Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (365 commits)
  ALSA: hda - Disable sticky PCM stream assignment for AD codecs
  ALSA: usb - Creative USB X-Fi volume knob support
  ALSA: ca0106: Use card specific dac id for mute controls.
  ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
  ALSA: ca0106: Create a nice spot for mapping channels to dacs.
  ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
  ALSA: ca0106: Pull out dac powering routine into separate function.
  ALSA: ca0106 - add Sound Blaster 5.1vx info.
  ASoC: tlv320dac33: Use usleep_range for delays
  ALSA: usb-audio: add Novation Launchpad support
  ALSA: hda - Add workarounds for CT-IBG controllers
  ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
  ASoC: tpa6130a2: Error handling for broken chip
  ASoC: max98088: Staticise m98088_eq_band
  ASoC: soc-core: Fix codec->name memory leak
  ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
  ALSA: hda - Add some workarounds for Creative IBG
  ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
  ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
  ALSA: hda - Add alc_init_jacks() call to other codecs
  ...

27 files changed, 784 insertions, 528 deletions

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index e30c8325f35e..37f191bbfdd9 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -117,6 +117,24 @@ config SND_PXA2XX_SOC_PALM27X
 	  Say Y if you want to add support for SoC audio on
 	  Palm T|X, T5, E2 or LifeDrive handheld computer.
 
+config SND_SOC_SAARB
+	tristate "SoC Audio support for Marvell Saarb"
+	depends on SND_PXA2XX_SOC && MACH_SAARB
+	select SND_PXA_SOC_SSP
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Saarb reference platform.
+
+config SND_SOC_TAVOREVB3
+	tristate "SoC Audio support for Marvell Tavor EVB3"
+	depends on SND_PXA2XX_SOC && MACH_TAVOREVB3
+	select SND_PXA_SOC_SSP
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Saarb reference platform.
+
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index caa03d8f4789..07660165ec8d 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -19,6 +19,8 @@ snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 snd-soc-palm27x-objs := palm27x.o
+snd-soc-saarb-objs := saarb.o
+snd-soc-tavorevb3-objs := tavorevb3.o
 snd-soc-zylonite-objs := zylonite.o
 snd-soc-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
@@ -38,6 +40,8 @@ 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_PXA2XX_SOC_Z2) += snd-soc-z2.o
+obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o
+obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
 obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
 obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index fefe1a57f31a..97e9423615c9 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -30,7 +30,6 @@
 #include <mach/audio.h>
 
 #include "../codecs/wm8731.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
 
 #define CORGI_HP        0
@@ -99,7 +98,7 @@ static void corgi_ext_control(struct snd_soc_codec *codec)
 static int corgi_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->card->codec;
+	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
 	corgi_ext_control(codec);
@@ -118,8 +117,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	unsigned int clk = 0;
 	int ret = 0;
 
@@ -150,7 +149,7 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -272,8 +271,9 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
 /*
  * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
  */
-static int corgi_wm8731_init(struct snd_soc_codec *codec)
+static int corgi_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
 	int err;
 
 	snd_soc_dapm_nc_pin(codec, "LLINEIN");
@@ -300,8 +300,10 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
 static struct snd_soc_dai_link corgi_dai = {
 	.name = "WM8731",
 	.stream_name = "WM8731",
-	.cpu_dai = &pxa_i2s_dai,
-	.codec_dai = &wm8731_dai,
+	.cpu_dai_name = "pxa-is2-dai",
+	.codec_dai_name = "wm8731-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8731-codec-0.001a",
 	.init = corgi_wm8731_init,
 	.ops = &corgi_ops,
 };
@@ -309,17 +311,10 @@ static struct snd_soc_dai_link corgi_dai = {
 /* corgi audio machine driver */
 static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = &corgi_dai,
 	.num_links = 1,
 };
 
-/* corgi audio subsystem */
-static struct snd_soc_device corgi_snd_devdata = {
-	.card = &snd_soc_corgi,
-	.codec_dev = &soc_codec_dev_wm8731,
-};
-
 static struct platform_device *corgi_snd_device;
 
 static int __init corgi_init(void)
@@ -334,8 +329,7 @@ static int __init corgi_init(void)
 	if (!corgi_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(corgi_snd_device, &corgi_snd_devdata);
-	corgi_snd_devdata.dev = &corgi_snd_device->dev;
+	platform_set_drvdata(corgi_snd_device, &snd_soc_corgi);
 	ret = platform_device_add(corgi_snd_device);
 
 	if (ret)
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 7cd2f89d7b10..c82cedb602fd 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -24,7 +24,6 @@
 #include <asm/mach-types.h>
 
 #include "../codecs/wm9705.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 
@@ -90,8 +89,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Mic Amp", NULL, "Mic (Internal)"},
 };
 
-static int e740_ac97_init(struct snd_soc_codec *codec)
+static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
+
 	snd_soc_dapm_nc_pin(codec, "HPOUTL");
 	snd_soc_dapm_nc_pin(codec, "HPOUTR");
 	snd_soc_dapm_nc_pin(codec, "PHONE");
@@ -116,30 +117,28 @@ static struct snd_soc_dai_link e740_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+		.cpu_dai_name = "pxa-ac97.0",
+		.codec_dai_name = "wm9705-hifi",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9705-codec",
 		.init = e740_ac97_init,
 	},
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+		.cpu_dai_name = "pxa-ac97.1",
+		.codec_dai_name = "wm9705-aux",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9705-codec",
 	},
 };
 
 static struct snd_soc_card e740 = {
 	.name = "Toshiba e740",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = e740_dai,
 	.num_links = ARRAY_SIZE(e740_dai),
 };
 
-static struct snd_soc_device e740_snd_devdata = {
-	.card = &e740,
-	.codec_dev = &soc_codec_dev_wm9705,
-};
-
 static struct platform_device *e740_snd_device;
 
 static int __init e740_init(void)
@@ -178,8 +177,7 @@ static int __init e740_init(void)
 		goto free_apwr_gpio;
 	}
 
-	platform_set_drvdata(e740_snd_device, &e740_snd_devdata);
-	e740_snd_devdata.dev = &e740_snd_device->dev;
+	platform_set_drvdata(e740_snd_device, &e740);
 	ret = platform_device_add(e740_snd_device);
 
 	if (!ret)
@@ -200,6 +198,9 @@ free_mic_amp_gpio:
 static void __exit e740_exit(void)
 {
 	platform_device_unregister(e740_snd_device);
+	gpio_free(GPIO_E740_WM9705_nAVDD2);
+	gpio_free(GPIO_E740_AMP_ON);
+	gpio_free(GPIO_E740_MIC_ON);
 }
 
 module_init(e740_init);
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 8dceccc5e059..4c143803a75e 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -24,7 +24,6 @@
 #include <asm/mach-types.h>
 
 #include "../codecs/wm9705.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 static int e750_spk_amp_event(struct snd_soc_dapm_widget *w,
@@ -72,8 +71,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"MIC1", NULL, "Mic (Internal)"},
 };
 
-static int e750_ac97_init(struct snd_soc_codec *codec)
+static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
+
 	snd_soc_dapm_nc_pin(codec, "LOUT");
 	snd_soc_dapm_nc_pin(codec, "ROUT");
 	snd_soc_dapm_nc_pin(codec, "PHONE");
@@ -98,31 +99,29 @@ static struct snd_soc_dai_link e750_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+		.cpu_dai_name = "pxa-ac97.0",
+		.codec_dai_name = "wm9705-hifi",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9705-codec",
 		.init = e750_ac97_init,
 		/* use ops to check startup state */
 	},
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+		.cpu_dai_name = "pxa-ac97.1",
+		.codec_dai_name ="wm9705-aux",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9705-codec",
 	},
 };
 
 static struct snd_soc_card e750 = {
 	.name = "Toshiba e750",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = e750_dai,
 	.num_links = ARRAY_SIZE(e750_dai),
 };
 
-static struct snd_soc_device e750_snd_devdata = {
-	.card = &e750,
-	.codec_dev = &soc_codec_dev_wm9705,
-};
-
 static struct platform_device *e750_snd_device;
 
 static int __init e750_init(void)
@@ -154,8 +153,7 @@ static int __init e750_init(void)
 		goto free_spk_amp_gpio;
 	}
 
-	platform_set_drvdata(e750_snd_device, &e750_snd_devdata);
-	e750_snd_devdata.dev = &e750_snd_device->dev;
+	platform_set_drvdata(e750_snd_device, &e750);
 	ret = platform_device_add(e750_snd_device);
 
 	if (!ret)
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index bc019cdce429..d42e5fe832c5 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -23,7 +23,6 @@
 #include <mach/eseries-gpio.h>
 
 #include "../codecs/wm9712.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
@@ -73,8 +72,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"MIC2", NULL, "Mic (Internal2)"},
 };
 
-static int e800_ac97_init(struct snd_soc_codec *codec)
+static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
+
 	snd_soc_dapm_new_controls(codec, e800_dapm_widgets,
 					ARRAY_SIZE(e800_dapm_widgets));
 
@@ -88,30 +89,28 @@ static struct snd_soc_dai_link e800_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+		.cpu_dai_name = "pxa-ac97.0",
+		.codec_dai_name = "wm9712-hifi",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9712-codec",
 		.init = e800_ac97_init,
 	},
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+		.cpu_dai_name = "pxa-ac97.1",
+		.codec_dai_name ="wm9712-aux",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9712-codec",
 	},
 };
 
 static struct snd_soc_card e800 = {
 	.name = "Toshiba e800",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = e800_dai,
 	.num_links = ARRAY_SIZE(e800_dai),
 };
 
-static struct snd_soc_device e800_snd_devdata = {
-	.card = &e800,
-	.codec_dev = &soc_codec_dev_wm9712,
-};
-
 static struct platform_device *e800_snd_device;
 
 static int __init e800_init(void)
@@ -141,8 +140,7 @@ static int __init e800_init(void)
 	if (!e800_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(e800_snd_device, &e800_snd_devdata);
-	e800_snd_devdata.dev = &e800_snd_device->dev;
+	platform_set_drvdata(e800_snd_device, &e800);
 	ret = platform_device_add(e800_snd_device);
 
 	if (!ret)
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index f4756e4025fd..eadf9d351a04 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -32,36 +32,33 @@
 #include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 static struct snd_soc_dai_link em_x270_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+		.cpu_dai_name = "pxa-ac97.0",
+		.codec_dai_name = "wm9712-hifi",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9712-codec",
 	},
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+		.cpu_dai_name = "pxa-ac97.1",
+		.codec_dai_name ="wm9712-aux",
+		.platform_name = "pxa-pcm-audio",
+		.codec_name = "wm9712-codec",
 	},
 };
 
 static struct snd_soc_card em_x270 = {
 	.name = "EM-X270",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = em_x270_dai,
 	.num_links = ARRAY_SIZE(em_x270_dai),
 };
 
-static struct snd_soc_device em_x270_snd_devdata = {
-	.card = &em_x270,
-	.codec_dev = &soc_codec_dev_wm9712,
-};
-
 static struct platform_device *em_x270_snd_device;
 
 static int __init em_x270_init(void)
@@ -76,8 +73,7 @@ static int __init em_x270_init(void)
 	if (!em_x270_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(em_x270_snd_device, &em_x270_snd_devdata);
-	em_x270_snd_devdata.dev = &em_x270_snd_device->dev;
+	platform_set_drvdata(em_x270_snd_device, &em_x270);
 	ret = platform_device_add(em_x270_snd_device);
 
 	if (ret)
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
index 405587a01160..154fc6f23438 100644
--- a/sound/soc/pxa/imote2.c
+++ b/sound/soc/pxa/imote2.c
@@ -6,14 +6,13 @@
 
 #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;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	unsigned int clk = 0;
 	int ret;
 
@@ -64,23 +63,19 @@ static struct snd_soc_ops imote2_asoc_ops = {
 static struct snd_soc_dai_link imote2_dai = {
 	.name = "WM8940",
 	.stream_name = "WM8940",
-	.cpu_dai = &pxa_i2s_dai,
-	.codec_dai = &wm8940_dai,
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "wm8940-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8940-codec.0-0034",
 	.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)
@@ -93,8 +88,7 @@ static int __init imote2_asoc_init(void)
 	if (!imote2_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(imote2_snd_device, &imote2_snd_devdata);
-	imote2_snd_devdata.dev = &imote2_snd_device->dev;
+	platform_set_drvdata(imote2_snd_device, &snd_soc_imote2);
 	ret = platform_device_add(imote2_snd_device);
 	if (ret)
 		platform_device_put(imote2_snd_device);
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 4c8d99a8d386..b8207ced4072 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -32,7 +32,6 @@
 #include <mach/magician.h>
 #include <asm/mach-types.h>
 #include "../codecs/uda1380.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
 #include "pxa-ssp.h"
 
@@ -71,7 +70,7 @@ static void magician_ext_control(struct snd_soc_codec *codec)
 static int magician_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->card->codec;
+	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
 	magician_ext_control(codec);
@@ -86,8 +85,8 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 				       struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	unsigned int acps, acds, width, rate;
 	unsigned int div4 = PXA_SSP_CLK_SCDB_4;
 	int ret = 0;
@@ -227,8 +226,8 @@ static int magician_capture_hw_params(struct snd_pcm_substream *substream,
 				      struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret = 0;
 
 	/* set codec DAI configuration */
@@ -393,8 +392,9 @@ static const struct snd_kcontrol_new uda1380_magician_controls[] = {
 /*
  * Logic for a uda1380 as connected on a HTC Magician
  */
-static int magician_uda1380_init(struct snd_soc_codec *codec)
+static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
 	int err;
 
 	/* NC codec pins */
@@ -427,16 +427,20 @@ static struct snd_soc_dai_link magician_dai[] = {
 {
 	.name = "uda1380",
 	.stream_name = "UDA1380 Playback",
-	.cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1],
-	.codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
+	.cpu_dai_name = "pxa-ssp-dai.0",
+	.codec_dai_name = "uda1380-hifi-playback",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "uda1380-codec.0-0018",
 	.init = magician_uda1380_init,
 	.ops = &magician_playback_ops,
 },
 {
 	.name = "uda1380",
 	.stream_name = "UDA1380 Capture",
-	.cpu_dai = &pxa_i2s_dai,
-	.codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "uda1380-hifi-capture",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "uda1380-codec.0-0018",
 	.ops = &magician_capture_ops,
 }
 };
@@ -446,13 +450,7 @@ static struct snd_soc_card snd_soc_card_magician = {
 	.name = "Magician",
 	.dai_link = magician_dai,
 	.num_links = ARRAY_SIZE(magician_dai),
-	.platform = &pxa2xx_soc_platform,
-};
 
-/* magician audio subsystem */
-static struct snd_soc_device magician_snd_devdata = {
-	.card = &snd_soc_card_magician,
-	.codec_dev = &soc_codec_dev_uda1380,
 };
 
 static struct platform_device *magician_snd_device;
@@ -514,8 +512,7 @@ static int __init magician_init(void)
 		goto err_pdev;
 	}
 
-	platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
-	magician_snd_devdata.dev = &magician_snd_device->dev;
+	platform_set_drvdata(magician_snd_device, &snd_soc_card_magician);
 	ret = platform_device_add(magician_snd_device);
 	if (ret) {
 		platform_device_put(magician_snd_device);
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 19eda8bbfdaf..f284cc54bc80 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -54,7 +54,6 @@
 #include <sound/initval.h>
 #include <sound/ac97_codec.h>
 
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 #include "../codecs/wm9713.h"
 
@@ -128,8 +127,9 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Rear Speaker", NULL, "SPKR"},
 };
 
-static int mioa701_wm9713_init(struct snd_soc_codec *codec)
+static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
 	unsigned short reg;
 
 	/* Add mioa701 specific widgets */
@@ -139,12 +139,12 @@ static int mioa701_wm9713_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map));
 
 	/* Prepare GPIO8 for rear speaker amplifier */
-	reg = codec->read(codec, AC97_GPIO_CFG);
-	codec->write(codec, AC97_GPIO_CFG, reg | 0x0100);
+	reg = codec->driver->read(codec, AC97_GPIO_CFG);
+	codec->driver->write(codec, AC97_GPIO_CFG, reg | 0x0100);
 
 	/* Prepare MIC input */
-	reg = codec->read(codec, AC97_3D_CONTROL);
-	codec->write(codec, AC97_3D_CONTROL, reg | 0xc000);
+	reg = codec->driver->read(codec, AC97_3D_CONTROL);
+	codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000);
 
 	snd_soc_dapm_enable_pin(codec, "Front Speaker");
 	snd_soc_dapm_enable_pin(codec, "Rear Speaker");
@@ -162,32 +162,30 @@ static struct snd_soc_dai_link mioa701_dai[] = {
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-		.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+		.cpu_dai_name = "pxa-ac97.0",
+		.codec_dai_name = "wm9713-hifi",
+		.codec_name = "wm9713-codec",
 		.init = mioa701_wm9713_init,
+		.platform_name = "pxa-pcm-audio",
 		.ops = &mioa701_ops,
 	},
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-		.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+		.cpu_dai_name = "pxa-ac97.1",
+		.codec_dai_name ="wm9713-aux",
+		.codec_name = "wm9713-codec",
+		.platform_name = "pxa-pcm-audio",
 		.ops = &mioa701_ops,
 	},
 };
 
 static struct snd_soc_card mioa701 = {
 	.name = "MioA701",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = mioa701_dai,
 	.num_links = ARRAY_SIZE(mioa701_dai),
 };
 
-static struct snd_soc_device mioa701_snd_devdata = {
-	.card = &mioa701,
-	.codec_dev = &soc_codec_dev_wm9713,
-};
-
 static struct platform_device *mioa701_snd_device;
 
 static int mioa701_wm9713_probe(struct platform_device *pdev)
@@ -205,8 +203,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev)
 	if (!mioa701_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(mioa701_snd_device, &mioa701_snd_devdata);
-	mioa701_snd_devdata.dev = &mioa701_snd_device->dev;
+	platform_set_drvdata(mioa701_snd_device, &mioa701);
 
 	ret = platform_device_add(mioa701_snd_device);
 	if (!ret)
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 1f96e3227be5..13f6d485d571 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -29,7 +29,6 @@
 #include <mach/palmasoc.h>
 
 #include "../codecs/wm9712.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 static struct snd_soc_jack hs_jack;
@@ -75,8 +74,9 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 static struct snd_soc_card palm27x_asoc;
 
-static int palm27x_ac97_init(struct snd_soc_codec *codec)
+static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
 	int err;
 
 	/* add palm27x specific widgets */
@@ -112,7 +112,7 @@ static int palm27x_ac97_init(struct snd_soc_codec *codec)
 		return err;
 
 	/* Jack detection API stuff */
-	err = snd_soc_jack_new(&palm27x_asoc, "Headphone Jack",
+	err = snd_soc_jack_new(codec, "Headphone Jack",
 				SND_JACK_HEADPHONE, &hs_jack);
 	if (err)
 		return err;
@@ -132,30 +132,28 @@ static struct snd_soc_dai_link palm27x_dai[] = {
 {
 	.name = "AC97 HiFi",
 	.stream_name = "AC97 HiFi",
-	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.cpu_dai_name = "pxa-ac97.0",
+	.codec_dai_name =  "wm9712-hifi",
+	.codec_name = "wm9712-codec",
+	.platform_name = "pxa-pcm-audio",
 	.init = palm27x_ac97_init,
 },
 {
 	.name = "AC97 Aux",
 	.stream_name = "AC97 Aux",
-	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	.cpu_dai_name = "pxa-ac97.1",
+	.codec_dai_name = "wm9712-aux",
+	.codec_name = "wm9712-codec",
+	.platform_name = "pxa-pcm-audio",
 },
 };
 
 static struct snd_soc_card palm27x_asoc = {
 	.name = "Palm/PXA27x",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = palm27x_dai,
 	.num_links = ARRAY_SIZE(palm27x_dai),
 };
 
-static struct snd_soc_device palm27x_snd_devdata = {
-	.card = &palm27x_asoc,
-	.codec_dev = &soc_codec_dev_wm9712,
-};
-
 static struct platform_device *palm27x_snd_device;
 
 static int palm27x_asoc_probe(struct platform_device *pdev)
@@ -178,8 +176,7 @@ static int palm27x_asoc_probe(struct platform_device *pdev)
 	if (!palm27x_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata);
-	palm27x_snd_devdata.dev = &palm27x_snd_device->dev;
+	platform_set_drvdata(palm27x_snd_device, &palm27x_asoc);
 	ret = platform_device_add(palm27x_snd_device);
 
 	if (ret != 0)
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index c5f36e0eab58..af84ee9c5e11 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -31,7 +31,6 @@
 #include <mach/audio.h>
 
 #include "../codecs/wm8731.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
 
 #define POODLE_HP        1
@@ -76,7 +75,7 @@ static void poodle_ext_control(struct snd_soc_codec *codec)
 static int poodle_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->card->codec;
+	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
 	poodle_ext_control(codec);
@@ -97,8 +96,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	unsigned int clk = 0;
 	int ret = 0;
 
@@ -129,7 +128,7 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -237,8 +236,9 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
 /*
  * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device
  */
-static int poodle_wm8731_init(struct snd_soc_codec *codec)
+static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
 	int err;
 
 	snd_soc_dapm_nc_pin(codec, "LLINEIN");
@@ -266,8 +266,10 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
 static struct snd_soc_dai_link poodle_dai = {
 	.name = "WM8731",
 	.stream_name = "WM8731",
-	.cpu_dai = &pxa_i2s_dai,
-	.codec_dai = &wm8731_dai,
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "wm8731-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8731-codec.0-001a",
 	.init = poodle_wm8731_init,
 	.ops = &poodle_ops,
 };
@@ -275,15 +277,9 @@ static struct snd_soc_dai_link poodle_dai = {
 /* poodle audio machine driver */
 static struct snd_soc_card snd_soc_poodle = {
 	.name = "Poodle",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = &poodle_dai,
 	.num_links = 1,
-};
-
-/* poodle audio subsystem */
-static struct snd_soc_device poodle_snd_devdata = {
-	.card = &snd_soc_poodle,
-	.codec_dev = &soc_codec_dev_wm8731,
+	.owner = THIS_MODULE,
 };
 
 static struct platform_device *poodle_snd_device;
@@ -307,8 +303,7 @@ static int __init poodle_init(void)
 	if (!poodle_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(poodle_snd_device, &poodle_snd_devdata);
-	poodle_snd_devdata.dev = &poodle_snd_device->dev;
+	platform_set_drvdata(poodle_snd_device, &snd_soc_poodle);
 	ret = platform_device_add(poodle_snd_device);
 
 	if (ret)
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index a1fd23e0e3d0..b439eee462cb 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -35,7 +35,7 @@
 #include <mach/audio.h>
 #include <plat/ssp.h>
 
-#include "pxa2xx-pcm.h"
+#include "../../arm/pxa2xx-pcm.h"
 #include "pxa-ssp.h"
 
 /*
@@ -108,11 +108,9 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out)
 }
 
 static int pxa_ssp_startup(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
+			   struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	int ret = 0;
 
@@ -128,11 +126,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 }
 
 static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
+			     struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 
 	if (!cpu_dai->active) {
@@ -148,7 +144,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 
 static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
 {
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 
 	if (!cpu_dai->active)
@@ -166,7 +162,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai)
 
 static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
 {
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE;
 
@@ -230,7 +226,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
 static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	int clk_id, unsigned int freq, int dir)
 {
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	int val;
 
@@ -287,7 +283,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
 	int div_id, int div)
 {
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	int val;
 
@@ -338,7 +334,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
 static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 	int source, unsigned int freq_in, unsigned int freq_out)
 {
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
 
@@ -407,7 +403,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
 	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
 {
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	u32 sscr0;
 
@@ -442,7 +438,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
 static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
 	int tristate)
 {
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	u32 sscr1;
 
@@ -464,11 +460,9 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai,
 static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		unsigned int fmt)
 {
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
-	u32 sscr0;
-	u32 sscr1;
-	u32 sspsp;
+	u32 sscr0, sscr1, sspsp, scfr;
 
 	/* check if we need to change anything at all */
 	if (priv->dai_fmt == fmt)
@@ -483,16 +477,16 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 
 	/* reset port settings */
 	sscr0 = pxa_ssp_read_reg(ssp, SSCR0) &
-		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 	sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
 	sspsp = 0;
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
-		sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR;
+		sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFS:
-		sscr1 |= SSCR1_SCLKDIR;
+		sscr1 |= SSCR1_SCLKDIR | SSCR1_SCFR;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
 		break;
@@ -538,6 +532,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	pxa_ssp_write_reg(ssp, SSCR1, sscr1);
 	pxa_ssp_write_reg(ssp, SSPSP, sspsp);
 
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
+		scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR;
+		pxa_ssp_write_reg(ssp, SSCR1, scfr);
+
+		while (pxa_ssp_read_reg(ssp, SSSR) & SSSR_BSY)
+			cpu_relax();
+		break;
+	}
+
 	dump_registers(ssp);
 
 	/* Since we are configuring the timings for the format by hand
@@ -555,11 +560,9 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
  */
 static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
-				struct snd_soc_dai *dai)
+				struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	int chn = params_channels(params);
 	u32 sscr0;
@@ -568,7 +571,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
 	struct pxa2xx_pcm_dma_params *dma_data;
 
-	dma_data = snd_soc_dai_get_dma_data(dai, substream);
+	dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
 
 	/* generate correct DMA params */
 	kfree(dma_data);
@@ -581,7 +584,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 			((chn == 2) && (ttsa != 1)) || (width == 32),
 			substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
-	snd_soc_dai_set_dma_data(dai, substream, dma_data);
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
 
 	/* we can only change the settings if the port is not in use */
 	if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE)
@@ -589,10 +592,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 
 	/* clear selected SSP bits */
 	sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS);
-	pxa_ssp_write_reg(ssp, SSCR0, sscr0);
 
 	/* bit size */
-	sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 #ifdef CONFIG_PXA3xx
@@ -668,12 +669,10 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
-			   struct snd_soc_dai *dai)
+			   struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	int ret = 0;
-	struct ssp_priv *priv = cpu_dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
 	struct ssp_device *ssp = priv->ssp;
 	int val;
 
@@ -729,8 +728,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
-static int pxa_ssp_probe(struct platform_device *pdev,
-			    struct snd_soc_dai *dai)
+static int pxa_ssp_probe(struct snd_soc_dai *dai)
 {
 	struct ssp_priv *priv;
 	int ret;
@@ -746,7 +744,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
 	}
 
 	priv->dai_fmt = (unsigned int) -1;
-	dai->private_data = priv;
+	snd_soc_dai_set_drvdata(dai, priv);
 
 	return 0;
 
@@ -755,11 +753,13 @@ err_priv:
 	return ret;
 }
 
-static void pxa_ssp_remove(struct platform_device *pdev,
-			      struct snd_soc_dai *dai)
+static int pxa_ssp_remove(struct snd_soc_dai *dai)
 {
-	struct ssp_priv *priv = dai->private_data;
+	struct ssp_priv *priv = snd_soc_dai_get_drvdata(dai);
+
 	pxa_ssp_free(priv->ssp);
+	kfree(priv);
+	return 0;
 }
 
 #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
@@ -784,10 +784,7 @@ static struct snd_soc_dai_ops pxa_ssp_dai_ops = {
 	.set_tristate	= pxa_ssp_set_dai_tristate,
 };
 
-struct snd_soc_dai pxa_ssp_dai[] = {
-	{
-		.name = "pxa2xx-ssp1",
-		.id = 0,
+static struct snd_soc_dai_driver pxa_ssp_dai = {
 		.probe = pxa_ssp_probe,
 		.remove = pxa_ssp_remove,
 		.suspend = pxa_ssp_suspend,
@@ -805,81 +802,38 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.formats = PXA_SSP_FORMATS,
 		 },
 		.ops = &pxa_ssp_dai_ops,
+};
+
+static __devinit int asoc_ssp_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_dai(&pdev->dev, &pxa_ssp_dai);
+}
+
+static int __devexit asoc_ssp_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver asoc_ssp_driver = {
+	.driver = {
+			.name = "pxa-ssp-dai",
+			.owner = THIS_MODULE,
 	},
-	{	.name = "pxa2xx-ssp2",
-		.id = 1,
-		.probe = pxa_ssp_probe,
-		.remove = pxa_ssp_remove,
-		.suspend = pxa_ssp_suspend,
-		.resume = pxa_ssp_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 8,
-			.rates = PXA_SSP_RATES,
-			.formats = PXA_SSP_FORMATS,
-		},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 8,
-			.rates = PXA_SSP_RATES,
-			.formats = PXA_SSP_FORMATS,
-		 },
-		.ops = &pxa_ssp_dai_ops,
-	},
-	{
-		.name = "pxa2xx-ssp3",
-		.id = 2,
-		.probe = pxa_ssp_probe,
-		.remove = pxa_ssp_remove,
-		.suspend = pxa_ssp_suspend,
-		.resume = pxa_ssp_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 8,
-			.rates = PXA_SSP_RATES,
-			.formats = PXA_SSP_FORMATS,
-		},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 8,
-			.rates = PXA_SSP_RATES,
-			.formats = PXA_SSP_FORMATS,
-		 },
-		.ops = &pxa_ssp_dai_ops,
-	},
-	{
-		.name = "pxa2xx-ssp4",
-		.id = 3,
-		.probe = pxa_ssp_probe,
-		.remove = pxa_ssp_remove,
-		.suspend = pxa_ssp_suspend,
-		.resume = pxa_ssp_resume,
-		.playback = {
-			.channels_min = 1,
-			.channels_max = 8,
-			.rates = PXA_SSP_RATES,
-			.formats = PXA_SSP_FORMATS,
-		},
-		.capture = {
-			.channels_min = 1,
-			.channels_max = 8,
-			.rates = PXA_SSP_RATES,
-			.formats = PXA_SSP_FORMATS,
-		 },
-		.ops = &pxa_ssp_dai_ops,
-	},
+
+	.probe = asoc_ssp_probe,
+	.remove = __devexit_p(asoc_ssp_remove),
 };
-EXPORT_SYMBOL_GPL(pxa_ssp_dai);
 
 static int __init pxa_ssp_init(void)
 {
-	return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+	return platform_driver_register(&asoc_ssp_driver);
 }
 module_init(pxa_ssp_init);
 
 static void __exit pxa_ssp_exit(void)
 {
-	snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai));
+	platform_driver_unregister(&asoc_ssp_driver);
 }
 module_exit(pxa_ssp_exit);
 
diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h
index 91deadd55675..bc79da221c0d 100644
--- a/sound/soc/pxa/pxa-ssp.h
+++ b/sound/soc/pxa/pxa-ssp.h
@@ -42,6 +42,4 @@
 
 #define PXA_SSP_PLL_OUT  0
 
-extern struct snd_soc_dai pxa_ssp_dai[4];
-
 #endif
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index d314115e3dd7..ac51c6d25c42 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -24,7 +24,6 @@
 #include <mach/dma.h>
 #include <mach/audio.h>
 
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
@@ -104,24 +103,21 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
 #define pxa2xx_ac97_resume	NULL
 #endif
 
-static int pxa2xx_ac97_probe(struct platform_device *pdev,
-			     struct snd_soc_dai *dai)
+static int pxa2xx_ac97_probe(struct snd_soc_dai *dai)
 {
 	return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev));
 }
 
-static void pxa2xx_ac97_remove(struct platform_device *pdev,
-			       struct snd_soc_dai *dai)
+static int pxa2xx_ac97_remove(struct snd_soc_dai *dai)
 {
 	pxa2xx_ac97_hw_remove(to_platform_device(dai->dev));
+	return 0;
 }
 
 static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *params,
-				 struct snd_soc_dai *dai)
+				 struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct pxa2xx_pcm_dma_params *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -136,10 +132,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
 
 static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 				     struct snd_pcm_hw_params *params,
-				     struct snd_soc_dai *dai)
+				     struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct pxa2xx_pcm_dma_params *dma_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -154,11 +148,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
 
 static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 				     struct snd_pcm_hw_params *params,
-				     struct snd_soc_dai *dai)
+				     struct snd_soc_dai *cpu_dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
-
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		return -ENODEV;
 	else
@@ -188,10 +179,9 @@ static struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
  * There is only 1 physical AC97 interface for pxa2xx, but it
  * has extra fifo's that can be used for aux DACs and ADCs.
  */
-struct snd_soc_dai pxa_ac97_dai[] = {
+static struct snd_soc_dai_driver pxa_ac97_dai[] = {
 {
 	.name = "pxa2xx-ac97",
-	.id = 0,
 	.ac97_control = 1,
 	.probe = pxa2xx_ac97_probe,
 	.remove = pxa2xx_ac97_remove,
@@ -213,7 +203,6 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 },
 {
 	.name = "pxa2xx-ac97-aux",
-	.id = 1,
 	.ac97_control = 1,
 	.playback = {
 		.stream_name = "AC97 Aux Playback",
@@ -231,7 +220,6 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 },
 {
 	.name = "pxa2xx-ac97-mic",
-	.id = 2,
 	.ac97_control = 1,
 	.capture = {
 		.stream_name = "AC97 Mic Capture",
@@ -243,36 +231,26 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 },
 };
 
-EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
-static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev)
+static __devinit int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
-	int i;
-	pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data;
-
-	if (pdev->id >= 0) {
+	if (pdev->id != -1) {
 		dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
 		return -ENXIO;
 	}
 
-	for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++) {
-		pxa_ac97_dai[i].dev = &pdev->dev;
-		if (pdata && pdata->codec_pdata[0])
-			pxa_ac97_dai[i].ac97_pdata = pdata->codec_pdata[0];
-	}
-
 	/* Punt most of the init to the SoC probe; we may need the machine
 	 * driver to do interesting things with the clocking to get us up
 	 * and running.
 	 */
-	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+	return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai,
+			ARRAY_SIZE(pxa_ac97_dai));
 }
 
 static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
-
+	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai));
 	return 0;
 }
 
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
index e390de8edcd4..eda891e6f31b 100644
--- a/sound/soc/pxa/pxa2xx-ac97.h
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -14,8 +14,6 @@
 #define PXA2XX_DAI_AC97_AUX		1
 #define PXA2XX_DAI_AC97_MIC		2
 
-extern struct snd_soc_dai pxa_ac97_dai[3];
-
 /* platform data */
 extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index c1a5275721e4..11be5952a506 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -27,7 +27,6 @@
 #include <mach/dma.h>
 #include <mach/audio.h>
 
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
 
 /*
@@ -80,6 +79,7 @@ struct pxa_i2s_port {
 };
 static struct pxa_i2s_port pxa_i2s;
 static struct clk *clk_i2s;
+static int clk_ena = 0;
 
 static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
 	.name			= "I2S PCM Stereo out",
@@ -101,7 +101,7 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 
 	if (IS_ERR(clk_i2s))
 		return PTR_ERR(clk_i2s);
@@ -162,13 +162,11 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct pxa2xx_pcm_dma_params *dma_data;
 
 	BUG_ON(IS_ERR(clk_i2s));
 	clk_enable(clk_i2s);
-	dai->private_data = dai;
+	clk_ena = 1;
 	pxa_i2s_wait();
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -176,7 +174,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	else
 		dma_data = &pxa2xx_i2s_pcm_stereo_in;
 
-	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
+	snd_soc_dai_set_dma_data(dai, substream, dma_data);
 
 	/* is port used by another stream */
 	if (!(SACR0 & SACR0_ENB)) {
@@ -259,9 +257,9 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream,
 	if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) {
 		SACR0 &= ~SACR0_ENB;
 		pxa_i2s_wait();
-		if (dai->private_data != NULL) {
+		if (clk_ena) {
 			clk_disable(clk_i2s);
-			dai->private_data = NULL;
+			clk_ena = 0;
 		}
 	}
 }
@@ -300,6 +298,35 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 #define pxa2xx_i2s_resume	NULL
 #endif
 
+static int pxa2xx_i2s_probe(struct snd_soc_dai *dai)
+{
+	clk_i2s = clk_get(dai->dev, "I2SCLK");
+	if (IS_ERR(clk_i2s))
+		return PTR_ERR(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 0;
+}
+
+static int  pxa2xx_i2s_remove(struct snd_soc_dai *dai)
+{
+	clk_put(clk_i2s);
+	clk_i2s = ERR_PTR(-ENOENT);
+	return 0;
+}
+
 #define PXA2XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
@@ -313,9 +340,9 @@ static struct snd_soc_dai_ops pxa_i2s_dai_ops = {
 	.set_sysclk	= pxa2xx_i2s_set_dai_sysclk,
 };
 
-struct snd_soc_dai pxa_i2s_dai = {
-	.name = "pxa2xx-i2s",
-	.id = 0,
+static struct snd_soc_dai_driver pxa_i2s_dai = {
+	.probe = pxa2xx_i2s_probe,
+	.remove = pxa2xx_i2s_remove,
 	.suspend = pxa2xx_i2s_suspend,
 	.resume = pxa2xx_i2s_resume,
 	.playback = {
@@ -332,49 +359,20 @@ struct snd_soc_dai pxa_i2s_dai = {
 	.symmetric_rates = 1,
 };
 
-EXPORT_SYMBOL_GPL(pxa_i2s_dai);
-
-static int pxa2xx_i2s_probe(struct platform_device *dev)
+static int pxa2xx_i2s_drv_probe(struct platform_device *pdev)
 {
-	int ret;
-
-	clk_i2s = clk_get(&dev->dev, "I2SCLK");
-	if (IS_ERR(clk_i2s))
-		return PTR_ERR(clk_i2s);
-
-	pxa_i2s_dai.dev = &dev->dev;
-	pxa_i2s_dai.private_data = NULL;
-	ret = snd_soc_register_dai(&pxa_i2s_dai);
-	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;
+	return snd_soc_register_dai(&pdev->dev, &pxa_i2s_dai);
 }
 
-static int __devexit pxa2xx_i2s_remove(struct platform_device *dev)
+static int __devexit pxa2xx_i2s_drv_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_dai(&pxa_i2s_dai);
-	clk_put(clk_i2s);
-	clk_i2s = ERR_PTR(-ENOENT);
+	snd_soc_unregister_dai(&pdev->dev);
 	return 0;
 }
 
 static struct platform_driver pxa2xx_i2s_driver = {
-	.probe = pxa2xx_i2s_probe,
-	.remove = __devexit_p(pxa2xx_i2s_remove),
+	.probe = pxa2xx_i2s_drv_probe,
+	.remove = __devexit_p(pxa2xx_i2s_drv_remove),
 
 	.driver = {
 		.name = "pxa2xx-i2s",
@@ -400,3 +398,4 @@ module_exit(pxa2xx_i2s_exit);
 MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
 MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa2xx-i2s");
diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h
index e2def441153e..070f3c6059fe 100644
--- a/sound/soc/pxa/pxa2xx-i2s.h
+++ b/sound/soc/pxa/pxa2xx-i2s.h
@@ -15,6 +15,4 @@
 /* I2S clock */
 #define PXA2XX_I2S_SYSCLK		0
 
-extern struct snd_soc_dai pxa_i2s_dai;
-
 #endif
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index adc7e6f15f93..02fb66416ddc 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -16,7 +16,6 @@
 #include <sound/soc.h>
 #include <sound/pxa2xx-lib.h>
 
-#include "pxa2xx-pcm.h"
 #include "../../arm/pxa2xx-pcm.h"
 
 static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
@@ -28,7 +27,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	struct pxa2xx_pcm_dma_params *dma;
 	int ret;
 
-	dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+	dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
@@ -95,14 +94,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
-	if (dai->playback.channels_min) {
+	if (dai->driver->playback.channels_min) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->capture.channels_min) {
+	if (dai->driver->capture.channels_min) {
 		ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
@@ -112,25 +111,44 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	return ret;
 }
 
-struct snd_soc_platform pxa2xx_soc_platform = {
-	.name		= "pxa2xx-audio",
-	.pcm_ops 	= &pxa2xx_pcm_ops,
+static struct snd_soc_platform_driver pxa2xx_soc_platform = {
+	.ops 	= &pxa2xx_pcm_ops,
 	.pcm_new	= pxa2xx_soc_pcm_new,
 	.pcm_free	= pxa2xx_pcm_free_dma_buffers,
 };
-EXPORT_SYMBOL_GPL(pxa2xx_soc_platform);
 
-static int __init pxa2xx_soc_platform_init(void)
+static int __devinit pxa2xx_soc_platform_probe(struct platform_device *pdev)
 {
-	return snd_soc_register_platform(&pxa2xx_soc_platform);
+	return snd_soc_register_platform(&pdev->dev, &pxa2xx_soc_platform);
 }
-module_init(pxa2xx_soc_platform_init);
 
-static void __exit pxa2xx_soc_platform_exit(void)
+static int __devexit pxa2xx_soc_platform_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_platform(&pxa2xx_soc_platform);
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver pxa_pcm_driver = {
+	.driver = {
+			.name = "pxa-pcm-audio",
+			.owner = THIS_MODULE,
+	},
+
+	.probe = pxa2xx_soc_platform_probe,
+	.remove = __devexit_p(pxa2xx_soc_platform_remove),
+};
+
+static int __init snd_pxa_pcm_init(void)
+{
+	return platform_driver_register(&pxa_pcm_driver);
+}
+module_init(snd_pxa_pcm_init);
+
+static void __exit snd_pxa_pcm_exit(void)
+{
+	platform_driver_unregister(&pxa_pcm_driver);
 }
-module_exit(pxa2xx_soc_platform_exit);
+module_exit(snd_pxa_pcm_exit);
 
 MODULE_AUTHOR("Nicolas Pitre");
 MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module");
diff --git a/sound/soc/pxa/pxa2xx-pcm.h b/sound/soc/pxa/pxa2xx-pcm.h
deleted file mode 100644
index 60c3b20aeeb4..000000000000
--- a/sound/soc/pxa/pxa2xx-pcm.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/*
- * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
- *
- * Author:	Nicolas Pitre
- * Created:	Nov 30, 2004
- * Copyright:	MontaVista Software, Inc.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef _PXA2XX_PCM_H
-#define _PXA2XX_PCM_H
-
-/* platform data */
-extern struct snd_soc_platform pxa2xx_soc_platform;
-
-#endif
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index 7e3f41696c41..2cda82bc5d2e 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -26,9 +26,6 @@
 
 #include <asm/mach-types.h>
 
-#include "../codecs/cs4270.h"
-#include "../codecs/ak4104.h"
-#include "pxa2xx-pcm.h"
 #include "pxa-ssp.h"
 
 #define GPIO_SPDIF_RESET	(38)
@@ -71,7 +68,7 @@ static void raumfeld_enable_audio(bool en)
 static int raumfeld_cs4270_startup(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 *codec_dai = rtd->codec_dai;
 
 	/* set freq to 0 to enable all possible codec sample rates */
 	return snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
@@ -80,7 +77,7 @@ static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream)
 static void raumfeld_cs4270_shutdown(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 *codec_dai = rtd->codec_dai;
 
 	/* set freq to 0 to enable all possible codec sample rates */
 	snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0);
@@ -90,8 +87,8 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream,
 				     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	unsigned int fmt, clk = 0;
 	int ret = 0;
 
@@ -167,32 +164,14 @@ static int raumfeld_line_resume(struct platform_device *pdev)
 	return 0;
 }
 
-static struct snd_soc_dai_link raumfeld_line_dai = {
-	.name		= "CS4270",
-	.stream_name	= "CS4270",
-	.cpu_dai	= &pxa_ssp_dai[PXA_DAI_SSP1],
-	.codec_dai	= &cs4270_dai,
-	.ops		= &raumfeld_cs4270_ops,
-};
-
-static struct snd_soc_card snd_soc_line_raumfeld = {
-	.name		= "Raumfeld analog",
-	.platform	= &pxa2xx_soc_platform,
-	.dai_link	= &raumfeld_line_dai,
-	.suspend_post	= raumfeld_line_suspend,
-	.resume_pre	= raumfeld_line_resume,
-	.num_links	= 1,
-};
-
-
 /* AK4104 */
 
 static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream,
 				     struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int fmt, ret = 0, clk = 0;
 
 	switch (params_rate(params)) {
@@ -247,34 +226,35 @@ static struct snd_soc_ops raumfeld_ak4104_ops = {
 	.hw_params = raumfeld_ak4104_hw_params,
 };
 
-static struct snd_soc_dai_link raumfeld_spdif_dai = {
+static struct snd_soc_dai_link raumfeld_dai[] = {
+{
 	.name		= "ak4104",
 	.stream_name	= "Playback",
-	.cpu_dai	= &pxa_ssp_dai[PXA_DAI_SSP2],
-	.codec_dai	= &ak4104_dai,
+	.cpu_dai_name = "pxa-ssp-dai.1",
+	.codec_dai_name = "ak4104-hifi",
+	.platform_name = "pxa-pcm-audio",
 	.ops		= &raumfeld_ak4104_ops,
-};
-
-static struct snd_soc_card snd_soc_spdif_raumfeld = {
-	.name		= "Raumfeld S/PDIF",
-	.platform	= &pxa2xx_soc_platform,
-	.dai_link	= &raumfeld_spdif_dai,
-	.num_links	= 1
-};
-
-/* raumfeld_audio audio subsystem */
-static struct snd_soc_device raumfeld_line_devdata = {
-	.card = &snd_soc_line_raumfeld,
-	.codec_dev = &soc_codec_device_cs4270,
-};
+	.codec_name = "ak4104-codec.0",
+},
+{
+	.name		= "CS4270",
+	.stream_name	= "CS4270",
+	.cpu_dai_name = "pxa-ssp-dai.0",
+	.platform_name = "pxa-pcm-audio",
+	.codec_dai_name = "cs4270-hifi",
+	.codec_name = "cs4270-codec.0-0048",
+	.ops		= &raumfeld_cs4270_ops,
+},};
 
-static struct snd_soc_device raumfeld_spdif_devdata = {
-	.card = &snd_soc_spdif_raumfeld,
-	.codec_dev = &soc_codec_device_ak4104,
+static struct snd_soc_card snd_soc_raumfeld = {
+	.name		= "Raumfeld",
+	.dai_link	= raumfeld_dai,
+	.suspend_post	= raumfeld_line_suspend,
+	.resume_pre	= raumfeld_line_resume,
+	.num_links	= ARRAY_SIZE(raumfeld_dai),
 };
 
-static struct platform_device *raumfeld_audio_line_device;
-static struct platform_device *raumfeld_audio_spdif_device;
+static struct platform_device *raumfeld_audio_device;
 
 static int __init raumfeld_audio_init(void)
 {
@@ -292,38 +272,19 @@ static int __init raumfeld_audio_init(void)
 
 	set_max9485_clk(MAX9485_MCLK_FREQ_122880);
 
-	/* LINE */
-	raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0);
-	if (!raumfeld_audio_line_device)
+	/* Register LINE and SPDIF */
+	raumfeld_audio_device = platform_device_alloc("soc-audio", 0);
+	if (!raumfeld_audio_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(raumfeld_audio_line_device,
-			     &raumfeld_line_devdata);
-	raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev;
-	ret = platform_device_add(raumfeld_audio_line_device);
-	if (ret)
-		platform_device_put(raumfeld_audio_line_device);
+	platform_set_drvdata(raumfeld_audio_device,
+			     &snd_soc_raumfeld);
+	ret = platform_device_add(raumfeld_audio_device);
 
 	/* no S/PDIF on Speakers */
 	if (machine_is_raumfeld_speaker())
 		return ret;
 
-	/* S/PDIF */
-	raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1);
-	if (!raumfeld_audio_spdif_device) {
-		platform_device_put(raumfeld_audio_line_device);
-		return -ENOMEM;
-	}
-
-	platform_set_drvdata(raumfeld_audio_spdif_device,
-			     &raumfeld_spdif_devdata);
-	raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev;
-	ret = platform_device_add(raumfeld_audio_spdif_device);
-	if (ret) {
-		platform_device_put(raumfeld_audio_line_device);
-		platform_device_put(raumfeld_audio_spdif_device);
-	}
-
 	raumfeld_enable_audio(true);
 
 	return ret;
@@ -333,10 +294,7 @@ static void __exit raumfeld_audio_exit(void)
 {
 	raumfeld_enable_audio(false);
 
-	platform_device_unregister(raumfeld_audio_line_device);
-
-	if (machine_is_raumfeld_connector())
-		platform_device_unregister(raumfeld_audio_spdif_device);
+	platform_device_unregister(raumfeld_audio_device);
 
 	i2c_unregister_device(max9486_client);
 
diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c
new file mode 100644
index 000000000000..d63cb474b4e1
--- /dev/null
+++ b/sound/soc/pxa/saarb.c
@@ -0,0 +1,200 @@
+/*
+ * saarb.c -- SoC audio for saarb
+ *
+ * Copyright (C) 2010 Marvell International Ltd.
+ * 	Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/88pm860x-codec.h"
+#include "pxa-ssp.h"
+
+static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd);
+
+static struct platform_device *saarb_snd_device;
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* saarb machine dapm widgets */
+static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* saarb machine audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int saarb_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int width = snd_pcm_format_physical_width(params_format(params));
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
+				     PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
+
+	return ret;
+}
+
+static struct snd_soc_ops saarb_i2s_ops = {
+	.hw_params	= saarb_i2s_hw_params,
+};
+
+static struct snd_soc_dai_link saarb_dai[] = {
+	{
+		.name		= "88PM860x I2S",
+		.stream_name	= "I2S Audio",
+		.cpu_dai_name	= "pxa-ssp-dai.1",
+		.codec_dai_name	= "88pm860x-i2s",
+		.platform_name	= "pxa-pcm-audio",
+		.codec_name	= "88pm860x-codec",
+		.init		= saarb_pm860x_init,
+		.ops		= &saarb_i2s_ops,
+	},
+};
+
+static struct snd_soc_card snd_soc_card_saarb = {
+	.name = "Saarb",
+	.dai_link = saarb_dai,
+	.num_links = ARRAY_SIZE(saarb_dai),
+};
+
+static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret;
+
+	snd_soc_dapm_new_controls(codec, saarb_dapm_widgets,
+				  ARRAY_SIZE(saarb_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Speaker");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+	return 0;
+}
+
+static int __init saarb_init(void)
+{
+	int ret;
+
+	if (!machine_is_saarb())
+		return -ENODEV;
+	saarb_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!saarb_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb);
+
+	ret = platform_device_add(saarb_snd_device);
+	if (ret)
+		platform_device_put(saarb_snd_device);
+
+	return ret;
+}
+
+static void __exit saarb_exit(void)
+{
+	platform_device_unregister(saarb_snd_device);
+}
+
+module_init(saarb_init);
+module_exit(saarb_exit);
+
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d256f5f313b5..f470f360f4dd 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -28,7 +28,6 @@
 #include <asm/mach-types.h>
 #include <mach/spitz.h>
 #include "../codecs/wm8750.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
 
 #define SPITZ_HP        0
@@ -107,7 +106,7 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
 static int spitz_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->card->codec;
+	struct snd_soc_codec *codec = rtd->codec;
 
 	/* check the jack status at stream startup */
 	spitz_ext_control(codec);
@@ -118,8 +117,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	unsigned int clk = 0;
 	int ret = 0;
 
@@ -274,8 +273,9 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
 /*
  * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device
  */
-static int spitz_wm8750_init(struct snd_soc_codec *codec)
+static int spitz_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
 	int err;
 
 	/* NC codec pins */
@@ -308,8 +308,10 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 static struct snd_soc_dai_link spitz_dai = {
 	.name = "wm8750",
 	.stream_name = "WM8750",
-	.cpu_dai = &pxa_i2s_dai,
-	.codec_dai = &wm8750_dai,
+	.cpu_dai_name = "pxa-is2",
+	.codec_dai_name = "wm8750-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm8750-codec.0-001a",
 	.init = spitz_wm8750_init,
 	.ops = &spitz_ops,
 };
@@ -317,17 +319,10 @@ static struct snd_soc_dai_link spitz_dai = {
 /* spitz audio machine driver */
 static struct snd_soc_card snd_soc_spitz = {
 	.name = "Spitz",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = &spitz_dai,
 	.num_links = 1,
 };
 
-/* spitz audio subsystem */
-static struct snd_soc_device spitz_snd_devdata = {
-	.card = &snd_soc_spitz,
-	.codec_dev = &soc_codec_dev_wm8750,
-};
-
 static struct platform_device *spitz_snd_device;
 
 static int __init spitz_init(void)
@@ -341,8 +336,7 @@ static int __init spitz_init(void)
 	if (!spitz_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(spitz_snd_device, &spitz_snd_devdata);
-	spitz_snd_devdata.dev = &spitz_snd_device->dev;
+	platform_set_drvdata(spitz_snd_device, &snd_soc_spitz);
 	ret = platform_device_add(spitz_snd_device);
 
 	if (ret)
diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c
new file mode 100644
index 000000000000..248c283fc4df
--- /dev/null
+++ b/sound/soc/pxa/tavorevb3.c
@@ -0,0 +1,200 @@
+/*
+ * tavorevb3.c -- SoC audio for Tavor EVB3
+ *
+ * Copyright (C) 2010 Marvell International Ltd.
+ * 	Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/88pm860x-codec.h"
+#include "pxa-ssp.h"
+
+static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd);
+
+static struct platform_device *evb3_snd_device;
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* tavorevb3 machine dapm widgets */
+static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* tavorevb3 machine audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int evb3_i2s_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int width = snd_pcm_format_physical_width(params_format(params));
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
+				     PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
+	return ret;
+}
+
+static struct snd_soc_ops evb3_i2s_ops = {
+	.hw_params	= evb3_i2s_hw_params,
+};
+
+static struct snd_soc_dai_link evb3_dai[] = {
+	{
+		.name		= "88PM860x I2S",
+		.stream_name	= "I2S Audio",
+		.cpu_dai_name	= "pxa-ssp-dai.1",
+		.codec_dai_name	= "88pm860x-i2s",
+		.platform_name	= "pxa-pcm-audio",
+		.codec_name	= "88pm860x-codec",
+		.init		= evb3_pm860x_init,
+		.ops		= &evb3_i2s_ops,
+	},
+};
+
+static struct snd_soc_card snd_soc_card_evb3 = {
+	.name = "Tavor EVB3",
+	.dai_link = evb3_dai,
+	.num_links = ARRAY_SIZE(evb3_dai),
+};
+
+static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret;
+
+	snd_soc_dapm_new_controls(codec, evb3_dapm_widgets,
+				  ARRAY_SIZE(evb3_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Speaker");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+	return 0;
+}
+
+static int __init tavorevb3_init(void)
+{
+	int ret;
+
+	if (!machine_is_tavorevb3())
+		return -ENODEV;
+	evb3_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!evb3_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3);
+
+	ret = platform_device_add(evb3_snd_device);
+	if (ret)
+		platform_device_put(evb3_snd_device);
+
+	return ret;
+}
+
+static void __exit tavorevb3_exit(void)
+{
+	platform_device_unregister(evb3_snd_device);
+}
+
+module_init(tavorevb3_init);
+module_exit(tavorevb3_exit);
+
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index dbbd3e9d1637..a3bfb2e8b70f 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -33,7 +33,6 @@
 #include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 static struct snd_soc_card tosa;
@@ -80,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->card->codec;
+	struct snd_soc_codec *codec = rtd->card->codec;
 
 	/* check the jack status at stream startup */
 	tosa_ext_control(codec);
@@ -184,8 +183,9 @@ static const struct snd_kcontrol_new tosa_controls[] = {
 		tosa_set_spk),
 };
 
-static int tosa_ac97_init(struct snd_soc_codec *codec)
+static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
 	int err;
 
 	snd_soc_dapm_nc_pin(codec, "OUT3");
@@ -212,16 +212,20 @@ static struct snd_soc_dai_link tosa_dai[] = {
 {
 	.name = "AC97",
 	.stream_name = "AC97 HiFi",
-	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+	.cpu_dai_name = "pxa-ac97.0",
+	.codec_dai_name = "wm9712-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm9712-codec",
 	.init = tosa_ac97_init,
 	.ops = &tosa_ops,
 },
 {
 	.name = "AC97 Aux",
 	.stream_name = "AC97 Aux",
-	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	.cpu_dai_name = "pxa-ac97.1",
+	.codec_dai_name = "wm9712-aux",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "wm9712-codec",
 	.ops = &tosa_ops,
 },
 };
@@ -248,18 +252,12 @@ static int tosa_remove(struct platform_device *dev)
 
 static struct snd_soc_card tosa = {
 	.name = "Tosa",
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = tosa_dai,
 	.num_links = ARRAY_SIZE(tosa_dai),
 	.probe = tosa_probe,
 	.remove = tosa_remove,
 };
 
-static struct snd_soc_device tosa_snd_devdata = {
-	.card = &tosa,
-	.codec_dev = &soc_codec_dev_wm9712,
-};
-
 static struct platform_device *tosa_snd_device;
 
 static int __init tosa_init(void)
@@ -275,8 +273,7 @@ static int __init tosa_init(void)
 		goto err_alloc;
 	}
 
-	platform_set_drvdata(tosa_snd_device, &tosa_snd_devdata);
-	tosa_snd_devdata.dev = &tosa_snd_device->dev;
+	platform_set_drvdata(tosa_snd_device, &tosa);
 	ret = platform_device_add(tosa_snd_device);
 
 	if (!ret)
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 4e4d2fa8ddc5..4cc841b44182 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -30,7 +30,6 @@
 #include <mach/z2.h>
 
 #include "../codecs/wm8750.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
 
 static struct snd_soc_card snd_soc_z2;
@@ -39,8 +38,8 @@ static int z2_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	unsigned int clk = 0;
 	int ret = 0;
 
@@ -138,8 +137,9 @@ static const struct snd_soc_dapm_route audio_map[] = {
 /*
  * Logic for a wm8750 as connected on a Z2 Device
  */
-static int z2_wm8750_init(struct snd_soc_codec *codec)
+static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
 	int ret;
 
 	/* NC codec pins */
@@ -160,7 +160,7 @@ static int z2_wm8750_init(struct snd_soc_codec *codec)
 		goto err;
 
 	/* Jack detection API stuff */
-	ret = snd_soc_jack_new(&snd_soc_z2, "Headset Jack", SND_JACK_HEADSET,
+	ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
 				&hs_jack);
 	if (ret)
 		goto err;
@@ -189,8 +189,10 @@ static struct snd_soc_ops z2_ops = {
 static struct snd_soc_dai_link z2_dai = {
 	.name		= "wm8750",
 	.stream_name	= "WM8750",
-	.cpu_dai	= &pxa_i2s_dai,
-	.codec_dai	= &wm8750_dai,
+	.cpu_dai_name	= "pxa2xx-i2s",
+	.codec_dai_name	= "wm8750-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name	= "wm8750-codec.0-001a",
 	.init		= z2_wm8750_init,
 	.ops		= &z2_ops,
 };
@@ -198,17 +200,10 @@ static struct snd_soc_dai_link z2_dai = {
 /* z2 audio machine driver */
 static struct snd_soc_card snd_soc_z2 = {
 	.name		= "Z2",
-	.platform	= &pxa2xx_soc_platform,
 	.dai_link	= &z2_dai,
 	.num_links	= 1,
 };
 
-/* z2 audio subsystem */
-static struct snd_soc_device z2_snd_devdata = {
-	.card		= &snd_soc_z2,
-	.codec_dev	= &soc_codec_dev_wm8750,
-};
-
 static struct platform_device *z2_snd_device;
 
 static int __init z2_init(void)
@@ -222,8 +217,7 @@ static int __init z2_init(void)
 	if (!z2_snd_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(z2_snd_device, &z2_snd_devdata);
-	z2_snd_devdata.dev = &z2_snd_device->dev;
+	platform_set_drvdata(z2_snd_device, &snd_soc_z2);
 	ret = platform_device_add(z2_snd_device);
 
 	if (ret)
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index dd678ae24398..d27e05af7759 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -23,7 +23,6 @@
 #include <sound/soc-dapm.h>
 
 #include "../codecs/wm9713.h"
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 #include "pxa-ssp.h"
 
@@ -71,10 +70,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{ "Multiactor", NULL, "SPKR" },
 };
 
-static int zylonite_wm9713_init(struct snd_soc_codec *codec)
+static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
 {
+	struct snd_soc_codec *codec = rtd->codec;
+
 	if (clk_pout)
-		snd_soc_dai_set_pll(&codec->dai[0], 0, 0,
+		snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
 				    clk_get_rate(pout), 0);
 
 	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
@@ -94,8 +95,8 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 				    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	unsigned int pll_out = 0;
 	unsigned int wm9713_div = 0;
 	int ret = 0;
@@ -163,21 +164,27 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 {
 	.name = "AC97",
 	.stream_name = "AC97 HiFi",
-	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
-	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+	.codec_name = "wm9713-codec",
+	.platform_name = "pxa-pcm-audio",
+	.cpu_dai_name = "pxa-ac97.0",
+	.codec_name = "wm9713-hifi",
 	.init = zylonite_wm9713_init,
 },
 {
 	.name = "AC97 Aux",
 	.stream_name = "AC97 Aux",
-	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-	.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+	.codec_name = "wm9713-codec",
+	.platform_name = "pxa-pcm-audio",
+	.cpu_dai_name = "pxa-ac97.1",
+	.codec_name = "wm9713-aux",
 },
 {
 	.name = "WM9713 Voice",
 	.stream_name = "WM9713 Voice",
-	.cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3],
-	.codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE],
+	.codec_name = "wm9713-codec",
+	.platform_name = "pxa-pcm-audio",
+	.cpu_dai_name = "pxa-ssp-dai.2",
+	.codec_name = "wm9713-voice",
 	.ops = &zylonite_voice_ops,
 },
 };
@@ -248,14 +255,9 @@ static struct snd_soc_card zylonite = {
 	.remove = &zylonite_remove,
 	.suspend_post = &zylonite_suspend_post,
 	.resume_pre = &zylonite_resume_pre,
-	.platform = &pxa2xx_soc_platform,
 	.dai_link = zylonite_dai,
 	.num_links = ARRAY_SIZE(zylonite_dai),
-};
-
-static struct snd_soc_device zylonite_snd_ac97_devdata = {
-	.card = &zylonite,
-	.codec_dev = &soc_codec_dev_wm9713,
+	.owner = THIS_MODULE,
 };
 
 static struct platform_device *zylonite_snd_ac97_device;
@@ -268,9 +270,7 @@ static int __init zylonite_init(void)
 	if (!zylonite_snd_ac97_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(zylonite_snd_ac97_device,
-			     &zylonite_snd_ac97_devdata);
-	zylonite_snd_ac97_devdata.dev = &zylonite_snd_ac97_device->dev;
+	platform_set_drvdata(zylonite_snd_ac97_device, &zylonite);
 
 	ret = platform_device_add(zylonite_snd_ac97_device);
 	if (ret != 0)