wm9081_read_hw(codec, reg);
+        else
+                return wm9081_read_reg_cache(codec, reg);
+}
+
+static int wm9081_write(struct snd_soc_codec *codec, unsigned int reg,
+                        unsigned int value)
+{
+        u16 *cache = codec->reg_cache;
+        u8 data[3];
+
+        BUG_ON(reg > WM9081_MAX_REGISTER);
+
+        if (!wm9081_reg_is_volatile(reg))
+                cache[reg] = value;
+
+        data[0] = reg;
+        data[1] = value >> 8;
+        data[2] = value & 0x00ff;
+
+        if (codec->hw_write(codec->control_data, data, 3) == 3)
+                return 0;
+        else
+                return -EIO;
+}
+
+static int wm9081_reset(struct snd_soc_codec *codec)
+{
+        return wm9081_write(codec, WM9081_SOFTWARE_RESET, 0);
+}
+
+static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_out_tlv, -2250, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
+static unsigned int drc_max_tlv[] = {
+        TLV_DB_RANGE_HEAD(4),
+        0, 0, TLV_DB_SCALE_ITEM(1200, 0, 0),
+        1, 1, TLV_DB_SCALE_ITEM(1800, 0, 0),
+        2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
+        3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(drc_qr_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(drc_startup_tlv, -300, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+
+static const DECLARE_TLV_DB_SCALE(in_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
+
+static const char *drc_high_text[] = {
+        "1",
+        "1/2",
+        "1/4",
+        "1/8",
+        "1/16",
+        "0",
+};
+
+static const struct soc_enum drc_high =
+        SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text);
+
+static const char *drc_low_text[] = {
+        "1",
+        "1/2",
+        "1/4",
+        "1/8",
+        "0",
+};
+
+static const struct soc_enum drc_low =
+        SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text);
+
+static const char *drc_atk_text[] = {
+        "181us",
+        "181us",
+        "363us",
+        "726us",
+        "1.45ms",
+        "2.9ms",
+        "5.8ms",
+        "11.6ms",
+        "23.2ms",
+        "46.4ms",
+        "92.8ms",
+        "185.6ms",
+};
+
+static const struct soc_enum drc_atk =
+        SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text);
+
+static const char *drc_dcy_text[] = {
+        "186ms",
+        "372ms",
+        "743ms",
+        "1.49s",
+        "2.97s",
+        "5.94s",
+        "11.89s",
+        "23.78s",
+        "47.56s",
+};
+
+static const struct soc_enum drc_dcy =
+        SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text);
+
+static const char *drc_qr_dcy_text[] = {
+        "0.725ms",
+        "1.45ms",
+        "5.8ms",
+};
+
+static const struct soc_enum drc_qr_dcy =
+        SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text);
+
+static const char *dac_deemph_text[] = {
+        "None",
+        "32kHz",
+        "44.1kHz",
+        "48kHz",
+};
+
+static const struct soc_enum dac_deemph =
+        SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text);
+
+static const char *speaker_mode_text[] = {
+        "Class D",
+        "Class AB",
+};
+
+static const struct soc_enum speaker_mode =
+        SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text);
+
+static int speaker_mode_get(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        unsigned int reg;
+
+        reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+        if (reg & WM9081_SPK_MODE)
+                ucontrol->value.integer.value[0] = 1;
+        else
+                ucontrol->value.integer.value[0] = 0;
+
+        return 0;
+}
+
+/*
+ * Stop any attempts to change speaker mode while the speaker is enabled.
+ *
+ * We also have some special anti-pop controls dependant on speaker
+ * mode which must be changed along with the mode.
+ */
+static int speaker_mode_put(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+        unsigned int reg_pwr = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+        unsigned int reg2 = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+
+        /* Are we changing anything? */
+        if (ucontrol->value.integer.value[0] ==
+            ((reg2 & WM9081_SPK_MODE) != 0))
+                return 0;
+
+        /* Don't try to change modes while enabled */
+        if (reg_pwr & WM9081_SPK_ENA)
+                return -EINVAL;
+
+        if (ucontrol->value.integer.value[0]) {
+                /* Class AB */
+                reg2 &= ~(WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL);
+                reg2 |= WM9081_SPK_MODE;
+        } else {
+                /* Class D */
+                reg2 |= WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL;
+                reg2 &= ~WM9081_SPK_MODE;
+        }
+
+        wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2);
+
+        return 0;
+}
+
+static const struct snd_kcontrol_new wm9081_snd_controls[] = {
+SOC_SINGLE_TLV("IN1 Volume", WM9081_ANALOGUE_MIXER, 1, 1, 1, in_tlv),
+SOC_SINGLE_TLV("IN2 Volume", WM9081_ANALOGUE_MIXER, 3, 1, 1, in_tlv),
+
+SOC_SINGLE_TLV("Playback Volume", WM9081_DAC_DIGITAL_1, 1, 96, 0, dac_tlv),
+
+SOC_SINGLE("LINEOUT Switch", WM9081_ANALOGUE_LINEOUT, 7, 1, 1),
+SOC_SINGLE("LINEOUT ZC Switch", WM9081_ANALOGUE_LINEOUT, 6, 1, 0),
+SOC_SINGLE_TLV("LINEOUT Volume", WM9081_ANALOGUE_LINEOUT, 0, 63, 0, out_tlv),
+
+SOC_SINGLE("DRC Switch", WM9081_DRC_1, 15, 1, 0),
+SOC_ENUM("DRC High Slope", drc_high),
+SOC_ENUM("DRC Low Slope", drc_low),
+SOC_SINGLE_TLV("DRC Input Volume", WM9081_DRC_4, 5, 60, 1, drc_in_tlv),
+SOC_SINGLE_TLV("DRC Output Volume", WM9081_DRC_4, 0, 30, 1, drc_out_tlv),
+SOC_SINGLE_TLV("DRC Minimum Volume", WM9081_DRC_2, 2, 3, 1, drc_min_tlv),
+SOC_SINGLE_TLV("DRC Maximum Volume", WM9081_DRC_2, 0, 3, 0, drc_max_tlv),
+SOC_ENUM("DRC Attack", drc_atk),
+SOC_ENUM("DRC Decay", drc_dcy),
+SOC_SINGLE("DRC Quick Release Switch", WM9081_DRC_1, 2, 1, 0),
+SOC_SINGLE_TLV("DRC Quick Release Volume", WM9081_DRC_2, 6, 3, 0, drc_qr_tlv),
+SOC_ENUM("DRC Quick Release Decay", drc_qr_dcy),
+SOC_SINGLE_TLV("DRC Startup Volume", WM9081_DRC_1, 6, 18, 0, drc_startup_tlv),
+
+SOC_SINGLE("EQ Switch", WM9081_EQ_1, 0, 1, 0),
+
+SOC_SINGLE("Speaker DC Volume", WM9081_ANALOGUE_SPEAKER_1, 3, 5, 0),
+SOC_SINGLE("Speaker AC Volume", WM9081_ANALOGUE_SPEAKER_1, 0, 5, 0),
+SOC_SINGLE("Speaker Switch", WM9081_ANALOGUE_SPEAKER_PGA, 7, 1, 1),
+SOC_SINGLE("Speaker ZC Switch", WM9081_ANALOGUE_SPEAKER_PGA, 6, 1, 0),
+SOC_SINGLE_TLV("Speaker Volume", WM9081_ANALOGUE_SPEAKER_PGA, 0, 63, 0,
+               out_tlv),
+SOC_ENUM("DAC Deemphasis", dac_deemph),
+SOC_ENUM_EXT("Speaker Mode", speaker_mode, speaker_mode_get, speaker_mode_put),
+};
+
+static const struct snd_kcontrol_new wm9081_eq_controls[] = {
+SOC_SINGLE_TLV("EQ1 Volume", WM9081_EQ_1, 11, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 Volume", WM9081_EQ_1, 6, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 Volume", WM9081_EQ_1, 1, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 Volume", WM9081_EQ_2, 11, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ5 Volume", WM9081_EQ_2, 6, 24, 0, eq_tlv),
+};
+
+static const struct snd_kcontrol_new mixer[] = {
+SOC_DAPM_SINGLE("IN1 Switch", WM9081_ANALOGUE_MIXER, 0, 1, 0),
+SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0),
+};
+
+static int speaker_event(struct snd_soc_dapm_widget *w,
+                         struct snd_kcontrol *kcontrol, int event)
+{
+        struct snd_soc_codec *codec = w->codec;
+        unsigned int reg = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+
+        switch (event) {
+        case SND_SOC_DAPM_POST_PMU:
+                reg |= WM9081_SPK_ENA;
+                break;
+
+        case SND_SOC_DAPM_PRE_PMD:
+                reg &= ~WM9081_SPK_ENA;
+                break;
+        }
+
+        wm9081_write(codec, WM9081_POWER_MANAGEMENT, reg);
+
+        return 0;
+}
+
+struct _fll_div {
+        u16 fll_fratio;
+        u16 fll_outdiv;
+        u16 fll_clk_ref_div;
+        u16 n;
+        u16 k;
+};
+
+/* The size in bits of the FLL divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static struct {
+        unsigned int min;
+        unsigned int max;
+        u16 fll_fratio;
+        int ratio;
+} fll_fratios[] = {
+        {       0,    64000, 4, 16 },
+        {   64000,   128000, 3,  8 },
+        {  128000,   256000, 2,  4 },
+        {  256000,  1000000, 1,  2 },
+        { 1000000, 13500000, 0,  1 },
+};
+
+static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
+                       unsigned int Fout)
+{
+        u64 Kpart;
+        unsigned int K, Ndiv, Nmod, target;
+        unsigned int div;
+        int i;
+
+        /* Fref must be <=13.5MHz */
+        div = 1;
+        while ((Fref / div) > 13500000) {
+                div *= 2;
+
+                if (div > 8) {
+                        pr_err("Can't scale %dMHz input down to <=13.5MHz\n",
+                               Fref);
+                        return -EINVAL;
+                }
+        }
+        fll_div->fll_clk_ref_div = div / 2;
+
+        pr_debug("Fref=%u Fout=%u\n", Fref, Fout);
+
+        /* Apply the division for our remaining calculations */
+        Fref /= div;
+
+        /* Fvco should be 90-100MHz; don't check the upper bound */
+        div = 0;
+        target = Fout * 2;
+        while (target < 90000000) {
+                div++;
+                target *= 2;
+                if (div > 7) {
+                        pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n",
+                               Fout);
+                        return -EINVAL;
+                }
+        }
+        fll_div->fll_outdiv = div;
+
+        pr_debug("Fvco=%dHz\n", target);
+
+        /* Find an appropraite FLL_FRATIO and factor it out of the target */
+        for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+                if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+                        fll_div->fll_fratio = fll_fratios[i].fll_fratio;
+                        target /= fll_fratios[i].ratio;
+                        break;
+                }
+        }
+        if (i == ARRAY_SIZE(fll_fratios)) {
+                pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref);
+                return -EINVAL;
+        }
+
+        /* Now, calculate N.K */
+        Ndiv = target / Fref;
+
+        fll_div->n = Ndiv;
+        Nmod = target % Fref;
+        pr_debug("Nmod=%d\n", Nmod);
+
+        /* Calculate fractional part - scale up so we can round. */
+        Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+        do_div(Kpart, Fref);
+
+        K = Kpart & 0xFFFFFFFF;
+
+        if ((K % 10) >= 5)
+                K += 5;
+
+        /* Move down to proper range now rounding is done */
+        fll_div->k = K / 10;
+
+        pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n",
+                 fll_div->n, fll_div->k,
+                 fll_div->fll_fratio, fll_div->fll_outdiv,
+                 fll_div->fll_clk_ref_div);
+
+        return 0;
+}
+
+static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
+                          unsigned int Fref, unsigned int Fout)
+{
+        struct wm9081_priv *wm9081 = codec->private_data;
+        u16 reg1, reg4, reg5;
+        struct _fll_div fll_div;
+        int ret;
+        int clk_sys_reg;
+
+        /* Any change? */
+        if (Fref == wm9081->fll_fref && Fout == wm9081->fll_fout)
+                return 0;
+
+        /* Disable the FLL */
+        if (Fout == 0) {
+                dev_dbg(codec->dev, "FLL disabled\n");
+                wm9081->fll_fref = 0;
+                wm9081->fll_fout = 0;
+
+                return 0;
+        }
+
+        ret = fll_factors(&fll_div, Fref, Fout);
+        if (ret != 0)
+                return ret;
+
+        reg5 = wm9081_read(codec, WM9081_FLL_CONTROL_5);
+        reg5 &= ~WM9081_FLL_CLK_SRC_MASK;
+
+        switch (fll_id) {
+        case WM9081_SYSCLK_FLL_MCLK:
+                reg5 |= 0x1;
+                break;
+
+        default:
+                dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id);
+                return -EINVAL;
+        }
+
+        /* Disable CLK_SYS while we reconfigure */
+        clk_sys_reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+        if (clk_sys_reg & WM9081_CLK_SYS_ENA)
+                wm9081_write(codec, WM9081_CLOCK_CONTROL_3,
+                             clk_sys_reg & ~WM9081_CLK_SYS_ENA);
+
+        /* Any FLL configuration change requires that the FLL be
+         * disabled first. */
+        reg1 = wm9081_read(codec, WM9081_FLL_CONTROL_1);
+        reg1 &= ~WM9081_FLL_ENA;
+        wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1);
+
+        /* Apply the configuration */
+        if (fll_div.k)
+                reg1 |= WM9081_FLL_FRAC_MASK;
+        else
+                reg1 &= ~WM9081_FLL_FRAC_MASK;
+        wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1);
+
+        wm9081_write(codec, WM9081_FLL_CONTROL_2,
+                     (fll_div.fll_outdiv << WM9081_FLL_OUTDIV_SHIFT) |
+                     (fll_div.fll_fratio << WM9081_FLL_FRATIO_SHIFT));
+        wm9081_write(codec, WM9081_FLL_CONTROL_3, fll_div.k);
+
+        reg4 = wm9081_read(codec, WM9081_FLL_CONTROL_4);
+        reg4 &= ~WM9081_FLL_N_MASK;
+        reg4 |= fll_div.n << WM9081_FLL_N_SHIFT;
+        wm9081_write(codec, WM9081_FLL_CONTROL_4, reg4);
+
+        reg5 &= ~WM9081_FLL_CLK_REF_DIV_MASK;
+        reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT;
+        wm9081_write(codec, WM9081_FLL_CONTROL_5, reg5);
+
+        /* Enable the FLL */
+        wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA);
+
+        /* Then bring CLK_SYS up again if it was disabled */
+        if (clk_sys_reg & WM9081_CLK_SYS_ENA)
+                wm9081_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg);
+
+        dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
+
+        wm9081->fll_fref = Fref;
+        wm9081->fll_fout = Fout;
+
+        return 0;
+}
+
+static int configure_clock(struct snd_soc_codec *codec)
+{
+        struct wm9081_priv *wm9081 = codec->private_data;
+        int new_sysclk, i, target;
+        unsigned int reg;
+        int ret = 0;
+        int mclkdiv = 0;
+        int fll = 0;
+
+        switch (wm9081->sysclk_source) {
+        case WM9081_SYSCLK_MCLK:
+                if (wm9081->mclk_rate > 12225000) {
+                        mclkdiv = 1;
+                        wm9081->sysclk_rate = wm9081->mclk_rate / 2;
+                } else {
+                        wm9081->sysclk_rate = wm9081->mclk_rate;
+                }
+                wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK, 0, 0);
+                break;
+
+        case WM9081_SYSCLK_FLL_MCLK:
+                /* If we have a sample rate calculate a CLK_SYS that
+                 * gives us a suitable DAC configuration, plus BCLK.
+                 * Ideally we would check to see if we can clock
+                 * directly from MCLK and only use the FLL if this is
+                 * not the case, though care must be taken with free
+                 * running mode.
+                 */
+                if (wm9081->master && wm9081->bclk) {
+                        /* Make sure we can generate CLK_SYS and BCLK
+                         * and that we've got 3MHz for optimal
+                         * performance. */
+                        for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) {
+                                target = wm9081->fs * clk_sys_rates[i].ratio;
+                                new_sysclk = target;
+                                if (target >= wm9081->bclk &&
+                                    target > 3000000)
+                                        break;
+                        }
+                } else if (wm9081->fs) {
+                        for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) {
+                                new_sysclk = clk_sys_rates[i].ratio
+                                        * wm9081->fs;
+                                if (new_sysclk > 3000000)
+                                        break;
+                        }
+                } else {
+                        new_sysclk = 12288000;
+                }
+
+                ret = wm9081_set_fll(codec, WM9081_SYSCLK_FLL_MCLK,
+                                     wm9081->mclk_rate, new_sysclk);
+                if (ret == 0) {
+                        wm9081->sysclk_rate = new_sysclk;
+
+                        /* Switch SYSCLK over to FLL */
+                        fll = 1;
+                } else {
+                        wm9081->sysclk_rate = wm9081->mclk_rate;
+                }
+                break;
+
+        default:
+                return -EINVAL;
+        }
+
+        reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_1);
+        if (mclkdiv)
+                reg |= WM9081_MCLKDIV2;
+        else
+                reg &= ~WM9081_MCLKDIV2;
+        wm9081_write(codec, WM9081_CLOCK_CONTROL_1, reg);
+
+        reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+        if (fll)
+                reg |= WM9081_CLK_SRC_SEL;
+        else
+                reg &= ~WM9081_CLK_SRC_SEL;
+        wm9081_write(codec, WM9081_CLOCK_CONTROL_3, reg);
+
+        dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm9081->sysclk_rate);
+
+        return ret;
+}
+
+static int clk_sys_event(struct snd_soc_dapm_widget *w,
+                         struct snd_kcontrol *kcontrol, int event)
+{
+        struct snd_soc_codec *codec = w->codec;
+        struct wm9081_priv *wm9081 = codec->private_data;
+
+        /* This should be done on init() for bypass paths */
+        switch (wm9081->sysclk_source) {
+        case WM9081_SYSCLK_MCLK:
+                dev_dbg(codec->dev, "Using %dHz MCLK\n", wm9081->mclk_rate);
+                break;
+        case WM9081_SYSCLK_FLL_MCLK:
+                dev_dbg(codec->dev, "Using %dHz MCLK with FLL\n",
+                        wm9081->mclk_rate);
+                break;
+        default:
+                dev_err(codec->dev, "System clock not configured\n");
+                return -EINVAL;
+        }
+
+        switch (event) {
+        case SND_SOC_DAPM_PRE_PMU:
+                configure_clock(codec);
+                break;
+
+        case SND_SOC_DAPM_POST_PMD:
+                /* Disable the FLL if it's running */
+                wm9081_set_fll(codec, 0, 0, 0);
+                break;
+        }
+
+        return 0;
+}
+
+static const struct snd_soc_dapm_widget wm9081_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("IN1"),
+SND_SOC_DAPM_INPUT("IN2"),
+
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM9081_POWER_MANAGEMENT, 0, 0),
+
+SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0,
+                             mixer, ARRAY_SIZE(mixer)),
+
+SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0,
+                   speaker_event,
+                   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+SND_SOC_DAPM_OUTPUT("LINEOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+
+SND_SOC_DAPM_SUPPLY("CLK_SYS", WM9081_CLOCK_CONTROL_3, 0, 0, clk_sys_event,
+                    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM9081_CLOCK_CONTROL_3, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0),
+};
+
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+        { "DAC", NULL, "CLK_SYS" },
+        { "DAC", NULL, "CLK_DSP" },
+
+        { "Mixer", "IN1 Switch", "IN1" },
+        { "Mixer", "IN2 Switch", "IN2" },
+        { "Mixer", "Playback Switch", "DAC" },
+
+        { "LINEOUT PGA", NULL, "Mixer" },
+        { "LINEOUT PGA", NULL, "TOCLK" },
+        { "LINEOUT PGA", NULL, "CLK_SYS" },
+
+        { "LINEOUT", NULL, "LINEOUT PGA" },
+
+        { "Speaker PGA", NULL, "Mixer" },
+        { "Speaker PGA", NULL, "TOCLK" },
+        { "Speaker PGA", NULL, "CLK_SYS" },
+
+        { "SPKN", NULL, "Speaker PGA" },
+        { "SPKP", NULL, "Speaker PGA" },
+};
+
+static int wm9081_set_bias_level(struct snd_soc_codec *codec,
+                                 enum snd_soc_bias_level level)
+{
+        u16 reg;
+
+        switch (level) {
+        case SND_SOC_BIAS_ON:
+                break;
+
+        case SND_SOC_BIAS_PREPARE:
+                /* VMID=2*40k */
+                reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+                reg &= ~WM9081_VMID_SEL_MASK;
+                reg |= 0x2;
+                wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+                /* Normal bias current */
+                reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+                reg &= ~WM9081_STBY_BIAS_ENA;
+                wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+                break;
+
+        case SND_SOC_BIAS_STANDBY:
+                /* Initial cold start */
+                if (codec->bias_level == SND_SOC_BIAS_OFF) {
+                        /* Disable LINEOUT discharge */
+                        reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL);
+                        reg &= ~WM9081_LINEOUT_DISCH;
+                        wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg);
+
+                        /* Select startup bias source */
+                        reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+                        reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA;
+                        wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+
+                        /* VMID 2*4k; Soft VMID ramp enable */
+                        reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+                        reg |= WM9081_VMID_RAMP | 0x6;
+                        wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+                        mdelay(100);
+
+                        /* Normal bias enable & soft start off */
+                        reg |= WM9081_BIAS_ENA;
+                        reg &= ~WM9081_VMID_RAMP;
+                        wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+                        /* Standard bias source */
+                        reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+                        reg &= ~WM9081_BIAS_SRC;
+                        wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+                }
+
+                /* VMID 2*240k */
+                reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+                reg &= ~WM9081_VMID_SEL_MASK;
+                reg |= 0x40;
+                wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+                /* Standby bias current on */
+                reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+                reg |= WM9081_STBY_BIAS_ENA;
+                wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+                break;
+
+        case SND_SOC_BIAS_OFF:
+                /* Startup bias source */
+                reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1);
+                reg |= WM9081_BIAS_SRC;
+                wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg);
+
+                /* Disable VMID and biases with soft ramping */
+                reg = wm9081_read(codec, WM9081_VMID_CONTROL);
+                reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA);
+                reg |= WM9081_VMID_RAMP;
+                wm9081_write(codec, WM9081_VMID_CONTROL, reg);
+
+                /* Actively discharge LINEOUT */
+                reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL);
+                reg |= WM9081_LINEOUT_DISCH;
+                wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg);
+                break;
+        }
+
+        codec->bias_level = level;
+
+        return 0;
+}
+
+static int wm9081_set_dai_fmt(struct snd_soc_dai *dai,
+                              unsigned int fmt)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        struct wm9081_priv *wm9081 = codec->private_data;
+        unsigned int aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2);
+
+        aif2 &= ~(WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV |
+                  WM9081_BCLK_DIR | WM9081_LRCLK_DIR | WM9081_AIF_FMT_MASK);
+
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBS_CFS:
+                wm9081->master = 0;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFM:
+                aif2 |= WM9081_LRCLK_DIR;
+                wm9081->master = 1;
+                break;
+        case SND_SOC_DAIFMT_CBM_CFS:
+                aif2 |= WM9081_BCLK_DIR;
+                wm9081->master = 1;
+                break;
+        case SND_SOC_DAIFMT_CBM_CFM:
+                aif2 |= WM9081_LRCLK_DIR | WM9081_BCLK_DIR;
+                wm9081->master = 1;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_DSP_B:
+                aif2 |= WM9081_AIF_LRCLK_INV;
+        case SND_SOC_DAIFMT_DSP_A:
+                aif2 |= 0x3;
+                break;
+        case SND_SOC_DAIFMT_I2S:
+                aif2 |= 0x2;
+                break;
+        case SND_SOC_DAIFMT_RIGHT_J:
+                break;
+        case SND_SOC_DAIFMT_LEFT_J:
+                aif2 |= 0x1;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        case SND_SOC_DAIFMT_DSP_A:
+        case SND_SOC_DAIFMT_DSP_B:
+                /* frame inversion not valid for DSP modes */
+                switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+                case SND_SOC_DAIFMT_NB_NF:
+                        break;
+                case SND_SOC_DAIFMT_IB_NF:
+                        aif2 |= WM9081_AIF_BCLK_INV;
+                        break;
+                default:
+                        return -EINVAL;
+                }
+                break;
+
+        case SND_SOC_DAIFMT_I2S:
+        case SND_SOC_DAIFMT_RIGHT_J:
+        case SND_SOC_DAIFMT_LEFT_J:
+                switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+                case SND_SOC_DAIFMT_NB_NF:
+                        break;
+                case SND_SOC_DAIFMT_IB_IF:
+                        aif2 |= WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV;
+                        break;
+                case SND_SOC_DAIFMT_IB_NF:
+                        aif2 |= WM9081_AIF_BCLK_INV;
+                        break;
+                case SND_SOC_DAIFMT_NB_IF:
+                        aif2 |= WM9081_AIF_LRCLK_INV;
+                        break;
+                default:
+                        return -EINVAL;
+                }
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
+
+        return 0;
+}
+
+static int wm9081_hw_params(struct snd_pcm_substream *substream,
+                            struct snd_pcm_hw_params *params,
+                            struct snd_soc_dai *dai)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        struct wm9081_priv *wm9081 = codec->private_data;
+        int ret, i, best, best_val, cur_val;
+        unsigned int clk_ctrl2, aif1, aif2, aif3, aif4;
+
+        clk_ctrl2 = wm9081_read(codec, WM9081_CLOCK_CONTROL_2);
+        clk_ctrl2 &= ~(WM9081_CLK_SYS_RATE_MASK | WM9081_SAMPLE_RATE_MASK);
+
+        aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1);
+
+        aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2);
+        aif2 &= ~WM9081_AIF_WL_MASK;
+
+        aif3 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_3);
+        aif3 &= ~WM9081_BCLK_DIV_MASK;
+
+        aif4 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_4);
+        aif4 &= ~WM9081_LRCLK_RATE_MASK;
+
+        /* What BCLK do we need? */
+        wm9081->fs = params_rate(params);
+        wm9081->bclk = 2 * wm9081->fs;
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                wm9081->bclk *= 16;
+                break;
+        case SNDRV_PCM_FORMAT_S20_3LE:
+                wm9081->bclk *= 20;
+                aif2 |= 0x4;
+                break;
+        case SNDRV_PCM_FORMAT_S24_LE:
+                wm9081->bclk *= 24;
+                aif2 |= 0x8;
+                break;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                wm9081->bclk *= 32;
+                aif2 |= 0xc;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        if (aif1 & WM9081_AIFDAC_TDM_MODE_MASK) {
+                int slots = ((aif1 & WM9081_AIFDAC_TDM_MODE_MASK) >>
+                             WM9081_AIFDAC_TDM_MODE_SHIFT) + 1;
+                wm9081->bclk *= slots;
+        }
+
+        dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm9081->bclk);
+
+        ret = configure_clock(codec);
+        if (ret != 0)
+                return ret;
+
+        /* Select nearest CLK_SYS_RATE */
+        best = 0;
+        best_val = abs((wm9081->sysclk_rate / clk_sys_rates[0].ratio)
+                       - wm9081->fs);
+        for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
+                cur_val = abs((wm9081->sysclk_rate /
+                               clk_sys_rates[i].ratio) - wm9081->fs);;
+                if (cur_val < best_val) {
+                        best = i;
+                        best_val = cur_val;
+                }
+        }
+        dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n",
+                clk_sys_rates[best].ratio);
+        clk_ctrl2 |= (clk_sys_rates[best].clk_sys_rate
+                      << WM9081_CLK_SYS_RATE_SHIFT);
+
+        /* SAMPLE_RATE */
+        best = 0;
+        best_val = abs(wm9081->fs - sample_rates[0].rate);
+        for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
+                /* Closest match */
+                cur_val = abs(wm9081->fs - sample_rates[i].rate);
+                if (cur_val < best_val) {
+                        best = i;
+                        best_val = cur_val;
+                }
+        }
+        dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n",
+                sample_rates[best].rate);
+        clk_ctrl2 |= (sample_rates[best].sample_rate
+                        << WM9081_SAMPLE_RATE_SHIFT);
+
+        /* BCLK_DIV */
+        best = 0;
+        best_val = INT_MAX;
+        for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
+                cur_val = ((wm9081->sysclk_rate * 10) / bclk_divs[i].div)
+                        - wm9081->bclk;
+                if (cur_val < 0) /* Table is sorted */
+                        break;
+                if (cur_val < best_val) {
+                        best = i;
+                        best_val = cur_val;
+                }
+        }
+        wm9081->bclk = (wm9081->sysclk_rate * 10) / bclk_divs[best].div;
+        dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n",
+                bclk_divs[best].div, wm9081->bclk);
+        aif3 |= bclk_divs[best].bclk_div;
+
+        /* LRCLK is a simple fraction of BCLK */
+        dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm9081->bclk / wm9081->fs);
+        aif4 |= wm9081->bclk / wm9081->fs;
+
+        /* Apply a ReTune Mobile configuration if it's in use */
+        if (wm9081->retune) {
+                struct wm9081_retune_mobile_config *retune = wm9081->retune;
+                struct wm9081_retune_mobile_setting *s;
+                int eq1;
+
+                best = 0;
+                best_val = abs(retune->configs[0].rate - wm9081->fs);
+                for (i = 0; i < retune->num_configs; i++) {
+                        cur_val = abs(retune->configs[i].rate - wm9081->fs);
+                        if (cur_val < best_val) {
+                                best_val = cur_val;
+                                best = i;
+                        }
+                }
+                s = &retune->configs[best];
+
+                dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n",
+                        s->name, s->rate);
+
+                /* If the EQ is enabled then disable it while we write out */
+                eq1 = wm9081_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA;
+                if (eq1 & WM9081_EQ_ENA)
+                        wm9081_write(codec, WM9081_EQ_1, 0);
+
+                /* Write out the other values */
+                for (i = 1; i < ARRAY_SIZE(s->config); i++)
+                        wm9081_write(codec, WM9081_EQ_1 + i, s->config[i]);
+
+                eq1 |= (s->config[0] & ~WM9081_EQ_ENA);
+                wm9081_write(codec, WM9081_EQ_1, eq1);
+        }
+
+        wm9081_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2);
+        wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2);
+        wm9081_write(codec, WM9081_AUDIO_INTERFACE_3, aif3);
+        wm9081_write(codec, WM9081_AUDIO_INTERFACE_4, aif4);
+
+        return 0;
+}
+
+static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        unsigned int reg;
+
+        reg = wm9081_read(codec, WM9081_DAC_DIGITAL_2);
+
+        if (mute)
+                reg |= WM9081_DAC_MUTE;
+        else
+                reg &= ~WM9081_DAC_MUTE;
+
+        wm9081_write(codec, WM9081_DAC_DIGITAL_2, reg);
+
+        return 0;
+}
+
+static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai,
+                             int clk_id, unsigned int freq, int dir)
+{
+        struct snd_soc_codec *codec = codec_dai->codec;
+        struct wm9081_priv *wm9081 = codec->private_data;
+
+        switch (clk_id) {
+        case WM9081_SYSCLK_MCLK:
+        case WM9081_SYSCLK_FLL_MCLK:
+                wm9081->sysclk_source = clk_id;
+                wm9081->mclk_rate = freq;
+                break;
+
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+static int wm9081_set_tdm_slot(struct snd_soc_dai *dai,
+                               unsigned int mask, int slots)
+{
+        struct snd_soc_codec *codec = dai->codec;
+        unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1);
+
+        aif1 &= ~(WM9081_AIFDAC_TDM_SLOT_MASK | WM9081_AIFDAC_TDM_MODE_MASK);
+
+        if (slots < 1 || slots > 4)
+                return -EINVAL;
+
+        aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT;
+
+        switch (mask) {
+        case 1:
+                break;
+        case 2:
+                aif1 |= 0x10;
+                break;
+        case 4:
+                aif1 |= 0x20;
+                break;
+        case 8:
+                aif1 |= 0x30;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        wm9081_write(codec, WM9081_AUDIO_INTERFACE_1, aif1);
+
+        return 0;
+}
+
+#define WM9081_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM9081_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 wm9081_dai_ops = {
+        .hw_params = wm9081_hw_params,
+        .set_sysclk = wm9081_set_sysclk,
+        .set_fmt = wm9081_set_dai_fmt,
+        .digital_mute = wm9081_digital_mute,
+        .set_tdm_slot = wm9081_set_tdm_slot,
+};
+
+/* We report two channels because the CODEC processes a stereo signal, even
+ * though it is only capable of handling a mono output.
+ */
+struct snd_soc_dai wm9081_dai = {
+        .name = "WM9081",
+        .playback = {
+                .stream_name = "HiFi Playback",
+                .channels_min = 1,
+                .channels_max = 2,
+                .rates = WM9081_RATES,
+                .formats = WM9081_FORMATS,
+        },
+        .ops = &wm9081_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm9081_dai);
+
+
+static struct snd_soc_codec *wm9081_codec;
+
+static int wm9081_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec;
+        struct wm9081_priv *wm9081;
+        int ret = 0;
+
+        if (wm9081_codec == NULL) {
+                dev_err(&pdev->dev, "Codec device not registered\n");
+                return -ENODEV;
+        }
+
+        socdev->card->codec = wm9081_codec;
+        codec = wm9081_codec;
+        wm9081 = codec->private_data;
+
+        /* 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, wm9081_snd_controls,
+                             ARRAY_SIZE(wm9081_snd_controls));
+        if (!wm9081->retune) {
+                dev_dbg(codec->dev,
+                        "No ReTune Mobile data, using normal EQ\n");
+                snd_soc_add_controls(codec, wm9081_eq_controls,
+                                     ARRAY_SIZE(wm9081_eq_controls));
+        }
+
+        snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets,
+                                  ARRAY_SIZE(wm9081_dapm_widgets));
+        snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+        snd_soc_dapm_new_widgets(codec);
+
+        ret = snd_soc_init_card(socdev);
+        if (ret < 0) {
+                dev_err(codec->dev, "failed to register card: %d\n", ret);
+                goto card_err;
+        }
+
+        return ret;
+
+card_err:
+        snd_soc_free_pcms(socdev);
+        snd_soc_dapm_free(socdev);
+pcm_err:
+        return ret;
+}
+
+static int wm9081_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
+static int wm9081_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;
+
+        wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+        return 0;
+}
+
+static int wm9081_resume(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct snd_soc_codec *codec = socdev->card->codec;
+        u16 *reg_cache = codec->reg_cache;
+        int i;
+
+        for (i = 0; i < codec->reg_cache_size; i++) {
+                if (i == WM9081_SOFTWARE_RESET)
+                        continue;
+
+                wm9081_write(codec, i, reg_cache[i]);
+        }
+
+        wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+        return 0;
+}
+#else
+#define wm9081_suspend NULL
+#define wm9081_resume NULL
+#endif
+
+struct snd_soc_codec_device soc_codec_dev_wm9081 = {
+        .probe =        wm9081_probe,
+        .remove =       wm9081_remove,
+        .suspend =      wm9081_suspend,
+        .resume =       wm9081_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9081);
+
+static int wm9081_register(struct wm9081_priv *wm9081)
+{
+        struct snd_soc_codec *codec = &wm9081->codec;
+        int ret;
+        u16 reg;
+
+        if (wm9081_codec) {
+                dev_err(codec->dev, "Another WM9081 is registered\n");
+                ret = -EINVAL;
+                goto err;
+        }
+
+        mutex_init(&codec->mutex);
+        INIT_LIST_HEAD(&codec->dapm_widgets);
+        INIT_LIST_HEAD(&codec->dapm_paths);
+
+        codec->private_data = wm9081;
+        codec->name = "WM9081";
+        codec->owner = THIS_MODULE;
+        codec->read = wm9081_read;
+        codec->write = wm9081_write;
+        codec->dai = &wm9081_dai;
+        codec->num_dai = 1;
+        codec->reg_cache_size = ARRAY_SIZE(wm9081->reg_cache);
+        codec->reg_cache = &wm9081->reg_cache;
+        codec->bias_level = SND_SOC_BIAS_OFF;
+        codec->set_bias_level = wm9081_set_bias_level;
+
+        memcpy(codec->reg_cache, wm9081_reg_defaults,
+               sizeof(wm9081_reg_defaults));
+
+        reg = wm9081_read_hw(codec, WM9081_SOFTWARE_RESET);
+        if (reg != 0x9081) {
+                dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg);
+                ret = -EINVAL;
+                goto err;
+        }
+
+        ret = wm9081_reset(codec);
+        if (ret < 0) {
+                dev_err(codec->dev, "Failed to issue reset\n");
+                return ret;
+        }
+
+        wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+        /* Enable zero cross by default */
+        reg = wm9081_read(codec, WM9081_ANALOGUE_LINEOUT);
+        wm9081_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC);
+        reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_PGA);
+        wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_PGA,
+                     reg | WM9081_SPKPGAZC);
+
+        wm9081_dai.dev = codec->dev;
+
+        wm9081_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(&wm9081_dai);
+        if (ret != 0) {
+                dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+                snd_soc_unregister_codec(codec);
+                return ret;
+        }
+
+        return 0;
+
+err:
+        kfree(wm9081);
+        return ret;
+}
+
+static void wm9081_unregister(struct wm9081_priv *wm9081)
+{
+        wm9081_set_bias_level(&wm9081->codec, SND_SOC_BIAS_OFF);
+        snd_soc_unregister_dai(&wm9081_dai);
+        snd_soc_unregister_codec(&wm9081->codec);
+        kfree(wm9081);
+        wm9081_codec = NULL;
+}
+
+static __devinit int wm9081_i2c_probe(struct i2c_client *i2c,
+                                      const struct i2c_device_id *id)
+{
+        struct wm9081_priv *wm9081;
+        struct snd_soc_codec *codec;
+
+        wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL);
+        if (wm9081 == NULL)
+                return -ENOMEM;
+
+        codec = &wm9081->codec;
+        codec->hw_write = (hw_write_t)i2c_master_send;
+        wm9081->retune = i2c->dev.platform_data;
+
+        i2c_set_clientdata(i2c, wm9081);
+        codec->control_data = i2c;
+
+        codec->dev = &i2c->dev;
+
+        return wm9081_register(wm9081);
+}
+
+static __devexit int wm9081_i2c_remove(struct i2c_client *client)
+{
+        struct wm9081_priv *wm9081 = i2c_get_clientdata(client);
+        wm9081_unregister(wm9081);
+        return 0;
+}
+
+static const struct i2c_device_id wm9081_i2c_id[] = {
+        { "wm9081", 0 },
+        { }
+};
+MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id);
+
+static struct i2c_driver wm9081_i2c_driver = {
+        .driver = {
+                .name = "wm9081",
+                .owner = THIS_MODULE,
+        },
+        .probe =    wm9081_i2c_probe,
+        .remove =   __devexit_p(wm9081_i2c_remove),
+        .id_table = wm9081_i2c_id,
+};
+
+static int __init wm9081_modinit(void)
+{
+        int ret;
+
+        ret = i2c_add_driver(&wm9081_i2c_driver);
+        if (ret != 0) {
+                printk(KERN_ERR "Failed to register WM9081 I2C driver: %d\n",
+                       ret);
+        }
+
+        return ret;
+}
+module_init(wm9081_modinit);
+
+static void __exit wm9081_exit(void)
+{
+        i2c_del_driver(&wm9081_i2c_driver);
+}
+module_exit(wm9081_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM9081 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm9081.h b/sound/soc/codecs/wm9081.h
new file mode 100644
index 000000000000..42d3bc757021
--- /dev/null
+++ b/sound/soc/codecs/wm9081.h
@@ -0,0 +1,787 @@
+#ifndef WM9081_H
+#define WM9081_H
+
+/*
+ * wm9081.c  --  WM9081 ALSA SoC Audio driver
+ *
+ * Author: Mark Brown
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm9081_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm9081;
+
+/*
+ * SYSCLK sources
+ */
+#define WM9081_SYSCLK_MCLK      1   /* Use MCLK without FLL */
+#define WM9081_SYSCLK_FLL_MCLK  2   /* Use MCLK, enabling FLL if required */
+
+/*
+ * Register values.
+ */
+#define WM9081_SOFTWARE_RESET                   0x00
+#define WM9081_ANALOGUE_LINEOUT                 0x02
+#define WM9081_ANALOGUE_SPEAKER_PGA             0x03
+#define WM9081_VMID_CONTROL                     0x04
+#define WM9081_BIAS_CONTROL_1                   0x05
+#define WM9081_ANALOGUE_MIXER                   0x07
+#define WM9081_ANTI_POP_CONTROL                 0x08
+#define WM9081_ANALOGUE_SPEAKER_1               0x09
+#define WM9081_ANALOGUE_SPEAKER_2               0x0A
+#define WM9081_POWER_MANAGEMENT                 0x0B
+#define WM9081_CLOCK_CONTROL_1                  0x0C
+#define WM9081_CLOCK_CONTROL_2                  0x0D
+#define WM9081_CLOCK_CONTROL_3                  0x0E
+#define WM9081_FLL_CONTROL_1                    0x10
+#define WM9081_FLL_CONTROL_2                    0x11
+#define WM9081_FLL_CONTROL_3                    0x12
+#define WM9081_FLL_CONTROL_4                    0x13
+#define WM9081_FLL_CONTROL_5                    0x14
+#define WM9081_AUDIO_INTERFACE_1                0x16
+#define WM9081_AUDIO_INTERFACE_2                0x17
+#define WM9081_AUDIO_INTERFACE_3                0x18
+#define WM9081_AUDIO_INTERFACE_4                0x19
+#define WM9081_INTERRUPT_STATUS                 0x1A
+#define WM9081_INTERRUPT_STATUS_MASK            0x1B
+#define WM9081_INTERRUPT_POLARITY               0x1C
+#define WM9081_INTERRUPT_CONTROL                0x1D
+#define WM9081_DAC_DIGITAL_1                    0x1E
+#define WM9081_DAC_DIGITAL_2                    0x1F
+#define WM9081_DRC_1                            0x20
+#define WM9081_DRC_2                            0x21
+#define WM9081_DRC_3                            0x22
+#define WM9081_DRC_4                            0x23
+#define WM9081_WRITE_SEQUENCER_1                0x26
+#define WM9081_WRITE_SEQUENCER_2                0x27
+#define WM9081_MW_SLAVE_1                       0x28
+#define WM9081_EQ_1                             0x2A
+#define WM9081_EQ_2                             0x2B
+#define WM9081_EQ_3                             0x2C
+#define WM9081_EQ_4                             0x2D
+#define WM9081_EQ_5                             0x2E
+#define WM9081_EQ_6                             0x2F
+#define WM9081_EQ_7                             0x30
+#define WM9081_EQ_8                             0x31
+#define WM9081_EQ_9                             0x32
+#define WM9081_EQ_10                            0x33
+#define WM9081_EQ_11                            0x34
+#define WM9081_EQ_12                            0x35
+#define WM9081_EQ_13                            0x36
+#define WM9081_EQ_14                            0x37
+#define WM9081_EQ_15                            0x38
+#define WM9081_EQ_16                            0x39
+#define WM9081_EQ_17                            0x3A
+#define WM9081_EQ_18                            0x3B
+#define WM9081_EQ_19                            0x3C
+#define WM9081_EQ_20                            0x3D
+
+#define WM9081_REGISTER_COUNT                   55
+#define WM9081_MAX_REGISTER                     0x3D
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * R0 (0x00) - Software Reset
+ */
+#define WM9081_SW_RST_DEV_ID1_MASK              0xFFFF  /* SW_RST_DEV_ID1 - [15:0] */
+#define WM9081_SW_RST_DEV_ID1_SHIFT                  0  /* SW_RST_DEV_ID1 - [15:0] */
+#define WM9081_SW_RST_DEV_ID1_WIDTH                 16  /* SW_RST_DEV_ID1 - [15:0] */
+
+/*
+ * R2 (0x02) - Analogue Lineout
+ */
+#define WM9081_LINEOUT_MUTE                     0x0080  /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_MASK                0x0080  /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_SHIFT                    7  /* LINEOUT_MUTE */
+#define WM9081_LINEOUT_MUTE_WIDTH                    1  /* LINEOUT_MUTE */
+#define WM9081_LINEOUTZC                        0x0040  /* LINEOUTZC */
+#define WM9081_LINEOUTZC_MASK                   0x0040  /* LINEOUTZC */
+#define WM9081_LINEOUTZC_SHIFT                       6  /* LINEOUTZC */
+#define WM9081_LINEOUTZC_WIDTH                       1  /* LINEOUTZC */
+#define WM9081_LINEOUT_VOL_MASK                 0x003F  /* LINEOUT_VOL - [5:0] */
+#define WM9081_LINEOUT_VOL_SHIFT                     0  /* LINEOUT_VOL - [5:0] */
+#define WM9081_LINEOUT_VOL_WIDTH                     6  /* LINEOUT_VOL - [5:0] */
+
+/*
+ * R3 (0x03) - Analogue Speaker PGA
+ */
+#define WM9081_SPKPGA_MUTE                      0x0080  /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_MASK                 0x0080  /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_SHIFT                     7  /* SPKPGA_MUTE */
+#define WM9081_SPKPGA_MUTE_WIDTH                     1  /* SPKPGA_MUTE */
+#define WM9081_SPKPGAZC                         0x0040  /* SPKPGAZC */
+#define WM9081_SPKPGAZC_MASK                    0x0040  /* SPKPGAZC */
+#define WM9081_SPKPGAZC_SHIFT                        6  /* SPKPGAZC */
+#define WM9081_SPKPGAZC_WIDTH                        1  /* SPKPGAZC */
+#define WM9081_SPKPGA_VOL_MASK                  0x003F  /* SPKPGA_VOL - [5:0] */
+#define WM9081_SPKPGA_VOL_SHIFT                      0  /* SPKPGA_VOL - [5:0] */
+#define WM9081_SPKPGA_VOL_WIDTH                      6  /* SPKPGA_VOL - [5:0] */
+
+/*
+ * R4 (0x04) - VMID Control
+ */
+#define WM9081_VMID_BUF_ENA                     0x0020  /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_MASK                0x0020  /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_SHIFT                    5  /* VMID_BUF_ENA */
+#define WM9081_VMID_BUF_ENA_WIDTH                    1  /* VMID_BUF_ENA */
+#define WM9081_VMID_RAMP                        0x0008  /* VMID_RAMP */
+#define WM9081_VMID_RAMP_MASK                   0x0008  /* VMID_RAMP */
+#define WM9081_VMID_RAMP_SHIFT                       3  /* VMID_RAMP */
+#define WM9081_VMID_RAMP_WIDTH                       1  /* VMID_RAMP */
+#define WM9081_VMID_SEL_MASK                    0x0006  /* VMID_SEL - [2:1] */
+#define WM9081_VMID_SEL_SHIFT                        1  /* VMID_SEL - [2:1] */
+#define WM9081_VMID_SEL_WIDTH                        2  /* VMID_SEL - [2:1] */
+#define WM9081_VMID_FAST_ST                     0x0001  /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_MASK                0x0001  /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_SHIFT                    0  /* VMID_FAST_ST */
+#define WM9081_VMID_FAST_ST_WIDTH                    1  /* VMID_FAST_ST */
+
+/*
+ * R5 (0x05) - Bias Control 1
+ */
+#define WM9081_BIAS_SRC                         0x0040  /* BIAS_SRC */
+#define WM9081_BIAS_SRC_MASK                    0x0040  /* BIAS_SRC */
+#define WM9081_BIAS_SRC_SHIFT                        6  /* BIAS_SRC */
+#define WM9081_BIAS_SRC_WIDTH                        1  /* BIAS_SRC */
+#define WM9081_STBY_BIAS_LVL                    0x0020  /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_MASK               0x0020  /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_SHIFT                   5  /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_LVL_WIDTH                   1  /* STBY_BIAS_LVL */
+#define WM9081_STBY_BIAS_ENA                    0x0010  /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_MASK               0x0010  /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_SHIFT                   4  /* STBY_BIAS_ENA */
+#define WM9081_STBY_BIAS_ENA_WIDTH                   1  /* STBY_BIAS_ENA */
+#define WM9081_BIAS_LVL_MASK                    0x000C  /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_LVL_SHIFT                        2  /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_LVL_WIDTH                        2  /* BIAS_LVL - [3:2] */
+#define WM9081_BIAS_ENA                         0x0002  /* BIAS_ENA */
+#define WM9081_BIAS_ENA_MASK                    0x0002  /* BIAS_ENA */
+#define WM9081_BIAS_ENA_SHIFT                        1  /* BIAS_ENA */
+#define WM9081_BIAS_ENA_WIDTH                        1  /* BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA                 0x0001  /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_MASK            0x0001  /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_SHIFT                0  /* STARTUP_BIAS_ENA */
+#define WM9081_STARTUP_BIAS_ENA_WIDTH                1  /* STARTUP_BIAS_ENA */
+
+/*
+ * R7 (0x07) - Analogue Mixer
+ */
+#define WM9081_DAC_SEL                          0x0010  /* DAC_SEL */
+#define WM9081_DAC_SEL_MASK                     0x0010  /* DAC_SEL */
+#define WM9081_DAC_SEL_SHIFT                         4  /* DAC_SEL */
+#define WM9081_DAC_SEL_WIDTH                         1  /* DAC_SEL */
+#define WM9081_IN2_VOL                          0x0008  /* IN2_VOL */
+#define WM9081_IN2_VOL_MASK                     0x0008  /* IN2_VOL */
+#define WM9081_IN2_VOL_SHIFT                         3  /* IN2_VOL */
+#define WM9081_IN2_VOL_WIDTH                         1  /* IN2_VOL */
+#define WM9081_IN2_ENA                          0x0004  /* IN2_ENA */
+#define WM9081_IN2_ENA_MASK                     0x0004  /* IN2_ENA */
+#define WM9081_IN2_ENA_SHIFT                         2  /* IN2_ENA */
+#define WM9081_IN2_ENA_WIDTH                         1  /* IN2_ENA */
+#define WM9081_IN1_VOL                          0x0002  /* IN1_VOL */
+#define WM9081_IN1_VOL_MASK                     0x0002  /* IN1_VOL */
+#define WM9081_IN1_VOL_SHIFT                         1  /* IN1_VOL */
+#define WM9081_IN1_VOL_WIDTH                         1  /* IN1_VOL */
+#define WM9081_IN1_ENA                          0x0001  /* IN1_ENA */
+#define WM9081_IN1_ENA_MASK                     0x0001  /* IN1_ENA */
+#define WM9081_IN1_ENA_SHIFT                         0  /* IN1_ENA */
+#define WM9081_IN1_ENA_WIDTH                         1  /* IN1_ENA */
+
+/*
+ * R8 (0x08) - Anti Pop Control
+ */
+#define WM9081_LINEOUT_DISCH                    0x0004  /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_MASK               0x0004  /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_SHIFT                   2  /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_DISCH_WIDTH                   1  /* LINEOUT_DISCH */
+#define WM9081_LINEOUT_VROI                     0x0002  /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_MASK                0x0002  /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_SHIFT                    1  /* LINEOUT_VROI */
+#define WM9081_LINEOUT_VROI_WIDTH                    1  /* LINEOUT_VROI */
+#define WM9081_LINEOUT_CLAMP                    0x0001  /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_MASK               0x0001  /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_SHIFT                   0  /* LINEOUT_CLAMP */
+#define WM9081_LINEOUT_CLAMP_WIDTH                   1  /* LINEOUT_CLAMP */
+
+/*
+ * R9 (0x09) - Analogue Speaker 1
+ */
+#define WM9081_SPK_DCGAIN_MASK                  0x0038  /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_DCGAIN_SHIFT                      3  /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_DCGAIN_WIDTH                      3  /* SPK_DCGAIN - [5:3] */
+#define WM9081_SPK_ACGAIN_MASK                  0x0007  /* SPK_ACGAIN - [2:0] */
+#define WM9081_SPK_ACGAIN_SHIFT                      0  /* SPK_ACGAIN - [2:0] */
+#define WM9081_SPK_ACGAIN_WIDTH                      3  /* SPK_ACGAIN - [2:0] */
+
+/*
+ * R10 (0x0A) - Analogue Speaker 2
+ */
+#define WM9081_SPK_MODE                         0x0040  /* SPK_MODE */
+#define WM9081_SPK_MODE_MASK                    0x0040  /* SPK_MODE */
+#define WM9081_SPK_MODE_SHIFT                        6  /* SPK_MODE */
+#define WM9081_SPK_MODE_WIDTH                        1  /* SPK_MODE */
+#define WM9081_SPK_INV_MUTE                     0x0010  /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_MASK                0x0010  /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_SHIFT                    4  /* SPK_INV_MUTE */
+#define WM9081_SPK_INV_MUTE_WIDTH                    1  /* SPK_INV_MUTE */
+#define WM9081_OUT_SPK_CTRL                     0x0008  /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_MASK                0x0008  /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_SHIFT                    3  /* OUT_SPK_CTRL */
+#define WM9081_OUT_SPK_CTRL_WIDTH                    1  /* OUT_SPK_CTRL */
+
+/*
+ * R11 (0x0B) - Power Management
+ */
+#define WM9081_TSHUT_ENA                        0x0100  /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_MASK                   0x0100  /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_SHIFT                       8  /* TSHUT_ENA */
+#define WM9081_TSHUT_ENA_WIDTH                       1  /* TSHUT_ENA */
+#define WM9081_TSENSE_ENA                       0x0080  /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_MASK                  0x0080  /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_SHIFT                      7  /* TSENSE_ENA */
+#define WM9081_TSENSE_ENA_WIDTH                      1  /* TSENSE_ENA */
+#define WM9081_TEMP_SHUT                        0x0040  /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_MASK                   0x0040  /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_SHIFT                       6  /* TEMP_SHUT */
+#define WM9081_TEMP_SHUT_WIDTH                       1  /* TEMP_SHUT */
+#define WM9081_LINEOUT_ENA                      0x0010  /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_MASK                 0x0010  /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_SHIFT                     4  /* LINEOUT_ENA */
+#define WM9081_LINEOUT_ENA_WIDTH                     1  /* LINEOUT_ENA */
+#define WM9081_SPKPGA_ENA                       0x0004  /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_MASK                  0x0004  /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_SHIFT                      2  /* SPKPGA_ENA */
+#define WM9081_SPKPGA_ENA_WIDTH                      1  /* SPKPGA_ENA */
+#define WM9081_SPK_ENA                          0x0002  /* SPK_ENA */
+#define WM9081_SPK_ENA_MASK                     0x0002  /* SPK_ENA */
+#define WM9081_SPK_ENA_SHIFT                         1  /* SPK_ENA */
+#define WM9081_SPK_ENA_WIDTH                         1  /* SPK_ENA */
+#define WM9081_DAC_ENA                          0x0001  /* DAC_ENA */
+#define WM9081_DAC_ENA_MASK                     0x0001  /* DAC_ENA */
+#define WM9081_DAC_ENA_SHIFT                         0  /* DAC_ENA */
+#define WM9081_DAC_ENA_WIDTH                         1  /* DAC_ENA */
+
+/*
+ * R12 (0x0C) - Clock Control 1
+ */
+#define WM9081_CLK_OP_DIV_MASK                  0x1C00  /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_OP_DIV_SHIFT                     10  /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_OP_DIV_WIDTH                      3  /* CLK_OP_DIV - [12:10] */
+#define WM9081_CLK_TO_DIV_MASK                  0x0300  /* CLK_TO_DIV - [9:8] */
+#define WM9081_CLK_TO_DIV_SHIFT                      8  /* CLK_TO_DIV - [9:8] */
+#define WM9081_CLK_TO_DIV_WIDTH                      2  /* CLK_TO_DIV - [9:8] */
+#define WM9081_MCLKDIV2                         0x0080  /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_MASK                    0x0080  /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_SHIFT                        7  /* MCLKDIV2 */
+#define WM9081_MCLKDIV2_WIDTH                        1  /* MCLKDIV2 */
+
+/*
+ * R13 (0x0D) - Clock Control 2
+ */
+#define WM9081_CLK_SYS_RATE_MASK                0x00F0  /* CLK_SYS_RATE - [7:4] */
+#define WM9081_CLK_SYS_RATE_SHIFT                    4  /* CLK_SYS_RATE - [7:4] */
+#define WM9081_CLK_SYS_RATE_WIDTH                    4  /* CLK_SYS_RATE - [7:4] */
+#define WM9081_SAMPLE_RATE_MASK                 0x000F  /* SAMPLE_RATE - [3:0] */
+#define WM9081_SAMPLE_RATE_SHIFT                     0  /* SAMPLE_RATE - [3:0] */
+#define WM9081_SAMPLE_RATE_WIDTH                     4  /* SAMPLE_RATE - [3:0] */
+
+/*
+ * R14 (0x0E) - Clock Control 3
+ */
+#define WM9081_CLK_SRC_SEL                      0x2000  /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_MASK                 0x2000  /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_SHIFT                    13  /* CLK_SRC_SEL */
+#define WM9081_CLK_SRC_SEL_WIDTH                     1  /* CLK_SRC_SEL */
+#define WM9081_CLK_OP_ENA                       0x0020  /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_MASK                  0x0020  /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_SHIFT                      5  /* CLK_OP_ENA */
+#define WM9081_CLK_OP_ENA_WIDTH                      1  /* CLK_OP_ENA */
+#define WM9081_CLK_TO_ENA                       0x0004  /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_MASK                  0x0004  /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_SHIFT                      2  /* CLK_TO_ENA */
+#define WM9081_CLK_TO_ENA_WIDTH                      1  /* CLK_TO_ENA */
+#define WM9081_CLK_DSP_ENA                      0x0002  /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_MASK                 0x0002  /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_SHIFT                     1  /* CLK_DSP_ENA */
+#define WM9081_CLK_DSP_ENA_WIDTH                     1  /* CLK_DSP_ENA */
+#define WM9081_CLK_SYS_ENA                      0x0001  /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_MASK                 0x0001  /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_SHIFT                     0  /* CLK_SYS_ENA */
+#define WM9081_CLK_SYS_ENA_WIDTH                     1  /* CLK_SYS_ENA */
+
+/*
+ * R16 (0x10) - FLL Control 1
+ */
+#define WM9081_FLL_HOLD                         0x0008  /* FLL_HOLD */
+#define WM9081_FLL_HOLD_MASK                    0x0008  /* FLL_HOLD */
+#define WM9081_FLL_HOLD_SHIFT                        3  /* FLL_HOLD */
+#define WM9081_FLL_HOLD_WIDTH                        1  /* FLL_HOLD */
+#define WM9081_FLL_FRAC                         0x0004  /* FLL_FRAC */
+#define WM9081_FLL_FRAC_MASK                    0x0004  /* FLL_FRAC */
+#define WM9081_FLL_FRAC_SHIFT                        2  /* FLL_FRAC */
+#define WM9081_FLL_FRAC_WIDTH                        1  /* FLL_FRAC */
+#define WM9081_FLL_ENA                          0x0001  /* FLL_ENA */
+#define WM9081_FLL_ENA_MASK                     0x0001  /* FLL_ENA */
+#define WM9081_FLL_ENA_SHIFT                         0  /* FLL_ENA */
+#define WM9081_FLL_ENA_WIDTH                         1  /* FLL_ENA */
+
+/*
+ * R17 (0x11) - FLL Control 2
+ */
+#define WM9081_FLL_OUTDIV_MASK                  0x0700  /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_OUTDIV_SHIFT                      8  /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_OUTDIV_WIDTH                      3  /* FLL_OUTDIV - [10:8] */
+#define WM9081_FLL_CTRL_RATE_MASK               0x0070  /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_CTRL_RATE_SHIFT                   4  /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_CTRL_RATE_WIDTH                   3  /* FLL_CTRL_RATE - [6:4] */
+#define WM9081_FLL_FRATIO_MASK                  0x0007  /* FLL_FRATIO - [2:0] */
+#define WM9081_FLL_FRATIO_SHIFT                      0  /* FLL_FRATIO - [2:0] */
+#define WM9081_FLL_FRATIO_WIDTH                      3  /* FLL_FRATIO - [2:0] */
+
+/*
+ * R18 (0x12) - FLL Control 3
+ */
+#define WM9081_FLL_K_MASK                       0xFFFF  /* FLL_K - [15:0] */
+#define WM9081_FLL_K_SHIFT                           0  /* FLL_K - [15:0] */
+#define WM9081_FLL_K_WIDTH                          16  /* FLL_K - [15:0] */
+
+/*
+ * R19 (0x13) - FLL Control 4
+ */
+#define WM9081_FLL_N_MASK                       0x7FE0  /* FLL_N - [14:5] */
+#define WM9081_FLL_N_SHIFT                           5  /* FLL_N - [14:5] */
+#define WM9081_FLL_N_WIDTH                          10  /* FLL_N - [14:5] */
+#define WM9081_FLL_GAIN_MASK                    0x000F  /* FLL_GAIN - [3:0] */
+#define WM9081_FLL_GAIN_SHIFT                        0  /* FLL_GAIN - [3:0] */
+#define WM9081_FLL_GAIN_WIDTH                        4  /* FLL_GAIN - [3:0] */
+
+/*
+ * R20 (0x14) - FLL Control 5
+ */
+#define WM9081_FLL_CLK_REF_DIV_MASK             0x0018  /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_REF_DIV_SHIFT                 3  /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_REF_DIV_WIDTH                 2  /* FLL_CLK_REF_DIV - [4:3] */
+#define WM9081_FLL_CLK_SRC_MASK                 0x0003  /* FLL_CLK_SRC - [1:0] */
+#define WM9081_FLL_CLK_SRC_SHIFT                     0  /* FLL_CLK_SRC - [1:0] */
+#define WM9081_FLL_CLK_SRC_WIDTH                     2  /* FLL_CLK_SRC - [1:0] */
+
+/*
+ * R22 (0x16) - Audio Interface 1
+ */
+#define WM9081_AIFDAC_CHAN                      0x0040  /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_MASK                 0x0040  /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_SHIFT                     6  /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_CHAN_WIDTH                     1  /* AIFDAC_CHAN */
+#define WM9081_AIFDAC_TDM_SLOT_MASK             0x0030  /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_SLOT_SHIFT                 4  /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_SLOT_WIDTH                 2  /* AIFDAC_TDM_SLOT - [5:4] */
+#define WM9081_AIFDAC_TDM_MODE_MASK             0x000C  /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_AIFDAC_TDM_MODE_SHIFT                 2  /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_AIFDAC_TDM_MODE_WIDTH                 2  /* AIFDAC_TDM_MODE - [3:2] */
+#define WM9081_DAC_COMP                         0x0002  /* DAC_COMP */
+#define WM9081_DAC_COMP_MASK                    0x0002  /* DAC_COMP */
+#define WM9081_DAC_COMP_SHIFT                        1  /* DAC_COMP */
+#define WM9081_DAC_COMP_WIDTH                        1  /* DAC_COMP */
+#define WM9081_DAC_COMPMODE                     0x0001  /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_MASK                0x0001  /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_SHIFT                    0  /* DAC_COMPMODE */
+#define WM9081_DAC_COMPMODE_WIDTH                    1  /* DAC_COMPMODE */
+
+/*
+ * R23 (0x17) - Audio Interface 2
+ */
+#define WM9081_AIF_TRIS                         0x0200  /* AIF_TRIS */
+#define WM9081_AIF_TRIS_MASK                    0x0200  /* AIF_TRIS */
+#define WM9081_AIF_TRIS_SHIFT                        9  /* AIF_TRIS */
+#define WM9081_AIF_TRIS_WIDTH                        1  /* AIF_TRIS */
+#define WM9081_DAC_DAT_INV                      0x0100  /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_MASK                 0x0100  /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_SHIFT                     8  /* DAC_DAT_INV */
+#define WM9081_DAC_DAT_INV_WIDTH                     1  /* DAC_DAT_INV */
+#define WM9081_AIF_BCLK_INV                     0x0080  /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_MASK                0x0080  /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_SHIFT                    7  /* AIF_BCLK_INV */
+#define WM9081_AIF_BCLK_INV_WIDTH                    1  /* AIF_BCLK_INV */
+#define WM9081_BCLK_DIR                         0x0040  /* BCLK_DIR */
+#define WM9081_BCLK_DIR_MASK                    0x0040  /* BCLK_DIR */
+#define WM9081_BCLK_DIR_SHIFT                        6  /* BCLK_DIR */
+#define WM9081_BCLK_DIR_WIDTH                        1  /* BCLK_DIR */
+#define WM9081_LRCLK_DIR                        0x0020  /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_MASK                   0x0020  /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_SHIFT                       5  /* LRCLK_DIR */
+#define WM9081_LRCLK_DIR_WIDTH                       1  /* LRCLK_DIR */
+#define WM9081_AIF_LRCLK_INV                    0x0010  /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_MASK               0x0010  /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_SHIFT                   4  /* AIF_LRCLK_INV */
+#define WM9081_AIF_LRCLK_INV_WIDTH                   1  /* AIF_LRCLK_INV */
+#define WM9081_AIF_WL_MASK                      0x000C  /* AIF_WL - [3:2] */
+#define WM9081_AIF_WL_SHIFT                          2  /* AIF_WL - [3:2] */
+#define WM9081_AIF_WL_WIDTH                          2  /* AIF_WL - [3:2] */
+#define WM9081_AIF_FMT_MASK                     0x0003  /* AIF_FMT - [1:0] */
+#define WM9081_AIF_FMT_SHIFT                         0  /* AIF_FMT - [1:0] */
+#define WM9081_AIF_FMT_WIDTH                         2  /* AIF_FMT - [1:0] */
+
+/*
+ * R24 (0x18) - Audio Interface 3
+ */
+#define WM9081_BCLK_DIV_MASK                    0x001F  /* BCLK_DIV - [4:0] */
+#define WM9081_BCLK_DIV_SHIFT                        0  /* BCLK_DIV - [4:0] */
+#define WM9081_BCLK_DIV_WIDTH                        5  /* BCLK_DIV - [4:0] */
+
+/*
+ * R25 (0x19) - Audio Interface 4
+ */
+#define WM9081_LRCLK_RATE_MASK                  0x07FF  /* LRCLK_RATE - [10:0] */
+#define WM9081_LRCLK_RATE_SHIFT                      0  /* LRCLK_RATE - [10:0] */
+#define WM9081_LRCLK_RATE_WIDTH                     11  /* LRCLK_RATE - [10:0] */
+
+/*
+ * R26 (0x1A) - Interrupt Status
+ */
+#define WM9081_WSEQ_BUSY_EINT                   0x0004  /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_MASK              0x0004  /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_SHIFT                  2  /* WSEQ_BUSY_EINT */
+#define WM9081_WSEQ_BUSY_EINT_WIDTH                  1  /* WSEQ_BUSY_EINT */
+#define WM9081_TSHUT_EINT                       0x0001  /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_MASK                  0x0001  /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_SHIFT                      0  /* TSHUT_EINT */
+#define WM9081_TSHUT_EINT_WIDTH                      1  /* TSHUT_EINT */
+
+/*
+ * R27 (0x1B) - Interrupt Status Mask
+ */
+#define WM9081_IM_WSEQ_BUSY_EINT                0x0004  /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_MASK           0x0004  /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_SHIFT               2  /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_WSEQ_BUSY_EINT_WIDTH               1  /* IM_WSEQ_BUSY_EINT */
+#define WM9081_IM_TSHUT_EINT                    0x0001  /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_MASK               0x0001  /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_SHIFT                   0  /* IM_TSHUT_EINT */
+#define WM9081_IM_TSHUT_EINT_WIDTH                   1  /* IM_TSHUT_EINT */
+
+/*
+ * R28 (0x1C) - Interrupt Polarity
+ */
+#define WM9081_TSHUT_INV                        0x0001  /* TSHUT_INV */
+#define WM9081_TSHUT_INV_MASK                   0x0001  /* TSHUT_INV */
+#define WM9081_TSHUT_INV_SHIFT                       0  /* TSHUT_INV */
+#define WM9081_TSHUT_INV_WIDTH                       1  /* TSHUT_INV */
+
+/*
+ * R29 (0x1D) - Interrupt Control
+ */
+#define WM9081_IRQ_POL                          0x8000  /* IRQ_POL */
+#define WM9081_IRQ_POL_MASK                     0x8000  /* IRQ_POL */
+#define WM9081_IRQ_POL_SHIFT                        15  /* IRQ_POL */
+#define WM9081_IRQ_POL_WIDTH                         1  /* IRQ_POL */
+#define WM9081_IRQ_OP_CTRL                      0x0001  /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_MASK                 0x0001  /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_SHIFT                     0  /* IRQ_OP_CTRL */
+#define WM9081_IRQ_OP_CTRL_WIDTH                     1  /* IRQ_OP_CTRL */
+
+/*
+ * R30 (0x1E) - DAC Digital 1
+ */
+#define WM9081_DAC_VOL_MASK                     0x00FF  /* DAC_VOL - [7:0] */
+#define WM9081_DAC_VOL_SHIFT                         0  /* DAC_VOL - [7:0] */
+#define WM9081_DAC_VOL_WIDTH                         8  /* DAC_VOL - [7:0] */
+
+/*
+ * R31 (0x1F) - DAC Digital 2
+ */
+#define WM9081_DAC_MUTERATE                     0x0400  /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_MASK                0x0400  /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_SHIFT                   10  /* DAC_MUTERATE */
+#define WM9081_DAC_MUTERATE_WIDTH                    1  /* DAC_MUTERATE */
+#define WM9081_DAC_MUTEMODE                     0x0200  /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_MASK                0x0200  /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_SHIFT                    9  /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTEMODE_WIDTH                    1  /* DAC_MUTEMODE */
+#define WM9081_DAC_MUTE                         0x0008  /* DAC_MUTE */
+#define WM9081_DAC_MUTE_MASK                    0x0008  /* DAC_MUTE */
+#define WM9081_DAC_MUTE_SHIFT                        3  /* DAC_MUTE */
+#define WM9081_DAC_MUTE_WIDTH                        1  /* DAC_MUTE */
+#define WM9081_DEEMPH_MASK                      0x0006  /* DEEMPH - [2:1] */
+#define WM9081_DEEMPH_SHIFT                          1  /* DEEMPH - [2:1] */
+#define WM9081_DEEMPH_WIDTH                          2  /* DEEMPH - [2:1] */
+
+/*
+ * R32 (0x20) - DRC 1
+ */
+#define WM9081_DRC_ENA                          0x8000  /* DRC_ENA */
+#define WM9081_DRC_ENA_MASK                     0x8000  /* DRC_ENA */
+#define WM9081_DRC_ENA_SHIFT                        15  /* DRC_ENA */
+#define WM9081_DRC_ENA_WIDTH                         1  /* DRC_ENA */
+#define WM9081_DRC_STARTUP_GAIN_MASK            0x07C0  /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_STARTUP_GAIN_SHIFT                6  /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_STARTUP_GAIN_WIDTH                5  /* DRC_STARTUP_GAIN - [10:6] */
+#define WM9081_DRC_FF_DLY                       0x0020  /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_MASK                  0x0020  /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_SHIFT                      5  /* DRC_FF_DLY */
+#define WM9081_DRC_FF_DLY_WIDTH                      1  /* DRC_FF_DLY */
+#define WM9081_DRC_QR                           0x0004  /* DRC_QR */
+#define WM9081_DRC_QR_MASK                      0x0004  /* DRC_QR */
+#define WM9081_DRC_QR_SHIFT                          2  /* DRC_QR */
+#define WM9081_DRC_QR_WIDTH                          1  /* DRC_QR */
+#define WM9081_DRC_ANTICLIP                     0x0002  /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_MASK                0x0002  /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_SHIFT                    1  /* DRC_ANTICLIP */
+#define WM9081_DRC_ANTICLIP_WIDTH                    1  /* DRC_ANTICLIP */
+
+/*
+ * R33 (0x21) - DRC 2
+ */
+#define WM9081_DRC_ATK_MASK                     0xF000  /* DRC_ATK - [15:12] */
+#define WM9081_DRC_ATK_SHIFT                        12  /* DRC_ATK - [15:12] */
+#define WM9081_DRC_ATK_WIDTH                         4  /* DRC_ATK - [15:12] */
+#define WM9081_DRC_DCY_MASK                     0x0F00  /* DRC_DCY - [11:8] */
+#define WM9081_DRC_DCY_SHIFT                         8  /* DRC_DCY - [11:8] */
+#define WM9081_DRC_DCY_WIDTH                         4  /* DRC_DCY - [11:8] */
+#define WM9081_DRC_QR_THR_MASK                  0x00C0  /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_THR_SHIFT                      6  /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_THR_WIDTH                      2  /* DRC_QR_THR - [7:6] */
+#define WM9081_DRC_QR_DCY_MASK                  0x0030  /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_QR_DCY_SHIFT                      4  /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_QR_DCY_WIDTH                      2  /* DRC_QR_DCY - [5:4] */
+#define WM9081_DRC_MINGAIN_MASK                 0x000C  /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MINGAIN_SHIFT                     2  /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MINGAIN_WIDTH                     2  /* DRC_MINGAIN - [3:2] */
+#define WM9081_DRC_MAXGAIN_MASK                 0x0003  /* DRC_MAXGAIN - [1:0] */
+#define WM9081_DRC_MAXGAIN_SHIFT                     0  /* DRC_MAXGAIN - [1:0] */
+#define WM9081_DRC_MAXGAIN_WIDTH                     2  /* DRC_MAXGAIN - [1:0] */
+
+/*
+ * R34 (0x22) - DRC 3
+ */
+#define WM9081_DRC_HI_COMP_MASK                 0x0038  /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_HI_COMP_SHIFT                     3  /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_HI_COMP_WIDTH                     3  /* DRC_HI_COMP - [5:3] */
+#define WM9081_DRC_LO_COMP_MASK                 0x0007  /* DRC_LO_COMP - [2:0] */
+#define WM9081_DRC_LO_COMP_SHIFT                     0  /* DRC_LO_COMP - [2:0] */
+#define WM9081_DRC_LO_COMP_WIDTH                     3  /* DRC_LO_COMP - [2:0] */
+
+/*
+ * R35 (0x23) - DRC 4
+ */
+#define WM9081_DRC_KNEE_IP_MASK                 0x07E0  /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_IP_SHIFT                     5  /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_IP_WIDTH                     6  /* DRC_KNEE_IP - [10:5] */
+#define WM9081_DRC_KNEE_OP_MASK                 0x001F  /* DRC_KNEE_OP - [4:0] */
+#define WM9081_DRC_KNEE_OP_SHIFT                     0  /* DRC_KNEE_OP - [4:0] */
+#define WM9081_DRC_KNEE_OP_WIDTH                     5  /* DRC_KNEE_OP - [4:0] */
+
+/*
+ * R38 (0x26) - Write Sequencer 1
+ */
+#define WM9081_WSEQ_ENA                         0x8000  /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_MASK                    0x8000  /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_SHIFT                       15  /* WSEQ_ENA */
+#define WM9081_WSEQ_ENA_WIDTH                        1  /* WSEQ_ENA */
+#define WM9081_WSEQ_ABORT                       0x0200  /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_MASK                  0x0200  /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_SHIFT                      9  /* WSEQ_ABORT */
+#define WM9081_WSEQ_ABORT_WIDTH                      1  /* WSEQ_ABORT */
+#define WM9081_WSEQ_START                       0x0100  /* WSEQ_START */
+#define WM9081_WSEQ_START_MASK                  0x0100  /* WSEQ_START */
+#define WM9081_WSEQ_START_SHIFT                      8  /* WSEQ_START */
+#define WM9081_WSEQ_START_WIDTH                      1  /* WSEQ_START */
+#define WM9081_WSEQ_START_INDEX_MASK            0x007F  /* WSEQ_START_INDEX - [6:0] */
+#define WM9081_WSEQ_START_INDEX_SHIFT                0  /* WSEQ_START_INDEX - [6:0] */
+#define WM9081_WSEQ_START_INDEX_WIDTH                7  /* WSEQ_START_INDEX - [6:0] */
+
+/*
+ * R39 (0x27) - Write Sequencer 2
+ */
+#define WM9081_WSEQ_CURRENT_INDEX_MASK          0x07F0  /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_CURRENT_INDEX_SHIFT              4  /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_CURRENT_INDEX_WIDTH              7  /* WSEQ_CURRENT_INDEX - [10:4] */
+#define WM9081_WSEQ_BUSY                        0x0001  /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_MASK                   0x0001  /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_SHIFT                       0  /* WSEQ_BUSY */
+#define WM9081_WSEQ_BUSY_WIDTH                       1  /* WSEQ_BUSY */
+
+/*
+ * R40 (0x28) - MW Slave 1
+ */
+#define WM9081_SPI_CFG                          0x0020  /* SPI_CFG */
+#define WM9081_SPI_CFG_MASK                     0x0020  /* SPI_CFG */
+#define WM9081_SPI_CFG_SHIFT                         5  /* SPI_CFG */
+#define WM9081_SPI_CFG_WIDTH                         1  /* SPI_CFG */
+#define WM9081_SPI_4WIRE                        0x0010  /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_MASK                   0x0010  /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_SHIFT                       4  /* SPI_4WIRE */
+#define WM9081_SPI_4WIRE_WIDTH                       1  /* SPI_4WIRE */
+#define WM9081_ARA_ENA                          0x0008  /* ARA_ENA */
+#define WM9081_ARA_ENA_MASK                     0x0008  /* ARA_ENA */
+#define WM9081_ARA_ENA_SHIFT                         3  /* ARA_ENA */
+#define WM9081_ARA_ENA_WIDTH                         1  /* ARA_ENA */
+#define WM9081_AUTO_INC                         0x0002  /* AUTO_INC */
+#define WM9081_AUTO_INC_MASK                    0x0002  /* AUTO_INC */
+#define WM9081_AUTO_INC_SHIFT                        1  /* AUTO_INC */
+#define WM9081_AUTO_INC_WIDTH                        1  /* AUTO_INC */
+
+/*
+ * R42 (0x2A) - EQ 1
+ */
+#define WM9081_EQ_B1_GAIN_MASK                  0xF800  /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B1_GAIN_SHIFT                     11  /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B1_GAIN_WIDTH                      5  /* EQ_B1_GAIN - [15:11] */
+#define WM9081_EQ_B2_GAIN_MASK                  0x07C0  /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B2_GAIN_SHIFT                      6  /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B2_GAIN_WIDTH                      5  /* EQ_B2_GAIN - [10:6] */
+#define WM9081_EQ_B4_GAIN_MASK                  0x003E  /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_B4_GAIN_SHIFT                      1  /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_B4_GAIN_WIDTH                      5  /* EQ_B4_GAIN - [5:1] */
+#define WM9081_EQ_ENA                           0x0001  /* EQ_ENA */
+#define WM9081_EQ_ENA_MASK                      0x0001  /* EQ_ENA */
+#define WM9081_EQ_ENA_SHIFT                          0  /* EQ_ENA */
+#define WM9081_EQ_ENA_WIDTH                          1  /* EQ_ENA */
+
+/*
+ * R43 (0x2B) - EQ 2
+ */
+#define WM9081_EQ_B3_GAIN_MASK                  0xF800  /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B3_GAIN_SHIFT                     11  /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B3_GAIN_WIDTH                      5  /* EQ_B3_GAIN - [15:11] */
+#define WM9081_EQ_B5_GAIN_MASK                  0x07C0  /* EQ_B5_GAIN - [10:6] */
+#define WM9081_EQ_B5_GAIN_SHIFT                      6  /* EQ_B5_GAIN - [10:6] */
+#define WM9081_EQ_B5_GAIN_WIDTH                      5  /* EQ_B5_GAIN - [10:6] */
+
+/*
+ * R44 (0x2C) - EQ 3
+ */
+#define WM9081_EQ_B1_A_MASK                     0xFFFF  /* EQ_B1_A - [15:0] */
+#define WM9081_EQ_B1_A_SHIFT                         0  /* EQ_B1_A - [15:0] */
+#define WM9081_EQ_B1_A_WIDTH                        16  /* EQ_B1_A - [15:0] */
+
+/*
+ * R45 (0x2D) - EQ 4
+ */
+#define WM9081_EQ_B1_B_MASK                     0xFFFF  /* EQ_B1_B - [15:0] */
+#define WM9081_EQ_B1_B_SHIFT                         0  /* EQ_B1_B - [15:0] */
+#define WM9081_EQ_B1_B_WIDTH                        16  /* EQ_B1_B - [15:0] */
+
+/*
+ * R46 (0x2E) - EQ 5
+ */
+#define WM9081_EQ_B1_PG_MASK                    0xFFFF  /* EQ_B1_PG - [15:0] */
+#define WM9081_EQ_B1_PG_SHIFT                        0  /* EQ_B1_PG - [15:0] */
+#define WM9081_EQ_B1_PG_WIDTH                       16  /* EQ_B1_PG - [15:0] */
+
+/*
+ * R47 (0x2F) - EQ 6
+ */
+#define WM9081_EQ_B2_A_MASK                     0xFFFF  /* EQ_B2_A - [15:0] */
+#define WM9081_EQ_B2_A_SHIFT                         0  /* EQ_B2_A - [15:0] */
+#define WM9081_EQ_B2_A_WIDTH                        16  /* EQ_B2_A - [15:0] */
+
+/*
+ * R48 (0x30) - EQ 7
+ */
+#define WM9081_EQ_B2_B_MASK                     0xFFFF  /* EQ_B2_B - [15:0] */
+#define WM9081_EQ_B2_B_SHIFT                         0  /* EQ_B2_B - [15:0] */
+#define WM9081_EQ_B2_B_WIDTH                        16  /* EQ_B2_B - [15:0] */
+
+/*
+ * R49 (0x31) - EQ 8
+ */
+#define WM9081_EQ_B2_C_MASK                     0xFFFF  /* EQ_B2_C - [15:0] */
+#define WM9081_EQ_B2_C_SHIFT                         0  /* EQ_B2_C - [15:0] */
+#define WM9081_EQ_B2_C_WIDTH                        16  /* EQ_B2_C - [15:0] */
+
+/*
+ * R50 (0x32) - EQ 9
+ */
+#define WM9081_EQ_B2_PG_MASK                    0xFFFF  /* EQ_B2_PG - [15:0] */
+#define WM9081_EQ_B2_PG_SHIFT                        0  /* EQ_B2_PG - [15:0] */
+#define WM9081_EQ_B2_PG_WIDTH                       16  /* EQ_B2_PG - [15:0] */
+
+/*
+ * R51 (0x33) - EQ 10
+ */
+#define WM9081_EQ_B4_A_MASK                     0xFFFF  /* EQ_B4_A - [15:0] */
+#define WM9081_EQ_B4_A_SHIFT                         0  /* EQ_B4_A - [15:0] */
+#define WM9081_EQ_B4_A_WIDTH                        16  /* EQ_B4_A - [15:0] */
+
+/*
+ * R52 (0x34) - EQ 11
+ */
+#define WM9081_EQ_B4_B_MASK                     0xFFFF  /* EQ_B4_B - [15:0] */
+#define WM9081_EQ_B4_B_SHIFT                         0  /* EQ_B4_B - [15:0] */
+#define WM9081_EQ_B4_B_WIDTH                        16  /* EQ_B4_B - [15:0] */
+
+/*
+ * R53 (0x35) - EQ 12
+ */
+#define WM9081_EQ_B4_C_MASK                     0xFFFF  /* EQ_B4_C - [15:0] */
+#define WM9081_EQ_B4_C_SHIFT                         0  /* EQ_B4_C - [15:0] */
+#define WM9081_EQ_B4_C_WIDTH                        16  /* EQ_B4_C - [15:0] */
+
+/*
+ * R54 (0x36) - EQ 13
+ */
+#define WM9081_EQ_B4_PG_MASK                    0xFFFF  /* EQ_B4_PG - [15:0] */
+#define WM9081_EQ_B4_PG_SHIFT                        0  /* EQ_B4_PG - [15:0] */
+#define WM9081_EQ_B4_PG_WIDTH                       16  /* EQ_B4_PG - [15:0] */
+
+/*
+ * R55 (0x37) - EQ 14
+ */
+#define WM9081_EQ_B3_A_MASK                     0xFFFF  /* EQ_B3_A - [15:0] */
+#define WM9081_EQ_B3_A_SHIFT                         0  /* EQ_B3_A - [15:0] */
+#define WM9081_EQ_B3_A_WIDTH                        16  /* EQ_B3_A - [15:0] */
+
+/*
+ * R56 (0x38) - EQ 15
+ */
+#define WM9081_EQ_B3_B_MASK                     0xFFFF  /* EQ_B3_B - [15:0] */
+#define WM9081_EQ_B3_B_SHIFT                         0  /* EQ_B3_B - [15:0] */
+#define WM9081_EQ_B3_B_WIDTH                        16  /* EQ_B3_B - [15:0] */
+
+/*
+ * R57 (0x39) - EQ 16
+ */
+#define WM9081_EQ_B3_C_MASK                     0xFFFF  /* EQ_B3_C - [15:0] */
+#define WM9081_EQ_B3_C_SHIFT                         0  /* EQ_B3_C - [15:0] */
+#define WM9081_EQ_B3_C_WIDTH                        16  /* EQ_B3_C - [15:0] */
+
+/*
+ * R58 (0x3A) - EQ 17
+ */
+#define WM9081_EQ_B3_PG_MASK                    0xFFFF  /* EQ_B3_PG - [15:0] */
+#define WM9081_EQ_B3_PG_SHIFT                        0  /* EQ_B3_PG - [15:0] */
+#define WM9081_EQ_B3_PG_WIDTH                       16  /* EQ_B3_PG - [15:0] */
+
+/*
+ * R59 (0x3B) - EQ 18
+ */
+#define WM9081_EQ_B5_A_MASK                     0xFFFF  /* EQ_B5_A - [15:0] */
+#define WM9081_EQ_B5_A_SHIFT                         0  /* EQ_B5_A - [15:0] */
+#define WM9081_EQ_B5_A_WIDTH                        16  /* EQ_B5_A - [15:0] */
+
+/*
+ * R60 (0x3C) - EQ 19
+ */
+#define WM9081_EQ_B5_B_MASK                     0xFFFF  /* EQ_B5_B - [15:0] */
+#define WM9081_EQ_B5_B_SHIFT                         0  /* EQ_B5_B - [15:0] */
+#define WM9081_EQ_B5_B_WIDTH                        16  /* EQ_B5_B - [15:0] */
+
+/*
+ * R61 (0x3D) - EQ 20
+ */
+#define WM9081_EQ_B5_PG_MASK                    0xFFFF  /* EQ_B5_PG - [15:0] */
+#define WM9081_EQ_B5_PG_SHIFT                        0  /* EQ_B5_PG - [15:0] */
+#define WM9081_EQ_B5_PG_WIDTH                       16  /* EQ_B5_PG - [15:0] */
+
+
+#endif
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 6e23a81dba78..fa88b463e71f 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -282,14 +282,14 @@ struct snd_soc_dai wm9705_dai[] = {
                         .channels_min = 1,
                         .channels_max = 2,
                         .rates = WM9705_AC97_RATES,
-                        .formats = SNDRV_PCM_FMTBIT_S16_LE,
+                        .formats = SND_SOC_STD_AC97_FMTS,
                 },
                 .capture = {
                         .stream_name = "HiFi Capture",
                         .channels_min = 1,
                         .channels_max = 2,
                         .rates = WM9705_AC97_RATES,
-                        .formats = SNDRV_PCM_FMTBIT_S16_LE,
+                        .formats = SND_SOC_STD_AC97_FMTS,
                 },
                 .ops = &wm9705_dai_ops,
         },
@@ -318,7 +318,7 @@ static int wm9705_reset(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int wm9705_soc_suspend(struct platform_device *pdev)
+static int wm9705_soc_suspend(struct platform_device *pdev, pm_message_t msg)
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_codec *codec = socdev->card->codec;
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 765cf1e7369e..1fd4e88f50cf 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -534,13 +534,13 @@ struct snd_soc_dai wm9712_dai[] = {
                 .channels_min = 1,
                 .channels_max = 2,
                 .rates = WM9712_AC97_RATES,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+                .formats = SND_SOC_STD_AC97_FMTS,},
         .capture = {
                 .stream_name = "HiFi Capture",
                 .channels_min = 1,
                 .channels_max = 2,
                 .rates = WM9712_AC97_RATES,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+                .formats = SND_SOC_STD_AC97_FMTS,},
         .ops = &wm9712_dai_ops_hifi,
 },
 {
@@ -550,7 +550,7 @@ struct snd_soc_dai wm9712_dai[] = {
                 .channels_min = 1,
                 .channels_max = 1,
                 .rates = WM9712_AC97_RATES,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+                .formats = SND_SOC_STD_AC97_FMTS,},
         .ops = &wm9712_dai_ops_aux,
 }
 };
@@ -585,6 +585,8 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
         }
 
         soc_ac97_ops.reset(codec->ac97);
+        if (soc_ac97_ops.warm_reset)
+                soc_ac97_ops.warm_reset(codec->ac97);
         if (ac97_read(codec, 0) != wm9712_reg[0])
                 goto err;
         return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 523bad077fa0..abed37acf787 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -189,6 +189,26 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
 SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
 };
 
+static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
+                                 struct snd_kcontrol *kcontrol, int event)
+{
+        struct snd_soc_codec *codec = w->codec;
+        u16 status, rate;
+
+        BUG_ON(event != SND_SOC_DAPM_PRE_PMD);
+
+        /* Gracefully shut down the voice interface. */
+        status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000;
+        rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
+        ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
+        schedule_timeout_interruptible(msecs_to_jiffies(1));
+        ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
+        ac97_write(codec, AC97_EXTENDED_MID, status);
+
+        return 0;
+}
+
+
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path using the current
@@ -400,7 +420,8 @@ SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
+SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1,
+                   wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
 SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
@@ -689,7 +710,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source)
         Ndiv = target / source;
         if ((Ndiv < 5) || (Ndiv > 12))
                 printk(KERN_WARNING
-                        "WM9713 PLL N value %d out of recommended range!\n",
+                        "WM9713 PLL N value %u out of recommended range!\n",
                         Ndiv);
 
         pll_div->n = Ndiv;
@@ -936,21 +957,6 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
         return 0;
 }
 
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
-                                 struct snd_soc_dai *dai)
-{
-        struct snd_soc_codec *codec = dai->codec;
-        u16 status, rate;
-
-        /* Gracefully shut down the voice interface. */
-        status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
-        rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
-        ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
-        schedule_timeout_interruptible(msecs_to_jiffies(1));
-        ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
-        ac97_write(codec, AC97_EXTENDED_MID, status);
-}
-
 static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
                              struct snd_soc_dai *dai)
 {
@@ -1019,7 +1025,6 @@ static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
 
 static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
         .hw_params      = wm9713_pcm_hw_params,
-        .shutdown       = wm9713_voiceshutdown,
         .set_clkdiv     = wm9713_set_dai_clkdiv,
         .set_pll        = wm9713_set_dai_pll,
         .set_fmt        = wm9713_set_dai_fmt,
@@ -1035,13 +1040,13 @@ struct snd_soc_dai wm9713_dai[] = {
                 .channels_min = 1,
                 .channels_max = 2,
                 .rates = WM9713_RATES,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+                .formats = SND_SOC_STD_AC97_FMTS,},
         .capture = {
                 .stream_name = "HiFi Capture",
                 .channels_min = 1,
                 .channels_max = 2,
                 .rates = WM9713_RATES,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+                .formats = SND_SOC_STD_AC97_FMTS,},
         .ops = &wm9713_dai_ops_hifi,
         },
         {
@@ -1051,7 +1056,7 @@ struct snd_soc_dai wm9713_dai[] = {
                 .channels_min = 1,
                 .channels_max = 1,
                 .rates = WM9713_RATES,
-                .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+                .formats = SND_SOC_STD_AC97_FMTS,},
         .ops = &wm9713_dai_ops_aux,
         },
         {
@@ -1069,6 +1074,7 @@ struct snd_soc_dai wm9713_dai[] = {
                 .rates = WM9713_PCM_RATES,
                 .formats = WM9713_PCM_FORMATS,},
         .ops = &wm9713_dai_ops_voice,
+        .symmetric_rates = 1,
         },
 };
 EXPORT_SYMBOL_GPL(wm9713_dai);
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index bd7392c9657e..411a710be660 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -10,13 +10,14 @@ config SND_DAVINCI_SOC_I2S
         tristate
 
 config SND_DAVINCI_SOC_EVM
-        tristate "SoC Audio support for DaVinci EVM"
-        depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM
+        tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+        depends on SND_DAVINCI_SOC
+        depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
         select SND_DAVINCI_SOC_I2S
         select SND_SOC_TLV320AIC3X
         help
           Say Y if you want to add support for SoC audio on TI
-          DaVinci EVM platform.
+          DaVinci DM6446 or DM355 EVM platforms.
 
 config SND_DAVINCI_SOC_SFFSDR
         tristate "SoC Audio support for SFFSDR"
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 9b90b347007c..58fd1cbedd88 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -20,7 +20,11 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/dma.h>
-#include <mach/hardware.h>
+#include <asm/mach-types.h>
+
+#include <mach/asp.h>
+#include <mach/edma.h>
+#include <mach/mux.h>
 
 #include "../codecs/tlv320aic3x.h"
 #include "davinci-pcm.h"
@@ -150,7 +154,7 @@ static struct snd_soc_card snd_soc_card_evm = {
 
 /* evm audio private data */
 static struct aic3x_setup_data evm_aic3x_setup = {
-        .i2c_bus = 0,
+        .i2c_bus = 1,
         .i2c_address = 0x1b,
 };
 
@@ -161,36 +165,73 @@ static struct snd_soc_device evm_snd_devdata = {
         .codec_data = &evm_aic3x_setup,
 };
 
+/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */
 static struct resource evm_snd_resources[] = {
         {
-                .start = DAVINCI_MCBSP_BASE,
-                .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
+                .start = DAVINCI_ASP0_BASE,
+                .end = DAVINCI_ASP0_BASE + SZ_8K - 1,
                 .flags = IORESOURCE_MEM,
         },
 };
 
 static struct evm_snd_platform_data evm_snd_data = {
-        .tx_dma_ch      = DM644X_DMACH_MCBSP_TX,
-        .rx_dma_ch      = DM644X_DMACH_MCBSP_RX,
+        .tx_dma_ch      = DAVINCI_DMA_ASP0_TX,
+        .rx_dma_ch      = DAVINCI_DMA_ASP0_RX,
+};
+
+/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */
+static struct resource dm335evm_snd_resources[] = {
+        {
+                .start = DAVINCI_ASP1_BASE,
+                .end = DAVINCI_ASP1_BASE + SZ_8K - 1,
+                .flags = IORESOURCE_MEM,
+        },
+};
+
+static struct evm_snd_platform_data dm335evm_snd_data = {
+        .tx_dma_ch      = DAVINCI_DMA_ASP1_TX,
+        .rx_dma_ch      = DAVINCI_DMA_ASP1_RX,
 };
 
 static struct platform_device *evm_snd_device;
 
 static int __init evm_init(void)
 {
+        struct resource *resources;
+        unsigned num_resources;
+        struct evm_snd_platform_data *data;
+        int index;
         int ret;
 
-        evm_snd_device = platform_device_alloc("soc-audio", 0);
+        if (machine_is_davinci_evm()) {
+                davinci_cfg_reg(DM644X_MCBSP);
+
+                resources = evm_snd_resources;
+                num_resources = ARRAY_SIZE(evm_snd_resources);
+                data = &evm_snd_data;
+                index = 0;
+        } else if (machine_is_davinci_dm355_evm()) {
+                /* we don't use ASP1 IRQs, or we'd need to mux them ... */
+                davinci_cfg_reg(DM355_EVT8_ASP1_TX);
+                davinci_cfg_reg(DM355_EVT9_ASP1_RX);
+
+                resources = dm335evm_snd_resources;
+                num_resources = ARRAY_SIZE(dm335evm_snd_resources);
+                data = &dm335evm_snd_data;
+                index = 1;
+        } else
+                return -EINVAL;
+
+        evm_snd_device = platform_device_alloc("soc-audio", index);
         if (!evm_snd_device)
                 return -ENOMEM;
 
         platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
         evm_snd_devdata.dev = &evm_snd_device->dev;
-        platform_device_add_data(evm_snd_device, &evm_snd_data,
-                                 sizeof(evm_snd_data));
+        platform_device_add_data(evm_snd_device, data, sizeof(*data));
 
-        ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
-                                            ARRAY_SIZE(evm_snd_resources));
+        ret = platform_device_add_resources(evm_snd_device, resources,
+                        num_resources);
         if (ret) {
                 platform_device_put(evm_snd_device);
                 return ret;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index ffdb9439d3d8..b1ea52fc83c7 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -24,6 +24,26 @@
 
 #include "davinci-pcm.h"
 
+
+/*
+ * NOTE:  terminology here is confusing.
+ *
+ *  - This driver supports the "Audio Serial Port" (ASP),
+ *    found on dm6446, dm355, and other DaVinci chips.
+ *
+ *  - But it labels it a "Multi-channel Buffered Serial Port"
+ *    (McBSP) as on older chips like the dm642 ... which was
+ *    backward-compatible, possibly explaining that confusion.
+ *
+ *  - OMAP chips have a controller called McBSP, which is
+ *    incompatible with the DaVinci flavor of McBSP.
+ *
+ *  - Newer DaVinci chips have a controller called McASP,
+ *    incompatible with ASP and with either McBSP.
+ *
+ * In short:  this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
 #define DAVINCI_MCBSP_DRR_REG   0x00
 #define DAVINCI_MCBSP_DXR_REG   0x04
 #define DAVINCI_MCBSP_SPCR_REG  0x08
@@ -421,7 +441,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_card *card = socdev->card;
-        struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+        struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
         struct davinci_mcbsp_dev *dev;
         struct resource *mem, *ioarea;
         struct evm_snd_platform_data *pdata;
@@ -448,7 +468,7 @@ static int davinci_i2s_probe(struct platform_device *pdev,
 
         cpu_dai->private_data = dev;
 
-        dev->clk = clk_get(&pdev->dev, "McBSPCLK");
+        dev->clk = clk_get(&pdev->dev, NULL);
         if (IS_ERR(dev->clk)) {
                 ret = -ENODEV;
                 goto err_free_mem;
@@ -483,7 +503,7 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_card *card = socdev->card;
-        struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai;
+        struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai;
         struct davinci_mcbsp_dev *dev = cpu_dai->private_data;
         struct resource *mem;
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 7af3b5b3a53d..a05996588489 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -22,6 +22,7 @@
 #include <sound/soc.h>
 
 #include <asm/dma.h>
+#include <mach/edma.h>
 
 #include "davinci-pcm.h"
 
@@ -51,7 +52,7 @@ struct davinci_runtime_data {
         spinlock_t lock;
         int period;             /* current DMA period */
         int master_lch;         /* Master DMA channel */
-        int slave_lch;          /* Slave DMA channel */
+        int slave_lch;          /* linked parameter RAM reload slot */
         struct davinci_pcm_dma_params *params;  /* DMA params */
 };
 
@@ -90,18 +91,18 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
                 dst_bidx = data_type;
         }
 
-        davinci_set_dma_src_params(lch, src, INCR, W8BIT);
-        davinci_set_dma_dest_params(lch, dst, INCR, W8BIT);
-        davinci_set_dma_src_index(lch, src_bidx, 0);
-        davinci_set_dma_dest_index(lch, dst_bidx, 0);
-        davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC);
+        edma_set_src(lch, src, INCR, W8BIT);
+        edma_set_dest(lch, dst, INCR, W8BIT);
+        edma_set_src_index(lch, src_bidx, 0);
+        edma_set_dest_index(lch, dst_bidx, 0);
+        edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC);
 
         prtd->period++;
         if (unlikely(prtd->period >= runtime->periods))
                 prtd->period = 0;
 }
 
-static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
 {
         struct snd_pcm_substream *substream = data;
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
@@ -125,7 +126,7 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data;
-        int tcc = TCC_ANY;
+        struct edmacc_param p_ram;
         int ret;
 
         if (!dma_data)
@@ -134,22 +135,34 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
         prtd->params = dma_data;
 
         /* Request master DMA channel */
-        ret = davinci_request_dma(prtd->params->channel, prtd->params->name,
+        ret = edma_alloc_channel(prtd->params->channel,
                                   davinci_pcm_dma_irq, substream,
-                                  &prtd->master_lch, &tcc, EVENTQ_0);
-        if (ret)
+                                  EVENTQ_0);
+        if (ret < 0)
                 return ret;
+        prtd->master_lch = ret;
 
-        /* Request slave DMA channel */
-        ret = davinci_request_dma(PARAM_ANY, "Link",
-                                  NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0);
-        if (ret) {
-                davinci_free_dma(prtd->master_lch);
+        /* Request parameter RAM reload slot */
+        ret = edma_alloc_slot(EDMA_SLOT_ANY);
+        if (ret < 0) {
+                edma_free_channel(prtd->master_lch);
                 return ret;
         }
-
-        /* Link slave DMA channel in loopback */
-        davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch);
+        prtd->slave_lch = ret;
+
+        /* Issue transfer completion IRQ when the channel completes a
+         * transfer, then always reload from the same slot (by a kind
+         * of loopback link).  The completion IRQ handler will update
+         * the reload slot with a new buffer.
+         *
+         * REVISIT save p_ram here after setting up everything except
+         * the buffer and its length (ccnt) ... use it as a template
+         * so davinci_pcm_enqueue_dma() takes less time in IRQ.
+         */
+        edma_read_slot(prtd->slave_lch, &p_ram);
+        p_ram.opt |= TCINTEN | EDMA_TCC(prtd->master_lch);
+        p_ram.link_bcntrld = prtd->slave_lch << 5;
+        edma_write_slot(prtd->slave_lch, &p_ram);
 
         return 0;
 }
@@ -165,12 +178,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
         case SNDRV_PCM_TRIGGER_START:
         case SNDRV_PCM_TRIGGER_RESUME:
         case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-                davinci_start_dma(prtd->master_lch);
+                edma_start(prtd->master_lch);
                 break;
         case SNDRV_PCM_TRIGGER_STOP:
         case SNDRV_PCM_TRIGGER_SUSPEND:
         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-                davinci_stop_dma(prtd->master_lch);
+                edma_stop(prtd->master_lch);
                 break;
         default:
                 ret = -EINVAL;
@@ -185,14 +198,14 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 static int davinci_pcm_prepare(struct snd_pcm_substream *substream)
 {
         struct davinci_runtime_data *prtd = substream->runtime->private_data;
-        struct paramentry_descriptor temp;
+        struct edmacc_param temp;
 
         prtd->period = 0;
         davinci_pcm_enqueue_dma(substream);
 
-        /* Get slave channel dma params for master channel startup */
-        davinci_get_dma_params(prtd->slave_lch, &temp);
-        davinci_set_dma_params(prtd->master_lch, &temp);
+        /* Copy self-linked parameter RAM entry into master channel */
+        edma_read_slot(prtd->slave_lch, &temp);
+        edma_write_slot(prtd->master_lch, &temp);
 
         return 0;
 }
@@ -208,7 +221,7 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
 
         spin_lock(&prtd->lock);
 
-        davinci_dma_getposition(prtd->master_lch, &src, &dst);
+        edma_get_position(prtd->master_lch, &src, &dst);
         if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
                 count = src - runtime->dma_addr;
         else
@@ -253,10 +266,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream)
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct davinci_runtime_data *prtd = runtime->private_data;
 
-        davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch);
+        edma_unlink(prtd->slave_lch);
 
-        davinci_free_dma(prtd->slave_lch);
-        davinci_free_dma(prtd->master_lch);
+        edma_free_slot(prtd->slave_lch);
+        edma_free_channel(prtd->master_lch);
 
         kfree(prtd);
 
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 9fc908283371..5dbebf82249c 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,5 +1,8 @@
 config SND_SOC_OF_SIMPLE
         tristate
+        
+config SND_MPC52xx_DMA
+        tristate
 
 # ASoC platform support for the Freescale MPC8610 SOC.  This compiles drivers
 # for the SSI and the Elo DMA controller.  You will still need to select
@@ -22,7 +25,34 @@ config SND_SOC_MPC8610_HPCD
 config SND_SOC_MPC5200_I2S
         tristate "Freescale MPC5200 PSC in I2S mode driver"
         depends on PPC_MPC52xx && PPC_BESTCOMM
-        select SND_SOC_OF_SIMPLE
+        select SND_MPC52xx_DMA
         select PPC_BESTCOMM_GEN_BD
         help
           Say Y here to support the MPC5200 PSCs in I2S mode.
+
+config SND_SOC_MPC5200_AC97
+        tristate "Freescale MPC5200 PSC in AC97 mode driver"
+        depends on PPC_MPC52xx && PPC_BESTCOMM
+        select AC97_BUS
+        select SND_MPC52xx_DMA
+        select PPC_BESTCOMM_GEN_BD
+        help
+          Say Y here to support the MPC5200 PSCs in AC97 mode.
+
+config SND_MPC52xx_SOC_PCM030
+        tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
+        depends on PPC_MPC5200_SIMPLE && BROKEN
+        select SND_SOC_MPC5200_AC97
+        select SND_SOC_WM9712
+        help
+          Say Y if you want to add support for sound on the Phytec pcm030
+          baseboard.
+
+config SND_MPC52xx_SOC_EFIKA
+        tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
+        depends on PPC_EFIKA && BROKEN
+        select SND_SOC_MPC5200_AC97
+        select SND_SOC_STAC9766
+        help
+          Say Y if you want to add support for sound on the Efika.
+
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index f85134c86387..a83a73967ec6 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -10,5 +10,12 @@ snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-dma-objs := fsl_dma.o
 obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
 
+# MPC5200 Platform Support
+obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
 obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
+obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
+
+# MPC5200 Machine Support
+obj-$(CONFIG_SND_MPC52xx_SOC_PCM030) += pcm030-audio-fabric.o
+obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o
 
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
new file mode 100644
index 000000000000..85b0e7569504
--- /dev/null
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -0,0 +1,90 @@
+/*
+ * Efika driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/stac9766.h"
+
+static struct snd_soc_device device;
+static struct snd_soc_card card;
+
+static struct snd_soc_dai_link efika_fabric_dai[] = {
+{
+        .name = "AC97",
+        .stream_name = "AC97 Analog",
+        .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_ANALOG],
+        .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
+},
+{
+        .name = "AC97",
+        .stream_name = "AC97 IEC958",
+        .codec_dai = &stac9766_dai[STAC9766_DAI_AC97_DIGITAL],
+        .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
+},
+};
+
+static __init int efika_fabric_init(void)
+{
+        struct platform_device *pdev;
+        int rc;
+
+        if (!machine_is_compatible("bplan,efika"))
+                return -ENODEV;
+
+        card.platform = &mpc5200_audio_dma_platform;
+        card.name = "Efika";
+        card.dai_link = efika_fabric_dai;
+        card.num_links = ARRAY_SIZE(efika_fabric_dai);
+
+        device.card = &card;
+        device.codec_dev = &soc_codec_dev_stac9766;
+
+        pdev = platform_device_alloc("soc-audio", 1);
+        if (!pdev) {
+                pr_err("efika_fabric_init: platform_device_alloc() failed\n");
+                return -ENODEV;
+        }
+
+        platform_set_drvdata(pdev, &device);
+        device.dev = &pdev->dev;
+
+        rc = platform_device_add(pdev);
+        if (rc) {
+                pr_err("efika_fabric_init: platform_device_add() failed\n");
+                return -ENODEV;
+        }
+        return 0;
+}
+
+module_init(efika_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 Efika fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 3711d8454d96..93f0f38a32c9 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -375,18 +375,14 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
                 struct snd_pcm_runtime *first_runtime =
                         ssi_private->first_stream->runtime;
 
-                if (!first_runtime->rate || !first_runtime->sample_bits) {
+                if (!first_runtime->sample_bits) {
                         dev_err(substream->pcm->card->dev,
-                                "set sample rate and size in %s stream first\n",
+                                "set sample size in %s stream first\n",
                                 substream->stream == SNDRV_PCM_STREAM_PLAYBACK
                                 ? "capture" : "playback");
                         return -EAGAIN;
                 }
 
-                snd_pcm_hw_constraint_minmax(substream->runtime,
-                        SNDRV_PCM_HW_PARAM_RATE,
-                        first_runtime->rate, first_runtime->rate);
-
                 /* If we're in synchronous mode, then we need to constrain
                  * the sample size as well.  We don't support independent sample
                  * rates in asynchronous mode.
@@ -674,7 +670,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
         ssi_private->dev = ssi_info->dev;
         ssi_private->asynchronous = ssi_info->asynchronous;
 
-        ssi_private->dev->driver_data = fsl_ssi_dai;
+        dev_set_drvdata(ssi_private->dev, fsl_ssi_dai);
 
         /* Initialize the the device_attribute structure */
         dev_attr->attr.name = "ssi-stats";
@@ -693,6 +689,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
         fsl_ssi_dai->name = ssi_private->name;
         fsl_ssi_dai->id = ssi_info->id;
         fsl_ssi_dai->dev = ssi_info->dev;
+        fsl_ssi_dai->symmetric_rates = 1;
 
         ret = snd_soc_register_dai(fsl_ssi_dai);
         if (ret != 0) {
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
new file mode 100644
index 000000000000..efec33a1c5bd
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -0,0 +1,564 @@
+/*
+ * Freescale MPC5200 PSC DMA
+ * ALSA SoC Platform driver
+ *
+ * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+
+#include <sound/soc.h>
+
+#include <sysdev/bestcomm/bestcomm.h>
+#include <sysdev/bestcomm/gen_bd.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+
+/*
+ * Interrupt handlers
+ */
+static irqreturn_t psc_dma_status_irq(int irq, void *_psc_dma)
+{
+        struct psc_dma *psc_dma = _psc_dma;
+        struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+        u16 isr;
+
+        isr = in_be16(&regs->mpc52xx_psc_isr);
+
+        /* Playback underrun error */
+        if (psc_dma->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
+                psc_dma->stats.underrun_count++;
+
+        /* Capture overrun error */
+        if (psc_dma->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
+                psc_dma->stats.overrun_count++;
+
+        out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+
+        return IRQ_HANDLED;
+}
+
+/**
+ * psc_dma_bcom_enqueue_next_buffer - Enqueue another audio buffer
+ * @s: pointer to stream private data structure
+ *
+ * Enqueues another audio period buffer into the bestcomm queue.
+ *
+ * Note: The routine must only be called when there is space available in
+ * the queue.  Otherwise the enqueue will fail and the audio ring buffer
+ * will get out of sync
+ */
+static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
+{
+        struct bcom_bd *bd;
+
+        /* Prepare and enqueue the next buffer descriptor */
+        bd = bcom_prepare_next_buffer(s->bcom_task);
+        bd->status = s->period_bytes;
+        bd->data[0] = s->period_next_pt;
+        bcom_submit_next_buffer(s->bcom_task, NULL);
+
+        /* Update for next period */
+        s->period_next_pt += s->period_bytes;
+        if (s->period_next_pt >= s->period_end)
+                s->period_next_pt = s->period_start;
+}
+
+static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s)
+{
+        while (s->appl_ptr < s->runtime->control->appl_ptr) {
+
+                if (bcom_queue_full(s->bcom_task))
+                        return;
+
+                s->appl_ptr += s->period_size;
+
+                psc_dma_bcom_enqueue_next_buffer(s);
+        }
+}
+
+/* Bestcomm DMA irq handler */
+static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream)
+{
+        struct psc_dma_stream *s = _psc_dma_stream;
+
+        spin_lock(&s->psc_dma->lock);
+        /* For each finished period, dequeue the completed period buffer
+         * and enqueue a new one in it's place. */
+        while (bcom_buffer_done(s->bcom_task)) {
+                bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+                s->period_current_pt += s->period_bytes;
+                if (s->period_current_pt >= s->period_end)
+                        s->period_current_pt = s->period_start;
+        }
+        psc_dma_bcom_enqueue_tx(s);
+        spin_unlock(&s->psc_dma->lock);
+
+        /* If the stream is active, then also inform the PCM middle layer
+         * of the period finished event. */
+        if (s->active)
+                snd_pcm_period_elapsed(s->stream);
+
+        return IRQ_HANDLED;
+}
+
+static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream)
+{
+        struct psc_dma_stream *s = _psc_dma_stream;
+
+        spin_lock(&s->psc_dma->lock);
+        /* For each finished period, dequeue the completed period buffer
+         * and enqueue a new one in it's place. */
+        while (bcom_buffer_done(s->bcom_task)) {
+                bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
+
+                s->period_current_pt += s->period_bytes;
+                if (s->period_current_pt >= s->period_end)
+                        s->period_current_pt = s->period_start;
+
+                psc_dma_bcom_enqueue_next_buffer(s);
+        }
+        spin_unlock(&s->psc_dma->lock);
+
+        /* If the stream is active, then also inform the PCM middle layer
+         * of the period finished event. */
+        if (s->active)
+                snd_pcm_period_elapsed(s->stream);
+
+        return IRQ_HANDLED;
+}
+
+static int psc_dma_hw_free(struct snd_pcm_substream *substream)
+{
+        snd_pcm_set_runtime_buffer(substream, NULL);
+        return 0;
+}
+
+/**
+ * psc_dma_trigger: start and stop the DMA transfer.
+ *
+ * This function is called by ALSA to start, stop, pause, and resume the DMA
+ * transfer of data.
+ */
+static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct psc_dma_stream *s;
+        struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+        u16 imr;
+        unsigned long flags;
+        int i;
+
+        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+                s = &psc_dma->capture;
+        else
+                s = &psc_dma->playback;
+
+        dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)"
+                " stream_id=%i\n",
+                substream, cmd, substream->pstr->stream);
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+                s->period_bytes = frames_to_bytes(runtime,
+                                                  runtime->period_size);
+                s->period_start = virt_to_phys(runtime->dma_area);
+                s->period_end = s->period_start +
+                                (s->period_bytes * runtime->periods);
+                s->period_next_pt = s->period_start;
+                s->period_current_pt = s->period_start;
+                s->period_size = runtime->period_size;
+                s->active = 1;
+
+                /* track appl_ptr so that we have a better chance of detecting
+                 * end of stream and not over running it.
+                 */
+                s->runtime = runtime;
+                s->appl_ptr = s->runtime->control->appl_ptr -
+                                (runtime->period_size * runtime->periods);
+
+                /* Fill up the bestcomm bd queue and enable DMA.
+                 * This will begin filling the PSC's fifo.
+                 */
+                spin_lock_irqsave(&psc_dma->lock, flags);
+
+                if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+                        bcom_gen_bd_rx_reset(s->bcom_task);
+                        for (i = 0; i < runtime->periods; i++)
+                                if (!bcom_queue_full(s->bcom_task))
+                                        psc_dma_bcom_enqueue_next_buffer(s);
+                } else {
+                        bcom_gen_bd_tx_reset(s->bcom_task);
+                        psc_dma_bcom_enqueue_tx(s);
+                }
+
+                bcom_enable(s->bcom_task);
+                spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+                out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
+
+                break;
+
+        case SNDRV_PCM_TRIGGER_STOP:
+                s->active = 0;
+
+                spin_lock_irqsave(&psc_dma->lock, flags);
+                bcom_disable(s->bcom_task);
+                if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+                        bcom_gen_bd_rx_reset(s->bcom_task);
+                else
+                        bcom_gen_bd_tx_reset(s->bcom_task);
+                spin_unlock_irqrestore(&psc_dma->lock, flags);
+
+                break;
+
+        default:
+                dev_dbg(psc_dma->dev, "invalid command\n");
+                return -EINVAL;
+        }
+
+        /* Update interrupt enable settings */
+        imr = 0;
+        if (psc_dma->playback.active)
+                imr |= MPC52xx_PSC_IMR_TXEMP;
+        if (psc_dma->capture.active)
+                imr |= MPC52xx_PSC_IMR_ORERR;
+        out_be16(&regs->isr_imr.imr, psc_dma->imr | imr);
+
+        return 0;
+}
+
+
+/* ---------------------------------------------------------------------
+ * The PSC DMA 'ASoC platform' driver
+ *
+ * Can be referenced by an 'ASoC machine' driver
+ * This driver only deals with the audio bus; it doesn't have any
+ * interaction with the attached codec
+ */
+
+static const struct snd_pcm_hardware psc_dma_hardware = {
+        .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_BATCH,
+        .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
+                SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
+        .rate_min = 8000,
+        .rate_max = 48000,
+        .channels_min = 1,
+        .channels_max = 2,
+        .period_bytes_max       = 1024 * 1024,
+        .period_bytes_min       = 32,
+        .periods_min            = 2,
+        .periods_max            = 256,
+        .buffer_bytes_max       = 2 * 1024 * 1024,
+        .fifo_size              = 512,
+};
+
+static int psc_dma_open(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+        struct psc_dma_stream *s;
+        int rc;
+
+        dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream);
+
+        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+                s = &psc_dma->capture;
+        else
+                s = &psc_dma->playback;
+
+        snd_soc_set_runtime_hwparams(substream, &psc_dma_hardware);
+
+        rc = snd_pcm_hw_constraint_integer(runtime,
+                SNDRV_PCM_HW_PARAM_PERIODS);
+        if (rc < 0) {
+                dev_err(substream->pcm->card->dev, "invalid buffer size\n");
+                return rc;
+        }
+
+        s->stream = substream;
+        return 0;
+}
+
+static int psc_dma_close(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+        struct psc_dma_stream *s;
+
+        dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
+
+        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+                s = &psc_dma->capture;
+        else
+                s = &psc_dma->playback;
+
+        if (!psc_dma->playback.active &&
+            !psc_dma->capture.active) {
+
+                /* Disable all interrupts and reset the PSC */
+                out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+                out_8(&psc_dma->psc_regs->command, 4 << 4); /* reset error */
+        }
+        s->stream = NULL;
+        return 0;
+}
+
+static snd_pcm_uframes_t
+psc_dma_pointer(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+        struct psc_dma_stream *s;
+        dma_addr_t count;
+
+        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+                s = &psc_dma->capture;
+        else
+                s = &psc_dma->playback;
+
+        count = s->period_current_pt - s->period_start;
+
+        return bytes_to_frames(substream->runtime, count);
+}
+
+static int
+psc_dma_hw_params(struct snd_pcm_substream *substream,
+                         struct snd_pcm_hw_params *params)
+{
+        snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+        return 0;
+}
+
+static struct snd_pcm_ops psc_dma_ops = {
+        .open           = psc_dma_open,
+        .close          = psc_dma_close,
+        .hw_free        = psc_dma_hw_free,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .pointer        = psc_dma_pointer,
+        .trigger        = psc_dma_trigger,
+        .hw_params      = psc_dma_hw_params,
+};
+
+static u64 psc_dma_dmamask = 0xffffffff;
+static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
+                           struct snd_pcm *pcm)
+{
+        struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+        struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+        size_t size = psc_dma_hardware.buffer_bytes_max;
+        int rc = 0;
+
+        dev_dbg(rtd->socdev->dev, "psc_dma_new(card=%p, dai=%p, pcm=%p)\n",
+                card, dai, pcm);
+
+        if (!card->dev->dma_mask)
+                card->dev->dma_mask = &psc_dma_dmamask;
+        if (!card->dev->coherent_dma_mask)
+                card->dev->coherent_dma_mask = 0xffffffff;
+
+        if (pcm->streams[0].substream) {
+                rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+                                size, &pcm->streams[0].substream->dma_buffer);
+                if (rc)
+                        goto playback_alloc_err;
+        }
+
+        if (pcm->streams[1].substream) {
+                rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev,
+                                size, &pcm->streams[1].substream->dma_buffer);
+                if (rc)
+                        goto capture_alloc_err;
+        }
+
+        if (rtd->socdev->card->codec->ac97)
+                rtd->socdev->card->codec->ac97->private_data = psc_dma;
+
+        return 0;
+
+ capture_alloc_err:
+        if (pcm->streams[0].substream)
+                snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+
+ playback_alloc_err:
+        dev_err(card->dev, "Cannot allocate buffer(s)\n");
+
+        return -ENOMEM;
+}
+
+static void psc_dma_free(struct snd_pcm *pcm)
+{
+        struct snd_soc_pcm_runtime *rtd = pcm->private_data;
+        struct snd_pcm_substream *substream;
+        int stream;
+
+        dev_dbg(rtd->socdev->dev, "psc_dma_free(pcm=%p)\n", pcm);
+
+        for (stream = 0; stream < 2; stream++) {
+                substream = pcm->streams[stream].substream;
+                if (substream) {
+                        snd_dma_free_pages(&substream->dma_buffer);
+                        substream->dma_buffer.area = NULL;
+                        substream->dma_buffer.addr = 0;
+                }
+        }
+}
+
+struct snd_soc_platform mpc5200_audio_dma_platform = {
+        .name           = "mpc5200-psc-audio",
+        .pcm_ops        = &psc_dma_ops,
+        .pcm_new        = &psc_dma_new,
+        .pcm_free       = &psc_dma_free,
+};
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_platform);
+
+int mpc5200_audio_dma_create(struct of_device *op)
+{
+        phys_addr_t fifo;
+        struct psc_dma *psc_dma;
+        struct resource res;
+        int size, irq, rc;
+        const __be32 *prop;
+        void __iomem *regs;
+
+        /* Fetch the registers and IRQ of the PSC */
+        irq = irq_of_parse_and_map(op->node, 0);
+        if (of_address_to_resource(op->node, 0, &res)) {
+                dev_err(&op->dev, "Missing reg property\n");
+                return -ENODEV;
+        }
+        regs = ioremap(res.start, 1 + res.end - res.start);
+        if (!regs) {
+                dev_err(&op->dev, "Could not map registers\n");
+                return -ENODEV;
+        }
+
+        /* Allocate and initialize the driver private data */
+        psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL);
+        if (!psc_dma) {
+                iounmap(regs);
+                return -ENOMEM;
+        }
+
+        /* Get the PSC ID */
+        prop = of_get_property(op->node, "cell-index", &size);
+        if (!prop || size < sizeof *prop)
+                return -ENODEV;
+
+        spin_lock_init(&psc_dma->lock);
+        psc_dma->id = be32_to_cpu(*prop);
+        psc_dma->irq = irq;
+        psc_dma->psc_regs = regs;
+        psc_dma->fifo_regs = regs + sizeof *psc_dma->psc_regs;
+        psc_dma->dev = &op->dev;
+        psc_dma->playback.psc_dma = psc_dma;
+        psc_dma->capture.psc_dma = psc_dma;
+        snprintf(psc_dma->name, sizeof psc_dma->name, "PSC%u", psc_dma->id);
+
+        /* Find the address of the fifo data registers and setup the
+         * DMA tasks */
+        fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
+        psc_dma->capture.bcom_task =
+                bcom_psc_gen_bd_rx_init(psc_dma->id, 10, fifo, 512);
+        psc_dma->playback.bcom_task =
+                bcom_psc_gen_bd_tx_init(psc_dma->id, 10, fifo);
+        if (!psc_dma->capture.bcom_task ||
+            !psc_dma->playback.bcom_task) {
+                dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
+                iounmap(regs);
+                kfree(psc_dma);
+                return -ENODEV;
+        }
+
+        /* Disable all interrupts and reset the PSC */
+        out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+         /* reset receiver */
+        out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_RX);
+         /* reset transmitter */
+        out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_TX);
+         /* reset error */
+        out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_RST_ERR_STAT);
+         /* reset mode */
+        out_8(&psc_dma->psc_regs->command, MPC52xx_PSC_SEL_MODE_REG_1);
+
+        /* Set up mode register;
+         * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
+         * Second write: register Normal mode for non loopback
+         */
+        out_8(&psc_dma->psc_regs->mode, 0);
+        out_8(&psc_dma->psc_regs->mode, 0);
+
+        /* Set the TX and RX fifo alarm thresholds */
+        out_be16(&psc_dma->fifo_regs->rfalarm, 0x100);
+        out_8(&psc_dma->fifo_regs->rfcntl, 0x4);
+        out_be16(&psc_dma->fifo_regs->tfalarm, 0x100);
+        out_8(&psc_dma->fifo_regs->tfcntl, 0x7);
+
+        /* Lookup the IRQ numbers */
+        psc_dma->playback.irq =
+                bcom_get_task_irq(psc_dma->playback.bcom_task);
+        psc_dma->capture.irq =
+                bcom_get_task_irq(psc_dma->capture.bcom_task);
+
+        rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED,
+                         "psc-dma-status", psc_dma);
+        rc |= request_irq(psc_dma->capture.irq,
+                          &psc_dma_bcom_irq_rx, IRQF_SHARED,
+                          "psc-dma-capture", &psc_dma->capture);
+        rc |= request_irq(psc_dma->playback.irq,
+                          &psc_dma_bcom_irq_tx, IRQF_SHARED,
+                          "psc-dma-playback", &psc_dma->playback);
+        if (rc) {
+                free_irq(psc_dma->irq, psc_dma);
+                free_irq(psc_dma->capture.irq,
+                         &psc_dma->capture);
+                free_irq(psc_dma->playback.irq,
+                         &psc_dma->playback);
+                return -ENODEV;
+        }
+
+        /* Save what we've done so it can be found again later */
+        dev_set_drvdata(&op->dev, psc_dma);
+
+        /* Tell the ASoC OF helpers about it */
+        return snd_soc_register_platform(&mpc5200_audio_dma_platform);
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);
+
+int mpc5200_audio_dma_destroy(struct of_device *op)
+{
+        struct psc_dma *psc_dma = dev_get_drvdata(&op->dev);
+
+        dev_dbg(&op->dev, "mpc5200_audio_dma_destroy()\n");
+
+        snd_soc_unregister_platform(&mpc5200_audio_dma_platform);
+
+        bcom_gen_bd_rx_release(psc_dma->capture.bcom_task);
+        bcom_gen_bd_tx_release(psc_dma->playback.bcom_task);
+
+        /* Release irqs */
+        free_irq(psc_dma->irq, psc_dma);
+        free_irq(psc_dma->capture.irq, &psc_dma->capture);
+        free_irq(psc_dma->playback.irq, &psc_dma->playback);
+
+        iounmap(psc_dma->psc_regs);
+        kfree(psc_dma);
+        dev_set_drvdata(&op->dev, NULL);
+
+        return 0;
+}
+EXPORT_SYMBOL_GPL(mpc5200_audio_dma_destroy);
+
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in DMA mode ASoC Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
new file mode 100644
index 000000000000..2000803f06a7
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -0,0 +1,80 @@
+/*
+ * Freescale MPC5200 Audio DMA driver
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC5200_DMA_H__
+#define __SOUND_SOC_FSL_MPC5200_DMA_H__
+
+#define PSC_STREAM_NAME_LEN 32
+
+/**
+ * psc_ac97_stream - Data specific to a single stream (playback or capture)
+ * @active:             flag indicating if the stream is active
+ * @psc_dma:            pointer back to parent psc_dma data structure
+ * @bcom_task:          bestcomm task structure
+ * @irq:                irq number for bestcomm task
+ * @period_start:       physical address of start of DMA region
+ * @period_end:         physical address of end of DMA region
+ * @period_next_pt:     physical address of next DMA buffer to enqueue
+ * @period_bytes:       size of DMA period in bytes
+ */
+struct psc_dma_stream {
+        struct snd_pcm_runtime *runtime;
+        snd_pcm_uframes_t appl_ptr;
+
+        int active;
+        struct psc_dma *psc_dma;
+        struct bcom_task *bcom_task;
+        int irq;
+        struct snd_pcm_substream *stream;
+        dma_addr_t period_start;
+        dma_addr_t period_end;
+        dma_addr_t period_next_pt;
+        dma_addr_t period_current_pt;
+        int period_bytes;
+        int period_size;
+};
+
+/**
+ * psc_dma - Private driver data
+ * @name: short name for this device ("PSC0", "PSC1", etc)
+ * @psc_regs: pointer to the PSC's registers
+ * @fifo_regs: pointer to the PSC's FIFO registers
+ * @irq: IRQ of this PSC
+ * @dev: struct device pointer
+ * @dai: the CPU DAI for this device
+ * @sicr: Base value used in serial interface control register; mode is ORed
+ *        with this value.
+ * @playback: Playback stream context data
+ * @capture: Capture stream context data
+ */
+struct psc_dma {
+        char name[32];
+        struct mpc52xx_psc __iomem *psc_regs;
+        struct mpc52xx_psc_fifo __iomem *fifo_regs;
+        unsigned int irq;
+        struct device *dev;
+        spinlock_t lock;
+        u32 sicr;
+        uint sysclk;
+        int imr;
+        int id;
+        unsigned int slots;
+
+        /* per-stream data */
+        struct psc_dma_stream playback;
+        struct psc_dma_stream capture;
+
+        /* Statistics */
+        struct {
+                unsigned long overrun_count;
+                unsigned long underrun_count;
+        } stats;
+};
+
+int mpc5200_audio_dma_create(struct of_device *op);
+int mpc5200_audio_dma_destroy(struct of_device *op);
+
+extern struct snd_soc_platform mpc5200_audio_dma_platform;
+
+#endif /* __SOUND_SOC_FSL_MPC5200_DMA_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
new file mode 100644
index 000000000000..794a247b3eb5
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -0,0 +1,329 @@
+/*
+ * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
+ *
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/time.h>
+#include <asm/delay.h>
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+
+#define DRV_NAME "mpc5200-psc-ac97"
+
+/* ALSA only supports a single AC97 device so static is recommend here */
+static struct psc_dma *psc_dma;
+
+static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+        int status;
+        unsigned int val;
+
+        /* Wait for command send status zero = ready */
+        status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+                                MPC52xx_PSC_SR_CMDSEND), 100, 0);
+        if (status == 0) {
+                pr_err("timeout on ac97 bus (rdy)\n");
+                return -ENODEV;
+        }
+        /* Send the read */
+        out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));
+
+        /* Wait for the answer */
+        status = spin_event_timeout((in_be16(&psc_dma->psc_regs->sr_csr.status) &
+                                MPC52xx_PSC_SR_DATA_VAL), 100, 0);
+        if (status == 0) {
+                pr_err("timeout on ac97 read (val) %x\n",
+                                in_be16(&psc_dma->psc_regs->sr_csr.status));
+                return -ENODEV;
+        }
+        /* Get the data */
+        val = in_be32(&psc_dma->psc_regs->ac97_data);
+        if (((val >> 24) & 0x7f) != reg) {
+                pr_err("reg echo error on ac97 read\n");
+                return -ENODEV;
+        }
+        val = (val >> 8) & 0xffff;
+
+        return (unsigned short) val;
+}
+
+static void psc_ac97_write(struct snd_ac97 *ac97,
+                                unsigned short reg, unsigned short val)
+{
+        int status;
+
+        /* Wait for command status zero = ready */
+        status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+                                MPC52xx_PSC_SR_CMDSEND), 100, 0);
+        if (status == 0) {
+                pr_err("timeout on ac97 bus (write)\n");
+                return;
+        }
+        /* Write data */
+        out_be32(&psc_dma->psc_regs->ac97_cmd,
+                        ((reg & 0x7f) << 24) | (val << 8));
+}
+
+static void psc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+        struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+        out_be32(&regs->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR);
+        udelay(3);
+        out_be32(&regs->sicr, psc_dma->sicr);
+}
+
+static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+        struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+        /* Do a cold reset */
+        out_8(&regs->op1, MPC52xx_PSC_OP_RES);
+        udelay(10);
+        out_8(&regs->op0, MPC52xx_PSC_OP_RES);
+        udelay(50);
+        psc_ac97_warm_reset(ac97);
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+        .read           = psc_ac97_read,
+        .write          = psc_ac97_write,
+        .reset          = psc_ac97_cold_reset,
+        .warm_reset     = psc_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
+                                 struct snd_pcm_hw_params *params,
+                                 struct snd_soc_dai *cpu_dai)
+{
+        struct psc_dma *psc_dma = cpu_dai->private_data;
+
+        dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+                " periods=%i buffer_size=%i  buffer_bytes=%i channels=%i"
+                " rate=%i format=%i\n",
+                __func__, substream, params_period_size(params),
+                params_period_bytes(params), params_periods(params),
+                params_buffer_size(params), params_buffer_bytes(params),
+                params_channels(params), params_rate(params),
+                params_format(params));
+
+
+        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+                if (params_channels(params) == 1)
+                        psc_dma->slots |= 0x00000100;
+                else
+                        psc_dma->slots |= 0x00000300;
+        } else {
+                if (params_channels(params) == 1)
+                        psc_dma->slots |= 0x01000000;
+                else
+                        psc_dma->slots |= 0x03000000;
+        }
+        out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+
+        return 0;
+}
+
+static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream,
+                                 struct snd_pcm_hw_params *params,
+                                 struct snd_soc_dai *cpu_dai)
+{
+        struct psc_dma *psc_dma = cpu_dai->private_data;
+
+        if (params_channels(params) == 1)
+                out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000);
+        else
+                out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000);
+
+        return 0;
+}
+
+static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+                                                        struct snd_soc_dai *dai)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_STOP:
+                if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+                        psc_dma->slots &= 0xFFFF0000;
+                else
+                        psc_dma->slots &= 0x0000FFFF;
+
+                out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+                break;
+        }
+        return 0;
+}
+
+static int psc_ac97_probe(struct platform_device *pdev,
+                                        struct snd_soc_dai *cpu_dai)
+{
+        struct psc_dma *psc_dma = cpu_dai->private_data;
+        struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+        /* Go */
+        out_8(&regs->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
+        return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_ac97_dai_template: template CPU Digital Audio Interface
+ */
+static struct snd_soc_dai_ops psc_ac97_analog_ops = {
+        .hw_params      = psc_ac97_hw_analog_params,
+        .trigger        = psc_ac97_trigger,
+};
+
+static struct snd_soc_dai_ops psc_ac97_digital_ops = {
+        .hw_params      = psc_ac97_hw_digital_params,
+};
+
+struct snd_soc_dai psc_ac97_dai[] = {
+{
+        .name   = "AC97",
+        .ac97_control = 1,
+        .probe  = psc_ac97_probe,
+        .playback = {
+                .channels_min   = 1,
+                .channels_max   = 6,
+                .rates          = SNDRV_PCM_RATE_8000_48000,
+                .formats = SNDRV_PCM_FMTBIT_S32_BE,
+        },
+        .capture = {
+                .channels_min   = 1,
+                .channels_max   = 2,
+                .rates          = SNDRV_PCM_RATE_8000_48000,
+                .formats = SNDRV_PCM_FMTBIT_S32_BE,
+        },
+        .ops = &psc_ac97_analog_ops,
+},
+{
+        .name   = "SPDIF",
+        .ac97_control = 1,
+        .playback = {
+                .channels_min   = 1,
+                .channels_max   = 2,
+                .rates          = SNDRV_PCM_RATE_32000 | \
+                        SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+                .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+        },
+        .ops = &psc_ac97_digital_ops,
+} };
+EXPORT_SYMBOL_GPL(psc_ac97_dai);
+
+
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int __devinit psc_ac97_of_probe(struct of_device *op,
+                                      const struct of_device_id *match)
+{
+        int rc, i;
+        struct snd_ac97 ac97;
+        struct mpc52xx_psc __iomem *regs;
+
+        rc = mpc5200_audio_dma_create(op);
+        if (rc != 0)
+                return rc;
+
+        for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+                psc_ac97_dai[i].dev = &op->dev;
+
+        rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
+        if (rc != 0) {
+                dev_err(&op->dev, "Failed to register DAI\n");
+                return rc;
+        }
+
+        psc_dma = dev_get_drvdata(&op->dev);
+        regs = psc_dma->psc_regs;
+        ac97.private_data = psc_dma;
+
+        for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+                psc_ac97_dai[i].private_data = psc_dma;
+
+        psc_dma->imr = 0;
+        out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+
+        /* Configure the serial interface mode to AC97 */
+        psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97;
+        out_be32(&regs->sicr, psc_dma->sicr);
+
+        /* No slots active */
+        out_be32(&regs->ac97_slots, 0x00000000);
+
+        return 0;
+}
+
+static int __devexit psc_ac97_of_remove(struct of_device *op)
+{
+        return mpc5200_audio_dma_destroy(op);
+}
+
+/* Match table for of_platform binding */
+static struct of_device_id psc_ac97_match[] __devinitdata = {
+        { .compatible = "fsl,mpc5200-psc-ac97", },
+        { .compatible = "fsl,mpc5200b-psc-ac97", },
+        {}
+};
+MODULE_DEVICE_TABLE(of, psc_ac97_match);
+
+static struct of_platform_driver psc_ac97_driver = {
+        .match_table = psc_ac97_match,
+        .probe = psc_ac97_of_probe,
+        .remove = __devexit_p(psc_ac97_of_remove),
+        .driver = {
+                .name = "mpc5200-psc-ac97",
+                .owner = THIS_MODULE,
+        },
+};
+
+/* ---------------------------------------------------------------------
+ * Module setup and teardown; simply register the of_platform driver
+ * for the PSC in AC97 mode.
+ */
+static int __init psc_ac97_init(void)
+{
+        return of_register_platform_driver(&psc_ac97_driver);
+}
+module_init(psc_ac97_init);
+
+static void __exit psc_ac97_exit(void)
+{
+        of_unregister_platform_driver(&psc_ac97_driver);
+}
+module_exit(psc_ac97_exit);
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION("mpc5200 AC97 module");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h
new file mode 100644
index 000000000000..4bc18c35c369
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.h
@@ -0,0 +1,15 @@
+/*
+ * Freescale MPC5200 PSC in AC97 mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+
+extern struct snd_soc_dai psc_ac97_dai[];
+
+#define MPC5200_AC97_NORMAL 0
+#define MPC5200_AC97_SPDIF 1
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b5..ce8de90fb94a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -3,31 +3,21 @@
  * ALSA SoC Digital Audio Interface (DAI) driver
  *
  * Copyright (C) 2008 Secret Lab Technologies Ltd.
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
  */
 
-#include <linux/init.h>
 #include <linux/module.h>
-#include <linux/interrupt.h>
-#include <linux/device.h>
-#include <linux/delay.h>
 #include <linux/of_device.h>
 #include <linux/of_platform.h>
-#include <linux/dma-mapping.h>
 
-#include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
-#include <sound/initval.h>
 #include <sound/soc.h>
-#include <sound/soc-of-simple.h>
 
-#include <sysdev/bestcomm/bestcomm.h>
-#include <sysdev/bestcomm/gen_bd.h>
 #include <asm/mpc52xx_psc.h>
 
-MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
-MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
-MODULE_LICENSE("GPL");
+#include "mpc5200_psc_i2s.h"
+#include "mpc5200_dma.h"
 
 /**
  * PSC_I2S_RATES: sample rates supported by the I2S
@@ -44,191 +34,17 @@ MODULE_LICENSE("GPL");
  * PSC_I2S_FORMATS: audio formats supported by the PSC I2S mode
  */
 #define PSC_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \
-                         SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE | \
-                         SNDRV_PCM_FMTBIT_S32_BE)
-
-/**
- * psc_i2s_stream - Data specific to a single stream (playback or capture)
- * @active:             flag indicating if the stream is active
- * @psc_i2s:            pointer back to parent psc_i2s data structure
- * @bcom_task:          bestcomm task structure
- * @irq:                irq number for bestcomm task
- * @period_start:       physical address of start of DMA region
- * @period_end:         physical address of end of DMA region
- * @period_next_pt:     physical address of next DMA buffer to enqueue
- * @period_bytes:       size of DMA period in bytes
- */
-struct psc_i2s_stream {
-        int active;
-        struct psc_i2s *psc_i2s;
-        struct bcom_task *bcom_task;
-        int irq;
-        struct snd_pcm_substream *stream;
-        dma_addr_t period_start;
-        dma_addr_t period_end;
-        dma_addr_t period_next_pt;
-        dma_addr_t period_current_pt;
-        int period_bytes;
-};
-
-/**
- * psc_i2s - Private driver data
- * @name: short name for this device ("PSC0", "PSC1", etc)
- * @psc_regs: pointer to the PSC's registers
- * @fifo_regs: pointer to the PSC's FIFO registers
- * @irq: IRQ of this PSC
- * @dev: struct device pointer
- * @dai: the CPU DAI for this device
- * @sicr: Base value used in serial interface control register; mode is ORed
- *        with this value.
- * @playback: Playback stream context data
- * @capture: Capture stream context data
- */
-struct psc_i2s {
-        char name[32];
-        struct mpc52xx_psc __iomem *psc_regs;
-        struct mpc52xx_psc_fifo __iomem *fifo_regs;
-        unsigned int irq;
-        struct device *dev;
-        struct snd_soc_dai dai;
-        spinlock_t lock;
-        u32 sicr;
-
-        /* per-stream data */
-        struct psc_i2s_stream playback;
-        struct psc_i2s_stream capture;
-
-        /* Statistics */
-        struct {
-                int overrun_count;
-                int underrun_count;
-        } stats;
-};
-
-/*
- * Interrupt handlers
- */
-static irqreturn_t psc_i2s_status_irq(int irq, void *_psc_i2s)
-{
-        struct psc_i2s *psc_i2s = _psc_i2s;
-        struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
-        u16 isr;
-
-        isr = in_be16(&regs->mpc52xx_psc_isr);
-
-        /* Playback underrun error */
-        if (psc_i2s->playback.active && (isr & MPC52xx_PSC_IMR_TXEMP))
-                psc_i2s->stats.underrun_count++;
-
-        /* Capture overrun error */
-        if (psc_i2s->capture.active && (isr & MPC52xx_PSC_IMR_ORERR))
-                psc_i2s->stats.overrun_count++;
-
-        out_8(&regs->command, 4 << 4);  /* reset the error status */
-
-        return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_bcom_enqueue_next_buffer - Enqueue another audio buffer
- * @s: pointer to stream private data structure
- *
- * Enqueues another audio period buffer into the bestcomm queue.
- *
- * Note: The routine must only be called when there is space available in
- * the queue.  Otherwise the enqueue will fail and the audio ring buffer
- * will get out of sync
- */
-static void psc_i2s_bcom_enqueue_next_buffer(struct psc_i2s_stream *s)
-{
-        struct bcom_bd *bd;
-
-        /* Prepare and enqueue the next buffer descriptor */
-        bd = bcom_prepare_next_buffer(s->bcom_task);
-        bd->status = s->period_bytes;
-        bd->data[0] = s->period_next_pt;
-        bcom_submit_next_buffer(s->bcom_task, NULL);
-
-        /* Update for next period */
-        s->period_next_pt += s->period_bytes;
-        if (s->period_next_pt >= s->period_end)
-                s->period_next_pt = s->period_start;
-}
-
-/* Bestcomm DMA irq handler */
-static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream)
-{
-        struct psc_i2s_stream *s = _psc_i2s_stream;
-
-        /* For each finished period, dequeue the completed period buffer
-         * and enqueue a new one in it's place. */
-        while (bcom_buffer_done(s->bcom_task)) {
-                bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
-                s->period_current_pt += s->period_bytes;
-                if (s->period_current_pt >= s->period_end)
-                        s->period_current_pt = s->period_start;
-                psc_i2s_bcom_enqueue_next_buffer(s);
-                bcom_enable(s->bcom_task);
-        }
-
-        /* If the stream is active, then also inform the PCM middle layer
-         * of the period finished event. */
-        if (s->active)
-                snd_pcm_period_elapsed(s->stream);
-
-        return IRQ_HANDLED;
-}
-
-/**
- * psc_i2s_startup: create a new substream
- *
- * This is the first function called when a stream is opened.
- *
- * If this is the first stream open, then grab the IRQ and program most of
- * the PSC registers.
- */
-static int psc_i2s_startup(struct snd_pcm_substream *substream,
-                           struct snd_soc_dai *dai)
-{
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-        int rc;
-
-        dev_dbg(psc_i2s->dev, "psc_i2s_startup(substream=%p)\n", substream);
-
-        if (!psc_i2s->playback.active &&
-            !psc_i2s->capture.active) {
-                /* Setup the IRQs */
-                rc = request_irq(psc_i2s->irq, &psc_i2s_status_irq, IRQF_SHARED,
-                                 "psc-i2s-status", psc_i2s);
-                rc |= request_irq(psc_i2s->capture.irq,
-                                  &psc_i2s_bcom_irq, IRQF_SHARED,
-                                  "psc-i2s-capture", &psc_i2s->capture);
-                rc |= request_irq(psc_i2s->playback.irq,
-                                  &psc_i2s_bcom_irq, IRQF_SHARED,
-                                  "psc-i2s-playback", &psc_i2s->playback);
-                if (rc) {
-                        free_irq(psc_i2s->irq, psc_i2s);
-                        free_irq(psc_i2s->capture.irq,
-                                 &psc_i2s->capture);
-                        free_irq(psc_i2s->playback.irq,
-                                 &psc_i2s->playback);
-                        return -ENODEV;
-                }
-        }
-
-        return 0;
-}
+                         SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
 
 static int psc_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 psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
+        struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
         u32 mode;
 
-        dev_dbg(psc_i2s->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+        dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
                 " periods=%i buffer_size=%i  buffer_bytes=%i\n",
                 __func__, substream, params_period_size(params),
                 params_period_bytes(params), params_periods(params),
@@ -248,175 +64,15 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream,
                 mode = MPC52xx_PSC_SICR_SIM_CODEC_32;
                 break;
         default:
-                dev_dbg(psc_i2s->dev, "invalid format\n");
-                return -EINVAL;
-        }
-        out_be32(&psc_i2s->psc_regs->sicr, psc_i2s->sicr | mode);
-
-        snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
-        return 0;
-}
-
-static int psc_i2s_hw_free(struct snd_pcm_substream *substream,
-                           struct snd_soc_dai *dai)
-{
-        snd_pcm_set_runtime_buffer(substream, NULL);
-        return 0;
-}
-
-/**
- * psc_i2s_trigger: start and stop the DMA transfer.
- *
- * This function is called by ALSA to start, stop, pause, and resume the DMA
- * transfer of data.
- */
-static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
-                           struct snd_soc_dai *dai)
-{
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-        struct snd_pcm_runtime *runtime = substream->runtime;
-        struct psc_i2s_stream *s;
-        struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs;
-        u16 imr;
-        u8 psc_cmd;
-        unsigned long flags;
-
-        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-                s = &psc_i2s->capture;
-        else
-                s = &psc_i2s->playback;
-
-        dev_dbg(psc_i2s->dev, "psc_i2s_trigger(substream=%p, cmd=%i)"
-                " stream_id=%i\n",
-                substream, cmd, substream->pstr->stream);
-
-        switch (cmd) {
-        case SNDRV_PCM_TRIGGER_START:
-                s->period_bytes = frames_to_bytes(runtime,
-                                                  runtime->period_size);
-                s->period_start = virt_to_phys(runtime->dma_area);
-                s->period_end = s->period_start +
-                                (s->period_bytes * runtime->periods);
-                s->period_next_pt = s->period_start;
-                s->period_current_pt = s->period_start;
-                s->active = 1;
-
-                /* First; reset everything */
-                if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
-                        out_8(&regs->command, MPC52xx_PSC_RST_RX);
-                        out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
-                } else {
-                        out_8(&regs->command, MPC52xx_PSC_RST_TX);
-                        out_8(&regs->command, MPC52xx_PSC_RST_ERR_STAT);
-                }
-
-                /* Next, fill up the bestcomm bd queue and enable DMA.
-                 * This will begin filling the PSC's fifo. */
-                if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-                        bcom_gen_bd_rx_reset(s->bcom_task);
-                else
-                        bcom_gen_bd_tx_reset(s->bcom_task);
-                while (!bcom_queue_full(s->bcom_task))
-                        psc_i2s_bcom_enqueue_next_buffer(s);
-                bcom_enable(s->bcom_task);
-
-                /* Due to errata in the i2s mode; need to line up enabling
-                 * the transmitter with a transition on the frame sync
-                 * line */
-
-                spin_lock_irqsave(&psc_i2s->lock, flags);
-                /* first make sure it is low */
-                while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
-                        ;
-                /* then wait for the transition to high */
-                while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
-                        ;
-                /* Finally, enable the PSC.
-                 * Receiver must always be enabled; even when we only want
-                 * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
-                psc_cmd = MPC52xx_PSC_RX_ENABLE;
-                if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK)
-                        psc_cmd |= MPC52xx_PSC_TX_ENABLE;
-                out_8(&regs->command, psc_cmd);
-                spin_unlock_irqrestore(&psc_i2s->lock, flags);
-
-                break;
-
-        case SNDRV_PCM_TRIGGER_STOP:
-                /* Turn off the PSC */
-                s->active = 0;
-                if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
-                        if (!psc_i2s->playback.active) {
-                                out_8(&regs->command, 2 << 4);  /* reset rx */
-                                out_8(&regs->command, 3 << 4);  /* reset tx */
-                                out_8(&regs->command, 4 << 4);  /* reset err */
-                        }
-                } else {
-                        out_8(&regs->command, 3 << 4);  /* reset tx */
-                        out_8(&regs->command, 4 << 4);  /* reset err */
-                        if (!psc_i2s->capture.active)
-                                out_8(&regs->command, 2 << 4);  /* reset rx */
-                }
-
-                bcom_disable(s->bcom_task);
-                while (!bcom_queue_empty(s->bcom_task))
-                        bcom_retrieve_buffer(s->bcom_task, NULL, NULL);
-
-                break;
-
-        default:
-                dev_dbg(psc_i2s->dev, "invalid command\n");
+                dev_dbg(psc_dma->dev, "invalid format\n");
                 return -EINVAL;
         }
-
-        /* Update interrupt enable settings */
-        imr = 0;
-        if (psc_i2s->playback.active)
-                imr |= MPC52xx_PSC_IMR_TXEMP;
-        if (psc_i2s->capture.active)
-                imr |= MPC52xx_PSC_IMR_ORERR;
-        out_be16(&regs->isr_imr.imr, imr);
+        out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr | mode);
 
         return 0;
 }
 
 /**
- * psc_i2s_shutdown: shutdown the data transfer on a stream
- *
- * Shutdown the PSC if there are no other substreams open.
- */
-static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
-                             struct snd_soc_dai *dai)
-{
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-
-        dev_dbg(psc_i2s->dev, "psc_i2s_shutdown(substream=%p)\n", substream);
-
-        /*
-         * If this is the last active substream, disable the PSC and release
-         * the IRQ.
-         */
-        if (!psc_i2s->playback.active &&
-            !psc_i2s->capture.active) {
-
-                /* Disable all interrupts and reset the PSC */
-                out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
-                out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset tx */
-                out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset rx */
-                out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
-                out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
-
-                /* Release irqs */
-                free_irq(psc_i2s->irq, psc_i2s);
-                free_irq(psc_i2s->capture.irq, &psc_i2s->capture);
-                free_irq(psc_i2s->playback.irq, &psc_i2s->playback);
-        }
-}
-
-/**
  * psc_i2s_set_sysclk: set the clock frequency and direction
  *
  * This function is called by the machine driver to tell us what the clock
@@ -433,8 +89,8 @@ static void psc_i2s_shutdown(struct snd_pcm_substream *substream,
 static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
                               int clk_id, unsigned int freq, int dir)
 {
-        struct psc_i2s *psc_i2s = cpu_dai->private_data;
-        dev_dbg(psc_i2s->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
+        struct psc_dma *psc_dma = cpu_dai->private_data;
+        dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n",
                                 cpu_dai, dir);
         return (dir == SND_SOC_CLOCK_IN) ? 0 : -EINVAL;
 }
@@ -452,8 +108,8 @@ static int psc_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
  */
 static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 {
-        struct psc_i2s *psc_i2s = cpu_dai->private_data;
-        dev_dbg(psc_i2s->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
+        struct psc_dma *psc_dma = cpu_dai->private_data;
+        dev_dbg(psc_dma->dev, "psc_i2s_set_fmt(cpu_dai=%p, format=%i)\n",
                                 cpu_dai, format);
         return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL;
 }
@@ -469,16 +125,13 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
 static struct snd_soc_dai_ops psc_i2s_dai_ops = {
-        .startup        = psc_i2s_startup,
         .hw_params      = psc_i2s_hw_params,
-        .hw_free        = psc_i2s_hw_free,
-        .shutdown       = psc_i2s_shutdown,
-        .trigger        = psc_i2s_trigger,
         .set_sysclk     = psc_i2s_set_sysclk,
         .set_fmt        = psc_i2s_set_fmt,
 };
 
-static struct snd_soc_dai psc_i2s_dai_template = {
+struct snd_soc_dai psc_i2s_dai[] = {{
+        .name   = "I2S",
         .playback = {
                 .channels_min = 2,
                 .channels_max = 2,
@@ -492,222 +145,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
                 .formats = PSC_I2S_FORMATS,
         },
         .ops = &psc_i2s_dai_ops,
-};
-
-/* ---------------------------------------------------------------------
- * The PSC I2S 'ASoC platform' driver
- *
- * Can be referenced by an 'ASoC machine' driver
- * This driver only deals with the audio bus; it doesn't have any
- * interaction with the attached codec
- */
-
-static const struct snd_pcm_hardware psc_i2s_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_S8 | SNDRV_PCM_FMTBIT_S16_BE |
-                   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
-        .rate_min = 8000,
-        .rate_max = 48000,
-        .channels_min = 2,
-        .channels_max = 2,
-        .period_bytes_max       = 1024 * 1024,
-        .period_bytes_min       = 32,
-        .periods_min            = 2,
-        .periods_max            = 256,
-        .buffer_bytes_max       = 2 * 1024 * 1024,
-        .fifo_size              = 0,
-};
-
-static int psc_i2s_pcm_open(struct snd_pcm_substream *substream)
-{
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-        struct psc_i2s_stream *s;
-
-        dev_dbg(psc_i2s->dev, "psc_i2s_pcm_open(substream=%p)\n", substream);
-
-        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-                s = &psc_i2s->capture;
-        else
-                s = &psc_i2s->playback;
-
-        snd_soc_set_runtime_hwparams(substream, &psc_i2s_pcm_hardware);
-
-        s->stream = substream;
-        return 0;
-}
-
-static int psc_i2s_pcm_close(struct snd_pcm_substream *substream)
-{
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-        struct psc_i2s_stream *s;
-
-        dev_dbg(psc_i2s->dev, "psc_i2s_pcm_close(substream=%p)\n", substream);
-
-        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-                s = &psc_i2s->capture;
-        else
-                s = &psc_i2s->playback;
-
-        s->stream = NULL;
-        return 0;
-}
-
-static snd_pcm_uframes_t
-psc_i2s_pcm_pointer(struct snd_pcm_substream *substream)
-{
-        struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data;
-        struct psc_i2s_stream *s;
-        dma_addr_t count;
-
-        if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-                s = &psc_i2s->capture;
-        else
-                s = &psc_i2s->playback;
-
-        count = s->period_current_pt - s->period_start;
-
-        return bytes_to_frames(substream->runtime, count);
-}
-
-static struct snd_pcm_ops psc_i2s_pcm_ops = {
-        .open           = psc_i2s_pcm_open,
-        .close          = psc_i2s_pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .pointer        = psc_i2s_pcm_pointer,
-};
-
-static u64 psc_i2s_pcm_dmamask = 0xffffffff;
-static int psc_i2s_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
-                           struct snd_pcm *pcm)
-{
-        struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-        size_t size = psc_i2s_pcm_hardware.buffer_bytes_max;
-        int rc = 0;
-
-        dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_new(card=%p, dai=%p, pcm=%p)\n",
-                card, dai, pcm);
-
-        if (!card->dev->dma_mask)
-                card->dev->dma_mask = &psc_i2s_pcm_dmamask;
-        if (!card->dev->coherent_dma_mask)
-                card->dev->coherent_dma_mask = 0xffffffff;
-
-        if (pcm->streams[0].substream) {
-                rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
-                                        &pcm->streams[0].substream->dma_buffer);
-                if (rc)
-                        goto playback_alloc_err;
-        }
-
-        if (pcm->streams[1].substream) {
-                rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->dev, size,
-                                        &pcm->streams[1].substream->dma_buffer);
-                if (rc)
-                        goto capture_alloc_err;
-        }
-
-        return 0;
-
- capture_alloc_err:
-        if (pcm->streams[0].substream)
-                snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
- playback_alloc_err:
-        dev_err(card->dev, "Cannot allocate buffer(s)\n");
-        return -ENOMEM;
-}
-
-static void psc_i2s_pcm_free(struct snd_pcm *pcm)
-{
-        struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-        struct snd_pcm_substream *substream;
-        int stream;
-
-        dev_dbg(rtd->socdev->dev, "psc_i2s_pcm_free(pcm=%p)\n", pcm);
-
-        for (stream = 0; stream < 2; stream++) {
-                substream = pcm->streams[stream].substream;
-                if (substream) {
-                        snd_dma_free_pages(&substream->dma_buffer);
-                        substream->dma_buffer.area = NULL;
-                        substream->dma_buffer.addr = 0;
-                }
-        }
-}
-
-struct snd_soc_platform psc_i2s_pcm_soc_platform = {
-        .name           = "mpc5200-psc-audio",
-        .pcm_ops        = &psc_i2s_pcm_ops,
-        .pcm_new        = &psc_i2s_pcm_new,
-        .pcm_free       = &psc_i2s_pcm_free,
-};
-
-/* ---------------------------------------------------------------------
- * Sysfs attributes for debugging
- */
-
-static ssize_t psc_i2s_status_show(struct device *dev,
-                           struct device_attribute *attr, char *buf)
-{
-        struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
-
-        return sprintf(buf, "status=%.4x sicr=%.8x rfnum=%i rfstat=0x%.4x "
-                        "tfnum=%i tfstat=0x%.4x\n",
-                        in_be16(&psc_i2s->psc_regs->sr_csr.status),
-                        in_be32(&psc_i2s->psc_regs->sicr),
-                        in_be16(&psc_i2s->fifo_regs->rfnum) & 0x1ff,
-                        in_be16(&psc_i2s->fifo_regs->rfstat),
-                        in_be16(&psc_i2s->fifo_regs->tfnum) & 0x1ff,
-                        in_be16(&psc_i2s->fifo_regs->tfstat));
-}
-
-static int *psc_i2s_get_stat_attr(struct psc_i2s *psc_i2s, const char *name)
-{
-        if (strcmp(name, "playback_underrun") == 0)
-                return &psc_i2s->stats.underrun_count;
-        if (strcmp(name, "capture_overrun") == 0)
-                return &psc_i2s->stats.overrun_count;
-
-        return NULL;
-}
-
-static ssize_t psc_i2s_stat_show(struct device *dev,
-                                 struct device_attribute *attr, char *buf)
-{
-        struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
-        int *attrib;
-
-        attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
-        if (!attrib)
-                return 0;
-
-        return sprintf(buf, "%i\n", *attrib);
-}
-
-static ssize_t psc_i2s_stat_store(struct device *dev,
-                                  struct device_attribute *attr,
-                                  const char *buf,
-                                  size_t count)
-{
-        struct psc_i2s *psc_i2s = dev_get_drvdata(dev);
-        int *attrib;
-
-        attrib = psc_i2s_get_stat_attr(psc_i2s, attr->attr.name);
-        if (!attrib)
-                return 0;
-
-        *attrib = simple_strtoul(buf, NULL, 0);
-        return count;
-}
-
-static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL);
-static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show,
-                        psc_i2s_stat_store);
-static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
-                        psc_i2s_stat_store);
+} };
+EXPORT_SYMBOL_GPL(psc_i2s_dai);
 
 /* ---------------------------------------------------------------------
  * OF platform bus binding code:
@@ -717,150 +156,65 @@ static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show,
 static int __devinit psc_i2s_of_probe(struct of_device *op,
                                       const struct of_device_id *match)
 {
-        phys_addr_t fifo;
-        struct psc_i2s *psc_i2s;
-        struct resource res;
-        int size, psc_id, irq, rc;
-        const __be32 *prop;
-        void __iomem *regs;
-
-        dev_dbg(&op->dev, "probing psc i2s device\n");
-
-        /* Get the PSC ID */
-        prop = of_get_property(op->node, "cell-index", &size);
-        if (!prop || size < sizeof *prop)
-                return -ENODEV;
-        psc_id = be32_to_cpu(*prop);
-
-        /* Fetch the registers and IRQ of the PSC */
-        irq = irq_of_parse_and_map(op->node, 0);
-        if (of_address_to_resource(op->node, 0, &res)) {
-                dev_err(&op->dev, "Missing reg property\n");
-                return -ENODEV;
-        }
-        regs = ioremap(res.start, 1 + res.end - res.start);
-        if (!regs) {
-                dev_err(&op->dev, "Could not map registers\n");
-                return -ENODEV;
-        }
+        int rc;
+        struct psc_dma *psc_dma;
+        struct mpc52xx_psc __iomem *regs;
 
-        /* Allocate and initialize the driver private data */
-        psc_i2s = kzalloc(sizeof *psc_i2s, GFP_KERNEL);
-        if (!psc_i2s) {
-                iounmap(regs);
-                return -ENOMEM;
-        }
-        spin_lock_init(&psc_i2s->lock);
-        psc_i2s->irq = irq;
-        psc_i2s->psc_regs = regs;
-        psc_i2s->fifo_regs = regs + sizeof *psc_i2s->psc_regs;
-        psc_i2s->dev = &op->dev;
-        psc_i2s->playback.psc_i2s = psc_i2s;
-        psc_i2s->capture.psc_i2s = psc_i2s;
-        snprintf(psc_i2s->name, sizeof psc_i2s->name, "PSC%u", psc_id+1);
-
-        /* Fill out the CPU DAI structure */
-        memcpy(&psc_i2s->dai, &psc_i2s_dai_template, sizeof psc_i2s->dai);
-        psc_i2s->dai.private_data = psc_i2s;
-        psc_i2s->dai.name = psc_i2s->name;
-        psc_i2s->dai.id = psc_id;
-
-        /* Find the address of the fifo data registers and setup the
-         * DMA tasks */
-        fifo = res.start + offsetof(struct mpc52xx_psc, buffer.buffer_32);
-        psc_i2s->capture.bcom_task =
-                bcom_psc_gen_bd_rx_init(psc_id, 10, fifo, 512);
-        psc_i2s->playback.bcom_task =
-                bcom_psc_gen_bd_tx_init(psc_id, 10, fifo);
-        if (!psc_i2s->capture.bcom_task ||
-            !psc_i2s->playback.bcom_task) {
-                dev_err(&op->dev, "Could not allocate bestcomm tasks\n");
-                iounmap(regs);
-                kfree(psc_i2s);
-                return -ENODEV;
+        rc = mpc5200_audio_dma_create(op);
+        if (rc != 0)
+                return rc;
+
+        rc = snd_soc_register_dais(psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
+        if (rc != 0) {
+                pr_err("Failed to register DAI\n");
+                return 0;
         }
 
-        /* Disable all interrupts and reset the PSC */
-        out_be16(&psc_i2s->psc_regs->isr_imr.imr, 0);
-        out_8(&psc_i2s->psc_regs->command, 3 << 4); /* reset transmitter */
-        out_8(&psc_i2s->psc_regs->command, 2 << 4); /* reset receiver */
-        out_8(&psc_i2s->psc_regs->command, 1 << 4); /* reset mode */
-        out_8(&psc_i2s->psc_regs->command, 4 << 4); /* reset error */
+        psc_dma = dev_get_drvdata(&op->dev);
+        regs = psc_dma->psc_regs;
 
         /* Configure the serial interface mode; defaulting to CODEC8 mode */
-        psc_i2s->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
+        psc_dma->sicr = MPC52xx_PSC_SICR_DTS1 | MPC52xx_PSC_SICR_I2S |
                         MPC52xx_PSC_SICR_CLKPOL;
-        if (of_get_property(op->node, "fsl,cellslave", NULL))
-                psc_i2s->sicr |= MPC52xx_PSC_SICR_CELLSLAVE |
-                                 MPC52xx_PSC_SICR_GENCLK;
-        out_be32(&psc_i2s->psc_regs->sicr,
-                 psc_i2s->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
+        out_be32(&psc_dma->psc_regs->sicr,
+                 psc_dma->sicr | MPC52xx_PSC_SICR_SIM_CODEC_8);
 
         /* Check for the codec handle.  If it is not present then we
          * are done */
         if (!of_get_property(op->node, "codec-handle", NULL))
                 return 0;
 
-        /* Set up mode register;
-         * First write: RxRdy (FIFO Alarm) generates rx FIFO irq
-         * Second write: register Normal mode for non loopback
-         */
-        out_8(&psc_i2s->psc_regs->mode, 0);
-        out_8(&psc_i2s->psc_regs->mode, 0);
-
-        /* Set the TX and RX fifo alarm thresholds */
-        out_be16(&psc_i2s->fifo_regs->rfalarm, 0x100);
-        out_8(&psc_i2s->fifo_regs->rfcntl, 0x4);
-        out_be16(&psc_i2s->fifo_regs->tfalarm, 0x100);
-        out_8(&psc_i2s->fifo_regs->tfcntl, 0x7);
-
-        /* Lookup the IRQ numbers */
-        psc_i2s->playback.irq =
-                bcom_get_task_irq(psc_i2s->playback.bcom_task);
-        psc_i2s->capture.irq =
-                bcom_get_task_irq(psc_i2s->capture.bcom_task);
-
-        /* Save what we've done so it can be found again later */
-        dev_set_drvdata(&op->dev, psc_i2s);
-
-        /* Register the SYSFS files */
-        rc = device_create_file(psc_i2s->dev, &dev_attr_status);
-        rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun);
-        rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun);
-        if (rc)
-                dev_info(psc_i2s->dev, "error creating sysfs files\n");
-
-        snd_soc_register_platform(&psc_i2s_pcm_soc_platform);
-
-        /* Tell the ASoC OF helpers about it */
-        of_snd_soc_register_platform(&psc_i2s_pcm_soc_platform, op->node,
-                                     &psc_i2s->dai);
+        /* Due to errata in the dma mode; need to line up enabling
+         * the transmitter with a transition on the frame sync
+         * line */
+
+        /* first make sure it is low */
+        while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) != 0)
+                ;
+        /* then wait for the transition to high */
+        while ((in_8(&regs->ipcr_acr.ipcr) & 0x80) == 0)
+                ;
+        /* Finally, enable the PSC.
+         * Receiver must always be enabled; even when we only want
+         * transmit.  (see 15.3.2.3 of MPC5200B User's Guide) */
+
+        /* Go */
+        out_8(&psc_dma->psc_regs->command,
+                        MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
 
         return 0;
+
 }
 
 static int __devexit psc_i2s_of_remove(struct of_device *op)
 {
-        struct psc_i2s *psc_i2s = dev_get_drvdata(&op->dev);
-
-        dev_dbg(&op->dev, "psc_i2s_remove()\n");
-
-        snd_soc_unregister_platform(&psc_i2s_pcm_soc_platform);
-
-        bcom_gen_bd_rx_release(psc_i2s->capture.bcom_task);
-        bcom_gen_bd_tx_release(psc_i2s->playback.bcom_task);
-
-        iounmap(psc_i2s->psc_regs);
-        iounmap(psc_i2s->fifo_regs);
-        kfree(psc_i2s);
-        dev_set_drvdata(&op->dev, NULL);
-
-        return 0;
+        return mpc5200_audio_dma_destroy(op);
 }
 
 /* Match table for of_platform binding */
 static struct of_device_id psc_i2s_match[] __devinitdata = {
         { .compatible = "fsl,mpc5200-psc-i2s", },
+        { .compatible = "fsl,mpc5200b-psc-i2s", },
         {}
 };
 MODULE_DEVICE_TABLE(of, psc_i2s_match);
@@ -891,4 +245,7 @@ static void __exit psc_i2s_exit(void)
 }
 module_exit(psc_i2s_exit);
 
+MODULE_AUTHOR("Grant Likely <grant.likely@secretlab.ca>");
+MODULE_DESCRIPTION("Freescale MPC5200 PSC in I2S mode ASoC Driver");
+MODULE_LICENSE("GPL");
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h
new file mode 100644
index 000000000000..ce55e070fdf3
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_i2s.h
@@ -0,0 +1,12 @@
+/*
+ * 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/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
new file mode 100644
index 000000000000..8766f7a3893d
--- /dev/null
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -0,0 +1,90 @@
+/*
+ * Phytec pcm030 driver for the PSC of the Freescale MPC52xx
+ * configured as AC97 interface
+ *
+ * Copyright 2008 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-of-simple.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+#include "../codecs/wm9712.h"
+
+static struct snd_soc_device device;
+static struct snd_soc_card card;
+
+static struct snd_soc_dai_link pcm030_fabric_dai[] = {
+{
+        .name = "AC97",
+        .stream_name = "AC97 Analog",
+        .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+        .cpu_dai = &psc_ac97_dai[MPC5200_AC97_NORMAL],
+},
+{
+        .name = "AC97",
+        .stream_name = "AC97 IEC958",
+        .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+        .cpu_dai = &psc_ac97_dai[MPC5200_AC97_SPDIF],
+},
+};
+
+static __init int pcm030_fabric_init(void)
+{
+        struct platform_device *pdev;
+        int rc;
+
+        if (!machine_is_compatible("phytec,pcm030"))
+                return -ENODEV;
+
+        card.platform = &mpc5200_audio_dma_platform;
+        card.name = "pcm030";
+        card.dai_link = pcm030_fabric_dai;
+        card.num_links = ARRAY_SIZE(pcm030_fabric_dai);
+
+        device.card = &card;
+        device.codec_dev = &soc_codec_dev_wm9712;
+
+        pdev = platform_device_alloc("soc-audio", 1);
+        if (!pdev) {
+                pr_err("pcm030_fabric_init: platform_device_alloc() failed\n");
+                return -ENODEV;
+        }
+
+        platform_set_drvdata(pdev, &device);
+        device.dev = &pdev->dev;
+
+        rc = platform_device_add(pdev);
+        if (rc) {
+                pr_err("pcm030_fabric_init: platform_device_add() failed\n");
+                return -ENODEV;
+        }
+        return 0;
+}
+
+module_init(pcm030_fabric_init);
+
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_DESCRIPTION(DRV_NAME ": mpc5200 pcm030 fabric driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 675732e724d5..b771238662b6 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -39,6 +39,14 @@ config SND_OMAP_SOC_OMAP2EVM
         help
           Say Y if you want to add support for SoC audio on the omap2evm board.
 
+config SND_OMAP_SOC_OMAP3EVM
+        tristate "SoC Audio support for OMAP3EVM board"
+        depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3EVM
+        select SND_OMAP_SOC_MCBSP
+        select SND_SOC_TWL4030
+        help
+          Say Y if you want to add support for SoC audio on the omap3evm board.
+
 config SND_OMAP_SOC_SDP3430
         tristate "SoC Audio support for Texas Instruments SDP3430"
         depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 0c9e4ac37660..a37f49862389 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -10,6 +10,7 @@ snd-soc-n810-objs := n810.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
 snd-soc-omap2evm-objs := omap2evm.o
+snd-soc-omap3evm-objs := omap3evm.o
 snd-soc-sdp3430-objs := sdp3430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
 snd-soc-omap3beagle-objs := omap3beagle.o
@@ -18,6 +19,7 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
+obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index a6d1178ce128..b60b1dfbc435 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -383,10 +383,9 @@ static int __init n810_soc_init(void)
         clk_set_parent(sys_clkout2_src, func96m_clk);
         clk_set_rate(sys_clkout2, 12000000);
 
-        if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0)
-                BUG();
-        if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)
-                BUG();
+        BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
+               (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
+
         gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
         gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
 
@@ -417,6 +416,6 @@ static void __exit n810_soc_exit(void)
 module_init(n810_soc_init);
 module_exit(n810_soc_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
 MODULE_DESCRIPTION("ALSA SoC Nokia N810");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 9c09b94f0cf8..a5d46a7b196a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -3,7 +3,8 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -214,8 +215,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
         struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
         struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
         int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
-        int wlen, channels;
+        int wlen, channels, wpf;
         unsigned long port;
+        unsigned int format;
 
         if (cpu_class_is_omap1()) {
                 dma = omap1_dma_reqs[bus_id][substream->stream];
@@ -243,18 +245,24 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
                 return 0;
         }
 
-        channels = params_channels(params);
+        format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+        wpf = channels = params_channels(params);
         switch (channels) {
         case 2:
-                /* Use dual-phase frames */
-                regs->rcr2      |= RPHASE;
-                regs->xcr2      |= XPHASE;
+                if (format == SND_SOC_DAIFMT_I2S) {
+                        /* Use dual-phase frames */
+                        regs->rcr2      |= RPHASE;
+                        regs->xcr2      |= XPHASE;
+                        /* Set 1 word per (McBSP) frame for phase1 and phase2 */
+                        wpf--;
+                        regs->rcr2      |= RFRLEN2(wpf - 1);
+                        regs->xcr2      |= XFRLEN2(wpf - 1);
+                }
         case 1:
-                /* Set 1 word per (McBSP) frame */
-                regs->rcr2      |= RFRLEN2(1 - 1);
-                regs->rcr1      |= RFRLEN1(1 - 1);
-                regs->xcr2      |= XFRLEN2(1 - 1);
-                regs->xcr1      |= XFRLEN1(1 - 1);
+        case 4:
+                /* Set word per (McBSP) frame for phase1 */
+                regs->rcr1      |= RFRLEN1(wpf - 1);
+                regs->xcr1      |= XFRLEN1(wpf - 1);
                 break;
         default:
                 /* Unsupported number of channels */
@@ -276,14 +284,15 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
         }
 
         /* Set FS period and length in terms of bit clock periods */
-        switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+        switch (format) {
         case SND_SOC_DAIFMT_I2S:
-                regs->srgr2     |= FPER(wlen * 2 - 1);
+                regs->srgr2     |= FPER(wlen * channels - 1);
                 regs->srgr1     |= FWID(wlen - 1);
                 break;
+        case SND_SOC_DAIFMT_DSP_A:
         case SND_SOC_DAIFMT_DSP_B:
                 regs->srgr2     |= FPER(wlen * channels - 1);
-                regs->srgr1     |= FWID(wlen * channels - 2);
+                regs->srgr1     |= FWID(0);
                 break;
         }
 
@@ -302,6 +311,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 {
         struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
         struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+        unsigned int temp_fmt = fmt;
 
         if (mcbsp_data->configured)
                 return 0;
@@ -324,10 +334,19 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
                 regs->rcr2      |= RDATDLY(1);
                 regs->xcr2      |= XDATDLY(1);
                 break;
+        case SND_SOC_DAIFMT_DSP_A:
+                /* 1-bit data delay */
+                regs->rcr2      |= RDATDLY(1);
+                regs->xcr2      |= XDATDLY(1);
+                /* Invert FS polarity configuration */
+                temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
+                break;
         case SND_SOC_DAIFMT_DSP_B:
                 /* 0-bit data delay */
                 regs->rcr2      |= RDATDLY(0);
                 regs->xcr2      |= XDATDLY(0);
+                /* Invert FS polarity configuration */
+                temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
                 break;
         default:
                 /* Unsupported data format */
@@ -351,7 +370,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
         }
 
         /* Set bit clock (CLKX/CLKR) and FS polarities */
-        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
         case SND_SOC_DAIFMT_NB_NF:
                 /*
                  * Normal BCLK + FS.
@@ -488,13 +507,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
         .id = (link_id),                                        \
         .playback = {                                           \
                 .channels_min = 1,                              \
-                .channels_max = 2,                              \
+                .channels_max = 4,                              \
                 .rates = OMAP_MCBSP_RATES,                      \
                 .formats = SNDRV_PCM_FMTBIT_S16_LE,             \
         },                                                      \
         .capture = {                                            \
                 .channels_min = 1,                              \
-                .channels_max = 2,                              \
+                .channels_max = 4,                              \
                 .rates = OMAP_MCBSP_RATES,                      \
                 .formats = SNDRV_PCM_FMTBIT_S16_LE,             \
         },                                                      \
@@ -529,6 +548,6 @@ static void __exit snd_omap_mcbsp_exit(void)
 }
 module_exit(snd_omap_mcbsp_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index df7ad13ba73d..c8147aace813 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -3,7 +3,8 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 1bdbb0427183..6454e15f7d28 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -3,7 +3,8 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -86,8 +87,10 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
         struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data;
         int err = 0;
 
+        /* return if this is a bufferless transfer e.g.
+         * codec <--> BT codec or GSM modem -- lg FIXME */
         if (!dma_data)
-                return -ENODEV;
+                return 0;
 
         snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
         runtime->dma_bytes = params_buffer_bytes(params);
@@ -133,6 +136,11 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
         struct omap_pcm_dma_data *dma_data = prtd->dma_data;
         struct omap_dma_channel_params dma_params;
 
+        /* return if this is a bufferless transfer e.g.
+         * codec <--> BT codec or GSM modem -- lg FIXME */
+        if (!prtd->dma_data)
+                return 0;
+
         memset(&dma_params, 0, sizeof(dma_params));
         /*
          * Note: Regardless of interface data formats supported by OMAP McBSP
@@ -367,6 +375,6 @@ static void __exit omap_soc_platform_exit(void)
 }
 module_exit(omap_soc_platform_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index e4369bdfd77d..8d9d26916b05 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -3,7 +3,8 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
index 0c2322dcf02a..027e1a40f8a1 100644
--- a/sound/soc/omap/omap2evm.c
+++ b/sound/soc/omap/omap2evm.c
@@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = {
         .name = "TWL4030",
         .stream_name = "TWL4030",
         .cpu_dai = &omap_mcbsp_dai[0],
-        .codec_dai = &twl4030_dai,
+        .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
         .ops = &omap2evm_ops,
 };
 
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
index fd24a4acd2f5..b0cff9f33b7e 100644
--- a/sound/soc/omap/omap3beagle.c
+++ b/sound/soc/omap/omap3beagle.c
@@ -41,23 +41,33 @@ static int omap3beagle_hw_params(struct snd_pcm_substream *substream,
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
         struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
         struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        unsigned int fmt;
         int ret;
 
+        switch (params_channels(params)) {
+        case 2: /* Stereo I2S mode */
+                fmt =   SND_SOC_DAIFMT_I2S |
+                        SND_SOC_DAIFMT_NB_NF |
+                        SND_SOC_DAIFMT_CBM_CFM;
+                break;
+        case 4: /* Four channel TDM mode */
+                fmt =   SND_SOC_DAIFMT_DSP_A |
+                        SND_SOC_DAIFMT_IB_NF |
+                        SND_SOC_DAIFMT_CBM_CFM;
+                break;
+        default:
+                return -EINVAL;
+        }
+
         /* Set codec DAI configuration */
-        ret = snd_soc_dai_set_fmt(codec_dai,
-                                  SND_SOC_DAIFMT_I2S |
-                                  SND_SOC_DAIFMT_NB_NF |
-                                  SND_SOC_DAIFMT_CBM_CFM);
+        ret = snd_soc_dai_set_fmt(codec_dai, fmt);
         if (ret < 0) {
                 printk(KERN_ERR "can't set codec DAI configuration\n");
                 return ret;
         }
 
         /* Set cpu DAI configuration */
-        ret = snd_soc_dai_set_fmt(cpu_dai,
-                                  SND_SOC_DAIFMT_I2S |
-                                  SND_SOC_DAIFMT_NB_NF |
-                                  SND_SOC_DAIFMT_CBM_CFM);
+        ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
         if (ret < 0) {
                 printk(KERN_ERR "can't set cpu DAI configuration\n");
                 return ret;
@@ -83,7 +93,7 @@ static struct snd_soc_dai_link omap3beagle_dai = {
         .name = "TWL4030",
         .stream_name = "TWL4030",
         .cpu_dai = &omap_mcbsp_dai[0],
-        .codec_dai = &twl4030_dai,
+        .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
         .ops = &omap3beagle_ops,
 };
 
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
new file mode 100644
index 000000000000..9114c263077b
--- /dev/null
+++ b/sound/soc/omap/omap3evm.c
@@ -0,0 +1,147 @@
+/*
+ * omap3evm.c  -- ALSA SoC support for OMAP3 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2008 Texas Instruments, Incorporated
+ *
+ * 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.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
+ * whether express or implied; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int omap3evm_hw_params(struct snd_pcm_substream *substream,
+        struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        int ret;
+
+        /* Set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai,
+                                  SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_NF |
+                                  SND_SOC_DAIFMT_CBM_CFM);
+        if (ret < 0) {
+                printk(KERN_ERR "Can't set codec DAI configuration\n");
+                return ret;
+        }
+
+        /* Set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai,
+                                  SND_SOC_DAIFMT_I2S |
+                                  SND_SOC_DAIFMT_NB_NF |
+                                  SND_SOC_DAIFMT_CBM_CFM);
+        if (ret < 0) {
+                printk(KERN_ERR "Can't set cpu DAI configuration\n");
+                return ret;
+        }
+
+        /* Set the codec system clock for DAC and ADC */
+        ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+                                     SND_SOC_CLOCK_IN);
+        if (ret < 0) {
+                printk(KERN_ERR "Can't set codec system clock\n");
+                return ret;
+        }
+
+        return 0;
+}
+
+static struct snd_soc_ops omap3evm_ops = {
+        .hw_params = omap3evm_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3evm_dai = {
+        .name           = "TWL4030",
+        .stream_name    = "TWL4030",
+        .cpu_dai        = &omap_mcbsp_dai[0],
+        .codec_dai      = &twl4030_dai[TWL4030_DAI_HIFI],
+        .ops            = &omap3evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_omap3evm = {
+        .name = "omap3evm",
+        .platform = &omap_soc_platform,
+        .dai_link = &omap3evm_dai,
+        .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device omap3evm_snd_devdata = {
+        .card = &snd_soc_omap3evm,
+        .codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *omap3evm_snd_device;
+
+static int __init omap3evm_soc_init(void)
+{
+        int ret;
+
+        if (!machine_is_omap3evm()) {
+                pr_err("Not OMAP3 EVM!\n");
+                return -ENODEV;
+        }
+        pr_info("OMAP3 EVM SoC init\n");
+
+        omap3evm_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!omap3evm_snd_device) {
+                printk(KERN_ERR "Platform device allocation failed\n");
+                return -ENOMEM;
+        }
+
+        platform_set_drvdata(omap3evm_snd_device, &omap3evm_snd_devdata);
+        omap3evm_snd_devdata.dev = &omap3evm_snd_device->dev;
+        *(unsigned int *)omap3evm_dai.cpu_dai->private_data = 1;
+
+        ret = platform_device_add(omap3evm_snd_device);
+        if (ret)
+                goto err1;
+
+        return 0;
+
+err1:
+        printk(KERN_ERR "Unable to add platform device\n");
+        platform_device_put(omap3evm_snd_device);
+
+        return ret;
+}
+
+static void __exit omap3evm_soc_exit(void)
+{
+        platform_device_unregister(omap3evm_snd_device);
+}
+
+module_init(omap3evm_soc_init);
+module_exit(omap3evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
+MODULE_LICENSE("GPLv2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index fe282d4ef422..ad219aaf7cb8 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
                 .name = "PCM1773",
                 .stream_name = "HiFi Out",
                 .cpu_dai = &omap_mcbsp_dai[0],
-                .codec_dai = &twl4030_dai,
+                .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
                 .ops = &omap3pandora_out_ops,
                 .init = omap3pandora_out_init,
         }, {
                 .name = "TWL4030",
                 .stream_name = "Line/Mic In",
                 .cpu_dai = &omap_mcbsp_dai[1],
-                .codec_dai = &twl4030_dai,
+                .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
                 .ops = &omap3pandora_in_ops,
                 .init = omap3pandora_in_init,
         }
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index a952a4eb3361..a4e149b7f0eb 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -62,7 +62,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
         /* Set codec DAI configuration */
         err = snd_soc_dai_set_fmt(codec_dai,
                                   SND_SOC_DAIFMT_DSP_B |
-                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_NB_NF |
                                   SND_SOC_DAIFMT_CBM_CFM);
         if (err < 0) {
                 printk(KERN_ERR "can't set codec DAI configuration\n");
@@ -72,7 +72,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
         /* Set cpu DAI configuration */
         err = snd_soc_dai_set_fmt(cpu_dai,
                                   SND_SOC_DAIFMT_DSP_B |
-                                  SND_SOC_DAIFMT_NB_IF |
+                                  SND_SOC_DAIFMT_NB_NF |
                                   SND_SOC_DAIFMT_CBM_CFM);
         if (err < 0) {
                 printk(KERN_ERR "can't set cpu DAI configuration\n");
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index a72dc4e159e5..ec4f8fd8b3a2 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = {
         .name = "TWL4030",
         .stream_name = "TWL4030",
         .cpu_dai = &omap_mcbsp_dai[0],
-        .codec_dai = &twl4030_dai,
+        .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
         .ops = &overo_ops,
 };
 
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index 10f1c867f11d..b719e5db4f57 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -84,6 +84,49 @@ static struct snd_soc_ops sdp3430_ops = {
         .hw_params = sdp3430_hw_params,
 };
 
+static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
+        struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        int ret;
+
+        /* Set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai,
+                                SND_SOC_DAIFMT_DSP_A |
+                                SND_SOC_DAIFMT_IB_NF |
+                                SND_SOC_DAIFMT_CBS_CFM);
+        if (ret) {
+                printk(KERN_ERR "can't set codec DAI configuration\n");
+                return ret;
+        }
+
+        /* Set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai,
+                                SND_SOC_DAIFMT_DSP_A |
+                                SND_SOC_DAIFMT_IB_NF |
+                                SND_SOC_DAIFMT_CBM_CFM);
+        if (ret < 0) {
+                printk(KERN_ERR "can't set cpu DAI configuration\n");
+                return ret;
+        }
+
+        /* Set the codec system clock for DAC and ADC */
+        ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+                                            SND_SOC_CLOCK_IN);
+        if (ret < 0) {
+                printk(KERN_ERR "can't set codec system clock\n");
+                return ret;
+        }
+
+        return 0;
+}
+
+static struct snd_soc_ops sdp3430_voice_ops = {
+        .hw_params = sdp3430_hw_voice_params,
+};
+
 /* Headset jack */
 static struct snd_soc_jack hs_jack;
 
@@ -192,28 +235,58 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec)
         return ret;
 }
 
+static int sdp3430_twl4030_voice_init(struct snd_soc_codec *codec)
+{
+        unsigned short reg;
+
+        /* Enable voice interface */
+        reg = codec->read(codec, TWL4030_REG_VOICE_IF);
+        reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+        codec->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+        return 0;
+}
+
+
 /* Digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link sdp3430_dai = {
-        .name = "TWL4030",
-        .stream_name = "TWL4030",
-        .cpu_dai = &omap_mcbsp_dai[0],
-        .codec_dai = &twl4030_dai,
-        .init = sdp3430_twl4030_init,
-        .ops = &sdp3430_ops,
+static struct snd_soc_dai_link sdp3430_dai[] = {
+        {
+                .name = "TWL4030 I2S",
+                .stream_name = "TWL4030 Audio",
+                .cpu_dai = &omap_mcbsp_dai[0],
+                .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+                .init = sdp3430_twl4030_init,
+                .ops = &sdp3430_ops,
+        },
+        {
+                .name = "TWL4030 PCM",
+                .stream_name = "TWL4030 Voice",
+                .cpu_dai = &omap_mcbsp_dai[1],
+                .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
+                .init = sdp3430_twl4030_voice_init,
+                .ops = &sdp3430_voice_ops,
+        },
 };
 
 /* Audio machine driver */
 static struct snd_soc_card snd_soc_sdp3430 = {
         .name = "SDP3430",
         .platform = &omap_soc_platform,
-        .dai_link = &sdp3430_dai,
-        .num_links = 1,
+        .dai_link = sdp3430_dai,
+        .num_links = ARRAY_SIZE(sdp3430_dai),
+};
+
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+        .ramp_delay_value = 3,
+        .sysclk = 26000,
 };
 
 /* Audio subsystem */
 static struct snd_soc_device sdp3430_snd_devdata = {
         .card = &snd_soc_sdp3430,
         .codec_dev = &soc_codec_dev_twl4030,
+        .codec_data = &twl4030_setup,
 };
 
 static struct platform_device *sdp3430_snd_device;
@@ -236,7 +309,8 @@ static int __init sdp3430_soc_init(void)
 
         platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata);
         sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev;
-        *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */
+        *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+        *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
 
         ret = platform_device_add(sdp3430_snd_device);
         if (ret)
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index ad8a10fe6298..dcd163a4ee9a 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -89,13 +89,13 @@ config SND_PXA2XX_SOC_E800
           Toshiba e800 PDA
 
 config SND_PXA2XX_SOC_EM_X270
-        tristate "SoC Audio support for CompuLab EM-x270"
+        tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
         depends on SND_PXA2XX_SOC && MACH_EM_X270
         select SND_PXA2XX_SOC_AC97
         select SND_SOC_WM9712
         help
           Say Y if you want to add support for SoC audio on
-          CompuLab EM-x270.
+          CompuLab EM-x270, eXeda and CM-X300 machines.
 
 config SND_PXA2XX_SOC_PALM27X
         bool "SoC Audio support for Palm T|X, T5 and LifeDrive"
@@ -134,3 +134,12 @@ config SND_PXA2XX_SOC_MIOA701
         help
           Say Y if you want to add support for SoC audio on the
           MIO A701.
+
+config SND_PXA2XX_SOC_IMOTE2
+       tristate "SoC Audio support for IMote 2"
+       depends on SND_PXA2XX_SOC && MACH_INTELMOTE2
+       select SND_PXA2XX_SOC_I2S
+       select SND_SOC_WM8940
+       help
+         Say Y if you want to add support for SoC audio on the
+         IMote 2.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 4b90c3ccae45..6e096b480335 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o
 snd-soc-zylonite-objs := zylonite.o
 snd-soc-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
+snd-soc-imote2-objs := imote2.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -35,3 +36,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
+obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index 949be9c2a01b..f4756e4025fd 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -1,7 +1,7 @@
 /*
- * em-x270.c  --  SoC audio for EM-X270
+ * SoC audio driver for EM-X270, eXeda and CM-X300
  *
- * Copyright 2007 CompuLab, Ltd.
+ * Copyright 2007, 2009 CompuLab, Ltd.
  *
  * Author: Mike Rapoport <mike@compulab.co.il>
  *
@@ -68,7 +68,8 @@ static int __init em_x270_init(void)
 {
         int ret;
 
-        if (!machine_is_em_x270())
+        if (!(machine_is_em_x270() || machine_is_exeda()
+              || machine_is_cm_x300()))
                 return -ENODEV;
 
         em_x270_snd_device = platform_device_alloc("soc-audio", -1);
@@ -95,5 +96,5 @@ module_exit(em_x270_exit);
 
 /* Module information */
 MODULE_AUTHOR("Mike Rapoport");
-MODULE_DESCRIPTION("ALSA SoC EM-X270");
+MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
new file mode 100644
index 000000000000..405587a01160
--- /dev/null
+++ b/sound/soc/pxa/imote2.c
@@ -0,0 +1,114 @@
+
+#include <linux/module.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm8940.h"
+#include "pxa2xx-i2s.h"
+#include "pxa2xx-pcm.h"
+
+static int imote2_asoc_hw_params(struct snd_pcm_substream *substream,
+                                 struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        unsigned int clk = 0;
+        int ret;
+
+        switch (params_rate(params)) {
+        case 8000:
+        case 16000:
+        case 48000:
+        case 96000:
+                clk = 12288000;
+                break;
+        case 11025:
+        case 22050:
+        case 44100:
+                clk = 11289600;
+                break;
+        }
+
+        /* set codec DAI configuration */
+        ret = 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;
+
+        /* CPU should be clock master */
+        ret = snd_soc_dai_set_fmt(cpu_dai,  SND_SOC_DAIFMT_I2S
+                                  | SND_SOC_DAIFMT_NB_NF
+                                  | SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
+                                     SND_SOC_CLOCK_IN);
+        if (ret < 0)
+                return ret;
+
+        /* set the I2S system clock as input (unused) */
+        ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, clk,
+                SND_SOC_CLOCK_OUT);
+
+        return ret;
+}
+
+static struct snd_soc_ops imote2_asoc_ops = {
+        .hw_params = imote2_asoc_hw_params,
+};
+
+static struct snd_soc_dai_link imote2_dai = {
+        .name = "WM8940",
+        .stream_name = "WM8940",
+        .cpu_dai = &pxa_i2s_dai,
+        .codec_dai = &wm8940_dai,
+        .ops = &imote2_asoc_ops,
+};
+
+static struct snd_soc_card snd_soc_imote2 = {
+        .name = "Imote2",
+        .platform = &pxa2xx_soc_platform,
+        .dai_link = &imote2_dai,
+        .num_links = 1,
+};
+
+static struct snd_soc_device imote2_snd_devdata = {
+        .card = &snd_soc_imote2,
+        .codec_dev = &soc_codec_dev_wm8940,
+};
+
+static struct platform_device *imote2_snd_device;
+
+static int __init imote2_asoc_init(void)
+{
+        int ret;
+
+        if (!machine_is_intelmote2())
+                return -ENODEV;
+        imote2_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!imote2_snd_device)
+                return -ENOMEM;
+
+        platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata);
+        imote2_snd_devdata.dev = &imote2_snd_device->dev;
+        ret = platform_device_add(imote2_snd_device);
+        if (ret)
+                platform_device_put(imote2_snd_device);
+
+        return ret;
+}
+module_init(imote2_asoc_init);
+
+static void __exit imote2_asoc_exit(void)
+{
+        platform_device_unregister(imote2_snd_device);
+}
+module_exit(imote2_asoc_exit);
+
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_DESCRIPTION("ALSA SoC Imote 2");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 0625c342a1c9..326955dea36c 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -106,7 +106,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
                         /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_16;
                         break;
-                case 32:
+                default: /* 32 */
                         /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_8;
                 }
@@ -118,7 +118,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
                         /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_4;
                         break;
-                case 32:
+                default: /* 32 */
                         /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_2;
                 }
@@ -130,7 +130,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
                         /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_2;
                         break;
-                case 32:
+                default: /* 32 */
                         /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_1;
                 }
@@ -142,7 +142,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
                         /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_2;
                         break;
-                case 32:
+                default: /* 32 */
                         /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_1;
                 }
@@ -154,19 +154,20 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
                         /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_2;
                         break;
-                case 32:
+                default: /* 32 */
                         /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_1;
                 }
                 break;
         case 96000:
+        default:
                 acps = 12235000;
                 switch (width) {
                 case 16:
                         /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_1;
                         break;
-                case 32:
+                default: /* 32 */
                         /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
                         acds = PXA_SSP_CLK_AUDIO_DIV_2;
                         div4 = PXA_SSP_CLK_SCDB_1;
@@ -183,7 +184,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 
         /* set cpu DAI configuration */
         ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
-                        SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS);
+                        SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS);
         if (ret < 0)
                 return ret;
 
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 48a73f64500b..44fcc4e01e08 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -200,7 +200,7 @@ static struct snd_soc_device palm27x_snd_devdata = {
 
 static struct platform_device *palm27x_snd_device;
 
-static int __init palm27x_asoc_init(void)
+static int palm27x_asoc_probe(struct platform_device *pdev)
 {
         int ret;
 
@@ -208,6 +208,10 @@ static int __init palm27x_asoc_init(void)
                 machine_is_palmld()))
                 return -ENODEV;
 
+        if (pdev->dev.platform_data)
+                palm27x_ep_gpio = ((struct palm27x_asoc_info *)
+                        (pdev->dev.platform_data))->jack_gpio;
+
         ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
         if (ret)
                 return ret;
@@ -245,16 +249,31 @@ err_alloc:
         return ret;
 }
 
-static void __exit palm27x_asoc_exit(void)
+static int __devexit palm27x_asoc_remove(struct platform_device *pdev)
 {
         free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
         gpio_free(palm27x_ep_gpio);
         platform_device_unregister(palm27x_snd_device);
+        return 0;
 }
 
-void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+static struct platform_driver palm27x_wm9712_driver = {
+        .probe          = palm27x_asoc_probe,
+        .remove         = __devexit_p(palm27x_asoc_remove),
+        .driver         = {
+                .name           = "palm27x-asoc",
+                .owner          = THIS_MODULE,
+        },
+};
+
+static int __init palm27x_asoc_init(void)
+{
+        return platform_driver_register(&palm27x_wm9712_driver);
+}
+
+static void __exit palm27x_asoc_exit(void)
 {
-        palm27x_ep_gpio = data->jack_gpio;
+        platform_driver_unregister(&palm27x_wm9712_driver);
 }
 
 module_init(palm27x_asoc_init);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 308a657928d2..19c45409d94c 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -50,139 +50,6 @@ struct ssp_priv {
 #endif
 };
 
-#define PXA2xx_SSP1_BASE        0x41000000
-#define PXA27x_SSP2_BASE        0x41700000
-#define PXA27x_SSP3_BASE        0x41900000
-#define PXA3xx_SSP4_BASE        0x41a00000
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = {
-        .name                   = "SSP1 PCM Mono out",
-        .dev_addr               = PXA2xx_SSP1_BASE + SSDR,
-        .drcmr                  = &DRCMR(14),
-        .dcmd                   = DCMD_INCSRCADDR | DCMD_FLOWTRG |
-                                  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = {
-        .name                   = "SSP1 PCM Mono in",
-        .dev_addr               = PXA2xx_SSP1_BASE + SSDR,
-        .drcmr                  = &DRCMR(13),
-        .dcmd                   = DCMD_INCTRGADDR | DCMD_FLOWSRC |
-                                  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = {
-        .name                   = "SSP1 PCM Stereo out",
-        .dev_addr               = PXA2xx_SSP1_BASE + SSDR,
-        .drcmr                  = &DRCMR(14),
-        .dcmd                   = DCMD_INCSRCADDR | DCMD_FLOWTRG |
-                                  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = {
-        .name                   = "SSP1 PCM Stereo in",
-        .dev_addr               = PXA2xx_SSP1_BASE + SSDR,
-        .drcmr                  = &DRCMR(13),
-        .dcmd                   = DCMD_INCTRGADDR | DCMD_FLOWSRC |
-                                  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = {
-        .name                   = "SSP2 PCM Mono out",
-        .dev_addr               = PXA27x_SSP2_BASE + SSDR,
-        .drcmr                  = &DRCMR(16),
-        .dcmd                   = DCMD_INCSRCADDR | DCMD_FLOWTRG |
-                                  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = {
-        .name                   = "SSP2 PCM Mono in",
-        .dev_addr               = PXA27x_SSP2_BASE + SSDR,
-        .drcmr                  = &DRCMR(15),
-        .dcmd                   = DCMD_INCTRGADDR | DCMD_FLOWSRC |
-                                  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = {
-        .name                   = "SSP2 PCM Stereo out",
-        .dev_addr               = PXA27x_SSP2_BASE + SSDR,
-        .drcmr                  = &DRCMR(16),
-        .dcmd                   = DCMD_INCSRCADDR | DCMD_FLOWTRG |
-                                  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = {
-        .name                   = "SSP2 PCM Stereo in",
-        .dev_addr               = PXA27x_SSP2_BASE + SSDR,
-        .drcmr                  = &DRCMR(15),
-        .dcmd                   = DCMD_INCTRGADDR | DCMD_FLOWSRC |
-                                  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = {
-        .name                   = "SSP3 PCM Mono out",
-        .dev_addr               = PXA27x_SSP3_BASE + SSDR,
-        .drcmr                  = &DRCMR(67),
-        .dcmd                   = DCMD_INCSRCADDR | DCMD_FLOWTRG |
-                                  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = {
-        .name                   = "SSP3 PCM Mono in",
-        .dev_addr               = PXA27x_SSP3_BASE + SSDR,
-        .drcmr                  = &DRCMR(66),
-        .dcmd                   = DCMD_INCTRGADDR | DCMD_FLOWSRC |
-                                  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = {
-        .name                   = "SSP3 PCM Stereo out",
-        .dev_addr               = PXA27x_SSP3_BASE + SSDR,
-        .drcmr                  = &DRCMR(67),
-        .dcmd                   = DCMD_INCSRCADDR | DCMD_FLOWTRG |
-                                  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = {
-        .name                   = "SSP3 PCM Stereo in",
-        .dev_addr               = PXA27x_SSP3_BASE + SSDR,
-        .drcmr                  = &DRCMR(66),
-        .dcmd                   = DCMD_INCTRGADDR | DCMD_FLOWSRC |
-                                  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = {
-        .name                   = "SSP4 PCM Mono out",
-        .dev_addr               = PXA3xx_SSP4_BASE + SSDR,
-        .drcmr                  = &DRCMR(67),
-        .dcmd                   = DCMD_INCSRCADDR | DCMD_FLOWTRG |
-                                  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = {
-        .name                   = "SSP4 PCM Mono in",
-        .dev_addr               = PXA3xx_SSP4_BASE + SSDR,
-        .drcmr                  = &DRCMR(66),
-        .dcmd                   = DCMD_INCTRGADDR | DCMD_FLOWSRC |
-                                  DCMD_BURST16 | DCMD_WIDTH2,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = {
-        .name                   = "SSP4 PCM Stereo out",
-        .dev_addr               = PXA3xx_SSP4_BASE + SSDR,
-        .drcmr                  = &DRCMR(67),
-        .dcmd                   = DCMD_INCSRCADDR | DCMD_FLOWTRG |
-                                  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
-static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = {
-        .name                   = "SSP4 PCM Stereo in",
-        .dev_addr               = PXA3xx_SSP4_BASE + SSDR,
-        .drcmr                  = &DRCMR(66),
-        .dcmd                   = DCMD_INCTRGADDR | DCMD_FLOWSRC |
-                                  DCMD_BURST16 | DCMD_WIDTH4,
-};
-
 static void dump_registers(struct ssp_device *ssp)
 {
         dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n",
@@ -194,25 +61,33 @@ static void dump_registers(struct ssp_device *ssp)
                  ssp_read_reg(ssp, SSACD));
 }
 
-static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = {
-        {
-                &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in,
-                &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in,
-        },
-        {
-                &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in,
-                &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in,
-        },
-        {
-                &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in,
-                &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in,
-        },
-        {
-                &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in,
-                &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in,
-        },
+struct pxa2xx_pcm_dma_data {
+        struct pxa2xx_pcm_dma_params params;
+        char name[20];
 };
 
+static struct pxa2xx_pcm_dma_params *
+ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
+{
+        struct pxa2xx_pcm_dma_data *dma;
+
+        dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
+        if (dma == NULL)
+                return NULL;
+
+        snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id,
+                        width4 ? "32-bit" : "16-bit", out ? "out" : "in");
+
+        dma->params.name = dma->name;
+        dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx);
+        dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) :
+                                  (DCMD_INCTRGADDR | DCMD_FLOWSRC)) |
+                        (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16;
+        dma->params.dev_addr = ssp->phys_base + SSDR;
+
+        return &dma->params;
+}
+
 static int pxa_ssp_startup(struct snd_pcm_substream *substream,
                            struct snd_soc_dai *dai)
 {
@@ -227,6 +102,11 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
                 clk_enable(priv->dev.ssp->clk);
                 ssp_disable(&priv->dev);
         }
+
+        if (cpu_dai->dma_data) {
+                kfree(cpu_dai->dma_data);
+                cpu_dai->dma_data = NULL;
+        }
         return ret;
 }
 
@@ -241,6 +121,11 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
                 ssp_disable(&priv->dev);
                 clk_disable(priv->dev.ssp->clk);
         }
+
+        if (cpu_dai->dma_data) {
+                kfree(cpu_dai->dma_data);
+                cpu_dai->dma_data = NULL;
+        }
 }
 
 #ifdef CONFIG_PM
@@ -280,12 +165,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
  * ssp_set_clkdiv - set SSP clock divider
  * @div: serial clock rate divider
  */
-static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+static void ssp_set_scr(struct ssp_device *ssp, u32 div)
+{
+        u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+
+        if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+                sscr0 &= ~0x0000ff00;
+                sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
+        } else {
+                sscr0 &= ~0x000fff00;
+                sscr0 |= (div - 1) << 8;     /* 1..4096 */
+        }
+        ssp_write_reg(ssp, SSCR0, sscr0);
+}
+
+/**
+ * ssp_get_clkdiv - get SSP clock divider
+ */
+static u32 ssp_get_scr(struct ssp_device *ssp)
 {
-        struct ssp_device *ssp = dev->ssp;
-        u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+        u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+        u32 div;
 
-        ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+        if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+                div = ((sscr0 >> 8) & 0xff) * 2 + 2;
+        else
+                div = ((sscr0 >> 8) & 0xfff) + 1;
+        return div;
 }
 
 /*
@@ -302,7 +208,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
                 ~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 
         dev_dbg(&ssp->pdev->dev,
-                "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
+                "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n",
                 cpu_dai->id, clk_id, freq);
 
         switch (clk_id) {
@@ -326,7 +232,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
                 break;
         case PXA_SSP_CLK_AUDIO:
                 priv->sysclk = 0;
-                ssp_set_scr(&priv->dev, 1);
+                ssp_set_scr(ssp, 1);
                 sscr0 |= SSCR0_ACS;
                 break;
         default:
@@ -387,7 +293,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
                 ssp_write_reg(ssp, SSACD, val);
                 break;
         case PXA_SSP_DIV_SCR:
-                ssp_set_scr(&priv->dev, div);
+                ssp_set_scr(ssp, div);
                 break;
         default:
                 return -ENODEV;
@@ -451,7 +357,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
                         ssacd |= (0x6 << 4);
 
                         dev_dbg(&ssp->pdev->dev,
-                                "Using SSACDD %x to supply %dHz\n",
+                                "Using SSACDD %x to supply %uHz\n",
                                 val, freq_out);
                         break;
                 }
@@ -568,7 +474,10 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
                 case SND_SOC_DAIFMT_NB_IF:
                         break;
                 case SND_SOC_DAIFMT_IB_IF:
-                        sspsp |= SSPSP_SCMODE(3);
+                        sspsp |= SSPSP_SCMODE(2);
+                        break;
+                case SND_SOC_DAIFMT_IB_NF:
+                        sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
                         break;
                 default:
                         return -EINVAL;
@@ -585,7 +494,13 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
                 case SND_SOC_DAIFMT_NB_NF:
                         sspsp |= SSPSP_SFRMP;
                         break;
+                case SND_SOC_DAIFMT_NB_IF:
+                        break;
                 case SND_SOC_DAIFMT_IB_IF:
+                        sspsp |= SSPSP_SCMODE(2);
+                        break;
+                case SND_SOC_DAIFMT_IB_NF:
+                        sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP;
                         break;
                 default:
                         return -EINVAL;
@@ -623,25 +538,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
         struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
         struct ssp_priv *priv = cpu_dai->private_data;
         struct ssp_device *ssp = priv->dev.ssp;
-        int dma = 0, chn = params_channels(params);
+        int chn = params_channels(params);
         u32 sscr0;
         u32 sspsp;
         int width = snd_pcm_format_physical_width(params_format(params));
         int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
 
-        /* select correct DMA params */
-        if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
-                dma = 1; /* capture DMA offset is 1,3 */
+        /* generate correct DMA params */
+        if (cpu_dai->dma_data)
+                kfree(cpu_dai->dma_data);
+
         /* Network mode with one active slot (ttsa == 1) can be used
          * to force 16-bit frame width on the wire (for S16_LE), even
          * with two channels. Use 16-bit DMA transfers for this case.
          */
-        if (((chn == 2) && (ttsa != 1)) || (width == 32))
-                dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
-
-        cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
-
-        dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
+        cpu_dai->dma_data = ssp_get_dma_params(ssp,
+                        ((chn == 2) && (ttsa != 1)) || (width == 32),
+                        substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
         /* we can only change the settings if the port is not in use */
         if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
@@ -674,8 +587,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
         case SND_SOC_DAIFMT_I2S:
                sspsp = ssp_read_reg(ssp, SSPSP);
 
-                if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
-                     (width == 16)) {
+                if ((ssp_get_scr(ssp) == 4) && (width == 16)) {
                         /* This is a special case where the bitclk is 64fs
                         * and we're not dealing with 2*32 bits of audio
                         * samples.
@@ -806,6 +718,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
                 goto err_priv;
         }
 
+        priv->dai_fmt = (unsigned int) -1;
         dai->private_data = priv;
 
         return 0;
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 2f4b6e489b78..4743e262895d 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -106,10 +106,8 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
         if (IS_ERR(clk_i2s))
                 return PTR_ERR(clk_i2s);
 
-        if (!cpu_dai->active) {
-                SACR0 |= SACR0_RST;
+        if (!cpu_dai->active)
                 SACR0 = 0;
-        }
 
         return 0;
 }
@@ -178,9 +176,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 
         /* is port used by another stream */
         if (!(SACR0 & SACR0_ENB)) {
-
                 SACR0 = 0;
-                SACR1 = 0;
                 if (pxa_i2s.master)
                         SACR0 |= SACR0_BCKD;
 
@@ -226,6 +222,10 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 
         switch (cmd) {
         case SNDRV_PCM_TRIGGER_START:
+                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                        SACR1 &= ~SACR1_DRPL;
+                else
+                        SACR1 &= ~SACR1_DREC;
                 SACR0 |= SACR0_ENB;
                 break;
         case SNDRV_PCM_TRIGGER_RESUME:
@@ -252,21 +252,16 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
                 SAIMR &= ~SAIMR_RFS;
         }
 
-        if (SACR1 & (SACR1_DREC | SACR1_DRPL)) {
+        if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
                 SACR0 &= ~SACR0_ENB;
                 pxa_i2s_wait();
                 clk_disable(clk_i2s);
         }
-
-        clk_put(clk_i2s);
 }
 
 #ifdef CONFIG_PM
 static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 {
-        if (!dai->active)
-                return 0;
-
         /* store registers */
         pxa_i2s.sacr0 = SACR0;
         pxa_i2s.sacr1 = SACR1;
@@ -281,16 +276,14 @@ static int pxa2xx_i2s_suspend(struct snd_soc_dai *dai)
 
 static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 {
-        if (!dai->active)
-                return 0;
-
         pxa_i2s_wait();
 
-        SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB;
+        SACR0 = pxa_i2s.sacr0 & ~SACR0_ENB;
         SACR1 = pxa_i2s.sacr1;
         SAIMR = pxa_i2s.saimr;
         SADIV = pxa_i2s.sadiv;
-        SACR0 |= SACR0_ENB;
+
+        SACR0 = pxa_i2s.sacr0;
 
         return 0;
 }
@@ -329,6 +322,7 @@ struct snd_soc_dai pxa_i2s_dai = {
                 .rates = PXA2XX_I2S_RATES,
                 .formats = SNDRV_PCM_FMTBIT_S16_LE,},
         .ops = &pxa_i2s_dai_ops,
+        .symmetric_rates = 1,
 };
 
 EXPORT_SYMBOL_GPL(pxa_i2s_dai);
@@ -346,6 +340,19 @@ static int pxa2xx_i2s_probe(struct platform_device *dev)
         if (ret != 0)
                 clk_put(clk_i2s);
 
+        /*
+         * PXA Developer's Manual:
+         * If SACR0[ENB] is toggled in the middle of a normal operation,
+         * the SACR0[RST] bit must also be set and cleared to reset all
+         * I2S controller registers.
+         */
+        SACR0 = SACR0_RST;
+        SACR0 = 0;
+        /* Make sure RPL and REC are disabled */
+        SACR1 = SACR1_DRPL | SACR1_DREC;
+        /* Along with FIFO servicing */
+        SAIMR &= ~(SAIMR_RFS | SAIMR_TFS);
+
         return ret;
 }
 
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 32063790d95b..93e6c87b7399 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -69,8 +69,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
                 break;
         }
 
-        s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
-                            s3c2412_get_iisclk());
+        s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
+                                s3c2412_get_iisclk());
 
         /* set codec DAI configuration */
         ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
@@ -145,8 +145,9 @@ static struct snd_soc_dai_link jive_dai = {
 };
 
 /* jive audio machine driver */
-static struct snd_soc_machine snd_soc_machine_jive = {
+static struct snd_soc_card snd_soc_machine_jive = {
         .name           = "Jive",
+        .platform       = &s3c24xx_soc_platform,
         .dai_link       = &jive_dai,
         .num_links      = 1,
 };
@@ -157,9 +158,8 @@ static struct wm8750_setup_data jive_wm8750_setup = {
 
 /* jive audio subsystem */
 static struct snd_soc_device jive_snd_devdata = {
-        .machine        = &snd_soc_machine_jive,
-        .platform       = &s3c24xx_soc_platform,
-        .codec_dev      = &soc_codec_dev_wm8750_spi,
+        .card           = &snd_soc_machine_jive,
+        .codec_dev      = &soc_codec_dev_wm8750,
         .codec_data     = &jive_wm8750_setup,
 };
 
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 289fadf60b10..906709e6dd5f 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -345,9 +345,11 @@ static void lm4857_write_regs(void)
 static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
         struct snd_ctl_elem_value *ucontrol)
 {
-        int reg = kcontrol->private_value & 0xFF;
-        int shift = (kcontrol->private_value >> 8) & 0x0F;
-        int mask = (kcontrol->private_value >> 16) & 0xFF;
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
+        int reg = mc->reg;
+        int shift = mc->shift;
+        int mask = mc->max;
 
         pr_debug("Entered %s\n", __func__);
 
@@ -358,9 +360,11 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
 static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
         struct snd_ctl_elem_value *ucontrol)
 {
-        int reg = kcontrol->private_value & 0xFF;
-        int shift = (kcontrol->private_value >> 8) & 0x0F;
-        int mask = (kcontrol->private_value >> 16) & 0xFF;
+        struct soc_mixer_control *mc =
+                (struct soc_mixer_control *)kcontrol->private_value;
+        int reg = mc->reg;
+        int shift = mc->shift;
+        int mask = mc->max;
 
         if (((lm4857_regs[reg] >> shift) & mask) ==
                 ucontrol->value.integer.value[0])
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 295a4c910262..1a283170ca92 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -37,6 +37,20 @@
 
 #include "s3c-i2s-v2.h"
 
+#undef S3C_IIS_V2_SUPPORTED
+
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
+#define S3C_IIS_V2_SUPPORTED
+#endif
+
+#ifdef CONFIG_PLAT_S3C64XX
+#define S3C_IIS_V2_SUPPORTED
+#endif
+
+#ifndef S3C_IIS_V2_SUPPORTED
+#error Unsupported CPU model
+#endif
+
 #define S3C2412_I2S_DEBUG_CON 0
 
 static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
@@ -75,7 +89,7 @@ static inline void dbg_showcon(const char *fn, u32 con)
 
 
 /* Turn on or off the transmission path. */
-void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
+static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
 {
         void __iomem *regs = i2s->regs;
         u32 fic, con, mod;
@@ -105,7 +119,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
                         break;
 
                 default:
-                        dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+                        dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n",
+                                mod & S3C2412_IISMOD_MODE_MASK);
+                        break;
                 }
 
                 writel(con, regs + S3C2412_IISCON);
@@ -132,7 +148,9 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
                         break;
 
                 default:
-                        dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+                        dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n",
+                                mod & S3C2412_IISMOD_MODE_MASK);
+                        break;
                 }
 
                 writel(mod, regs + S3C2412_IISMOD);
@@ -143,9 +161,8 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
         dbg_showcon(__func__, con);
         pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
 }
-EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl);
 
-void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
+static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
 {
         void __iomem *regs = i2s->regs;
         u32 fic, con, mod;
@@ -175,7 +192,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
                         break;
 
                 default:
-                        dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+                        dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n",
+                                mod & S3C2412_IISMOD_MODE_MASK);
                 }
 
                 writel(mod, regs + S3C2412_IISMOD);
@@ -199,7 +217,8 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
                         break;
 
                 default:
-                        dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+                        dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n",
+                                mod & S3C2412_IISMOD_MODE_MASK);
                 }
 
                 writel(con, regs + S3C2412_IISCON);
@@ -209,7 +228,6 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
         fic = readl(regs + S3C2412_IISFIC);
         pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
 }
-EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl);
 
 /*
  * Wait for the LR signal to allow synchronisation to the L/R clock
@@ -266,7 +284,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
  */
 #define IISMOD_MASTER_MASK (1 << 11)
 #define IISMOD_SLAVE (1 << 11)
-#define IISMOD_MASTER (0x0)
+#define IISMOD_MASTER (0 << 11)
 #endif
 
         switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -281,7 +299,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
                 iismod |= IISMOD_MASTER;
                 break;
         default:
-                pr_debug("unknwon master/slave format\n");
+                pr_err("unknwon master/slave format\n");
                 return -EINVAL;
         }
 
@@ -298,7 +316,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
                 iismod |= S3C2412_IISMOD_SDF_IIS;
                 break;
         default:
-                pr_debug("Unknown data format\n");
+                pr_err("Unknown data format\n");
                 return -EINVAL;
         }
 
@@ -327,6 +345,7 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
         iismod = readl(i2s->regs + S3C2412_IISMOD);
         pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
 
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
         switch (params_format(params)) {
         case SNDRV_PCM_FORMAT_S8:
                 iismod |= S3C2412_IISMOD_8BIT;
@@ -335,6 +354,25 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
                 iismod &= ~S3C2412_IISMOD_8BIT;
                 break;
         }
+#endif
+
+#ifdef CONFIG_PLAT_S3C64XX
+        iismod &= ~0x606;
+        /* Sample size */
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S8:
+                /* 8 bit sample, 16fs BCLK */
+                iismod |= 0x2004;
+                break;
+        case SNDRV_PCM_FORMAT_S16_LE:
+                /* 16 bit sample, 32fs BCLK */
+                break;
+        case SNDRV_PCM_FORMAT_S24_LE:
+                /* 24 bit sample, 48fs BCLK */
+                iismod |= 0x4002;
+                break;
+        }
+#endif
 
         writel(iismod, i2s->regs + S3C2412_IISMOD);
         pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
@@ -473,9 +511,9 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 /* default table of all avaialable root fs divisors */
 static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
 
-int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
-                          unsigned int *fstab,
-                          unsigned int rate, struct clk *clk)
+int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+                            unsigned int *fstab,
+                            unsigned int rate, struct clk *clk)
 {
         unsigned long clkrate = clk_get_rate(clk);
         unsigned int div;
@@ -489,6 +527,8 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
         unsigned int best_rate = 0;
         unsigned int best_deviation = INT_MAX;
 
+        pr_debug("Input clock rate %ldHz\n", clkrate);
+
         if (fstab == NULL)
                 fstab = iis_fs_tab;
 
@@ -507,7 +547,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
                 actual = clkrate / (fsdiv * div);
                 deviation = actual - rate;
 
-                printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
+                printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n",
                        fsdiv, div, actual, deviation);
 
                 deviation = abs(deviation);
@@ -523,7 +563,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
                         break;
         }
 
-        printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
+        printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n",
                best_fs, best_div, best_rate);
 
         info->fs_div = best_fs;
@@ -531,7 +571,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
 
         return 0;
 }
-EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate);
 
 int s3c_i2sv2_probe(struct platform_device *pdev,
                     struct snd_soc_dai *dai,
@@ -539,12 +579,31 @@ int s3c_i2sv2_probe(struct platform_device *pdev,
                     unsigned long base)
 {
         struct device *dev = &pdev->dev;
+        unsigned int iismod;
 
         i2s->dev = dev;
 
         /* record our i2s structure for later use in the callbacks */
         dai->private_data = i2s;
 
+        if (!base) {
+                struct resource *res = platform_get_resource(pdev,
+                                                             IORESOURCE_MEM,
+                                                             0);
+                if (!res) {
+                        dev_err(dev, "Unable to get register resource\n");
+                        return -ENXIO;
+                }
+
+                if (!request_mem_region(res->start, resource_size(res),
+                                        "s3c64xx-i2s-v4")) {
+                        dev_err(dev, "Unable to request register region\n");
+                        return -EBUSY;
+                }
+
+                base = res->start;
+        }
+
         i2s->regs = ioremap(base, 0x100);
         if (i2s->regs == NULL) {
                 dev_err(dev, "cannot ioremap registers\n");
@@ -560,12 +619,16 @@ int s3c_i2sv2_probe(struct platform_device *pdev,
 
         clk_enable(i2s->iis_pclk);
 
+        /* Mark ourselves as in TXRX mode so we can run through our cleanup
+         * process without warnings. */
+        iismod = readl(i2s->regs + S3C2412_IISMOD);
+        iismod |= S3C2412_IISMOD_MODE_TXRX;
+        writel(iismod, i2s->regs + S3C2412_IISMOD);
         s3c2412_snd_txctrl(i2s, 0);
         s3c2412_snd_rxctrl(i2s, 0);
 
         return 0;
 }
-
 EXPORT_SYMBOL_GPL(s3c_i2sv2_probe);
 
 #ifdef CONFIG_PM
@@ -624,15 +687,18 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
 
 int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 {
-        dai->ops.trigger = s3c2412_i2s_trigger;
-        dai->ops.hw_params = s3c2412_i2s_hw_params;
-        dai->ops.set_fmt = s3c2412_i2s_set_fmt;
-        dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv;
+        struct snd_soc_dai_ops *ops = dai->ops;
+
+        ops->trigger = s3c2412_i2s_trigger;
+        ops->hw_params = s3c2412_i2s_hw_params;
+        ops->set_fmt = s3c2412_i2s_set_fmt;
+        ops->set_clkdiv = s3c2412_i2s_set_clkdiv;
 
         dai->suspend = s3c2412_i2s_suspend;
         dai->resume = s3c2412_i2s_resume;
 
         return snd_soc_register_dai(dai);
 }
-
 EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 1ca3cdaa8213..168a088ba761 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -33,8 +33,8 @@
 
 #include <plat/regs-s3c2412-iis.h>
 
-#include <plat/regs-gpio.h>
 #include <plat/audio.h>
+#include <mach/regs-gpio.h>
 #include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
@@ -120,7 +120,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 
         s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk");
         if (s3c2412_i2s.iis_cclk == NULL) {
-                pr_debug("failed to get i2sclk clock\n");
+                pr_err("failed to get i2sclk clock\n");
                 iounmap(s3c2412_i2s.regs);
                 return -ENODEV;
         }
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 33c5de7e255f..3c06c401d0fb 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -108,48 +108,19 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
         return 0;
 }
 
-
-unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai)
+struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai)
 {
         struct s3c_i2sv2_info *i2s = to_info(dai);
 
-        return clk_get_rate(i2s->iis_cclk);
+        return i2s->iis_cclk;
 }
-EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate);
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock);
 
 static int s3c64xx_i2s_probe(struct platform_device *pdev,
                              struct snd_soc_dai *dai)
 {
-        struct device *dev = &pdev->dev;
-        struct s3c_i2sv2_info *i2s;
-        int ret;
-
-        dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id);
-
-        if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) {
-                dev_err(dev, "id %d out of range\n", pdev->id);
-                return -EINVAL;
-        }
-
-        i2s = &s3c64xx_i2s[pdev->id];
-
-        ret = s3c_i2sv2_probe(pdev, dai, i2s,
-                              pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
-        if (ret)
-                return ret;
-
-        i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
-        i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
-
-        i2s->iis_cclk = clk_get(dev, "audio-bus");
-        if (IS_ERR(i2s->iis_cclk)) {
-                dev_err(dev, "failed to get audio-bus");
-                iounmap(i2s->regs);
-                return -ENODEV;
-        }
-
         /* configure GPIO for i2s port */
-        switch (pdev->id) {
+        switch (dai->id) {
         case 0:
                 s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK);
                 s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK);
@@ -175,41 +146,122 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev,
         SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define S3C64XX_I2S_FMTS \
-        (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE)
+        (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\
+         SNDRV_PCM_FMTBIT_S24_LE)
 
 static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
         .set_sysclk     = s3c64xx_i2s_set_sysclk,       
 };
 
-struct snd_soc_dai s3c64xx_i2s_dai = {
-        .name           = "s3c64xx-i2s",
-        .id             = 0,
-        .probe          = s3c64xx_i2s_probe,
-        .playback = {
-                .channels_min   = 2,
-                .channels_max   = 2,
-                .rates          = S3C64XX_I2S_RATES,
-                .formats        = S3C64XX_I2S_FMTS,
+struct snd_soc_dai s3c64xx_i2s_dai[] = {
+        {
+                .name           = "s3c64xx-i2s",
+                .id             = 0,
+                .probe          = s3c64xx_i2s_probe,
+                .playback = {
+                        .channels_min   = 2,
+                        .channels_max   = 2,
+                        .rates          = S3C64XX_I2S_RATES,
+                        .formats        = S3C64XX_I2S_FMTS,
+                },
+                .capture = {
+                         .channels_min  = 2,
+                         .channels_max  = 2,
+                         .rates         = S3C64XX_I2S_RATES,
+                         .formats       = S3C64XX_I2S_FMTS,
+                 },
+                .ops = &s3c64xx_i2s_dai_ops,
+                .symmetric_rates = 1,
         },
-        .capture = {
-                .channels_min   = 2,
-                .channels_max   = 2,
-                .rates          = S3C64XX_I2S_RATES,
-                .formats        = S3C64XX_I2S_FMTS,
+        {
+                .name           = "s3c64xx-i2s",
+                .id             = 1,
+                .probe          = s3c64xx_i2s_probe,
+                .playback = {
+                        .channels_min   = 2,
+                        .channels_max   = 2,
+                        .rates          = S3C64XX_I2S_RATES,
+                        .formats        = S3C64XX_I2S_FMTS,
+                },
+                .capture = {
+                         .channels_min  = 2,
+                         .channels_max  = 2,
+                         .rates         = S3C64XX_I2S_RATES,
+                         .formats       = S3C64XX_I2S_FMTS,
+                 },
+                .ops = &s3c64xx_i2s_dai_ops,
+                .symmetric_rates = 1,
         },
-        .ops = &s3c64xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai);
 
+static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
+{
+        struct s3c_i2sv2_info *i2s;
+        struct snd_soc_dai *dai;
+        int ret;
+
+        if (pdev->id >= ARRAY_SIZE(s3c64xx_i2s)) {
+                dev_err(&pdev->dev, "id %d out of range\n", pdev->id);
+                return -EINVAL;
+        }
+
+        i2s = &s3c64xx_i2s[pdev->id];
+        dai = &s3c64xx_i2s_dai[pdev->id];
+        dai->dev = &pdev->dev;
+
+        i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
+        i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
+
+        i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus");
+        if (IS_ERR(i2s->iis_cclk)) {
+                dev_err(&pdev->dev, "failed to get audio-bus\n");
+                ret = PTR_ERR(i2s->iis_cclk);
+                goto err;
+        }
+
+        ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
+        if (ret)
+                goto err_clk;
+
+        ret = s3c_i2sv2_register_dai(dai);
+        if (ret != 0)
+                goto err_i2sv2;
+
+        return 0;
+
+err_i2sv2:
+        /* Not implemented for I2Sv2 core yet */
+err_clk:
+        clk_put(i2s->iis_cclk);
+err:
+        return ret;
+}
+
+static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev)
+{
+        dev_err(&pdev->dev, "Device removal not yet supported\n");
+        return 0;
+}
+
+static struct platform_driver s3c64xx_iis_driver = {
+        .probe  = s3c64xx_iis_dev_probe,
+        .remove = s3c64xx_iis_dev_remove,
+        .driver = {
+                .name = "s3c64xx-iis",
+                .owner = THIS_MODULE,
+        },
+};
+
 static int __init s3c64xx_i2s_init(void)
 {
-        return  s3c_i2sv2_register_dai(&s3c64xx_i2s_dai);
+        return platform_driver_register(&s3c64xx_iis_driver);
 }
 module_init(s3c64xx_i2s_init);
 
 static void __exit s3c64xx_i2s_exit(void)
 {
-        snd_soc_unregister_dai(&s3c64xx_i2s_dai);
+        platform_driver_unregister(&s3c64xx_iis_driver);
 }
 module_exit(s3c64xx_i2s_exit);
 
@@ -217,6 +269,3 @@ module_exit(s3c64xx_i2s_exit);
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C64XX I2S SoC Interface");
 MODULE_LICENSE("GPL");
-
-
-
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index b7ffe3c38b66..02148cee2613 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -15,6 +15,8 @@
 #ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H
 #define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__
 
+struct clk;
+
 #include "s3c-i2s-v2.h"
 
 #define S3C64XX_DIV_BCLK        S3C_I2SV2_DIV_BCLK
@@ -24,8 +26,8 @@
 #define S3C64XX_CLKSRC_PCLK     (0)
 #define S3C64XX_CLKSRC_MUX      (1)
 
-extern struct snd_soc_dai s3c64xx_i2s_dai;
+extern struct snd_soc_dai s3c64xx_i2s_dai[];
 
-extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai);
+extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai);
 
 #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */
diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig
new file mode 100644
index 000000000000..c74eb3d4a47c
--- /dev/null
+++ b/sound/soc/s6000/Kconfig
@@ -0,0 +1,19 @@
+config SND_S6000_SOC
+        tristate "SoC Audio for the Stretch s6000 family"
+        depends on XTENSA_VARIANT_S6000
+        help
+          Say Y or M if you want to add support for codecs attached to
+          s6000 family chips. You will also need to select the platform
+          to support below.
+
+config SND_S6000_SOC_I2S
+        tristate
+
+config SND_S6000_SOC_S6IPCAM
+        tristate "SoC Audio support for Stretch 6105 IP Camera"
+        depends on SND_S6000_SOC && XTENSA_PLATFORM_S6105
+        select SND_S6000_SOC_I2S
+        select SND_SOC_TLV320AIC3X
+        help
+          Say Y if you want to add support for SoC audio on the
+          Stretch s6105 IP Camera Reference Design.
diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile
new file mode 100644
index 000000000000..7a613612e010
--- /dev/null
+++ b/sound/soc/s6000/Makefile
@@ -0,0 +1,11 @@
+# s6000 Platform Support
+snd-soc-s6000-objs := s6000-pcm.o
+snd-soc-s6000-i2s-objs := s6000-i2s.o
+
+obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o
+obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o
+
+# s6105 Machine Support
+snd-soc-s6ipcam-objs := s6105-ipcam.o
+
+obj-$(CONFIG_SND_S6000_SOC_S6IPCAM) += snd-soc-s6ipcam.o
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
new file mode 100644
index 000000000000..c5cda187ecab
--- /dev/null
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -0,0 +1,629 @@
+/*
+ * ALSA SoC I2S Audio Layer for the Stretch S6000 family
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/interrupt.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 "s6000-i2s.h"
+#include "s6000-pcm.h"
+
+struct s6000_i2s_dev {
+        dma_addr_t sifbase;
+        u8 __iomem *scbbase;
+        unsigned int wide;
+        unsigned int channel_in;
+        unsigned int channel_out;
+        unsigned int lines_in;
+        unsigned int lines_out;
+        struct s6000_pcm_dma_params dma_params;
+};
+
+#define S6_I2S_INTERRUPT_STATUS 0x00
+#define   S6_I2S_INT_OVERRUN    1
+#define   S6_I2S_INT_UNDERRUN   2
+#define   S6_I2S_INT_ALIGNMENT  4
+#define S6_I2S_INTERRUPT_ENABLE 0x04
+#define S6_I2S_INTERRUPT_RAW    0x08
+#define S6_I2S_INTERRUPT_CLEAR  0x0C
+#define S6_I2S_INTERRUPT_SET    0x10
+#define S6_I2S_MODE             0x20
+#define   S6_I2S_DUAL           0
+#define   S6_I2S_WIDE           1
+#define S6_I2S_TX_DEFAULT       0x24
+#define S6_I2S_DATA_CFG(c)      (0x40 + 0x10 * (c))
+#define   S6_I2S_IN             0
+#define   S6_I2S_OUT            1
+#define   S6_I2S_UNUSED         2
+#define S6_I2S_INTERFACE_CFG(c) (0x44 + 0x10 * (c))
+#define   S6_I2S_DIV_MASK       0x001fff
+#define   S6_I2S_16BIT          0x000000
+#define   S6_I2S_20BIT          0x002000
+#define   S6_I2S_24BIT          0x004000
+#define   S6_I2S_32BIT          0x006000
+#define   S6_I2S_BITS_MASK      0x006000
+#define   S6_I2S_MEM_16BIT      0x000000
+#define   S6_I2S_MEM_32BIT      0x008000
+#define   S6_I2S_MEM_MASK       0x008000
+#define   S6_I2S_CHANNELS_SHIFT 16
+#define   S6_I2S_CHANNELS_MASK  0x030000
+#define   S6_I2S_SCK_IN         0x000000
+#define   S6_I2S_SCK_OUT        0x040000
+#define   S6_I2S_SCK_DIR        0x040000
+#define   S6_I2S_WS_IN          0x000000
+#define   S6_I2S_WS_OUT         0x080000
+#define   S6_I2S_WS_DIR         0x080000
+#define   S6_I2S_LEFT_FIRST     0x000000
+#define   S6_I2S_RIGHT_FIRST    0x100000
+#define   S6_I2S_FIRST          0x100000
+#define   S6_I2S_CUR_SCK        0x200000
+#define   S6_I2S_CUR_WS         0x400000
+#define S6_I2S_ENABLE(c)        (0x48 + 0x10 * (c))
+#define   S6_I2S_DISABLE_IF     0x02
+#define   S6_I2S_ENABLE_IF      0x03
+#define   S6_I2S_IS_BUSY        0x04
+#define   S6_I2S_DMA_ACTIVE     0x08
+#define   S6_I2S_IS_ENABLED     0x10
+
+#define S6_I2S_NUM_LINES        4
+
+#define S6_I2S_SIF_PORT0        0x0000000
+#define S6_I2S_SIF_PORT1        0x0000080 /* docs say 0x0000010 */
+
+static inline void s6_i2s_write_reg(struct s6000_i2s_dev *dev, int reg, u32 val)
+{
+        writel(val, dev->scbbase + reg);
+}
+
+static inline u32 s6_i2s_read_reg(struct s6000_i2s_dev *dev, int reg)
+{
+        return readl(dev->scbbase + reg);
+}
+
+static inline void s6_i2s_mod_reg(struct s6000_i2s_dev *dev, int reg,
+                                  u32 mask, u32 val)
+{
+        val ^= s6_i2s_read_reg(dev, reg) & ~mask;
+        s6_i2s_write_reg(dev, reg, val);
+}
+
+static void s6000_i2s_start_channel(struct s6000_i2s_dev *dev, int channel)
+{
+        int i, j, cur, prev;
+
+        /*
+         * Wait for WCLK to toggle 5 times before enabling the channel
+         * s6000 Family Datasheet 3.6.4:
+         *   "At least two cycles of WS must occur between commands
+         *    to disable or enable the interface"
+         */
+        j = 0;
+        prev = ~S6_I2S_CUR_WS;
+        for (i = 1000000; --i && j < 6; ) {
+                cur = s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(channel))
+                       & S6_I2S_CUR_WS;
+                if (prev != cur) {
+                        prev = cur;
+                        j++;
+                }
+        }
+        if (j < 6)
+                printk(KERN_WARNING "s6000-i2s: timeout waiting for WCLK\n");
+
+        s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_ENABLE_IF);
+}
+
+static void s6000_i2s_stop_channel(struct s6000_i2s_dev *dev, int channel)
+{
+        s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_DISABLE_IF);
+}
+
+static void s6000_i2s_start(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data;
+        int channel;
+
+        channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+                        dev->channel_out : dev->channel_in;
+
+        s6000_i2s_start_channel(dev, channel);
+}
+
+static void s6000_i2s_stop(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data;
+        int channel;
+
+        channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+                        dev->channel_out : dev->channel_in;
+
+        s6000_i2s_stop_channel(dev, channel);
+}
+
+static int s6000_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+                             int after)
+{
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ^ !after)
+                        s6000_i2s_start(substream);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                if (!after)
+                        s6000_i2s_stop(substream);
+        }
+        return 0;
+}
+
+static unsigned int s6000_i2s_int_sources(struct s6000_i2s_dev *dev)
+{
+        unsigned int pending;
+        pending = s6_i2s_read_reg(dev, S6_I2S_INTERRUPT_RAW);
+        pending &= S6_I2S_INT_ALIGNMENT |
+                   S6_I2S_INT_UNDERRUN |
+                   S6_I2S_INT_OVERRUN;
+        s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, pending);
+
+        return pending;
+}
+
+static unsigned int s6000_i2s_check_xrun(struct snd_soc_dai *cpu_dai)
+{
+        struct s6000_i2s_dev *dev = cpu_dai->private_data;
+        unsigned int errors;
+        unsigned int ret;
+
+        errors = s6000_i2s_int_sources(dev);
+        if (likely(!errors))
+                return 0;
+
+        ret = 0;
+        if (errors & S6_I2S_INT_ALIGNMENT)
+                printk(KERN_ERR "s6000-i2s: WCLK misaligned\n");
+        if (errors & S6_I2S_INT_UNDERRUN)
+                ret |= 1 << SNDRV_PCM_STREAM_PLAYBACK;
+        if (errors & S6_I2S_INT_OVERRUN)
+                ret |= 1 << SNDRV_PCM_STREAM_CAPTURE;
+        return ret;
+}
+
+static void s6000_i2s_wait_disabled(struct s6000_i2s_dev *dev)
+{
+        int channel;
+        int n = 50;
+        for (channel = 0; channel < 2; channel++) {
+                while (--n >= 0) {
+                        int v = s6_i2s_read_reg(dev, S6_I2S_ENABLE(channel));
+                        if ((v & S6_I2S_IS_ENABLED)
+                            || !(v & (S6_I2S_DMA_ACTIVE | S6_I2S_IS_BUSY)))
+                                break;
+                        udelay(20);
+                }
+        }
+        if (n < 0)
+                printk(KERN_WARNING "s6000-i2s: timeout disabling interfaces");
+}
+
+static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+                                   unsigned int fmt)
+{
+        struct s6000_i2s_dev *dev = cpu_dai->private_data;
+        u32 w;
+
+        switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+        case SND_SOC_DAIFMT_CBM_CFM:
+                w = S6_I2S_SCK_IN | S6_I2S_WS_IN;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFM:
+                w = S6_I2S_SCK_OUT | S6_I2S_WS_IN;
+                break;
+        case SND_SOC_DAIFMT_CBM_CFS:
+                w = S6_I2S_SCK_IN | S6_I2S_WS_OUT;
+                break;
+        case SND_SOC_DAIFMT_CBS_CFS:
+                w = S6_I2S_SCK_OUT | S6_I2S_WS_OUT;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+        case SND_SOC_DAIFMT_NB_NF:
+                w |= S6_I2S_LEFT_FIRST;
+                break;
+        case SND_SOC_DAIFMT_NB_IF:
+                w |= S6_I2S_RIGHT_FIRST;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(0),
+                       S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w);
+        s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(1),
+                       S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w);
+
+        return 0;
+}
+
+static int s6000_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+{
+        struct s6000_i2s_dev *dev = dai->private_data;
+
+        if (!div || (div & 1) || div > (S6_I2S_DIV_MASK + 1) * 2)
+                return -EINVAL;
+
+        s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(div_id),
+                       S6_I2S_DIV_MASK, div / 2 - 1);
+        return 0;
+}
+
+static int s6000_i2s_hw_params(struct snd_pcm_substream *substream,
+                               struct snd_pcm_hw_params *params,
+                               struct snd_soc_dai *dai)
+{
+        struct s6000_i2s_dev *dev = dai->private_data;
+        int interf;
+        u32 w = 0;
+
+        if (dev->wide)
+                interf = 0;
+        else {
+                w |= (((params_channels(params) - 2) / 2)
+                      << S6_I2S_CHANNELS_SHIFT) & S6_I2S_CHANNELS_MASK;
+                interf = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                                ? dev->channel_out : dev->channel_in;
+        }
+
+        switch (params_format(params)) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                w |= S6_I2S_16BIT | S6_I2S_MEM_16BIT;
+                break;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                w |= S6_I2S_32BIT | S6_I2S_MEM_32BIT;
+                break;
+        default:
+                printk(KERN_WARNING "s6000-i2s: unsupported PCM format %x\n",
+                       params_format(params));
+                return -EINVAL;
+        }
+
+        if (s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(interf))
+             & S6_I2S_IS_ENABLED) {
+                printk(KERN_ERR "s6000-i2s: interface already enabled\n");
+                return -EBUSY;
+        }
+
+        s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(interf),
+                       S6_I2S_CHANNELS_MASK|S6_I2S_MEM_MASK|S6_I2S_BITS_MASK,
+                       w);
+
+        return 0;
+}
+
+static int s6000_i2s_dai_probe(struct platform_device *pdev,
+                               struct snd_soc_dai *dai)
+{
+        struct s6000_i2s_dev *dev = dai->private_data;
+        struct s6000_snd_platform_data *pdata = pdev->dev.platform_data;
+
+        if (!pdata)
+                return -EINVAL;
+
+        dev->wide = pdata->wide;
+        dev->channel_in = pdata->channel_in;
+        dev->channel_out = pdata->channel_out;
+        dev->lines_in = pdata->lines_in;
+        dev->lines_out = pdata->lines_out;
+
+        s6_i2s_write_reg(dev, S6_I2S_MODE,
+                         dev->wide ? S6_I2S_WIDE : S6_I2S_DUAL);
+
+        if (dev->wide) {
+                int i;
+
+                if (dev->lines_in + dev->lines_out > S6_I2S_NUM_LINES)
+                        return -EINVAL;
+
+                dev->channel_in = 0;
+                dev->channel_out = 1;
+                dai->capture.channels_min = 2 * dev->lines_in;
+                dai->capture.channels_max = dai->capture.channels_min;
+                dai->playback.channels_min = 2 * dev->lines_out;
+                dai->playback.channels_max = dai->playback.channels_min;
+
+                for (i = 0; i < dev->lines_out; i++)
+                        s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_OUT);
+
+                for (; i < S6_I2S_NUM_LINES - dev->lines_in; i++)
+                        s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i),
+                                         S6_I2S_UNUSED);
+
+                for (; i < S6_I2S_NUM_LINES; i++)
+                        s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_IN);
+        } else {
+                unsigned int cfg[2] = {S6_I2S_UNUSED, S6_I2S_UNUSED};
+
+                if (dev->lines_in > 1 || dev->lines_out > 1)
+                        return -EINVAL;
+
+                dai->capture.channels_min = 2 * dev->lines_in;
+                dai->capture.channels_max = 8 * dev->lines_in;
+                dai->playback.channels_min = 2 * dev->lines_out;
+                dai->playback.channels_max = 8 * dev->lines_out;
+
+                if (dev->lines_in)
+                        cfg[dev->channel_in] = S6_I2S_IN;
+                if (dev->lines_out)
+                        cfg[dev->channel_out] = S6_I2S_OUT;
+
+                s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(0), cfg[0]);
+                s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(1), cfg[1]);
+        }
+
+        if (dev->lines_out) {
+                if (dev->lines_in) {
+                        if (!dev->dma_params.dma_out)
+                                return -ENODEV;
+                } else {
+                        dev->dma_params.dma_out = dev->dma_params.dma_in;
+                        dev->dma_params.dma_in = 0;
+                }
+        }
+        dev->dma_params.sif_in = dev->sifbase + (dev->channel_in ?
+                                        S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0);
+        dev->dma_params.sif_out = dev->sifbase + (dev->channel_out ?
+                                        S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0);
+        dev->dma_params.same_rate = pdata->same_rate | pdata->wide;
+        return 0;
+}
+
+#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \
+                         SNDRV_PCM_RATE_8000_192000)
+#define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops s6000_i2s_dai_ops = {
+        .set_fmt = s6000_i2s_set_dai_fmt,
+        .set_clkdiv = s6000_i2s_set_clkdiv,
+        .hw_params = s6000_i2s_hw_params,
+};
+
+struct snd_soc_dai s6000_i2s_dai = {
+        .name = "s6000-i2s",
+        .id = 0,
+        .probe = s6000_i2s_dai_probe,
+        .playback = {
+                .channels_min = 2,
+                .channels_max = 8,
+                .formats = S6000_I2S_FORMATS,
+                .rates = S6000_I2S_RATES,
+                .rate_min = 0,
+                .rate_max = 1562500,
+        },
+        .capture = {
+                .channels_min = 2,
+                .channels_max = 8,
+                .formats = S6000_I2S_FORMATS,
+                .rates = S6000_I2S_RATES,
+                .rate_min = 0,
+                .rate_max = 1562500,
+        },
+        .ops = &s6000_i2s_dai_ops,
+}
+EXPORT_SYMBOL_GPL(s6000_i2s_dai);
+
+static int __devinit s6000_i2s_probe(struct platform_device *pdev)
+{
+        struct s6000_i2s_dev *dev;
+        struct resource *scbmem, *sifmem, *region, *dma1, *dma2;
+        u8 __iomem *mmio;
+        int ret;
+
+        scbmem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!scbmem) {
+                dev_err(&pdev->dev, "no mem resource?\n");
+                ret = -ENODEV;
+                goto err_release_none;
+        }
+
+        region = request_mem_region(scbmem->start,
+                                    scbmem->end - scbmem->start + 1,
+                                    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);
+        if (!mmio) {
+                dev_err(&pdev->dev, "can't ioremap SCB region\n");
+                ret = -ENOMEM;
+                goto err_release_scb;
+        }
+
+        sifmem = platform_get_resource(pdev, IORESOURCE_MEM, 1);
+        if (!sifmem) {
+                dev_err(&pdev->dev, "no second mem resource?\n");
+                ret = -ENODEV;
+                goto err_release_map;
+        }
+
+        region = request_mem_region(sifmem->start,
+                                    sifmem->end - sifmem->start + 1,
+                                    pdev->name);
+        if (!region) {
+                dev_err(&pdev->dev, "I2S SIF region already claimed\n");
+                ret = -EBUSY;
+                goto err_release_map;
+        }
+
+        dma1 = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+        if (!dma1) {
+                dev_err(&pdev->dev, "no dma resource?\n");
+                ret = -ENODEV;
+                goto err_release_sif;
+        }
+
+        region = request_mem_region(dma1->start, dma1->end - dma1->start + 1,
+                                    pdev->name);
+        if (!region) {
+                dev_err(&pdev->dev, "I2S DMA region already claimed\n");
+                ret = -EBUSY;
+                goto err_release_sif;
+        }
+
+        dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+        if (dma2) {
+                region = request_mem_region(dma2->start,
+                                            dma2->end - dma2->start + 1,
+                                            pdev->name);
+                if (!region) {
+                        dev_err(&pdev->dev,
+                                "I2S DMA region already claimed\n");
+                        ret = -EBUSY;
+                        goto err_release_dma1;
+                }
+        }
+
+        dev = kzalloc(sizeof(struct s6000_i2s_dev), GFP_KERNEL);
+        if (!dev) {
+                ret = -ENOMEM;
+                goto err_release_dma2;
+        }
+
+        s6000_i2s_dai.dev = &pdev->dev;
+        s6000_i2s_dai.private_data = dev;
+        s6000_i2s_dai.dma_data = &dev->dma_params;
+
+        dev->sifbase = sifmem->start;
+        dev->scbbase = mmio;
+
+        s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0);
+        s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR,
+                         S6_I2S_INT_ALIGNMENT |
+                         S6_I2S_INT_UNDERRUN |
+                         S6_I2S_INT_OVERRUN);
+
+        s6000_i2s_stop_channel(dev, 0);
+        s6000_i2s_stop_channel(dev, 1);
+        s6000_i2s_wait_disabled(dev);
+
+        dev->dma_params.check_xrun = s6000_i2s_check_xrun;
+        dev->dma_params.trigger = s6000_i2s_trigger;
+        dev->dma_params.dma_in = dma1->start;
+        dev->dma_params.dma_out = dma2 ? dma2->start : 0;
+        dev->dma_params.irq = platform_get_irq(pdev, 0);
+        if (dev->dma_params.irq < 0) {
+                dev_err(&pdev->dev, "no irq resource?\n");
+                ret = -ENODEV;
+                goto err_release_dev;
+        }
+
+        s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE,
+                         S6_I2S_INT_ALIGNMENT |
+                         S6_I2S_INT_UNDERRUN |
+                         S6_I2S_INT_OVERRUN);
+
+        ret = snd_soc_register_dai(&s6000_i2s_dai);
+        if (ret)
+                goto err_release_dev;
+
+        return 0;
+
+err_release_dev:
+        kfree(dev);
+err_release_dma2:
+        if (dma2)
+                release_mem_region(dma2->start, dma2->end - dma2->start + 1);
+err_release_dma1:
+        release_mem_region(dma1->start, dma1->end - dma1->start + 1);
+err_release_sif:
+        release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1);
+err_release_map:
+        iounmap(mmio);
+err_release_scb:
+        release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1);
+err_release_none:
+        return ret;
+}
+
+static void __devexit s6000_i2s_remove(struct platform_device *pdev)
+{
+        struct s6000_i2s_dev *dev = s6000_i2s_dai.private_data;
+        struct resource *region;
+        void __iomem *mmio = dev->scbbase;
+
+        snd_soc_unregister_dai(&s6000_i2s_dai);
+
+        s6000_i2s_stop_channel(dev, 0);
+        s6000_i2s_stop_channel(dev, 1);
+
+        s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0);
+        s6000_i2s_dai.private_data = 0;
+        kfree(dev);
+
+        region = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+        release_mem_region(region->start, region->end - region->start + 1);
+
+        region = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+        if (region)
+                release_mem_region(region->start,
+                                   region->end - region->start + 1);
+
+        region = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        release_mem_region(region->start, (region->end - region->start) + 1);
+
+        iounmap(mmio);
+        region = platform_get_resource(pdev, IORESOURCE_IO, 0);
+        release_mem_region(region->start, (region->end - region->start) + 1);
+}
+
+static struct platform_driver s6000_i2s_driver = {
+        .probe  = s6000_i2s_probe,
+        .remove = __devexit_p(s6000_i2s_remove),
+        .driver = {
+                .name   = "s6000-i2s",
+                .owner  = THIS_MODULE,
+        },
+};
+
+static int __init s6000_i2s_init(void)
+{
+        return platform_driver_register(&s6000_i2s_driver);
+}
+module_init(s6000_i2s_init);
+
+static void __exit s6000_i2s_exit(void)
+{
+        platform_driver_unregister(&s6000_i2s_driver);
+}
+module_exit(s6000_i2s_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6000-i2s.h b/sound/soc/s6000/s6000-i2s.h
new file mode 100644
index 000000000000..2375fdfe6dba
--- /dev/null
+++ b/sound/soc/s6000/s6000-i2s.h
@@ -0,0 +1,25 @@
+/*
+ * ALSA SoC I2S Audio Layer for the Stretch s6000 family
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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 _S6000_I2S_H
+#define _S6000_I2S_H
+
+extern struct snd_soc_dai s6000_i2s_dai;
+
+struct s6000_snd_platform_data {
+        int lines_in;
+        int lines_out;
+        int channel_in;
+        int channel_out;
+        int wide;
+        int same_rate;
+};
+#endif
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
new file mode 100644
index 000000000000..83b8028e209d
--- /dev/null
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -0,0 +1,497 @@
+/*
+ * ALSA PCM interface for the Stetch s6000 family
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+#include <variant/dmac.h>
+
+#include "s6000-pcm.h"
+
+#define S6_PCM_PREALLOCATE_SIZE (96 * 1024)
+#define S6_PCM_PREALLOCATE_MAX  (2048 * 1024)
+
+static struct snd_pcm_hardware s6000_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_JOINT_DUPLEX),
+        .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE),
+        .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \
+                  SNDRV_PCM_RATE_8000_192000),
+        .rate_min = 0,
+        .rate_max = 1562500,
+        .channels_min = 2,
+        .channels_max = 8,
+        .buffer_bytes_max = 0x7ffffff0,
+        .period_bytes_min = 16,
+        .period_bytes_max = 0xfffff0,
+        .periods_min = 2,
+        .periods_max = 1024, /* no limit */
+        .fifo_size = 0,
+};
+
+struct s6000_runtime_data {
+        spinlock_t lock;
+        int period;             /* current DMA period */
+};
+
+static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct s6000_runtime_data *prtd = runtime->private_data;
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+        int channel;
+        unsigned int period_size;
+        unsigned int dma_offset;
+        dma_addr_t dma_pos;
+        dma_addr_t src, dst;
+
+        period_size = snd_pcm_lib_period_bytes(substream);
+        dma_offset = prtd->period * period_size;
+        dma_pos = runtime->dma_addr + dma_offset;
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                src = dma_pos;
+                dst = par->sif_out;
+                channel = par->dma_out;
+        } else {
+                src = par->sif_in;
+                dst = dma_pos;
+                channel = par->dma_in;
+        }
+
+        if (!s6dmac_channel_enabled(DMA_MASK_DMAC(channel),
+                                    DMA_INDEX_CHNL(channel)))
+                return;
+
+        if (s6dmac_fifo_full(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel))) {
+                printk(KERN_ERR "s6000-pcm: fifo full\n");
+                return;
+        }
+
+        BUG_ON(period_size & 15);
+        s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel),
+                        src, dst, period_size);
+
+        prtd->period++;
+        if (unlikely(prtd->period >= runtime->periods))
+                prtd->period = 0;
+}
+
+static irqreturn_t s6000_pcm_irq(int irq, void *data)
+{
+        struct snd_pcm *pcm = data;
+        struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+        struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+        struct s6000_runtime_data *prtd;
+        unsigned int has_xrun;
+        int i, ret = IRQ_NONE;
+        u32 channel[2] = {
+                [SNDRV_PCM_STREAM_PLAYBACK] = params->dma_out,
+                [SNDRV_PCM_STREAM_CAPTURE] = params->dma_in
+        };
+
+        has_xrun = params->check_xrun(runtime->dai->cpu_dai);
+
+        for (i = 0; i < ARRAY_SIZE(channel); ++i) {
+                struct snd_pcm_substream *substream = pcm->streams[i].substream;
+                unsigned int pending;
+
+                if (!channel[i])
+                        continue;
+
+                if (unlikely(has_xrun & (1 << i)) &&
+                    substream->runtime &&
+                    snd_pcm_running(substream)) {
+                        dev_dbg(pcm->dev, "xrun\n");
+                        snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+                        ret = IRQ_HANDLED;
+                }
+
+                pending = s6dmac_int_sources(DMA_MASK_DMAC(channel[i]),
+                                             DMA_INDEX_CHNL(channel[i]));
+
+                if (pending & 1) {
+                        ret = IRQ_HANDLED;
+                        if (likely(substream->runtime &&
+                                   snd_pcm_running(substream))) {
+                                snd_pcm_period_elapsed(substream);
+                                dev_dbg(pcm->dev, "period elapsed %x %x\n",
+                                       s6dmac_cur_src(DMA_MASK_DMAC(channel[i]),
+                                                   DMA_INDEX_CHNL(channel[i])),
+                                       s6dmac_cur_dst(DMA_MASK_DMAC(channel[i]),
+                                                   DMA_INDEX_CHNL(channel[i])));
+                                prtd = substream->runtime->private_data;
+                                spin_lock(&prtd->lock);
+                                s6000_pcm_enqueue_dma(substream);
+                                spin_unlock(&prtd->lock);
+                        }
+                }
+
+                if (unlikely(pending & ~7)) {
+                        if (pending & (1 << 3))
+                                printk(KERN_WARNING
+                                       "s6000-pcm: DMA %x Underflow\n",
+                                       channel[i]);
+                        if (pending & (1 << 4))
+                                printk(KERN_WARNING
+                                       "s6000-pcm: DMA %x Overflow\n",
+                                       channel[i]);
+                        if (pending & 0x1e0)
+                                printk(KERN_WARNING
+                                       "s6000-pcm: DMA %x Master Error "
+                                       "(mask %x)\n",
+                                       channel[i], pending >> 5);
+
+                }
+        }
+
+        return ret;
+}
+
+static int s6000_pcm_start(struct snd_pcm_substream *substream)
+{
+        struct s6000_runtime_data *prtd = substream->runtime->private_data;
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+        unsigned long flags;
+        int srcinc;
+        u32 dma;
+
+        spin_lock_irqsave(&prtd->lock, flags);
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                srcinc = 1;
+                dma = par->dma_out;
+        } else {
+                srcinc = 0;
+                dma = par->dma_in;
+        }
+        s6dmac_enable_chan(DMA_MASK_DMAC(dma), DMA_INDEX_CHNL(dma),
+                           1 /* priority 1 (0 is max) */,
+                           0 /* peripheral requests w/o xfer length mode */,
+                           srcinc /* source address increment */,
+                           srcinc^1 /* destination address increment */,
+                           0 /* chunksize 0 (skip impossible on this dma) */,
+                           0 /* source skip after chunk (impossible) */,
+                           0 /* destination skip after chunk (impossible) */,
+                           4 /* 16 byte burst size */,
+                           -1 /* don't conserve bandwidth */,
+                           0 /* low watermark irq descriptor theshold */,
+                           0 /* disable hardware timestamps */,
+                           1 /* enable channel */);
+
+        s6000_pcm_enqueue_dma(substream);
+        s6000_pcm_enqueue_dma(substream);
+
+        spin_unlock_irqrestore(&prtd->lock, flags);
+
+        return 0;
+}
+
+static int s6000_pcm_stop(struct snd_pcm_substream *substream)
+{
+        struct s6000_runtime_data *prtd = substream->runtime->private_data;
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+        unsigned long flags;
+        u32 channel;
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                channel = par->dma_out;
+        else
+                channel = par->dma_in;
+
+        s6dmac_set_terminal_count(DMA_MASK_DMAC(channel),
+                                  DMA_INDEX_CHNL(channel), 0);
+
+        spin_lock_irqsave(&prtd->lock, flags);
+
+        s6dmac_disable_chan(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel));
+
+        spin_unlock_irqrestore(&prtd->lock, flags);
+
+        return 0;
+}
+
+static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+        int ret;
+
+        ret = par->trigger(substream, cmd, 0);
+        if (ret < 0)
+                return ret;
+
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_RESUME:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                ret = s6000_pcm_start(substream);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                ret = s6000_pcm_stop(substream);
+                break;
+        default:
+                ret = -EINVAL;
+        }
+        if (ret < 0)
+                return ret;
+
+        return par->trigger(substream, cmd, 1);
+}
+
+static int s6000_pcm_prepare(struct snd_pcm_substream *substream)
+{
+        struct s6000_runtime_data *prtd = substream->runtime->private_data;
+
+        prtd->period = 0;
+
+        return 0;
+}
+
+static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct s6000_runtime_data *prtd = runtime->private_data;
+        unsigned long flags;
+        unsigned int offset;
+        dma_addr_t count;
+
+        spin_lock_irqsave(&prtd->lock, flags);
+
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                count = s6dmac_cur_src(DMA_MASK_DMAC(par->dma_out),
+                                       DMA_INDEX_CHNL(par->dma_out));
+        else
+                count = s6dmac_cur_dst(DMA_MASK_DMAC(par->dma_in),
+                                       DMA_INDEX_CHNL(par->dma_in));
+
+        count -= runtime->dma_addr;
+
+        spin_unlock_irqrestore(&prtd->lock, flags);
+
+        offset = bytes_to_frames(runtime, count);
+        if (unlikely(offset >= runtime->buffer_size))
+                offset = 0;
+
+        return offset;
+}
+
+static int s6000_pcm_open(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct s6000_runtime_data *prtd;
+        int ret;
+
+        snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware);
+
+        ret = snd_pcm_hw_constraint_step(runtime, 0,
+                                         SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 16);
+        if (ret < 0)
+                return ret;
+        ret = snd_pcm_hw_constraint_step(runtime, 0,
+                                         SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16);
+        if (ret < 0)
+                return ret;
+        ret = snd_pcm_hw_constraint_integer(runtime,
+                                            SNDRV_PCM_HW_PARAM_PERIODS);
+        if (ret < 0)
+                return ret;
+
+        if (par->same_rate) {
+                int rate;
+                spin_lock(&par->lock); /* needed? */
+                rate = par->rate;
+                spin_unlock(&par->lock);
+                if (rate != -1) {
+                        ret = snd_pcm_hw_constraint_minmax(runtime,
+                                                        SNDRV_PCM_HW_PARAM_RATE,
+                                                        rate, rate);
+                        if (ret < 0)
+                                return ret;
+                }
+        }
+
+        prtd = kzalloc(sizeof(struct s6000_runtime_data), GFP_KERNEL);
+        if (prtd == NULL)
+                return -ENOMEM;
+
+        spin_lock_init(&prtd->lock);
+
+        runtime->private_data = prtd;
+
+        return 0;
+}
+
+static int s6000_pcm_close(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct s6000_runtime_data *prtd = runtime->private_data;
+
+        kfree(prtd);
+
+        return 0;
+}
+
+static int s6000_pcm_hw_params(struct snd_pcm_substream *substream,
+                                 struct snd_pcm_hw_params *hw_params)
+{
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+        int ret;
+        ret = snd_pcm_lib_malloc_pages(substream,
+                                       params_buffer_bytes(hw_params));
+        if (ret < 0) {
+                printk(KERN_WARNING "s6000-pcm: allocation of memory failed\n");
+                return ret;
+        }
+
+        if (par->same_rate) {
+                spin_lock(&par->lock);
+                if (par->rate == -1 ||
+                    !(par->in_use & ~(1 << substream->stream))) {
+                        par->rate = params_rate(hw_params);
+                        par->in_use |= 1 << substream->stream;
+                } else if (params_rate(hw_params) != par->rate) {
+                        snd_pcm_lib_free_pages(substream);
+                        par->in_use &= ~(1 << substream->stream);
+                        ret = -EBUSY;
+                }
+                spin_unlock(&par->lock);
+        }
+        return ret;
+}
+
+static int s6000_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+        struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data;
+
+        spin_lock(&par->lock);
+        par->in_use &= ~(1 << substream->stream);
+        if (!par->in_use)
+                par->rate = -1;
+        spin_unlock(&par->lock);
+
+        return snd_pcm_lib_free_pages(substream);
+}
+
+static struct snd_pcm_ops s6000_pcm_ops = {
+        .open =         s6000_pcm_open,
+        .close =        s6000_pcm_close,
+        .ioctl =        snd_pcm_lib_ioctl,
+        .hw_params =    s6000_pcm_hw_params,
+        .hw_free =      s6000_pcm_hw_free,
+        .trigger =      s6000_pcm_trigger,
+        .prepare =      s6000_pcm_prepare,
+        .pointer =      s6000_pcm_pointer,
+};
+
+static void s6000_pcm_free(struct snd_pcm *pcm)
+{
+        struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+        struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+
+        free_irq(params->irq, pcm);
+        snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static u64 s6000_pcm_dmamask = DMA_32BIT_MASK;
+
+static int s6000_pcm_new(struct snd_card *card,
+                         struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+        struct snd_soc_pcm_runtime *runtime = pcm->private_data;
+        struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data;
+        int res;
+
+        if (!card->dev->dma_mask)
+                card->dev->dma_mask = &s6000_pcm_dmamask;
+        if (!card->dev->coherent_dma_mask)
+                card->dev->coherent_dma_mask = DMA_32BIT_MASK;
+
+        if (params->dma_in) {
+                s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in),
+                                    DMA_INDEX_CHNL(params->dma_in));
+                s6dmac_int_sources(DMA_MASK_DMAC(params->dma_in),
+                                   DMA_INDEX_CHNL(params->dma_in));
+        }
+
+        if (params->dma_out) {
+                s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_out),
+                                    DMA_INDEX_CHNL(params->dma_out));
+                s6dmac_int_sources(DMA_MASK_DMAC(params->dma_out),
+                                   DMA_INDEX_CHNL(params->dma_out));
+        }
+
+        res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED,
+                          s6000_soc_platform.name, pcm);
+        if (res) {
+                printk(KERN_ERR "s6000-pcm couldn't get IRQ\n");
+                return res;
+        }
+
+        res = snd_pcm_lib_preallocate_pages_for_all(pcm,
+                                                    SNDRV_DMA_TYPE_DEV,
+                                                    card->dev,
+                                                    S6_PCM_PREALLOCATE_SIZE,
+                                                    S6_PCM_PREALLOCATE_MAX);
+        if (res)
+                printk(KERN_WARNING "s6000-pcm: preallocation failed\n");
+
+        spin_lock_init(&params->lock);
+        params->in_use = 0;
+        params->rate = -1;
+        return 0;
+}
+
+struct snd_soc_platform s6000_soc_platform = {
+        .name =         "s6000-audio",
+        .pcm_ops =      &s6000_pcm_ops,
+        .pcm_new =      s6000_pcm_new,
+        .pcm_free =     s6000_pcm_free,
+};
+EXPORT_SYMBOL_GPL(s6000_soc_platform);
+
+static int __init s6000_pcm_init(void)
+{
+        return snd_soc_register_platform(&s6000_soc_platform);
+}
+module_init(s6000_pcm_init);
+
+static void __exit s6000_pcm_exit(void)
+{
+        snd_soc_unregister_platform(&s6000_soc_platform);
+}
+module_exit(s6000_pcm_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6000-pcm.h b/sound/soc/s6000/s6000-pcm.h
new file mode 100644
index 000000000000..96f23f6f52bf
--- /dev/null
+++ b/sound/soc/s6000/s6000-pcm.h
@@ -0,0 +1,35 @@
+/*
+ * ALSA PCM interface for the Stretch s6000 family
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.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 _S6000_PCM_H
+#define _S6000_PCM_H
+
+struct snd_soc_dai;
+struct snd_pcm_substream;
+
+struct s6000_pcm_dma_params {
+        unsigned int (*check_xrun)(struct snd_soc_dai *cpu_dai);
+        int (*trigger)(struct snd_pcm_substream *substream, int cmd, int after);
+        dma_addr_t sif_in;
+        dma_addr_t sif_out;
+        u32 dma_in;
+        u32 dma_out;
+        int irq;
+        int same_rate;
+
+        spinlock_t lock;
+        int in_use;
+        int rate;
+};
+
+extern struct snd_soc_platform s6000_soc_platform;
+
+#endif
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
new file mode 100644
index 000000000000..b5f95f9781c1
--- /dev/null
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -0,0 +1,244 @@
+/*
+ * ASoC driver for Stretch s6105 IP camera platform
+ *
+ * Author:      Daniel Gloeckner, <dg@emlix.com>
+ * Copyright:   (C) 2009 emlix GmbH <info@emlix.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <variant/dmac.h>
+
+#include "../codecs/tlv320aic3x.h"
+#include "s6000-pcm.h"
+#include "s6000-i2s.h"
+
+#define S6105_CAM_CODEC_CLOCK 12288000
+
+static int s6105_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 = 0;
+
+        /* set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+                                             SND_SOC_DAIFMT_CBM_CFM);
+        if (ret < 0)
+                return ret;
+
+        /* set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM |
+                                           SND_SOC_DAIFMT_NB_NF);
+        if (ret < 0)
+                return ret;
+
+        /* set the codec system clock */
+        ret = snd_soc_dai_set_sysclk(codec_dai, 0, S6105_CAM_CODEC_CLOCK,
+                                            SND_SOC_CLOCK_OUT);
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+static struct snd_soc_ops s6105_ops = {
+        .hw_params = s6105_hw_params,
+};
+
+/* s6105 machine dapm widgets */
+static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
+        SND_SOC_DAPM_LINE("Audio Out Differential", NULL),
+        SND_SOC_DAPM_LINE("Audio Out Stereo", NULL),
+        SND_SOC_DAPM_LINE("Audio In", NULL),
+};
+
+/* s6105 machine audio_mapnections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
+        /* Audio Out connected to HPLOUT, HPLCOM, HPROUT */
+        {"Audio Out Differential", NULL, "HPLOUT"},
+        {"Audio Out Differential", NULL, "HPLCOM"},
+        {"Audio Out Stereo", NULL, "HPLOUT"},
+        {"Audio Out Stereo", NULL, "HPROUT"},
+
+        /* Audio In connected to LINE1L, LINE1R */
+        {"LINE1L", NULL, "Audio In"},
+        {"LINE1R", NULL, "Audio In"},
+};
+
+static int output_type_info(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+        uinfo->count = 1;
+        uinfo->value.enumerated.items = 2;
+        if (uinfo->value.enumerated.item) {
+                uinfo->value.enumerated.item = 1;
+                strcpy(uinfo->value.enumerated.name, "HPLOUT/HPROUT");
+        } else {
+                strcpy(uinfo->value.enumerated.name, "HPLOUT/HPLCOM");
+        }
+        return 0;
+}
+
+static int output_type_get(struct snd_kcontrol *kcontrol,
+                           struct snd_ctl_elem_value *ucontrol)
+{
+        ucontrol->value.enumerated.item[0] = kcontrol->private_value;
+        return 0;
+}
+
+static int output_type_put(struct snd_kcontrol *kcontrol,
+                           struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = kcontrol->private_data;
+        unsigned int val = (ucontrol->value.enumerated.item[0] != 0);
+        char *differential = "Audio Out Differential";
+        char *stereo = "Audio Out Stereo";
+
+        if (kcontrol->private_value == val)
+                return 0;
+        kcontrol->private_value = val;
+        snd_soc_dapm_disable_pin(codec, val ? differential : stereo);
+        snd_soc_dapm_sync(codec);
+        snd_soc_dapm_enable_pin(codec, val ? stereo : differential);
+        snd_soc_dapm_sync(codec);
+
+        return 1;
+}
+
+static const struct snd_kcontrol_new audio_out_mux = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .name = "Master Output Mux",
+        .index = 0,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .info = output_type_info,
+        .get = output_type_get,
+        .put = output_type_put,
+        .private_value = 1 /* default to stereo */
+};
+
+/* Logic for a aic3x as connected on the s6105 ip camera ref design */
+static int s6105_aic3x_init(struct snd_soc_codec *codec)
+{
+        /* Add s6105 specific widgets */
+        snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
+                                  ARRAY_SIZE(aic3x_dapm_widgets));
+
+        /* Set up s6105 specific audio path audio_map */
+        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+        /* not present */
+        snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
+        snd_soc_dapm_nc_pin(codec, "LINE2L");
+        snd_soc_dapm_nc_pin(codec, "LINE2R");
+
+        /* not connected */
+        snd_soc_dapm_nc_pin(codec, "MIC3L"); /* LINE2L on this chip */
+        snd_soc_dapm_nc_pin(codec, "MIC3R"); /* LINE2R on this chip */
+        snd_soc_dapm_nc_pin(codec, "LLOUT");
+        snd_soc_dapm_nc_pin(codec, "RLOUT");
+        snd_soc_dapm_nc_pin(codec, "HPRCOM");
+
+        /* always connected */
+        snd_soc_dapm_enable_pin(codec, "Audio In");
+
+        /* must correspond to audio_out_mux.private_value initializer */
+        snd_soc_dapm_disable_pin(codec, "Audio Out Differential");
+        snd_soc_dapm_sync(codec);
+        snd_soc_dapm_enable_pin(codec, "Audio Out Stereo");
+
+        snd_soc_dapm_sync(codec);
+
+        snd_ctl_add(codec->card, snd_ctl_new1(&audio_out_mux, codec));
+
+        return 0;
+}
+
+/* s6105 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link s6105_dai = {
+        .name = "TLV320AIC31",
+        .stream_name = "AIC31",
+        .cpu_dai = &s6000_i2s_dai,
+        .codec_dai = &aic3x_dai,
+        .init = s6105_aic3x_init,
+        .ops = &s6105_ops,
+};
+
+/* s6105 audio machine driver */
+static struct snd_soc_card snd_soc_card_s6105 = {
+        .name = "Stretch IP Camera",
+        .platform = &s6000_soc_platform,
+        .dai_link = &s6105_dai,
+        .num_links = 1,
+};
+
+/* s6105 audio private data */
+static struct aic3x_setup_data s6105_aic3x_setup = {
+        .i2c_bus = 0,
+        .i2c_address = 0x18,
+};
+
+/* s6105 audio subsystem */
+static struct snd_soc_device s6105_snd_devdata = {
+        .card = &snd_soc_card_s6105,
+        .codec_dev = &soc_codec_dev_aic3x,
+        .codec_data = &s6105_aic3x_setup,
+};
+
+static struct s6000_snd_platform_data __initdata s6105_snd_data = {
+        .wide           = 0,
+        .channel_in     = 0,
+        .channel_out    = 1,
+        .lines_in       = 1,
+        .lines_out      = 1,
+        .same_rate      = 1,
+};
+
+static struct platform_device *s6105_snd_device;
+
+static int __init s6105_init(void)
+{
+        int ret;
+
+        s6105_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!s6105_snd_device)
+                return -ENOMEM;
+
+        platform_set_drvdata(s6105_snd_device, &s6105_snd_devdata);
+        s6105_snd_devdata.dev = &s6105_snd_device->dev;
+        platform_device_add_data(s6105_snd_device, &s6105_snd_data,
+                                 sizeof(s6105_snd_data));
+
+        ret = platform_device_add(s6105_snd_device);
+        if (ret)
+                platform_device_put(s6105_snd_device);
+
+        return ret;
+}
+
+static void __exit s6105_exit(void)
+{
+        platform_device_unregister(s6105_snd_device);
+}
+
+module_init(s6105_init);
+module_exit(s6105_exit);
+
+MODULE_AUTHOR("Daniel Gloeckner");
+MODULE_DESCRIPTION("Stretch s6105 IP camera ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb920..baddb1242c71 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
         .info = (SNDRV_PCM_INFO_MMAP |
                 SNDRV_PCM_INFO_INTERLEAVED |
                 SNDRV_PCM_INFO_BLOCK_TRANSFER |
-                SNDRV_PCM_INFO_MMAP_VALID),
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_BATCH),
         .formats =      DMABRG_FMTS,
         .rates =        DMABRG_RATES,
         .rate_min =             8000,
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 56fa0872abbb..b378096cadb1 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -145,7 +145,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream,
         recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1;
 
         pr_debug("ssi_hw_params() enter\nssicr was    %08lx\n", ssicr);
-        pr_debug("bits: %d channels: %d\n", bits, channels);
+        pr_debug("bits: %u channels: %u\n", bits, channels);
 
         ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA |
                    CR_SWL_MASK);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 99712f652d0d..1d70829464ef 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -113,6 +113,35 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec)
 }
 #endif
 
+static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_soc_card *card = socdev->card;
+        struct snd_soc_dai_link *machine = rtd->dai;
+        struct snd_soc_dai *cpu_dai = machine->cpu_dai;
+        struct snd_soc_dai *codec_dai = machine->codec_dai;
+        int ret;
+
+        if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates ||
+            machine->symmetric_rates) {
+                dev_dbg(card->dev, "Symmetry forces %dHz rate\n", 
+                        machine->rate);
+
+                ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+                                                   SNDRV_PCM_HW_PARAM_RATE,
+                                                   machine->rate,
+                                                   machine->rate);
+                if (ret < 0) {
+                        dev_err(card->dev,
+                                "Unable to apply rate symmetry constraint: %d\n", ret);
+                        return ret;
+                }
+        }
+
+        return 0;
+}
+
 /*
  * Called by ALSA when a PCM substream is opened, the runtime->hw record is
  * then initialized and any private data can be allocated. This also calls
@@ -221,6 +250,13 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
                 goto machine_err;
         }
 
+        /* Symmetry only applies if we've already got an active stream. */
+        if (cpu_dai->active || codec_dai->active) {
+                ret = soc_pcm_apply_symmetry(substream);
+                if (ret != 0)
+                        goto machine_err;
+        }
+
         pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
         pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
         pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
@@ -263,7 +299,6 @@ static void close_delayed_work(struct work_struct *work)
 {
         struct snd_soc_card *card = container_of(work, struct snd_soc_card,
                                                  delayed_work.work);
-        struct snd_soc_device *socdev = card->socdev;
         struct snd_soc_codec *codec = card->codec;
         struct snd_soc_dai *codec_dai;
         int i;
@@ -279,27 +314,10 @@ static void close_delayed_work(struct work_struct *work)
 
                 /* are we waiting on this codec DAI stream */
                 if (codec_dai->pop_wait == 1) {
-
-                        /* Reduce power if no longer active */
-                        if (codec->active == 0) {
-                                pr_debug("pop wq D1 %s %s\n", codec->name,
-                                         codec_dai->playback.stream_name);
-                                snd_soc_dapm_set_bias_level(socdev,
-                                        SND_SOC_BIAS_PREPARE);
-                        }
-
                         codec_dai->pop_wait = 0;
                         snd_soc_dapm_stream_event(codec,
                                 codec_dai->playback.stream_name,
                                 SND_SOC_DAPM_STREAM_STOP);
-
-                        /* Fall into standby if no longer active */
-                        if (codec->active == 0) {
-                                pr_debug("pop wq D3 %s %s\n", codec->name,
-                                         codec_dai->playback.stream_name);
-                                snd_soc_dapm_set_bias_level(socdev,
-                                        SND_SOC_BIAS_STANDBY);
-                        }
                 }
         }
         mutex_unlock(&pcm_mutex);
@@ -363,10 +381,6 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
                 snd_soc_dapm_stream_event(codec,
                         codec_dai->capture.stream_name,
                         SND_SOC_DAPM_STREAM_STOP);
-
-                if (codec->active == 0 && codec_dai->pop_wait == 0)
-                        snd_soc_dapm_set_bias_level(socdev,
-                                                SND_SOC_BIAS_STANDBY);
         }
 
         mutex_unlock(&pcm_mutex);
@@ -431,36 +445,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
                 cancel_delayed_work(&card->delayed_work);
         }
 
-        /* do we need to power up codec */
-        if (codec->bias_level != SND_SOC_BIAS_ON) {
-                snd_soc_dapm_set_bias_level(socdev,
-                                            SND_SOC_BIAS_PREPARE);
-
-                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-                        snd_soc_dapm_stream_event(codec,
-                                        codec_dai->playback.stream_name,
-                                        SND_SOC_DAPM_STREAM_START);
-                else
-                        snd_soc_dapm_stream_event(codec,
-                                        codec_dai->capture.stream_name,
-                                        SND_SOC_DAPM_STREAM_START);
-
-                snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
-                snd_soc_dai_digital_mute(codec_dai, 0);
-
-        } else {
-                /* codec already powered - power on widgets */
-                if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-                        snd_soc_dapm_stream_event(codec,
-                                        codec_dai->playback.stream_name,
-                                        SND_SOC_DAPM_STREAM_START);
-                else
-                        snd_soc_dapm_stream_event(codec,
-                                        codec_dai->capture.stream_name,
-                                        SND_SOC_DAPM_STREAM_START);
+        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+                snd_soc_dapm_stream_event(codec,
+                                          codec_dai->playback.stream_name,
+                                          SND_SOC_DAPM_STREAM_START);
+        else
+                snd_soc_dapm_stream_event(codec,
+                                          codec_dai->capture.stream_name,
+                                          SND_SOC_DAPM_STREAM_START);
 
-                snd_soc_dai_digital_mute(codec_dai, 0);
-        }
+        snd_soc_dai_digital_mute(codec_dai, 0);
 
 out:
         mutex_unlock(&pcm_mutex);
@@ -521,6 +515,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
                 }
         }
 
+        machine->rate = params_rate(params);
+
 out:
         mutex_unlock(&pcm_mutex);
         return ret;
@@ -632,6 +628,12 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
         struct snd_soc_codec *codec = card->codec;
         int i;
 
+        /* If the initialization of this soc device failed, there is no codec
+         * associated with it. Just bail out in this case.
+         */
+        if (!codec)
+                return 0;
+
         /* Due to the resume being scheduled into a workqueue we could
         * suspend before that's finished - wait for it to complete.
          */
@@ -954,6 +956,9 @@ static int soc_remove(struct platform_device *pdev)
         struct snd_soc_platform *platform = card->platform;
         struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
+        if (!card->instantiated)
+                return 0;
+
         run_delayed_work(&card->delayed_work);
 
         if (platform->remove)
@@ -1331,6 +1336,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
                 return ret;
         }
 
+        codec->socdev = socdev;
         codec->card->dev = socdev->dev;
         codec->card->private_data = codec;
         strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
@@ -1383,6 +1389,9 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
         snprintf(codec->card->longname, sizeof(codec->card->longname),
                  "%s (%s)", card->name, codec->name);
 
+        /* Make sure all DAPM widgets are instantiated */
+        snd_soc_dapm_new_widgets(codec);
+
         ret = snd_card_register(codec->card);
         if (ret < 0) {
                 printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
@@ -1741,7 +1750,7 @@ int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
 {
         int max = kcontrol->private_value;
 
-        if (max == 1)
+        if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
                 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
         else
                 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -1771,7 +1780,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
         unsigned int shift = mc->shift;
         unsigned int rshift = mc->rshift;
 
-        if (max == 1)
+        if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
                 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
         else
                 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -1878,7 +1887,7 @@ int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
                 (struct soc_mixer_control *)kcontrol->private_value;
         int max = mc->max;
 
-        if (max == 1)
+        if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
                 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
         else
                 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -2062,7 +2071,7 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
         unsigned int freq, int dir)
 {
-        if (dai->ops->set_sysclk)
+        if (dai->ops && dai->ops->set_sysclk)
                 return dai->ops->set_sysclk(dai, clk_id, freq, dir);
         else
                 return -EINVAL;
@@ -2082,7 +2091,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
         int div_id, int div)
 {
-        if (dai->ops->set_clkdiv)
+        if (dai->ops && dai->ops->set_clkdiv)
                 return dai->ops->set_clkdiv(dai, div_id, div);
         else
                 return -EINVAL;
@@ -2101,7 +2110,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
         int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-        if (dai->ops->set_pll)
+        if (dai->ops && dai->ops->set_pll)
                 return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
         else
                 return -EINVAL;
@@ -2117,7 +2126,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-        if (dai->ops->set_fmt)
+        if (dai->ops && dai->ops->set_fmt)
                 return dai->ops->set_fmt(dai, fmt);
         else
                 return -EINVAL;
@@ -2136,7 +2145,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
         unsigned int mask, int slots)
 {
-        if (dai->ops->set_sysclk)
+        if (dai->ops && dai->ops->set_tdm_slot)
                 return dai->ops->set_tdm_slot(dai, mask, slots);
         else
                 return -EINVAL;
@@ -2152,7 +2161,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-        if (dai->ops->set_sysclk)
+        if (dai->ops && dai->ops->set_tristate)
                 return dai->ops->set_tristate(dai, tristate);
         else
                 return -EINVAL;
@@ -2168,7 +2177,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-        if (dai->ops->digital_mute)
+        if (dai->ops && dai->ops->digital_mute)
                 return dai->ops->digital_mute(dai, mute);
         else
                 return -EINVAL;
@@ -2349,6 +2358,39 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform)
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
 
+static u64 codec_format_map[] = {
+        SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE,
+        SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE,
+        SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE,
+        SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
+        SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
+        SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE,
+        SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
+        SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
+        SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE,
+        SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE,
+        SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE,
+        SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE,
+        SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE,
+        SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE,
+        SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
+        | SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+};
+
+/* Fix up the DAI formats for endianness: codecs don't actually see
+ * the endianness of the data but we're using the CPU format
+ * definitions which do need to include endianness so we ensure that
+ * codec DAIs always have both big and little endian variants set.
+ */
+static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
+{
+        int i;
+
+        for (i = 0; i < ARRAY_SIZE(codec_format_map); i++)
+                if (stream->formats & codec_format_map[i])
+                        stream->formats |= codec_format_map[i];
+}
+
 /**
  * snd_soc_register_codec - Register a codec with the ASoC core
  *
@@ -2356,6 +2398,8 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
  */
 int snd_soc_register_codec(struct snd_soc_codec *codec)
 {
+        int i;
+
         if (!codec->name)
                 return -EINVAL;
 
@@ -2365,6 +2409,11 @@ int snd_soc_register_codec(struct snd_soc_codec *codec)
 
         INIT_LIST_HEAD(&codec->list);
 
+        for (i = 0; i < codec->num_dai; i++) {
+                fixup_codec_formats(&codec->dai[i].playback);
+                fixup_codec_formats(&codec->dai[i].capture);
+        }
+
         mutex_lock(&client_mutex);
         list_add(&codec->list, &codec_list);
         snd_soc_instantiate_cards();
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 735903a74675..21c69074aa17 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -12,7 +12,7 @@
  *  Features:
  *    o Changes power status of internal codec blocks depending on the
  *      dynamic configuration of codec internal audio paths and active
- *      DAC's/ADC's.
+ *      DACs/ADCs.
  *    o Platform power domain - can support external components i.e. amps and
  *      mic/meadphone insertion events.
  *    o Automatic Mic Bias support
@@ -52,23 +52,21 @@
 
 /* dapm power sequences - make this per codec in the future */
 static int dapm_up_seq[] = {
-        snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
-        snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac,
-        snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga,
-        snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
+        snd_soc_dapm_pre, snd_soc_dapm_supply, snd_soc_dapm_micbias,
+        snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux,
+        snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl,
+        snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
+        snd_soc_dapm_post
 };
 
 static int dapm_down_seq[] = {
         snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
         snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer,
         snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias,
-        snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post
+        snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_supply,
+        snd_soc_dapm_post
 };
 
-static int dapm_status = 1;
-module_param(dapm_status, int, 0);
-MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
-
 static void pop_wait(u32 pop_time)
 {
         if (pop_time)
@@ -96,6 +94,48 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
         return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
 }
 
+/**
+ * snd_soc_dapm_set_bias_level - set the bias level for the system
+ * @socdev: audio device
+ * @level: level to configure
+ *
+ * Configure the bias (power) levels for the SoC audio device.
+ *
+ * Returns 0 for success else error.
+ */
+static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
+                                       enum snd_soc_bias_level level)
+{
+        struct snd_soc_card *card = socdev->card;
+        struct snd_soc_codec *codec = socdev->card->codec;
+        int ret = 0;
+
+        switch (level) {
+        case SND_SOC_BIAS_ON:
+                dev_dbg(socdev->dev, "Setting full bias\n");
+                break;
+        case SND_SOC_BIAS_PREPARE:
+                dev_dbg(socdev->dev, "Setting bias prepare\n");
+                break;
+        case SND_SOC_BIAS_STANDBY:
+                dev_dbg(socdev->dev, "Setting standby bias\n");
+                break;
+        case SND_SOC_BIAS_OFF:
+                dev_dbg(socdev->dev, "Setting bias off\n");
+                break;
+        default:
+                dev_err(socdev->dev, "Setting invalid bias %d\n", level);
+                return -EINVAL;
+        }
+
+        if (card->set_bias_level)
+                ret = card->set_bias_level(card, level);
+        if (ret == 0 && codec->set_bias_level)
+                ret = codec->set_bias_level(codec, level);
+
+        return ret;
+}
+
 /* set up initial codec paths */
 static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
         struct snd_soc_dapm_path *p, int i)
@@ -165,6 +205,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
         case snd_soc_dapm_dac:
         case snd_soc_dapm_micbias:
         case snd_soc_dapm_vmid:
+        case snd_soc_dapm_supply:
                 p->connect = 1;
         break;
         /* does effect routing - dynamically connected */
@@ -179,7 +220,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
         }
 }
 
-/* connect mux widget to it's interconnecting audio paths */
+/* connect mux widget to its interconnecting audio paths */
 static int dapm_connect_mux(struct snd_soc_codec *codec,
         struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
         struct snd_soc_dapm_path *path, const char *control_name,
@@ -202,7 +243,7 @@ static int dapm_connect_mux(struct snd_soc_codec *codec,
         return -ENODEV;
 }
 
-/* connect mixer widget to it's interconnecting audio paths */
+/* connect mixer widget to its interconnecting audio paths */
 static int dapm_connect_mixer(struct snd_soc_codec *codec,
         struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
         struct snd_soc_dapm_path *path, const char *control_name)
@@ -357,8 +398,9 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
                                 path->long_name);
                         ret = snd_ctl_add(codec->card, path->kcontrol);
                         if (ret < 0) {
-                                printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n",
-                                                path->long_name);
+                                printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n",
+                                       path->long_name,
+                                       ret);
                                 kfree(path->long_name);
                                 path->long_name = NULL;
                                 return ret;
@@ -434,6 +476,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
         struct snd_soc_dapm_path *path;
         int con = 0;
 
+        if (widget->id == snd_soc_dapm_supply)
+                return 0;
+
         if (widget->id == snd_soc_dapm_adc && widget->active)
                 return 1;
 
@@ -470,6 +515,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
         struct snd_soc_dapm_path *path;
         int con = 0;
 
+        if (widget->id == snd_soc_dapm_supply)
+                return 0;
+
         /* active stream ? */
         if (widget->id == snd_soc_dapm_dac && widget->active)
                 return 1;
@@ -521,84 +569,12 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 }
 EXPORT_SYMBOL_GPL(dapm_reg_event);
 
-/*
- * Scan a single DAPM widget for a complete audio path and update the
- * power status appropriately.
+/* Standard power change method, used to apply power changes to most
+ * widgets.
  */
-static int dapm_power_widget(struct snd_soc_codec *codec, int event,
-                             struct snd_soc_dapm_widget *w)
+static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w)
 {
-        int in, out, power_change, power, ret;
-
-        /* vmid - no action */
-        if (w->id == snd_soc_dapm_vmid)
-                return 0;
-
-        /* active ADC */
-        if (w->id == snd_soc_dapm_adc && w->active) {
-                in = is_connected_input_ep(w);
-                dapm_clear_walk(w->codec);
-                w->power = (in != 0) ? 1 : 0;
-                dapm_update_bits(w);
-                return 0;
-        }
-
-        /* active DAC */
-        if (w->id == snd_soc_dapm_dac && w->active) {
-                out = is_connected_output_ep(w);
-                dapm_clear_walk(w->codec);
-                w->power = (out != 0) ? 1 : 0;
-                dapm_update_bits(w);
-                return 0;
-        }
-
-        /* pre and post event widgets */
-        if (w->id == snd_soc_dapm_pre) {
-                if (!w->event)
-                        return 0;
-
-                if (event == SND_SOC_DAPM_STREAM_START) {
-                        ret = w->event(w,
-                                       NULL, SND_SOC_DAPM_PRE_PMU);
-                        if (ret < 0)
-                                return ret;
-                } else if (event == SND_SOC_DAPM_STREAM_STOP) {
-                        ret = w->event(w,
-                                       NULL, SND_SOC_DAPM_PRE_PMD);
-                        if (ret < 0)
-                                return ret;
-                }
-                return 0;
-        }
-        if (w->id == snd_soc_dapm_post) {
-                if (!w->event)
-                        return 0;
-
-                if (event == SND_SOC_DAPM_STREAM_START) {
-                        ret = w->event(w,
-                                       NULL, SND_SOC_DAPM_POST_PMU);
-                        if (ret < 0)
-                                return ret;
-                } else if (event == SND_SOC_DAPM_STREAM_STOP) {
-                        ret = w->event(w,
-                                       NULL, SND_SOC_DAPM_POST_PMD);
-                        if (ret < 0)
-                                return ret;
-                }
-                return 0;
-        }
-
-        /* all other widgets */
-        in = is_connected_input_ep(w);
-        dapm_clear_walk(w->codec);
-        out = is_connected_output_ep(w);
-        dapm_clear_walk(w->codec);
-        power = (out != 0 && in != 0) ? 1 : 0;
-        power_change = (w->power == power) ? 0 : 1;
-        w->power = power;
-
-        if (!power_change)
-                return 0;
+        int ret;
 
         /* call any power change event handlers */
         if (w->event)
@@ -607,7 +583,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
                          w->name, w->event_flags);
 
         /* power up pre event */
-        if (power && w->event &&
+        if (w->power && w->event &&
             (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
                 ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
                 if (ret < 0)
@@ -615,7 +591,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
         }
 
         /* power down pre event */
-        if (!power && w->event &&
+        if (!w->power && w->event &&
             (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
                 ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
                 if (ret < 0)
@@ -623,17 +599,17 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
         }
 
         /* Lower PGA volume to reduce pops */
-        if (w->id == snd_soc_dapm_pga && !power)
-                dapm_set_pga(w, power);
+        if (w->id == snd_soc_dapm_pga && !w->power)
+                dapm_set_pga(w, w->power);
 
         dapm_update_bits(w);
 
         /* Raise PGA volume to reduce pops */
-        if (w->id == snd_soc_dapm_pga && power)
-                dapm_set_pga(w, power);
+        if (w->id == snd_soc_dapm_pga && w->power)
+                dapm_set_pga(w, w->power);
 
         /* power up post event */
-        if (power && w->event &&
+        if (w->power && w->event &&
             (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
                 ret = w->event(w,
                                NULL, SND_SOC_DAPM_POST_PMU);
@@ -642,7 +618,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
         }
 
         /* power down post event */
-        if (!power && w->event &&
+        if (!w->power && w->event &&
             (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
                 ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
                 if (ret < 0)
@@ -652,6 +628,116 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
         return 0;
 }
 
+/* Generic check to see if a widget should be powered.
+ */
+static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
+{
+        int in, out;
+
+        in = is_connected_input_ep(w);
+        dapm_clear_walk(w->codec);
+        out = is_connected_output_ep(w);
+        dapm_clear_walk(w->codec);
+        return out != 0 && in != 0;
+}
+
+/* Check to see if an ADC has power */
+static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
+{
+        int in;
+
+        if (w->active) {
+                in = is_connected_input_ep(w);
+                dapm_clear_walk(w->codec);
+                return in != 0;
+        } else {
+                return dapm_generic_check_power(w);
+        }
+}
+
+/* Check to see if a DAC has power */
+static int dapm_dac_check_power(struct snd_soc_dapm_widget *w)
+{
+        int out;
+
+        if (w->active) {
+                out = is_connected_output_ep(w);
+                dapm_clear_walk(w->codec);
+                return out != 0;
+        } else {
+                return dapm_generic_check_power(w);
+        }
+}
+
+/* Check to see if a power supply is needed */
+static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
+{
+        struct snd_soc_dapm_path *path;
+        int power = 0;
+
+        /* Check if one of our outputs is connected */
+        list_for_each_entry(path, &w->sinks, list_source) {
+                if (path->sink && path->sink->power_check &&
+                    path->sink->power_check(path->sink)) {
+                        power = 1;
+                        break;
+                }
+        }
+
+        dapm_clear_walk(w->codec);
+
+        return power;
+}
+
+/*
+ * Scan a single DAPM widget for a complete audio path and update the
+ * power status appropriately.
+ */
+static int dapm_power_widget(struct snd_soc_codec *codec, int event,
+                             struct snd_soc_dapm_widget *w)
+{
+        int ret;
+
+        switch (w->id) {
+        case snd_soc_dapm_pre:
+                if (!w->event)
+                        return 0;
+
+                if (event == SND_SOC_DAPM_STREAM_START) {
+                        ret = w->event(w,
+                                       NULL, SND_SOC_DAPM_PRE_PMU);
+                        if (ret < 0)
+                                return ret;
+                } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+                        ret = w->event(w,
+                                       NULL, SND_SOC_DAPM_PRE_PMD);
+                        if (ret < 0)
+                                return ret;
+                }
+                return 0;
+
+        case snd_soc_dapm_post:
+                if (!w->event)
+                        return 0;
+
+                if (event == SND_SOC_DAPM_STREAM_START) {
+                        ret = w->event(w,
+                                       NULL, SND_SOC_DAPM_POST_PMU);
+                        if (ret < 0)
+                                return ret;
+                } else if (event == SND_SOC_DAPM_STREAM_STOP) {
+                        ret = w->event(w,
+                                       NULL, SND_SOC_DAPM_POST_PMD);
+                        if (ret < 0)
+                                return ret;
+                }
+                return 0;
+
+        default:
+                return dapm_generic_apply_power(w);
+        }
+}
+
 /*
  * Scan each dapm widget for complete audio path.
  * A complete path is a route that has valid endpoints i.e.:-
@@ -663,31 +749,102 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event,
  */
 static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 {
+        struct snd_soc_device *socdev = codec->socdev;
         struct snd_soc_dapm_widget *w;
-        int i, c = 1, *seq = NULL, ret = 0;
-
-        /* do we have a sequenced stream event */
-        if (event == SND_SOC_DAPM_STREAM_START) {
-                c = ARRAY_SIZE(dapm_up_seq);
-                seq = dapm_up_seq;
-        } else if (event == SND_SOC_DAPM_STREAM_STOP) {
-                c = ARRAY_SIZE(dapm_down_seq);
-                seq = dapm_down_seq;
+        int ret = 0;
+        int i, power;
+        int sys_power = 0;
+
+        INIT_LIST_HEAD(&codec->up_list);
+        INIT_LIST_HEAD(&codec->down_list);
+
+        /* Check which widgets we need to power and store them in
+         * lists indicating if they should be powered up or down.
+         */
+        list_for_each_entry(w, &codec->dapm_widgets, list) {
+                switch (w->id) {
+                case snd_soc_dapm_pre:
+                        list_add_tail(&codec->down_list, &w->power_list);
+                        break;
+                case snd_soc_dapm_post:
+                        list_add_tail(&codec->up_list, &w->power_list);
+                        break;
+
+                default:
+                        if (!w->power_check)
+                                continue;
+
+                        power = w->power_check(w);
+                        if (power)
+                                sys_power = 1;
+
+                        if (w->power == power)
+                                continue;
+
+                        if (power)
+                                list_add_tail(&w->power_list, &codec->up_list);
+                        else
+                                list_add_tail(&w->power_list,
+                                              &codec->down_list);
+
+                        w->power = power;
+                        break;
+                }
         }
 
-        for (i = 0; i < c; i++) {
-                list_for_each_entry(w, &codec->dapm_widgets, list) {
+        /* If we're changing to all on or all off then prepare */
+        if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) ||
+            (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) {
+                ret = snd_soc_dapm_set_bias_level(socdev,
+                                                  SND_SOC_BIAS_PREPARE);
+                if (ret != 0)
+                        pr_err("Failed to prepare bias: %d\n", ret);
+        }
 
+        /* Power down widgets first; try to avoid amplifying pops. */
+        for (i = 0; i < ARRAY_SIZE(dapm_down_seq); i++) {
+                list_for_each_entry(w, &codec->down_list, power_list) {
                         /* is widget in stream order */
-                        if (seq && seq[i] && w->id != seq[i])
+                        if (w->id != dapm_down_seq[i])
                                 continue;
 
                         ret = dapm_power_widget(codec, event, w);
                         if (ret != 0)
-                                return ret;
+                                pr_err("Failed to power down %s: %d\n",
+                                       w->name, ret);
                 }
         }
 
+        /* Now power up. */
+        for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) {
+                list_for_each_entry(w, &codec->up_list, power_list) {
+                        /* is widget in stream order */
+                        if (w->id != dapm_up_seq[i])
+                                continue;
+
+                        ret = dapm_power_widget(codec, event, w);
+                        if (ret != 0)
+                                pr_err("Failed to power up %s: %d\n",
+                                       w->name, ret);
+                }
+        }
+
+        /* If we just powered the last thing off drop to standby bias */
+        if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) {
+                ret = snd_soc_dapm_set_bias_level(socdev,
+                                                  SND_SOC_BIAS_STANDBY);
+                if (ret != 0)
+                        pr_err("Failed to apply standby bias: %d\n", ret);
+        }
+
+        /* If we just powered up then move to active bias */
+        if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) {
+                ret = snd_soc_dapm_set_bias_level(socdev,
+                                                  SND_SOC_BIAS_ON);
+                if (ret != 0)
+                        pr_err("Failed to apply active bias: %d\n", ret);
+        }
+
         return 0;
 }
 
@@ -723,6 +880,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
                 case snd_soc_dapm_pga:
                 case snd_soc_dapm_mixer:
                 case snd_soc_dapm_mixer_named_ctl:
+                case snd_soc_dapm_supply:
                         if (w->name) {
                                 in = is_connected_input_ep(w);
                                 dapm_clear_walk(w->codec);
@@ -851,6 +1009,7 @@ static ssize_t dapm_widget_show(struct device *dev,
                 case snd_soc_dapm_pga:
                 case snd_soc_dapm_mixer:
                 case snd_soc_dapm_mixer_named_ctl:
+                case snd_soc_dapm_supply:
                         if (w->name)
                                 count += sprintf(buf + count, "%s: %s\n",
                                         w->name, w->power ? "On":"Off");
@@ -883,16 +1042,12 @@ static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
 
 int snd_soc_dapm_sys_add(struct device *dev)
 {
-        if (!dapm_status)
-                return 0;
         return device_create_file(dev, &dev_attr_dapm_widget);
 }
 
 static void snd_soc_dapm_sys_remove(struct device *dev)
 {
-        if (dapm_status) {
-                device_remove_file(dev, &dev_attr_dapm_widget);
-        }
+        device_remove_file(dev, &dev_attr_dapm_widget);
 }
 
 /* free all dapm widgets and resources */
@@ -1015,6 +1170,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
         case snd_soc_dapm_vmid:
         case snd_soc_dapm_pre:
         case snd_soc_dapm_post:
+        case snd_soc_dapm_supply:
                 list_add(&path->list, &codec->dapm_paths);
                 list_add(&path->list_sink, &wsink->sources);
                 list_add(&path->list_source, &wsource->sinks);
@@ -1108,15 +1264,22 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
                 case snd_soc_dapm_switch:
                 case snd_soc_dapm_mixer:
                 case snd_soc_dapm_mixer_named_ctl:
+                        w->power_check = dapm_generic_check_power;
                         dapm_new_mixer(codec, w);
                         break;
                 case snd_soc_dapm_mux:
                 case snd_soc_dapm_value_mux:
+                        w->power_check = dapm_generic_check_power;
                         dapm_new_mux(codec, w);
                         break;
                 case snd_soc_dapm_adc:
+                        w->power_check = dapm_adc_check_power;
+                        break;
                 case snd_soc_dapm_dac:
+                        w->power_check = dapm_dac_check_power;
+                        break;
                 case snd_soc_dapm_pga:
+                        w->power_check = dapm_generic_check_power;
                         dapm_new_pga(codec, w);
                         break;
                 case snd_soc_dapm_input:
@@ -1126,6 +1289,10 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
                 case snd_soc_dapm_hp:
                 case snd_soc_dapm_mic:
                 case snd_soc_dapm_line:
+                        w->power_check = dapm_generic_check_power;
+                        break;
+                case snd_soc_dapm_supply:
+                        w->power_check = dapm_supply_check_power;
                 case snd_soc_dapm_vmid:
                 case snd_soc_dapm_pre:
                 case snd_soc_dapm_post:
@@ -1626,35 +1793,11 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
 EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 
 /**
- * snd_soc_dapm_set_bias_level - set the bias level for the system
- * @socdev: audio device
- * @level: level to configure
- *
- * Configure the bias (power) levels for the SoC audio device.
- *
- * Returns 0 for success else error.
- */
-int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
-                                enum snd_soc_bias_level level)
-{
-        struct snd_soc_card *card = socdev->card;
-        struct snd_soc_codec *codec = socdev->card->codec;
-        int ret = 0;
-
-        if (card->set_bias_level)
-                ret = card->set_bias_level(card, level);
-        if (ret == 0 && codec->set_bias_level)
-                ret = codec->set_bias_level(codec, level);
-
-        return ret;
-}
-
-/**
  * snd_soc_dapm_enable_pin - enable pin.
  * @codec: SoC codec
  * @pin: pin name
  *
- * Enables input/output pin and it's parents or children widgets iff there is
+ * Enables input/output pin and its parents or children widgets iff there is
  * a valid audio route and active audio stream.
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
@@ -1670,7 +1813,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
  * @codec: SoC codec
  * @pin: pin name
  *
- * Disables input/output pin and it's parents or children widgets.
+ * Disables input/output pin and its parents or children widgets.
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
diff --git a/sound/soc/txx9/Kconfig b/sound/soc/txx9/Kconfig
new file mode 100644
index 000000000000..ebc9327eae71
--- /dev/null
+++ b/sound/soc/txx9/Kconfig
@@ -0,0 +1,29 @@
+##
+## TXx9 ACLC
+##
+config SND_SOC_TXX9ACLC
+        tristate "SoC Audio for TXx9"
+        depends on HAS_TXX9_ACLC && TXX9_DMAC
+        help
+          This option enables support for the AC Link Controllers in TXx9 SoC.
+
+config HAS_TXX9_ACLC
+        bool
+
+config SND_SOC_TXX9ACLC_AC97
+        tristate
+        select AC97_BUS
+        select SND_AC97_CODEC
+        select SND_SOC_AC97_BUS
+
+
+##
+## Boards
+##
+config SND_SOC_TXX9ACLC_GENERIC
+        tristate "Generic TXx9 ACLC sound machine"
+        depends on SND_SOC_TXX9ACLC
+        select SND_SOC_TXX9ACLC_AC97
+        select SND_SOC_AC97_CODEC
+        help
+          This is a generic AC97 sound machine for use in TXx9 based systems.
diff --git a/sound/soc/txx9/Makefile b/sound/soc/txx9/Makefile
new file mode 100644
index 000000000000..551f16c0c4f9
--- /dev/null
+++ b/sound/soc/txx9/Makefile
@@ -0,0 +1,11 @@
+# Platform
+snd-soc-txx9aclc-objs := txx9aclc.o
+snd-soc-txx9aclc-ac97-objs := txx9aclc-ac97.o
+
+obj-$(CONFIG_SND_SOC_TXX9ACLC) += snd-soc-txx9aclc.o
+obj-$(CONFIG_SND_SOC_TXX9ACLC_AC97) += snd-soc-txx9aclc-ac97.o
+
+# Machine
+snd-soc-txx9aclc-generic-objs := txx9aclc-generic.o
+
+obj-$(CONFIG_SND_SOC_TXX9ACLC_GENERIC) += snd-soc-txx9aclc-generic.o
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
new file mode 100644
index 000000000000..0f83bdb9b16f
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -0,0 +1,255 @@
+/*
+ * TXx9 ACLC AC97 driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include "txx9aclc.h"
+
+#define AC97_DIR        \
+        (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE)
+
+#define AC97_RATES      \
+        SNDRV_PCM_RATE_8000_48000
+
+#ifdef __BIG_ENDIAN
+#define AC97_FMTS       SNDRV_PCM_FMTBIT_S16_BE
+#else
+#define AC97_FMTS       SNDRV_PCM_FMTBIT_S16_LE
+#endif
+
+static DECLARE_WAIT_QUEUE_HEAD(ac97_waitq);
+
+/* REVISIT: How to find txx9aclc_soc_device from snd_ac97? */
+static struct txx9aclc_soc_device *txx9aclc_soc_dev;
+
+static int txx9aclc_regready(struct txx9aclc_soc_device *dev)
+{
+        struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+
+        return __raw_readl(drvdata->base + ACINTSTS) & ACINT_REGACCRDY;
+}
+
+/* AC97 controller reads codec register */
+static unsigned short txx9aclc_ac97_read(struct snd_ac97 *ac97,
+                                         unsigned short reg)
+{
+        struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+        struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+        void __iomem *base = drvdata->base;
+        u32 dat;
+
+        if (!(__raw_readl(base + ACINTSTS) & ACINT_CODECRDY(ac97->num)))
+                return 0xffff;
+        reg |= ac97->num << 7;
+        dat = (reg << ACREGACC_REG_SHIFT) | ACREGACC_READ;
+        __raw_writel(dat, base + ACREGACC);
+        __raw_writel(ACINT_REGACCRDY, base + ACINTEN);
+        if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) {
+                __raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+                dev_err(dev->soc_dev.dev, "ac97 read timeout (reg %#x)\n", reg);
+                dat = 0xffff;
+                goto done;
+        }
+        dat = __raw_readl(base + ACREGACC);
+        if (((dat >> ACREGACC_REG_SHIFT) & 0xff) != reg) {
+                dev_err(dev->soc_dev.dev, "reg mismatch %x with %x\n",
+                        dat, reg);
+                dat = 0xffff;
+                goto done;
+        }
+        dat = (dat >> ACREGACC_DAT_SHIFT) & 0xffff;
+done:
+        __raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+        return dat;
+}
+
+/* AC97 controller writes to codec register */
+static void txx9aclc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
+                                unsigned short val)
+{
+        struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+        struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+        void __iomem *base = drvdata->base;
+
+        __raw_writel(((reg | (ac97->num << 7)) << ACREGACC_REG_SHIFT) |
+                     (val << ACREGACC_DAT_SHIFT),
+                     base + ACREGACC);
+        __raw_writel(ACINT_REGACCRDY, base + ACINTEN);
+        if (!wait_event_timeout(ac97_waitq, txx9aclc_regready(dev), HZ)) {
+                dev_err(dev->soc_dev.dev,
+                        "ac97 write timeout (reg %#x)\n", reg);
+        }
+        __raw_writel(ACINT_REGACCRDY, base + ACINTDIS);
+}
+
+static void txx9aclc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+        struct txx9aclc_soc_device *dev = txx9aclc_soc_dev;
+        struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+        void __iomem *base = drvdata->base;
+        u32 ready = ACINT_CODECRDY(ac97->num) | ACINT_REGACCRDY;
+
+        __raw_writel(ACCTL_ENLINK, base + ACCTLDIS);
+        mmiowb();
+        udelay(1);
+        __raw_writel(ACCTL_ENLINK, base + ACCTLEN);
+        /* wait for primary codec ready status */
+        __raw_writel(ready, base + ACINTEN);
+        if (!wait_event_timeout(ac97_waitq,
+                                (__raw_readl(base + ACINTSTS) & ready) == ready,
+                                HZ)) {
+                dev_err(&ac97->dev, "primary codec is not ready "
+                        "(status %#x)\n",
+                        __raw_readl(base + ACINTSTS));
+        }
+        __raw_writel(ACINT_REGACCRDY, base + ACINTSTS);
+        __raw_writel(ready, base + ACINTDIS);
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+        .read           = txx9aclc_ac97_read,
+        .write          = txx9aclc_ac97_write,
+        .reset          = txx9aclc_ac97_cold_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+static irqreturn_t txx9aclc_ac97_irq(int irq, void *dev_id)
+{
+        struct txx9aclc_plat_drvdata *drvdata = dev_id;
+        void __iomem *base = drvdata->base;
+
+        __raw_writel(__raw_readl(base + ACINTMSTS), base + ACINTDIS);
+        wake_up(&ac97_waitq);
+        return IRQ_HANDLED;
+}
+
+static int txx9aclc_ac97_probe(struct platform_device *pdev,
+                               struct snd_soc_dai *dai)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct txx9aclc_soc_device *dev =
+                container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+
+        dev->aclc_pdev = to_platform_device(dai->dev);
+        txx9aclc_soc_dev = dev;
+        return 0;
+}
+
+static void txx9aclc_ac97_remove(struct platform_device *pdev,
+                                 struct snd_soc_dai *dai)
+{
+        struct platform_device *aclc_pdev = to_platform_device(dai->dev);
+        struct txx9aclc_plat_drvdata *drvdata = platform_get_drvdata(aclc_pdev);
+
+        /* disable AC-link */
+        __raw_writel(ACCTL_ENLINK, drvdata->base + ACCTLDIS);
+        txx9aclc_soc_dev = NULL;
+}
+
+struct snd_soc_dai txx9aclc_ac97_dai = {
+        .name                   = "txx9aclc_ac97",
+        .ac97_control           = 1,
+        .probe                  = txx9aclc_ac97_probe,
+        .remove                 = txx9aclc_ac97_remove,
+        .playback = {
+                .rates          = AC97_RATES,
+                .formats        = AC97_FMTS,
+                .channels_min   = 2,
+                .channels_max   = 2,
+        },
+        .capture = {
+                .rates          = AC97_RATES,
+                .formats        = AC97_FMTS,
+                .channels_min   = 2,
+                .channels_max   = 2,
+        },
+};
+EXPORT_SYMBOL_GPL(txx9aclc_ac97_dai);
+
+static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev)
+{
+        struct txx9aclc_plat_drvdata *drvdata;
+        struct resource *r;
+        int err;
+        int irq;
+
+        irq = platform_get_irq(pdev, 0);
+        if (irq < 0)
+                return irq;
+        r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!r)
+                return -EBUSY;
+
+        if (!devm_request_mem_region(&pdev->dev, r->start, resource_size(r),
+                                     dev_name(&pdev->dev)))
+                return -EBUSY;
+
+        drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
+        if (!drvdata)
+                return -ENOMEM;
+        platform_set_drvdata(pdev, drvdata);
+        drvdata->physbase = r->start;
+        if (sizeof(drvdata->physbase) > sizeof(r->start) &&
+            r->start >= TXX9_DIRECTMAP_BASE &&
+            r->start < TXX9_DIRECTMAP_BASE + 0x400000)
+                drvdata->physbase |= 0xf00000000ull;
+        drvdata->base = devm_ioremap(&pdev->dev, r->start, resource_size(r));
+        if (!drvdata->base)
+                return -EBUSY;
+        err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq,
+                               IRQF_DISABLED, dev_name(&pdev->dev), drvdata);
+        if (err < 0)
+                return err;
+
+        txx9aclc_ac97_dai.dev = &pdev->dev;
+        return snd_soc_register_dai(&txx9aclc_ac97_dai);
+}
+
+static int __devexit txx9aclc_ac97_dev_remove(struct platform_device *pdev)
+{
+        snd_soc_unregister_dai(&txx9aclc_ac97_dai);
+        return 0;
+}
+
+static struct platform_driver txx9aclc_ac97_driver = {
+        .probe          = txx9aclc_ac97_dev_probe,
+        .remove         = __devexit_p(txx9aclc_ac97_dev_remove),
+        .driver         = {
+                .name   = "txx9aclc-ac97",
+                .owner  = THIS_MODULE,
+        },
+};
+
+static int __init txx9aclc_ac97_init(void)
+{
+        return platform_driver_register(&txx9aclc_ac97_driver);
+}
+
+static void __exit txx9aclc_ac97_exit(void)
+{
+        platform_driver_unregister(&txx9aclc_ac97_driver);
+}
+
+module_init(txx9aclc_ac97_init);
+module_exit(txx9aclc_ac97_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("TXx9 ACLC AC97 driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc-generic.c b/sound/soc/txx9/txx9aclc-generic.c
new file mode 100644
index 000000000000..3175de9a92cb
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc-generic.c
@@ -0,0 +1,98 @@
+/*
+ * Generic TXx9 ACLC machine driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * 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 is a very generic AC97 sound machine driver for boards which
+ * have (AC97) audio at ACLC (e.g. RBTX49XX boards).
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include "../codecs/ac97.h"
+#include "txx9aclc.h"
+
+static struct snd_soc_dai_link txx9aclc_generic_dai = {
+        .name = "AC97",
+        .stream_name = "AC97 HiFi",
+        .cpu_dai = &txx9aclc_ac97_dai,
+        .codec_dai = &ac97_dai,
+};
+
+static struct snd_soc_card txx9aclc_generic_card = {
+        .name           = "Generic TXx9 ACLC Audio",
+        .platform       = &txx9aclc_soc_platform,
+        .dai_link       = &txx9aclc_generic_dai,
+        .num_links      = 1,
+};
+
+static struct txx9aclc_soc_device txx9aclc_generic_soc_device = {
+        .soc_dev = {
+                .card           = &txx9aclc_generic_card,
+                .codec_dev      = &soc_codec_dev_ac97,
+        },
+};
+
+static int __init txx9aclc_generic_probe(struct platform_device *pdev)
+{
+        struct txx9aclc_soc_device *dev = &txx9aclc_generic_soc_device;
+        struct platform_device *soc_pdev;
+        int ret;
+
+        soc_pdev = platform_device_alloc("soc-audio", -1);
+        if (!soc_pdev)
+                return -ENOMEM;
+        platform_set_drvdata(soc_pdev, &dev->soc_dev);
+        dev->soc_dev.dev = &soc_pdev->dev;
+        ret = platform_device_add(soc_pdev);
+        if (ret) {
+                platform_device_put(soc_pdev);
+                return ret;
+        }
+        platform_set_drvdata(pdev, soc_pdev);
+        return 0;
+}
+
+static int __exit txx9aclc_generic_remove(struct platform_device *pdev)
+{
+        struct platform_device *soc_pdev = platform_get_drvdata(pdev);
+
+        platform_device_unregister(soc_pdev);
+        return 0;
+}
+
+static struct platform_driver txx9aclc_generic_driver = {
+        .remove = txx9aclc_generic_remove,
+        .driver = {
+                .name = "txx9aclc-generic",
+                .owner = THIS_MODULE,
+        },
+};
+
+static int __init txx9aclc_generic_init(void)
+{
+        return platform_driver_probe(&txx9aclc_generic_driver,
+                                     txx9aclc_generic_probe);
+}
+
+static void __exit txx9aclc_generic_exit(void)
+{
+        platform_driver_unregister(&txx9aclc_generic_driver);
+}
+
+module_init(txx9aclc_generic_init);
+module_exit(txx9aclc_generic_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("Generic TXx9 ACLC ALSA SoC audio driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
new file mode 100644
index 000000000000..938a58a5a244
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc.c
@@ -0,0 +1,430 @@
+/*
+ * Generic TXx9 ACLC platform driver
+ *
+ * Copyright (C) 2009 Atsushi Nemoto
+ *
+ * Based on RBTX49xx patch from CELF patch archive.
+ * (C) Copyright TOSHIBA CORPORATION 2004-2006
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/scatterlist.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "txx9aclc.h"
+
+static const struct snd_pcm_hardware txx9aclc_pcm_hardware = {
+        /*
+         * REVISIT: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID
+         * needs more works for noncoherent MIPS.
+         */
+        .info             = SNDRV_PCM_INFO_INTERLEAVED |
+                            SNDRV_PCM_INFO_BATCH |
+                            SNDRV_PCM_INFO_PAUSE,
+#ifdef __BIG_ENDIAN
+        .formats          = SNDRV_PCM_FMTBIT_S16_BE,
+#else
+        .formats          = SNDRV_PCM_FMTBIT_S16_LE,
+#endif
+        .period_bytes_min = 1024,
+        .period_bytes_max = 8 * 1024,
+        .periods_min      = 2,
+        .periods_max      = 4096,
+        .buffer_bytes_max = 32 * 1024,
+};
+
+static int txx9aclc_pcm_hw_params(struct snd_pcm_substream *substream,
+                                  struct snd_pcm_hw_params *params)
+{
+        struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+        struct snd_soc_device *socdev = rtd->socdev;
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct txx9aclc_dmadata *dmadata = runtime->private_data;
+        int ret;
+
+        ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+        if (ret < 0)
+                return ret;
+
+        dev_dbg(socdev->dev,
+                "runtime->dma_area = %#lx dma_addr = %#lx dma_bytes = %zd "
+                "runtime->min_align %ld\n",
+                (unsigned long)runtime->dma_area,
+                (unsigned long)runtime->dma_addr, runtime->dma_bytes,
+                runtime->min_align);
+        dev_dbg(socdev->dev,
+                "periods %d period_bytes %d stream %d\n",
+                params_periods(params), params_period_bytes(params),
+                substream->stream);
+
+        dmadata->substream = substream;
+        dmadata->pos = 0;
+        return 0;
+}
+
+static int txx9aclc_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        return snd_pcm_lib_free_pages(substream);
+}
+
+static int txx9aclc_pcm_prepare(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct txx9aclc_dmadata *dmadata = runtime->private_data;
+
+        dmadata->dma_addr = runtime->dma_addr;
+        dmadata->buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
+        dmadata->period_bytes = snd_pcm_lib_period_bytes(substream);
+
+        if (dmadata->buffer_bytes == dmadata->period_bytes) {
+                dmadata->frag_bytes = dmadata->period_bytes >> 1;
+                dmadata->frags = 2;
+        } else {
+                dmadata->frag_bytes = dmadata->period_bytes;
+                dmadata->frags = dmadata->buffer_bytes / dmadata->period_bytes;
+        }
+        dmadata->frag_count = 0;
+        dmadata->pos = 0;
+        return 0;
+}
+
+static void txx9aclc_dma_complete(void *arg)
+{
+        struct txx9aclc_dmadata *dmadata = arg;
+        unsigned long flags;
+
+        /* dma completion handler cannot submit new operations */
+        spin_lock_irqsave(&dmadata->dma_lock, flags);
+        if (dmadata->frag_count >= 0) {
+                dmadata->dmacount--;
+                BUG_ON(dmadata->dmacount < 0);
+                tasklet_schedule(&dmadata->tasklet);
+        }
+        spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+}
+
+static struct dma_async_tx_descriptor *
+txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
+{
+        struct dma_chan *chan = dmadata->dma_chan;
+        struct dma_async_tx_descriptor *desc;
+        struct scatterlist sg;
+
+        sg_init_table(&sg, 1);
+        sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf_dma_addr)),
+                    dmadata->frag_bytes, buf_dma_addr & (PAGE_SIZE - 1));
+        sg_dma_address(&sg) = buf_dma_addr;
+        desc = chan->device->device_prep_slave_sg(chan, &sg, 1,
+                dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+                DMA_TO_DEVICE : DMA_FROM_DEVICE,
+                DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+        if (!desc) {
+                dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+                return NULL;
+        }
+        desc->callback = txx9aclc_dma_complete;
+        desc->callback_param = dmadata;
+        desc->tx_submit(desc);
+        return desc;
+}
+
+#define NR_DMA_CHAIN            2
+
+static void txx9aclc_dma_tasklet(unsigned long data)
+{
+        struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data;
+        struct dma_chan *chan = dmadata->dma_chan;
+        struct dma_async_tx_descriptor *desc;
+        struct snd_pcm_substream *substream = dmadata->substream;
+        u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+                ACCTL_AUDODMA : ACCTL_AUDIDMA;
+        int i;
+        unsigned long flags;
+
+        spin_lock_irqsave(&dmadata->dma_lock, flags);
+        if (dmadata->frag_count < 0) {
+                struct txx9aclc_soc_device *dev =
+                        container_of(dmadata, struct txx9aclc_soc_device,
+                                     dmadata[substream->stream]);
+                struct txx9aclc_plat_drvdata *drvdata =
+                        txx9aclc_get_plat_drvdata(dev);
+                void __iomem *base = drvdata->base;
+
+                spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+                chan->device->device_terminate_all(chan);
+                /* first time */
+                for (i = 0; i < NR_DMA_CHAIN; i++) {
+                        desc = txx9aclc_dma_submit(dmadata,
+                                dmadata->dma_addr + i * dmadata->frag_bytes);
+                        if (!desc)
+                                return;
+                }
+                dmadata->dmacount = NR_DMA_CHAIN;
+                chan->device->device_issue_pending(chan);
+                spin_lock_irqsave(&dmadata->dma_lock, flags);
+                __raw_writel(ctlbit, base + ACCTLEN);
+                dmadata->frag_count = NR_DMA_CHAIN % dmadata->frags;
+                spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+                return;
+        }
+        BUG_ON(dmadata->dmacount >= NR_DMA_CHAIN);
+        while (dmadata->dmacount < NR_DMA_CHAIN) {
+                dmadata->dmacount++;
+                spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+                desc = txx9aclc_dma_submit(dmadata,
+                        dmadata->dma_addr +
+                        dmadata->frag_count * dmadata->frag_bytes);
+                if (!desc)
+                        return;
+                chan->device->device_issue_pending(chan);
+
+                spin_lock_irqsave(&dmadata->dma_lock, flags);
+                dmadata->frag_count++;
+                dmadata->frag_count %= dmadata->frags;
+                dmadata->pos += dmadata->frag_bytes;
+                dmadata->pos %= dmadata->buffer_bytes;
+                if ((dmadata->frag_count * dmadata->frag_bytes) %
+                    dmadata->period_bytes == 0)
+                        snd_pcm_period_elapsed(substream);
+        }
+        spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+}
+
+static int txx9aclc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct txx9aclc_soc_device *dev =
+                container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev);
+        struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+        void __iomem *base = drvdata->base;
+        unsigned long flags;
+        int ret = 0;
+        u32 ctlbit = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+                ACCTL_AUDODMA : ACCTL_AUDIDMA;
+
+        spin_lock_irqsave(&dmadata->dma_lock, flags);
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+                dmadata->frag_count = -1;
+                tasklet_schedule(&dmadata->tasklet);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+        case SNDRV_PCM_TRIGGER_SUSPEND:
+                __raw_writel(ctlbit, base + ACCTLDIS);
+                break;
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+        case SNDRV_PCM_TRIGGER_RESUME:
+                __raw_writel(ctlbit, base + ACCTLEN);
+                break;
+        default:
+                ret = -EINVAL;
+        }
+        spin_unlock_irqrestore(&dmadata->dma_lock, flags);
+        return ret;
+}
+
+static snd_pcm_uframes_t
+txx9aclc_pcm_pointer(struct snd_pcm_substream *substream)
+{
+        struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+
+        return bytes_to_frames(substream->runtime, dmadata->pos);
+}
+
+static int txx9aclc_pcm_open(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct txx9aclc_soc_device *dev =
+                container_of(rtd->socdev, struct txx9aclc_soc_device, soc_dev);
+        struct txx9aclc_dmadata *dmadata = &dev->dmadata[substream->stream];
+        int ret;
+
+        ret = snd_soc_set_runtime_hwparams(substream, &txx9aclc_pcm_hardware);
+        if (ret)
+                return ret;
+        /* ensure that buffer size is a multiple of period size */
+        ret = snd_pcm_hw_constraint_integer(substream->runtime,
+                                            SNDRV_PCM_HW_PARAM_PERIODS);
+        if (ret < 0)
+                return ret;
+        substream->runtime->private_data = dmadata;
+        return 0;
+}
+
+static int txx9aclc_pcm_close(struct snd_pcm_substream *substream)
+{
+        struct txx9aclc_dmadata *dmadata = substream->runtime->private_data;
+        struct dma_chan *chan = dmadata->dma_chan;
+
+        dmadata->frag_count = -1;
+        chan->device->device_terminate_all(chan);
+        return 0;
+}
+
+static struct snd_pcm_ops txx9aclc_pcm_ops = {
+        .open           = txx9aclc_pcm_open,
+        .close          = txx9aclc_pcm_close,
+        .ioctl          = snd_pcm_lib_ioctl,
+        .hw_params      = txx9aclc_pcm_hw_params,
+        .hw_free        = txx9aclc_pcm_hw_free,
+        .prepare        = txx9aclc_pcm_prepare,
+        .trigger        = txx9aclc_pcm_trigger,
+        .pointer        = txx9aclc_pcm_pointer,
+};
+
+static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+        snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+                            struct snd_pcm *pcm)
+{
+        return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+                card->dev, 64 * 1024, 4 * 1024 * 1024);
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+        struct txx9aclc_dmadata *dmadata = param;
+        char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */
+
+        snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name,
+                (int)dmadata->dma_res->start);
+        if (strcmp(dev_name(chan->device->dev), devname) == 0) {
+                chan->private = &dmadata->dma_slave;
+                return true;
+        }
+        return false;
+}
+
+static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev,
+                             struct txx9aclc_dmadata *dmadata)
+{
+        struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+        struct txx9dmac_slave *ds = &dmadata->dma_slave;
+        dma_cap_mask_t mask;
+
+        spin_lock_init(&dmadata->dma_lock);
+
+        ds->reg_width = sizeof(u32);
+        if (dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+                ds->tx_reg = drvdata->physbase + ACAUDODAT;
+                ds->rx_reg = 0;
+        } else {
+                ds->tx_reg = 0;
+                ds->rx_reg = drvdata->physbase + ACAUDIDAT;
+        }
+
+        /* Try to grab a DMA channel */
+        dma_cap_zero(mask);
+        dma_cap_set(DMA_SLAVE, mask);
+        dmadata->dma_chan = dma_request_channel(mask, filter, dmadata);
+        if (!dmadata->dma_chan) {
+                dev_err(dev->soc_dev.dev,
+                        "DMA channel for %s is not available\n",
+                        dmadata->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+                        "playback" : "capture");
+                return -EBUSY;
+        }
+        tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet,
+                     (unsigned long)dmadata);
+        return 0;
+}
+
+static int txx9aclc_pcm_probe(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct txx9aclc_soc_device *dev =
+                container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+        struct resource *r;
+        int i;
+        int ret;
+
+        dev->dmadata[0].stream = SNDRV_PCM_STREAM_PLAYBACK;
+        dev->dmadata[1].stream = SNDRV_PCM_STREAM_CAPTURE;
+        for (i = 0; i < 2; i++) {
+                r = platform_get_resource(dev->aclc_pdev, IORESOURCE_DMA, i);
+                if (!r) {
+                        ret = -EBUSY;
+                        goto exit;
+                }
+                dev->dmadata[i].dma_res = r;
+                ret = txx9aclc_dma_init(dev, &dev->dmadata[i]);
+                if (ret)
+                        goto exit;
+        }
+        return 0;
+
+exit:
+        for (i = 0; i < 2; i++) {
+                if (dev->dmadata[i].dma_chan)
+                        dma_release_channel(dev->dmadata[i].dma_chan);
+                dev->dmadata[i].dma_chan = NULL;
+        }
+        return ret;
+}
+
+static int txx9aclc_pcm_remove(struct platform_device *pdev)
+{
+        struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+        struct txx9aclc_soc_device *dev =
+                container_of(socdev, struct txx9aclc_soc_device, soc_dev);
+        struct txx9aclc_plat_drvdata *drvdata = txx9aclc_get_plat_drvdata(dev);
+        void __iomem *base = drvdata->base;
+        int i;
+
+        /* disable all FIFO DMAs */
+        __raw_writel(ACCTL_AUDODMA | ACCTL_AUDIDMA, base + ACCTLDIS);
+        /* dummy R/W to clear pending DMAREQ if any */
+        __raw_writel(__raw_readl(base + ACAUDIDAT), base + ACAUDODAT);
+
+        for (i = 0; i < 2; i++) {
+                struct txx9aclc_dmadata *dmadata = &dev->dmadata[i];
+                struct dma_chan *chan = dmadata->dma_chan;
+                if (chan) {
+                        dmadata->frag_count = -1;
+                        chan->device->device_terminate_all(chan);
+                        dma_release_channel(chan);
+                }
+                dev->dmadata[i].dma_chan = NULL;
+        }
+        return 0;
+}
+
+struct snd_soc_platform txx9aclc_soc_platform = {
+        .name           = "txx9aclc-audio",
+        .probe          = txx9aclc_pcm_probe,
+        .remove         = txx9aclc_pcm_remove,
+        .pcm_ops        = &txx9aclc_pcm_ops,
+        .pcm_new        = txx9aclc_pcm_new,
+        .pcm_free       = txx9aclc_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(txx9aclc_soc_platform);
+
+static int __init txx9aclc_soc_platform_init(void)
+{
+        return snd_soc_register_platform(&txx9aclc_soc_platform);
+}
+
+static void __exit txx9aclc_soc_platform_exit(void)
+{
+        snd_soc_unregister_platform(&txx9aclc_soc_platform);
+}
+
+module_init(txx9aclc_soc_platform_init);
+module_exit(txx9aclc_soc_platform_exit);
+
+MODULE_AUTHOR("Atsushi Nemoto <anemo@mba.ocn.ne.jp>");
+MODULE_DESCRIPTION("TXx9 ACLC Audio DMA driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/txx9/txx9aclc.h b/sound/soc/txx9/txx9aclc.h
new file mode 100644
index 000000000000..6769aab41b33
--- /dev/null
+++ b/sound/soc/txx9/txx9aclc.h
@@ -0,0 +1,83 @@
+/*
+ * TXx9 SoC AC Link Controller
+ *
+ * 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 __TXX9ACLC_H
+#define __TXX9ACLC_H
+
+#include <linux/interrupt.h>
+#include <asm/txx9/dmac.h>
+
+#define ACCTLEN                 0x00    /* control enable */
+#define ACCTLDIS                0x04    /* control disable */
+#define   ACCTL_ENLINK          0x00000001      /* enable/disable AC-link */
+#define   ACCTL_AUDODMA         0x00000100      /* AUDODMA enable/disable */
+#define   ACCTL_AUDIDMA         0x00001000      /* AUDIDMA enable/disable */
+#define   ACCTL_AUDOEHLT        0x00010000      /* AUDO error halt
+                                                   enable/disable */
+#define   ACCTL_AUDIEHLT        0x00100000      /* AUDI error halt
+                                                   enable/disable */
+#define ACREGACC                0x08    /* codec register access */
+#define   ACREGACC_DAT_SHIFT    0       /* data field */
+#define   ACREGACC_REG_SHIFT    16      /* address field */
+#define   ACREGACC_CODECID_SHIFT        24      /* CODEC ID field */
+#define   ACREGACC_READ         0x80000000      /* CODEC read */
+#define   ACREGACC_WRITE        0x00000000      /* CODEC write */
+#define ACINTSTS                0x10    /* interrupt status */
+#define ACINTMSTS               0x14    /* interrupt masked status */
+#define ACINTEN                 0x18    /* interrupt enable */
+#define ACINTDIS                0x1c    /* interrupt disable */
+#define   ACINT_CODECRDY(n)     (0x00000001 << (n))     /* CODECn ready */
+#define   ACINT_REGACCRDY       0x00000010      /* ACREGACC ready */
+#define   ACINT_AUDOERR         0x00000100      /* AUDO underrun error */
+#define   ACINT_AUDIERR         0x00001000      /* AUDI overrun error */
+#define ACDMASTS                0x80    /* DMA request status */
+#define   ACDMA_AUDO            0x00000001      /* AUDODMA pending */
+#define   ACDMA_AUDI            0x00000010      /* AUDIDMA pending */
+#define ACAUDODAT               0xa0    /* audio out data */
+#define ACAUDIDAT               0xb0    /* audio in data */
+#define ACREVID                 0xfc    /* revision ID */
+
+struct txx9aclc_dmadata {
+        struct resource *dma_res;
+        struct txx9dmac_slave dma_slave;
+        struct dma_chan *dma_chan;
+        struct tasklet_struct tasklet;
+        spinlock_t dma_lock;
+        int stream; /* SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE */
+        struct snd_pcm_substream *substream;
+        unsigned long pos;
+        dma_addr_t dma_addr;
+        unsigned long buffer_bytes;
+        unsigned long period_bytes;
+        unsigned long frag_bytes;
+        int frags;
+        int frag_count;
+        int dmacount;
+};
+
+struct txx9aclc_plat_drvdata {
+        void __iomem *base;
+        u64 physbase;
+};
+
+struct txx9aclc_soc_device {
+        struct snd_soc_device soc_dev;
+        struct platform_device *aclc_pdev;      /* for ioresources, drvdata */
+        struct txx9aclc_dmadata dmadata[2];
+};
+
+static inline struct txx9aclc_plat_drvdata *txx9aclc_get_plat_drvdata(
+        struct txx9aclc_soc_device *sdev)
+{
+        return platform_get_drvdata(sdev->aclc_pdev);
+}
+
+extern struct snd_soc_platform txx9aclc_soc_platform;
+extern struct snd_soc_dai txx9aclc_ac97_dai;
+
+#endif /* __TXX9ACLC_H */