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: (294 commits)
  S3C64XX: Staticise platform data for PCM devices
  ASoC: Rename controls with a / in wm_hubs
  snd-fm801: autodetect SF64-PCR (tuner-only) card
  ALSA: tea575x-tuner: fix mute
  ASoC: au1x: dbdma2: plug memleak in pcm device creation error path
  ASoC: au1x: dbdma2: fix oops on soc device removal.
  ALSA: hda - Fix memory leaks in the previous patch
  ALSA: hda - Add ALC661/259, ALC892/888VD support
  ALSA: opti9xx: remove snd_opti9xx fields
  ALSA: aaci - Clean up duplicate code
  ALSA: usb - Fix mixer map for Hercules Gamesurround Muse Pocket LT
  ALSA: hda - Add position_fix quirk for HP dv3
  ALSA: hda - Add a pin-fix for FSC Amilo Pi1505
  ALSA: hda - Fix Cxt5047 test mode
  ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API
  ASoC: sh: fsi: Add runtime PM support
  sh: ms7724se: Add runtime PM support for FSI
  ALSA: hda - Add a position_fix quirk for MSI Wind U115
  ALSA: opti-miro: add PnP detection
  ALSA: opti-miro: separate comon probing code
  ...

130 files changed, 8009 insertions, 2313 deletions

diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 0c5eac01bf2e..1470141d4167 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
-snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o
+snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 9eb610c2ba91..9df4c68ef000 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -268,7 +268,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
 #endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
 
 
-	ret = snd_soc_dai_set_pll(codec_dai, 0,
+	ret = snd_soc_dai_set_pll(codec_dai, 0, 0,
 					 clk_get_rate(CODEC_CLK), pll_out);
 	if (ret < 0) {
 		pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 885ba012557e..e028744c32ce 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -207,7 +207,7 @@ static int __init at91sam9g20ek_init(void)
 	struct clk *pllb;
 	int ret;
 
-	if (!machine_is_at91sam9g20ek())
+	if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc()))
 		return -ENODEV;
 
 	/*
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 594c6c5b7838..19e4d37eba1c 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -2,7 +2,7 @@
  * Au12x0/Au1550 PSC ALSA ASoC audio support.
  *
  * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ *	Manuel Lauss <manuel.lauss@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
@@ -333,6 +333,30 @@ static int au1xpsc_pcm_new(struct snd_card *card,
 
 static int au1xpsc_pcm_probe(struct platform_device *pdev)
 {
+	if (!au1xpsc_audio_pcmdma[PCM_TX] || !au1xpsc_audio_pcmdma[PCM_RX])
+		return -ENODEV;
+
+	return 0;
+}
+
+static int au1xpsc_pcm_remove(struct platform_device *pdev)
+{
+	return 0;
+}
+
+/* au1xpsc audio platform */
+struct snd_soc_platform au1xpsc_soc_platform = {
+	.name		= "au1xpsc-pcm-dbdma",
+	.probe		= au1xpsc_pcm_probe,
+	.remove		= au1xpsc_pcm_remove,
+	.pcm_ops 	= &au1xpsc_pcm_ops,
+	.pcm_new	= au1xpsc_pcm_new,
+	.pcm_free	= au1xpsc_pcm_free_dma_buffers,
+};
+EXPORT_SYMBOL_GPL(au1xpsc_soc_platform);
+
+static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev)
+{
 	struct resource *r;
 	int ret;
 
@@ -365,7 +389,9 @@ static int au1xpsc_pcm_probe(struct platform_device *pdev)
 	}
 	(au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start;
 
-	return 0;
+	ret = snd_soc_register_platform(&au1xpsc_soc_platform);
+	if (!ret)
+		return ret;
 
 out2:
 	kfree(au1xpsc_audio_pcmdma[PCM_RX]);
@@ -376,10 +402,12 @@ out1:
 	return ret;
 }
 
-static int au1xpsc_pcm_remove(struct platform_device *pdev)
+static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev)
 {
 	int i;
 
+	snd_soc_unregister_platform(&au1xpsc_soc_platform);
+
 	for (i = 0; i < 2; i++) {
 		if (au1xpsc_audio_pcmdma[i]) {
 			au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]);
@@ -391,32 +419,81 @@ static int au1xpsc_pcm_remove(struct platform_device *pdev)
 	return 0;
 }
 
-/* au1xpsc audio platform */
-struct snd_soc_platform au1xpsc_soc_platform = {
-	.name		= "au1xpsc-pcm-dbdma",
-	.probe		= au1xpsc_pcm_probe,
-	.remove		= au1xpsc_pcm_remove,
-	.pcm_ops 	= &au1xpsc_pcm_ops,
-	.pcm_new	= au1xpsc_pcm_new,
-	.pcm_free	= au1xpsc_pcm_free_dma_buffers,
+static struct platform_driver au1xpsc_pcm_driver = {
+	.driver	= {
+		.name	= "au1xpsc-pcm",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= au1xpsc_pcm_drvprobe,
+	.remove		= __devexit_p(au1xpsc_pcm_drvremove),
 };
-EXPORT_SYMBOL_GPL(au1xpsc_soc_platform);
 
-static int __init au1xpsc_audio_dbdma_init(void)
+static int __init au1xpsc_audio_dbdma_load(void)
 {
 	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
 	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
-	return snd_soc_register_platform(&au1xpsc_soc_platform);
+	return platform_driver_register(&au1xpsc_pcm_driver);
 }
 
-static void __exit au1xpsc_audio_dbdma_exit(void)
+static void __exit au1xpsc_audio_dbdma_unload(void)
 {
-	snd_soc_unregister_platform(&au1xpsc_soc_platform);
+	platform_driver_unregister(&au1xpsc_pcm_driver);
 }
 
-module_init(au1xpsc_audio_dbdma_init);
-module_exit(au1xpsc_audio_dbdma_exit);
+module_init(au1xpsc_audio_dbdma_load);
+module_exit(au1xpsc_audio_dbdma_unload);
+
+
+struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev)
+{
+	struct resource *res, *r;
+	struct platform_device *pd;
+	int id[2];
+	int ret;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!r)
+		return NULL;
+	id[0] = r->start;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!r)
+		return NULL;
+	id[1] = r->start;
+
+	res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL);
+	if (!res)
+		return NULL;
+
+	res[0].start = res[0].end = id[0];
+	res[1].start = res[1].end = id[1];
+	res[0].flags = res[1].flags = IORESOURCE_DMA;
+
+	pd = platform_device_alloc("au1xpsc-pcm", -1);
+	if (!pd)
+		goto out;
+
+	pd->resource = res;
+	pd->num_resources = 2;
+
+	ret = platform_device_add(pd);
+	if (!ret)
+		return pd;
+
+	platform_device_put(pd);
+out:
+	kfree(res);
+	return NULL;
+}
+EXPORT_SYMBOL_GPL(au1xpsc_pcm_add);
+
+void au1xpsc_pcm_destroy(struct platform_device *dmapd)
+{
+	if (dmapd)
+		platform_device_unregister(dmapd);
+}
+EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index a521aa90ddee..340311d7fed5 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -61,7 +61,8 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
-	unsigned short data, retry, tmo;
+	unsigned short retry, tmo;
+	unsigned long data;
 
 	au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
 	au_sync();
@@ -74,20 +75,26 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
 			  AC97_CDC(pscdata));
 		au_sync();
 
-		tmo = 2000;
-		while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD))
-			&& --tmo)
-			udelay(2);
+		tmo = 20;
+		do {
+			udelay(21);
+			if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)
+				break;
+		} while (--tmo);
 
-		data = au_readl(AC97_CDC(pscdata)) & 0xffff;
+		data = au_readl(AC97_CDC(pscdata));
 
 		au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
 		au_sync();
 
 		mutex_unlock(&pscdata->lock);
+
+		if (reg != ((data >> 16) & 0x7f))
+			tmo = 1;	/* wrong register, try again */
+
 	} while (--retry && !tmo);
 
-	return retry ? data : 0xffff;
+	return retry ? data & 0xffff : 0xffff;
 }
 
 /* AC97 controller writes to codec register */
@@ -109,10 +116,12 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 			  AC97_CDC(pscdata));
 		au_sync();
 
-		tmo = 2000;
-		while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD))
-		       && --tmo)
-			udelay(2);
+		tmo = 20;
+		do {
+			udelay(21);
+			if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)
+				break;
+		} while (--tmo);
 
 		au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
 		au_sync();
@@ -195,7 +204,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
 	unsigned long r, ro, stat;
-	int chans, stype = SUBSTREAM_TYPE(substream);
+	int chans, t, stype = SUBSTREAM_TYPE(substream);
 
 	chans = params_channels(params);
 
@@ -237,8 +246,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 		au_sync();
 
 		/* ...wait for it... */
-		while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)
-			asm volatile ("nop");
+		t = 100;
+		while ((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t)
+			msleep(1);
+
+		if (!t)
+			printk(KERN_ERR "PSC-AC97: can't disable!\n");
 
 		/* ...write config... */
 		au_writel(r, AC97_CFG(pscdata));
@@ -249,8 +262,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 		au_sync();
 
 		/* ...and wait for ready bit */
-		while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR))
-			asm volatile ("nop");
+		t = 100;
+		while ((!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && --t)
+			msleep(1);
+
+		if (!t)
+			printk(KERN_ERR "PSC-AC97: can't enable!\n");
 
 		mutex_unlock(&pscdata->lock);
 
@@ -300,19 +317,55 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
 static int au1xpsc_ac97_probe(struct platform_device *pdev,
 			      struct snd_soc_dai *dai)
 {
+	return au1xpsc_ac97_workdata ? 0 : -ENODEV;
+}
+
+static void au1xpsc_ac97_remove(struct platform_device *pdev,
+				struct snd_soc_dai *dai)
+{
+}
+
+static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.trigger	= au1xpsc_ac97_trigger,
+	.hw_params	= au1xpsc_ac97_hw_params,
+};
+
+struct snd_soc_dai au1xpsc_ac97_dai = {
+	.name			= "au1xpsc_ac97",
+	.ac97_control		= 1,
+	.probe			= au1xpsc_ac97_probe,
+	.remove			= au1xpsc_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,
+	},
+	.ops = &au1xpsc_ac97_dai_ops,
+};
+EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
+
+static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
+{
 	int ret;
 	struct resource *r;
 	unsigned long sel;
+	struct au1xpsc_audio_data *wd;
 
 	if (au1xpsc_ac97_workdata)
 		return -EBUSY;
 
-	au1xpsc_ac97_workdata =
-		kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
-	if (!au1xpsc_ac97_workdata)
+	wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
+	if (!wd)
 		return -ENOMEM;
 
-	mutex_init(&au1xpsc_ac97_workdata->lock);
+	mutex_init(&wd->lock);
 
 	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	if (!r) {
@@ -321,81 +374,95 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev,
 	}
 
 	ret = -EBUSY;
-	au1xpsc_ac97_workdata->ioarea =
-		request_mem_region(r->start, r->end - r->start + 1,
+	wd->ioarea = request_mem_region(r->start, r->end - r->start + 1,
 					"au1xpsc_ac97");
-	if (!au1xpsc_ac97_workdata->ioarea)
+	if (!wd->ioarea)
 		goto out0;
 
-	au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff);
-	if (!au1xpsc_ac97_workdata->mmio)
+	wd->mmio = ioremap(r->start, 0xffff);
+	if (!wd->mmio)
 		goto out1;
 
 	/* configuration: max dma trigger threshold, enable ac97 */
-	au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 |
-				     PSC_AC97CFG_TT_FIFO8 |
-				     PSC_AC97CFG_DE_ENABLE;
+	wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
+		  PSC_AC97CFG_DE_ENABLE;
 
-	/* preserve PSC clock source set up by platform (dev.platform_data
-	 * is already occupied by soc layer)
-	 */
-	sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK;
-	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+	/* preserve PSC clock source set up by platform	 */
+	sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
 	au_sync();
-	au_writel(0, PSC_SEL(au1xpsc_ac97_workdata));
+	au_writel(0, PSC_SEL(wd));
 	au_sync();
-	au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata));
+	au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd));
 	au_sync();
-	/* next up: cold reset.  Dont check for PSC-ready now since
-	 * there may not be any codec clock yet.
-	 */
 
-	return 0;
+	ret = snd_soc_register_dai(&au1xpsc_ac97_dai);
+	if (ret)
+		goto out1;
 
+	wd->dmapd = au1xpsc_pcm_add(pdev);
+	if (wd->dmapd) {
+		platform_set_drvdata(pdev, wd);
+		au1xpsc_ac97_workdata = wd;	/* MDEV */
+		return 0;
+	}
+
+	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
 out1:
-	release_resource(au1xpsc_ac97_workdata->ioarea);
-	kfree(au1xpsc_ac97_workdata->ioarea);
+	release_resource(wd->ioarea);
+	kfree(wd->ioarea);
 out0:
-	kfree(au1xpsc_ac97_workdata);
-	au1xpsc_ac97_workdata = NULL;
+	kfree(wd);
 	return ret;
 }
 
-static void au1xpsc_ac97_remove(struct platform_device *pdev,
-				struct snd_soc_dai *dai)
+static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev)
 {
+	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
+
+	if (wd->dmapd)
+		au1xpsc_pcm_destroy(wd->dmapd);
+
+	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
+
 	/* disable PSC completely */
-	au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+	au_writel(0, AC97_CFG(wd));
 	au_sync();
-	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
 	au_sync();
 
-	iounmap(au1xpsc_ac97_workdata->mmio);
-	release_resource(au1xpsc_ac97_workdata->ioarea);
-	kfree(au1xpsc_ac97_workdata->ioarea);
-	kfree(au1xpsc_ac97_workdata);
-	au1xpsc_ac97_workdata = NULL;
+	iounmap(wd->mmio);
+	release_resource(wd->ioarea);
+	kfree(wd->ioarea);
+	kfree(wd);
+
+	au1xpsc_ac97_workdata = NULL;	/* MDEV */
+
+	return 0;
 }
 
-static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
+#ifdef CONFIG_PM
+static int au1xpsc_ac97_drvsuspend(struct device *dev)
 {
+	struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
+
 	/* save interesting registers and disable PSC */
-	au1xpsc_ac97_workdata->pm[0] =
-			au_readl(PSC_SEL(au1xpsc_ac97_workdata));
+	wd->pm[0] = au_readl(PSC_SEL(wd));
 
-	au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+	au_writel(0, AC97_CFG(wd));
 	au_sync();
-	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata));
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
 	au_sync();
 
 	return 0;
 }
 
-static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
+static int au1xpsc_ac97_drvresume(struct device *dev)
 {
+	struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
+
 	/* restore PSC clock config */
-	au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
-			PSC_SEL(au1xpsc_ac97_workdata));
+	au_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd));
 	au_sync();
 
 	/* after this point the ac97 core will cold-reset the codec.
@@ -405,48 +472,44 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 	return 0;
 }
 
-static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
-	.trigger	= au1xpsc_ac97_trigger,
-	.hw_params	= au1xpsc_ac97_hw_params,
+static struct dev_pm_ops au1xpscac97_pmops = {
+	.suspend	= au1xpsc_ac97_drvsuspend,
+	.resume		= au1xpsc_ac97_drvresume,
 };
 
-struct snd_soc_dai au1xpsc_ac97_dai = {
-	.name			= "au1xpsc_ac97",
-	.ac97_control		= 1,
-	.probe			= au1xpsc_ac97_probe,
-	.remove			= au1xpsc_ac97_remove,
-	.suspend		= au1xpsc_ac97_suspend,
-	.resume			= au1xpsc_ac97_resume,
-	.playback = {
-		.rates		= AC97_RATES,
-		.formats	= AC97_FMTS,
-		.channels_min	= 2,
-		.channels_max	= 2,
-	},
-	.capture = {
-		.rates		= AC97_RATES,
-		.formats	= AC97_FMTS,
-		.channels_min	= 2,
-		.channels_max	= 2,
+#define AU1XPSCAC97_PMOPS &au1xpscac97_pmops
+
+#else
+
+#define AU1XPSCAC97_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xpsc_ac97_driver = {
+	.driver	= {
+		.name	= "au1xpsc_ac97",
+		.owner	= THIS_MODULE,
+		.pm	= AU1XPSCAC97_PMOPS,
 	},
-	.ops = &au1xpsc_ac97_dai_ops,
+	.probe		= au1xpsc_ac97_drvprobe,
+	.remove		= __devexit_p(au1xpsc_ac97_drvremove),
 };
-EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 
-static int __init au1xpsc_ac97_init(void)
+static int __init au1xpsc_ac97_load(void)
 {
 	au1xpsc_ac97_workdata = NULL;
-	return snd_soc_register_dai(&au1xpsc_ac97_dai);
+	return platform_driver_register(&au1xpsc_ac97_driver);
 }
 
-static void __exit au1xpsc_ac97_exit(void)
+static void __exit au1xpsc_ac97_unload(void)
 {
-	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
+	platform_driver_unregister(&au1xpsc_ac97_driver);
 }
 
-module_init(au1xpsc_ac97_init);
-module_exit(au1xpsc_ac97_exit);
+module_init(au1xpsc_ac97_load);
+module_exit(au1xpsc_ac97_unload);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver");
-MODULE_AUTHOR("Manuel Lauss <manuel.lauss@gmail.com>");
+MODULE_AUTHOR("Manuel Lauss");
+
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index bb589327ee32..0cf2ca61c776 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -2,7 +2,7 @@
  * Au12x0/Au1550 PSC ALSA ASoC audio support.
  *
  * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ *	Manuel Lauss <manuel.lauss@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
@@ -265,16 +265,52 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 static int au1xpsc_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
+	return 	au1xpsc_i2s_workdata ? 0 : -ENODEV;
+}
+
+static void au1xpsc_i2s_remove(struct platform_device *pdev,
+			       struct snd_soc_dai *dai)
+{
+}
+
+static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+	.trigger	= au1xpsc_i2s_trigger,
+	.hw_params	= au1xpsc_i2s_hw_params,
+	.set_fmt	= au1xpsc_i2s_set_fmt,
+};
+
+struct snd_soc_dai au1xpsc_i2s_dai = {
+	.name			= "au1xpsc_i2s",
+	.probe			= au1xpsc_i2s_probe,
+	.remove			= au1xpsc_i2s_remove,
+	.playback = {
+		.rates		= AU1XPSC_I2S_RATES,
+		.formats	= AU1XPSC_I2S_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 8,	/* 2 without external help */
+	},
+	.capture = {
+		.rates		= AU1XPSC_I2S_RATES,
+		.formats	= AU1XPSC_I2S_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 8,	/* 2 without external help */
+	},
+	.ops = &au1xpsc_i2s_dai_ops,
+};
+EXPORT_SYMBOL(au1xpsc_i2s_dai);
+
+static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev)
+{
 	struct resource *r;
 	unsigned long sel;
 	int ret;
+	struct au1xpsc_audio_data *wd;
 
 	if (au1xpsc_i2s_workdata)
 		return -EBUSY;
 
-	au1xpsc_i2s_workdata =
-		kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
-	if (!au1xpsc_i2s_workdata)
+	wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL);
+	if (!wd)
 		return -ENOMEM;
 
 	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
@@ -284,131 +320,146 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev,
 	}
 
 	ret = -EBUSY;
-	au1xpsc_i2s_workdata->ioarea =
-		request_mem_region(r->start, r->end - r->start + 1,
+	wd->ioarea = request_mem_region(r->start, r->end - r->start + 1,
 					"au1xpsc_i2s");
-	if (!au1xpsc_i2s_workdata->ioarea)
+	if (!wd->ioarea)
 		goto out0;
 
-	au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff);
-	if (!au1xpsc_i2s_workdata->mmio)
+	wd->mmio = ioremap(r->start, 0xffff);
+	if (!wd->mmio)
 		goto out1;
 
 	/* preserve PSC clock source set up by platform (dev.platform_data
 	 * is already occupied by soc layer)
 	 */
-	sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK;
-	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+	sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
 	au_sync();
-	au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata));
-	au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+	au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd));
+	au_writel(0, I2S_CFG(wd));
 	au_sync();
 
 	/* preconfigure: set max rx/tx fifo depths */
-	au1xpsc_i2s_workdata->cfg |=
-			PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8;
+	wd->cfg |= PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8;
 
 	/* don't wait for I2S core to become ready now; clocks may not
 	 * be running yet; depending on clock input for PSC a wait might
 	 * time out.
 	 */
 
-	return 0;
+	ret = snd_soc_register_dai(&au1xpsc_i2s_dai);
+	if (ret)
+		goto out1;
 
+	/* finally add the DMA device for this PSC */
+	wd->dmapd = au1xpsc_pcm_add(pdev);
+	if (wd->dmapd) {
+		platform_set_drvdata(pdev, wd);
+		au1xpsc_i2s_workdata = wd;
+		return 0;
+	}
+
+	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
 out1:
-	release_resource(au1xpsc_i2s_workdata->ioarea);
-	kfree(au1xpsc_i2s_workdata->ioarea);
+	release_resource(wd->ioarea);
+	kfree(wd->ioarea);
 out0:
-	kfree(au1xpsc_i2s_workdata);
-	au1xpsc_i2s_workdata = NULL;
+	kfree(wd);
 	return ret;
 }
 
-static void au1xpsc_i2s_remove(struct platform_device *pdev,
-			       struct snd_soc_dai *dai)
+static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
 {
-	au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
+
+	if (wd->dmapd)
+		au1xpsc_pcm_destroy(wd->dmapd);
+
+	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
+
+	au_writel(0, I2S_CFG(wd));
 	au_sync();
-	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
 	au_sync();
 
-	iounmap(au1xpsc_i2s_workdata->mmio);
-	release_resource(au1xpsc_i2s_workdata->ioarea);
-	kfree(au1xpsc_i2s_workdata->ioarea);
-	kfree(au1xpsc_i2s_workdata);
-	au1xpsc_i2s_workdata = NULL;
+	iounmap(wd->mmio);
+	release_resource(wd->ioarea);
+	kfree(wd->ioarea);
+	kfree(wd);
+
+	au1xpsc_i2s_workdata = NULL;	/* MDEV */
+
+	return 0;
 }
 
-static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
+#ifdef CONFIG_PM
+static int au1xpsc_i2s_drvsuspend(struct device *dev)
 {
+	struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
+
 	/* save interesting register and disable PSC */
-	au1xpsc_i2s_workdata->pm[0] =
-		au_readl(PSC_SEL(au1xpsc_i2s_workdata));
+	wd->pm[0] = au_readl(PSC_SEL(wd));
 
-	au_writel(0, I2S_CFG(au1xpsc_i2s_workdata));
+	au_writel(0, I2S_CFG(wd));
 	au_sync();
-	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
 	au_sync();
 
 	return 0;
 }
 
-static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_drvresume(struct device *dev)
 {
+	struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
+
 	/* select I2S mode and PSC clock */
-	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
+	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
 	au_sync();
-	au_writel(0, PSC_SEL(au1xpsc_i2s_workdata));
+	au_writel(0, PSC_SEL(wd));
 	au_sync();
-	au_writel(au1xpsc_i2s_workdata->pm[0],
-			PSC_SEL(au1xpsc_i2s_workdata));
+	au_writel(wd->pm[0], PSC_SEL(wd));
 	au_sync();
 
 	return 0;
 }
 
-static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
-	.trigger	= au1xpsc_i2s_trigger,
-	.hw_params	= au1xpsc_i2s_hw_params,
-	.set_fmt	= au1xpsc_i2s_set_fmt,
+static struct dev_pm_ops au1xpsci2s_pmops = {
+	.suspend	= au1xpsc_i2s_drvsuspend,
+	.resume		= au1xpsc_i2s_drvresume,
 };
 
-struct snd_soc_dai au1xpsc_i2s_dai = {
-	.name			= "au1xpsc_i2s",
-	.probe			= au1xpsc_i2s_probe,
-	.remove			= au1xpsc_i2s_remove,
-	.suspend		= au1xpsc_i2s_suspend,
-	.resume			= au1xpsc_i2s_resume,
-	.playback = {
-		.rates		= AU1XPSC_I2S_RATES,
-		.formats	= AU1XPSC_I2S_FMTS,
-		.channels_min	= 2,
-		.channels_max	= 8,	/* 2 without external help */
-	},
-	.capture = {
-		.rates		= AU1XPSC_I2S_RATES,
-		.formats	= AU1XPSC_I2S_FMTS,
-		.channels_min	= 2,
-		.channels_max	= 8,	/* 2 without external help */
+#define AU1XPSCI2S_PMOPS &au1xpsci2s_pmops
+
+#else
+
+#define AU1XPSCI2S_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xpsc_i2s_driver = {
+	.driver		= {
+		.name	= "au1xpsc_i2s",
+		.owner	= THIS_MODULE,
+		.pm	= AU1XPSCI2S_PMOPS,
 	},
-	.ops = &au1xpsc_i2s_dai_ops,
+	.probe		= au1xpsc_i2s_drvprobe,
+	.remove		= __devexit_p(au1xpsc_i2s_drvremove),
 };
-EXPORT_SYMBOL(au1xpsc_i2s_dai);
 
-static int __init au1xpsc_i2s_init(void)
+static int __init au1xpsc_i2s_load(void)
 {
 	au1xpsc_i2s_workdata = NULL;
-	return snd_soc_register_dai(&au1xpsc_i2s_dai);
+	return platform_driver_register(&au1xpsc_i2s_driver);
 }
 
-static void __exit au1xpsc_i2s_exit(void)
+static void __exit au1xpsc_i2s_unload(void)
 {
-	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
+	platform_driver_unregister(&au1xpsc_i2s_driver);
 }
 
-module_init(au1xpsc_i2s_init);
-module_exit(au1xpsc_i2s_exit);
+module_init(au1xpsc_i2s_load);
+module_exit(au1xpsc_i2s_unload);
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver");
-MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index 3f474e8ed4f6..32d3807d3f5a 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -2,7 +2,7 @@
  * Au12x0/Au1550 PSC ALSA ASoC audio support.
  *
  * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
- *	Manuel Lauss <mano@roarinelk.homelinux.net>
+ *	Manuel Lauss <manuel.lauss@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
@@ -21,6 +21,10 @@ extern struct snd_soc_dai au1xpsc_i2s_dai;
 extern struct snd_soc_platform au1xpsc_soc_platform;
 extern struct snd_ac97_bus_ops soc_ac97_ops;
 
+/* DBDMA helpers */
+extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev);
+extern void au1xpsc_pcm_destroy(struct platform_device *dmapd);
+
 struct au1xpsc_audio_data {
 	void __iomem *mmio;
 
@@ -30,6 +34,7 @@ struct au1xpsc_audio_data {
 	unsigned long pm[2];
 	struct resource *ioarea;
 	struct mutex lock;
+	struct platform_device *dmapd;
 };
 
 #define PCM_TX	0
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index cd361e304b0f..0f45a3f56be8 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7};
 	int ret = 0;
 	/* set cpu DAI configuration */
 	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
@@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
+	/* set cpu DAI channel mapping */
+	ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
+		channel_map, ARRAY_SIZE(channel_map), channel_map);
+	if (ret < 0)
+		return ret;
+
 	return 0;
 }
 
diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c
index 08269e91810c..2ef1e5013b8c 100644
--- a/sound/soc/blackfin/bf5xx-ad1938.c
+++ b/sound/soc/blackfin/bf5xx-ad1938.c
@@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7};
 	int ret = 0;
 	/* set cpu DAI configuration */
 	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
@@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* set codec DAI slots, 8 channels, all channels are enabled */
-	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8);
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI channel mapping */
+	ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
+		channel_map, ARRAY_SIZE(channel_map), channel_map);
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index 084b68884ada..3e6ada0dd1c4 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -49,7 +49,6 @@ struct bf5xx_i2s_port {
 	u16 rcr1;
 	u16 tcr2;
 	u16 rcr2;
-	int counter;
 	int configured;
 };
 
@@ -133,16 +132,6 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	return ret;
 }
 
-static int bf5xx_i2s_startup(struct snd_pcm_substream *substream,
-			     struct snd_soc_dai *dai)
-{
-	pr_debug("%s enter\n", __func__);
-
-	/*this counter is used for counting how many pcm streams are opened*/
-	bf5xx_i2s.counter++;
-	return 0;
-}
-
 static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
@@ -201,9 +190,8 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream,
 			       struct snd_soc_dai *dai)
 {
 	pr_debug("%s enter\n", __func__);
-	bf5xx_i2s.counter--;
 	/* No active stream, SPORT is allowed to be configured again. */
-	if (!bf5xx_i2s.counter)
+	if (!dai->active)
 		bf5xx_i2s.configured = 0;
 }
 
@@ -284,7 +272,6 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai)
 	SNDRV_PCM_FMTBIT_S32_LE)
 
 static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = {
-	.startup	= bf5xx_i2s_startup,
 	.shutdown	= bf5xx_i2s_shutdown,
 	.hw_params	= bf5xx_i2s_hw_params,
 	.set_fmt	= bf5xx_i2s_set_dai_fmt,
diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c
index ccb5e823bd18..a8c73cbbd685 100644
--- a/sound/soc/blackfin/bf5xx-tdm-pcm.c
+++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c
@@ -43,7 +43,7 @@
 #include "bf5xx-tdm.h"
 #include "bf5xx-sport.h"
 
-#define PCM_BUFFER_MAX  0x10000
+#define PCM_BUFFER_MAX  0x8000
 #define FRAGMENT_SIZE_MIN  (4*1024)
 #define FRAGMENTS_MIN  2
 #define FRAGMENTS_MAX  32
@@ -177,6 +177,9 @@ out:
 static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 	snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count)
 {
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct sport_device *sport = runtime->private_data;
+	struct bf5xx_tdm_port *tdm_port = sport->private_data;
 	unsigned int *src;
 	unsigned int *dst;
 	int i;
@@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 		dst += pos * 8;
 		while (count--) {
 			for (i = 0; i < substream->runtime->channels; i++)
-				*(dst + i) = *src++;
+				*(dst + tdm_port->tx_map[i]) = *src++;
 			dst += 8;
 		}
 	} else {
@@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 		src += pos * 8;
 		while (count--) {
 			for (i = 0; i < substream->runtime->channels; i++)
-				*dst++ = *(src+i);
+				*dst++ = *(src + tdm_port->rx_map[i]);
 			src += 8;
 		}
 	}
diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c
index ff546e91a22e..4b360124083e 100644
--- a/sound/soc/blackfin/bf5xx-tdm.c
+++ b/sound/soc/blackfin/bf5xx-tdm.c
@@ -46,14 +46,6 @@
 #include "bf5xx-sport.h"
 #include "bf5xx-tdm.h"
 
-struct bf5xx_tdm_port {
-	u16 tcr1;
-	u16 rcr1;
-	u16 tcr2;
-	u16 rcr2;
-	int configured;
-};
-
 static struct bf5xx_tdm_port bf5xx_tdm;
 static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
 
@@ -181,6 +173,40 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream,
 		bf5xx_tdm.configured = 0;
 }
 
+static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai,
+		unsigned int tx_num, unsigned int *tx_slot,
+		unsigned int rx_num, unsigned int *rx_slot)
+{
+	int i;
+	unsigned int slot;
+	unsigned int tx_mapped = 0, rx_mapped = 0;
+
+	if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) ||
+			(rx_num > BFIN_TDM_DAI_MAX_SLOTS))
+		return -EINVAL;
+
+	for (i = 0; i < tx_num; i++) {
+		slot = tx_slot[i];
+		if ((slot < BFIN_TDM_DAI_MAX_SLOTS) &&
+				(!(tx_mapped & (1 << slot)))) {
+			bf5xx_tdm.tx_map[i] = slot;
+			tx_mapped |= 1 << slot;
+		} else
+			return -EINVAL;
+	}
+	for (i = 0; i < rx_num; i++) {
+		slot = rx_slot[i];
+		if ((slot < BFIN_TDM_DAI_MAX_SLOTS) &&
+				(!(rx_mapped & (1 << slot)))) {
+			bf5xx_tdm.rx_map[i] = slot;
+			rx_mapped |= 1 << slot;
+		} else
+			return -EINVAL;
+	}
+
+	return 0;
+}
+
 #ifdef CONFIG_PM
 static int bf5xx_tdm_suspend(struct snd_soc_dai *dai)
 {
@@ -235,6 +261,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = {
 	.hw_params      = bf5xx_tdm_hw_params,
 	.set_fmt        = bf5xx_tdm_set_dai_fmt,
 	.shutdown       = bf5xx_tdm_shutdown,
+	.set_channel_map   = bf5xx_tdm_set_channel_map,
 };
 
 struct snd_soc_dai bf5xx_tdm_dai = {
@@ -300,6 +327,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev)
 		pr_err("Failed to register DAI: %d\n", ret);
 		goto sport_config_err;
 	}
+
+	sport_handle->private_data = &bf5xx_tdm;
 	return 0;
 
 sport_config_err:
diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h
index 618ec3d90cd4..04189a18c1ba 100644
--- a/sound/soc/blackfin/bf5xx-tdm.h
+++ b/sound/soc/blackfin/bf5xx-tdm.h
@@ -9,6 +9,17 @@
 #ifndef _BF5XX_TDM_H
 #define _BF5XX_TDM_H
 
+#define BFIN_TDM_DAI_MAX_SLOTS 8
+struct bf5xx_tdm_port {
+	u16 tcr1;
+	u16 rcr1;
+	u16 tcr2;
+	u16 rcr2;
+	unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS];
+	unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS];
+	int configured;
+};
+
 extern struct snd_soc_dai bf5xx_tdm_dai;
 
 #endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0edca93af3b0..52b005f8fed4 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -15,10 +15,12 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AD1836 if SPI_MASTER
 	select SND_SOC_AD1938 if SPI_MASTER
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
+	select SND_SOC_ADS117X
 	select SND_SOC_AD73311 if I2C
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_AK4642 if I2C
+	select SND_SOC_AK4671 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008
@@ -28,6 +30,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
 	select SND_SOC_TLV320AIC3X if I2C
+	select SND_SOC_TPA6130A2 if I2C
+	select SND_SOC_TLV320DAC33 if I2C
 	select SND_SOC_TWL4030 if TWL4030_CORE
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
@@ -36,6 +40,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8523 if I2C
 	select SND_SOC_WM8580 if I2C
+	select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI
+	select SND_SOC_WM8727
 	select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI
@@ -86,6 +92,9 @@ config SND_SOC_AD1980
 
 config SND_SOC_AD73311
 	tristate
+	
+config SND_SOC_ADS117X
+	tristate
 
 config SND_SOC_AK4104
 	tristate
@@ -96,6 +105,9 @@ config SND_SOC_AK4535
 config SND_SOC_AK4642
 	tristate
 
+config SND_SOC_AK4671
+	tristate
+
 # Cirrus Logic CS4270 Codec
 config SND_SOC_CS4270
 	tristate
@@ -136,7 +148,11 @@ config SND_SOC_TLV320AIC26
 config SND_SOC_TLV320AIC3X
 	tristate
 
+config SND_SOC_TLV320DAC33
+	tristate
+
 config SND_SOC_TWL4030
+	select TWL4030_CODEC
 	tristate
 
 config SND_SOC_UDA134X
@@ -160,6 +176,12 @@ config SND_SOC_WM8523
 config SND_SOC_WM8580
 	tristate
 
+config SND_SOC_WM8711
+	tristate
+
+config SND_SOC_WM8727
+	tristate
+
 config SND_SOC_WM8728
 	tristate
 
@@ -220,3 +242,6 @@ config SND_SOC_WM9713
 # Amp
 config SND_SOC_MAX9877
 	tristate
+
+config SND_SOC_TPA6130A2
+	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fb4af28486ba..dbaecb133ac7 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -3,9 +3,11 @@ snd-soc-ad1836-objs := ad1836.o
 snd-soc-ad1938-objs := ad1938.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
+snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-ak4642-objs := ak4642.o
+snd-soc-ak4671-objs := ak4671.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-l3-objs := l3.o
@@ -16,6 +18,7 @@ snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-tlv320dac33-objs := tlv320dac33.o
 snd-soc-twl4030-objs := twl4030.o
 snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
@@ -24,6 +27,8 @@ snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8523-objs := wm8523.o
 snd-soc-wm8580-objs := wm8580.o
+snd-soc-wm8711-objs := wm8711.o
+snd-soc-wm8727-objs := wm8727.o
 snd-soc-wm8728-objs := wm8728.o
 snd-soc-wm8731-objs := wm8731.o
 snd-soc-wm8750-objs := wm8750.o
@@ -47,15 +52,18 @@ snd-soc-wm-hubs-objs := wm_hubs.o
 
 # Amp
 snd-soc-max9877-objs := max9877.o
+snd-soc-tpa6130a2-objs := tpa6130a2.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
 obj-$(CONFIG_SND_SOC_AD1938)	+= snd-soc-ad1938.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
 obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
+obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
@@ -66,6 +74,7 @@ obj-$(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-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
@@ -74,6 +83,8 @@ obj-$(CONFIG_SND_SOC_WM8400)	+= snd-soc-wm8400.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8523)	+= snd-soc-wm8523.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o
+obj-$(CONFIG_SND_SOC_WM8711)	+= snd-soc-wm8711.o
+obj-$(CONFIG_SND_SOC_WM8727)	+= snd-soc-wm8727.o
 obj-$(CONFIG_SND_SOC_WM8728)	+= snd-soc-wm8728.o
 obj-$(CONFIG_SND_SOC_WM8731)	+= snd-soc-wm8731.o
 obj-$(CONFIG_SND_SOC_WM8750)	+= snd-soc-wm8750.o
@@ -97,3 +108,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
 
 # Amp
 obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
+obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 932299bb5d1e..69bd0acc81c8 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -117,9 +117,6 @@ static int ac97_soc_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto bus_err;
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0)
-		goto bus_err;
 	return 0;
 
 bus_err:
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index c48485f2c55d..2c18e3d1b71e 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -385,19 +385,7 @@ static int ad1836_probe(struct platform_device *pdev)
 	snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets,
 				  ARRAY_SIZE(ad1836_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;
 }
diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c
index 34b30efc3cb0..5d489186c05b 100644
--- a/sound/soc/codecs/ad1938.c
+++ b/sound/soc/codecs/ad1938.c
@@ -592,21 +592,9 @@ static int ad1938_probe(struct platform_device *pdev)
 	snd_soc_dapm_new_controls(codec, ad1938_dapm_widgets,
 				  ARRAY_SIZE(ad1938_dapm_widgets));
 	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
-	snd_soc_dapm_new_widgets(codec);
 
 	ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	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;
 }
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index d7440a982d22..39c0f7584e65 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -257,11 +257,6 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 
 	snd_soc_add_controls(codec, ad1980_snd_ac97_controls,
 				ARRAY_SIZE(ad1980_snd_ac97_controls));
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "ad1980: failed to register card\n");
-		goto reset_err;
-	}
 
 	return 0;
 
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index e61dac5e7b8f..d2fcc601722c 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -64,16 +64,8 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "ad73311: failed to register card\n");
-		goto register_err;
-	}
-
 	return ret;
 
-register_err:
-	snd_soc_free_pcms(socdev);
 pcm_err:
 	kfree(socdev->card->codec);
 	socdev->card->codec = NULL;
diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c
new file mode 100644
index 000000000000..cc96411ca3e6
--- /dev/null
+++ b/sound/soc/codecs/ads117x.c
@@ -0,0 +1,123 @@
+/*
+ * ads117x.c  --  Driver for ads1174/8 ADC chips
+ *
+ * Copyright 2009 ShotSpotter Inc.
+ * Author: Graeme Gregory <gg@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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "ads117x.h"
+
+#define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000)
+
+#define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+
+struct snd_soc_dai ads117x_dai = {
+/* ADC */
+	.name = "ADS117X ADC",
+	.id = 1,
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 32,
+		.rates = ADS117X_RATES,
+		.formats = ADS117X_FORMATS,},
+};
+EXPORT_SYMBOL_GPL(ads117x_dai);
+
+static int ads117x_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret;
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->card->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	codec->name = "ADS117X";
+	codec->owner = THIS_MODULE;
+	codec->dai = &ads117x_dai;
+	codec->num_dai = 1;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "ads117x: failed to create pcms\n");
+		kfree(codec);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int ads117x_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	snd_soc_free_pcms(socdev);
+	kfree(codec);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ads117x = {
+	.probe =	ads117x_probe,
+	.remove =	ads117x_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ads117x);
+
+static __devinit int ads117x_platform_probe(struct platform_device *pdev)
+{
+	ads117x_dai.dev = &pdev->dev;
+	return snd_soc_register_dai(&ads117x_dai);
+}
+
+static int __devexit ads117x_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&ads117x_dai);
+	return 0;
+}
+
+static struct platform_driver ads117x_codec_driver = {
+	.driver = {
+			.name = "ads117x",
+			.owner = THIS_MODULE,
+	},
+
+	.probe = ads117x_platform_probe,
+	.remove = __devexit_p(ads117x_platform_remove),
+};
+
+static int __init ads117x_init(void)
+{
+	return platform_driver_register(&ads117x_codec_driver);
+}
+module_init(ads117x_init);
+
+static void __exit ads117x_exit(void)
+{
+	platform_driver_unregister(&ads117x_codec_driver);
+}
+module_exit(ads117x_exit);
+
+MODULE_DESCRIPTION("ASoC ads117x driver");
+MODULE_AUTHOR("Graeme Gregory");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ads117x.h b/sound/soc/codecs/ads117x.h
new file mode 100644
index 000000000000..dbcf50ec9bd1
--- /dev/null
+++ b/sound/soc/codecs/ads117x.h
@@ -0,0 +1,13 @@
+/*
+ * ads117x.h  --  Driver for ads1174/8 ADC chips
+ *
+ * Copyright 2009 ShotSpotter Inc.
+ * Author: Graeme Gregory <gg@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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+extern struct snd_soc_dai ads117x_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ads117x;
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index 4d47bc4f7428..3a14c6fc4f5e 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -313,14 +313,6 @@ static int ak4104_probe(struct platform_device *pdev)
 		return ret;
 	}
 
-	/* Register the socdev */
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(codec->dev, "failed to register card\n");
-		snd_soc_free_pcms(socdev);
-		return ret;
-	}
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 0abec0d29a96..ff966567e2ba 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -294,7 +294,6 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -485,17 +484,9 @@ static int ak4535_init(struct snd_soc_device *socdev)
 	snd_soc_add_controls(codec, ak4535_snd_controls,
 				ARRAY_SIZE(ak4535_snd_controls));
 	ak4535_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "ak4535: failed to register card\n");
-		goto card_err;
-	}
 
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	kfree(codec->reg_cache);
 
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index e057c7b578df..b69861d52161 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -442,18 +442,9 @@ static int ak4642_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "ak4642: failed to register card\n");
-		goto card_err;
-	}
-
 	dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	return ret;
 
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
new file mode 100644
index 000000000000..82fca284d007
--- /dev/null
+++ b/sound/soc/codecs/ak4671.c
@@ -0,0 +1,815 @@
+/*
+ * ak4671.c  --  audio driver for AK4671
+ *
+ * Copyright (C) 2009 Samsung Electronics Co.Ltd
+ * Author: Joonyoung Shim <jy0922.shim@samsung.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "ak4671.h"
+
+static struct snd_soc_codec *ak4671_codec;
+
+/* codec private data */
+struct ak4671_priv {
+	struct snd_soc_codec codec;
+	u8 reg_cache[AK4671_CACHEREGNUM];
+};
+
+/* ak4671 register cache & default register settings */
+static const u8 ak4671_reg[AK4671_CACHEREGNUM] = {
+	0x00,	/* AK4671_AD_DA_POWER_MANAGEMENT	(0x00)	*/
+	0xf6,	/* AK4671_PLL_MODE_SELECT0		(0x01)	*/
+	0x00,	/* AK4671_PLL_MODE_SELECT1		(0x02)	*/
+	0x02,	/* AK4671_FORMAT_SELECT			(0x03)	*/
+	0x00,	/* AK4671_MIC_SIGNAL_SELECT		(0x04)	*/
+	0x55,	/* AK4671_MIC_AMP_GAIN			(0x05)	*/
+	0x00,	/* AK4671_MIXING_POWER_MANAGEMENT0	(0x06)	*/
+	0x00,	/* AK4671_MIXING_POWER_MANAGEMENT1	(0x07)	*/
+	0xb5,	/* AK4671_OUTPUT_VOLUME_CONTROL		(0x08)	*/
+	0x00,	/* AK4671_LOUT1_SIGNAL_SELECT		(0x09)	*/
+	0x00,	/* AK4671_ROUT1_SIGNAL_SELECT		(0x0a)	*/
+	0x00,	/* AK4671_LOUT2_SIGNAL_SELECT		(0x0b)	*/
+	0x00,	/* AK4671_ROUT2_SIGNAL_SELECT		(0x0c)	*/
+	0x00,	/* AK4671_LOUT3_SIGNAL_SELECT		(0x0d)	*/
+	0x00,	/* AK4671_ROUT3_SIGNAL_SELECT		(0x0e)	*/
+	0x00,	/* AK4671_LOUT1_POWER_MANAGERMENT	(0x0f)	*/
+	0x00,	/* AK4671_LOUT2_POWER_MANAGERMENT	(0x10)	*/
+	0x80,	/* AK4671_LOUT3_POWER_MANAGERMENT	(0x11)	*/
+	0x91,	/* AK4671_LCH_INPUT_VOLUME_CONTROL	(0x12)	*/
+	0x91,	/* AK4671_RCH_INPUT_VOLUME_CONTROL	(0x13)	*/
+	0xe1,	/* AK4671_ALC_REFERENCE_SELECT		(0x14)	*/
+	0x00,	/* AK4671_DIGITAL_MIXING_CONTROL	(0x15)	*/
+	0x00,	/* AK4671_ALC_TIMER_SELECT		(0x16)	*/
+	0x00,	/* AK4671_ALC_MODE_CONTROL		(0x17)	*/
+	0x02,	/* AK4671_MODE_CONTROL1			(0x18)	*/
+	0x01,	/* AK4671_MODE_CONTROL2			(0x19)	*/
+	0x18,	/* AK4671_LCH_OUTPUT_VOLUME_CONTROL	(0x1a)	*/
+	0x18,	/* AK4671_RCH_OUTPUT_VOLUME_CONTROL	(0x1b)	*/
+	0x00,	/* AK4671_SIDETONE_A_CONTROL		(0x1c)	*/
+	0x02,	/* AK4671_DIGITAL_FILTER_SELECT		(0x1d)	*/
+	0x00,	/* AK4671_FIL3_COEFFICIENT0		(0x1e)	*/
+	0x00,	/* AK4671_FIL3_COEFFICIENT1		(0x1f)	*/
+	0x00,	/* AK4671_FIL3_COEFFICIENT2		(0x20)	*/
+	0x00,	/* AK4671_FIL3_COEFFICIENT3		(0x21)	*/
+	0x00,	/* AK4671_EQ_COEFFICIENT0		(0x22)	*/
+	0x00,	/* AK4671_EQ_COEFFICIENT1		(0x23)	*/
+	0x00,	/* AK4671_EQ_COEFFICIENT2		(0x24)	*/
+	0x00,	/* AK4671_EQ_COEFFICIENT3		(0x25)	*/
+	0x00,	/* AK4671_EQ_COEFFICIENT4		(0x26)	*/
+	0x00,	/* AK4671_EQ_COEFFICIENT5		(0x27)	*/
+	0xa9,	/* AK4671_FIL1_COEFFICIENT0		(0x28)	*/
+	0x1f,	/* AK4671_FIL1_COEFFICIENT1		(0x29)	*/
+	0xad,	/* AK4671_FIL1_COEFFICIENT2		(0x2a)	*/
+	0x20,	/* AK4671_FIL1_COEFFICIENT3		(0x2b)	*/
+	0x00,	/* AK4671_FIL2_COEFFICIENT0		(0x2c)	*/
+	0x00,	/* AK4671_FIL2_COEFFICIENT1		(0x2d)	*/
+	0x00,	/* AK4671_FIL2_COEFFICIENT2		(0x2e)	*/
+	0x00,	/* AK4671_FIL2_COEFFICIENT3		(0x2f)	*/
+	0x00,	/* AK4671_DIGITAL_FILTER_SELECT2	(0x30)	*/
+	0x00,	/* this register not used			*/
+	0x00,	/* AK4671_E1_COEFFICIENT0		(0x32)	*/
+	0x00,	/* AK4671_E1_COEFFICIENT1		(0x33)	*/
+	0x00,	/* AK4671_E1_COEFFICIENT2		(0x34)	*/
+	0x00,	/* AK4671_E1_COEFFICIENT3		(0x35)	*/
+	0x00,	/* AK4671_E1_COEFFICIENT4		(0x36)	*/
+	0x00,	/* AK4671_E1_COEFFICIENT5		(0x37)	*/
+	0x00,	/* AK4671_E2_COEFFICIENT0		(0x38)	*/
+	0x00,	/* AK4671_E2_COEFFICIENT1		(0x39)	*/
+	0x00,	/* AK4671_E2_COEFFICIENT2		(0x3a)	*/
+	0x00,	/* AK4671_E2_COEFFICIENT3		(0x3b)	*/
+	0x00,	/* AK4671_E2_COEFFICIENT4		(0x3c)	*/
+	0x00,	/* AK4671_E2_COEFFICIENT5		(0x3d)	*/
+	0x00,	/* AK4671_E3_COEFFICIENT0		(0x3e)	*/
+	0x00,	/* AK4671_E3_COEFFICIENT1		(0x3f)	*/
+	0x00,	/* AK4671_E3_COEFFICIENT2		(0x40)	*/
+	0x00,	/* AK4671_E3_COEFFICIENT3		(0x41)	*/
+	0x00,	/* AK4671_E3_COEFFICIENT4		(0x42)	*/
+	0x00,	/* AK4671_E3_COEFFICIENT5		(0x43)	*/
+	0x00,	/* AK4671_E4_COEFFICIENT0		(0x44)	*/
+	0x00,	/* AK4671_E4_COEFFICIENT1		(0x45)	*/
+	0x00,	/* AK4671_E4_COEFFICIENT2		(0x46)	*/
+	0x00,	/* AK4671_E4_COEFFICIENT3		(0x47)	*/
+	0x00,	/* AK4671_E4_COEFFICIENT4		(0x48)	*/
+	0x00,	/* AK4671_E4_COEFFICIENT5		(0x49)	*/
+	0x00,	/* AK4671_E5_COEFFICIENT0		(0x4a)	*/
+	0x00,	/* AK4671_E5_COEFFICIENT1		(0x4b)	*/
+	0x00,	/* AK4671_E5_COEFFICIENT2		(0x4c)	*/
+	0x00,	/* AK4671_E5_COEFFICIENT3		(0x4d)	*/
+	0x00,	/* AK4671_E5_COEFFICIENT4		(0x4e)	*/
+	0x00,	/* AK4671_E5_COEFFICIENT5		(0x4f)	*/
+	0x88,	/* AK4671_EQ_CONTROL_250HZ_100HZ	(0x50)	*/
+	0x88,	/* AK4671_EQ_CONTROL_3500HZ_1KHZ	(0x51)	*/
+	0x08,	/* AK4671_EQ_CONTRO_10KHZ		(0x52)	*/
+	0x00,	/* AK4671_PCM_IF_CONTROL0		(0x53)	*/
+	0x00,	/* AK4671_PCM_IF_CONTROL1		(0x54)	*/
+	0x00,	/* AK4671_PCM_IF_CONTROL2		(0x55)	*/
+	0x18,	/* AK4671_DIGITAL_VOLUME_B_CONTROL	(0x56)	*/
+	0x18,	/* AK4671_DIGITAL_VOLUME_C_CONTROL	(0x57)	*/
+	0x00,	/* AK4671_SIDETONE_VOLUME_CONTROL	(0x58)	*/
+	0x00,	/* AK4671_DIGITAL_MIXING_CONTROL2	(0x59)	*/
+	0x00,	/* AK4671_SAR_ADC_CONTROL		(0x5a)	*/
+};
+
+/*
+ * LOUT1/ROUT1 output volume control:
+ * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB)
+ */
+static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1);
+
+/*
+ * LOUT2/ROUT2 output volume control:
+ * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB)
+ */
+static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1);
+
+/*
+ * LOUT3/ROUT3 output volume control:
+ * from -6 to 3 dB in 3 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0);
+
+/*
+ * Mic amp gain control:
+ * from -15 to 30 dB in 3 dB steps
+ * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not
+ * available
+ */
+static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0);
+
+static const struct snd_kcontrol_new ak4671_snd_controls[] = {
+	/* Common playback gain controls */
+	SOC_SINGLE_TLV("Line Output1 Playback Volume",
+			AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv),
+	SOC_SINGLE_TLV("Headphone Output2 Playback Volume",
+			AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv),
+	SOC_SINGLE_TLV("Line Output3 Playback Volume",
+			AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv),
+
+	/* Common capture gain controls */
+	SOC_DOUBLE_TLV("Mic Amp Capture Volume",
+			AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv),
+};
+
+/* event handlers */
+static int ak4671_out2_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	u8 reg;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT);
+		reg |= AK4671_MUTEN;
+		snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg);
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT);
+		reg &= ~AK4671_MUTEN;
+		snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg);
+		break;
+	}
+
+	return 0;
+}
+
+/* Output Mixers */
+static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0),
+	SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0),
+	SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0),
+	SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0),
+	SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0),
+	SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0),
+	SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0),
+	SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0),
+	SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0),
+	SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0),
+	SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0),
+	SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0),
+	SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0),
+	SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0),
+	SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0),
+	SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0),
+	SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0),
+	SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0),
+	SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0),
+	SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0),
+	SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0),
+	SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0),
+	SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0),
+	SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0),
+	SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0),
+	SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0),
+	SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0),
+	SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0),
+	SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0),
+	SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0),
+	SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0),
+};
+
+/* Input MUXs */
+static const char *ak4671_lin_mux_texts[] =
+		{"LIN1", "LIN2", "LIN3", "LIN4"};
+static const struct soc_enum ak4671_lin_mux_enum =
+	SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0,
+			ARRAY_SIZE(ak4671_lin_mux_texts),
+			ak4671_lin_mux_texts);
+static const struct snd_kcontrol_new ak4671_lin_mux_control =
+	SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum);
+
+static const char *ak4671_rin_mux_texts[] =
+		{"RIN1", "RIN2", "RIN3", "RIN4"};
+static const struct soc_enum ak4671_rin_mux_enum =
+	SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2,
+			ARRAY_SIZE(ak4671_rin_mux_texts),
+			ak4671_rin_mux_texts);
+static const struct snd_kcontrol_new ak4671_rin_mux_control =
+	SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum);
+
+static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = {
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("LIN1"),
+	SND_SOC_DAPM_INPUT("RIN1"),
+	SND_SOC_DAPM_INPUT("LIN2"),
+	SND_SOC_DAPM_INPUT("RIN2"),
+	SND_SOC_DAPM_INPUT("LIN3"),
+	SND_SOC_DAPM_INPUT("RIN3"),
+	SND_SOC_DAPM_INPUT("LIN4"),
+	SND_SOC_DAPM_INPUT("RIN4"),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("LOUT1"),
+	SND_SOC_DAPM_OUTPUT("ROUT1"),
+	SND_SOC_DAPM_OUTPUT("LOUT2"),
+	SND_SOC_DAPM_OUTPUT("ROUT2"),
+	SND_SOC_DAPM_OUTPUT("LOUT3"),
+	SND_SOC_DAPM_OUTPUT("ROUT3"),
+
+	/* DAC */
+	SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback",
+			AK4671_AD_DA_POWER_MANAGEMENT, 6, 0),
+	SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback",
+			AK4671_AD_DA_POWER_MANAGEMENT, 7, 0),
+
+	/* ADC */
+	SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture",
+			AK4671_AD_DA_POWER_MANAGEMENT, 4, 0),
+	SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture",
+			AK4671_AD_DA_POWER_MANAGEMENT, 5, 0),
+
+	/* PGA */
+	SND_SOC_DAPM_PGA("LOUT2 Mix Amp",
+			AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ROUT2 Mix Amp",
+			AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("LIN1 Mixing Circuit",
+			AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("RIN1 Mixing Circuit",
+			AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("LIN2 Mixing Circuit",
+			AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("RIN2 Mixing Circuit",
+			AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("LIN3 Mixing Circuit",
+			AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("RIN3 Mixing Circuit",
+			AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("LIN4 Mixing Circuit",
+			AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("RIN4 Mixing Circuit",
+			AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0),
+
+	/* Output Mixers */
+	SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0,
+			&ak4671_lout1_mixer_controls[0],
+			ARRAY_SIZE(ak4671_lout1_mixer_controls)),
+	SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0,
+			&ak4671_rout1_mixer_controls[0],
+			ARRAY_SIZE(ak4671_rout1_mixer_controls)),
+	SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT,
+			0, 0, &ak4671_lout2_mixer_controls[0],
+			ARRAY_SIZE(ak4671_lout2_mixer_controls),
+			ak4671_out2_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT,
+			1, 0, &ak4671_rout2_mixer_controls[0],
+			ARRAY_SIZE(ak4671_rout2_mixer_controls),
+			ak4671_out2_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0,
+			&ak4671_lout3_mixer_controls[0],
+			ARRAY_SIZE(ak4671_lout3_mixer_controls)),
+	SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0,
+			&ak4671_rout3_mixer_controls[0],
+			ARRAY_SIZE(ak4671_rout3_mixer_controls)),
+
+	/* Input MUXs */
+	SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0,
+			&ak4671_lin_mux_control),
+	SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0,
+			&ak4671_rin_mux_control),
+
+	/* Mic Power */
+	SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0),
+
+	/* Supply */
+	SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"DAC Left", "NULL", "PMPLL"},
+	{"DAC Right", "NULL", "PMPLL"},
+	{"ADC Left", "NULL", "PMPLL"},
+	{"ADC Right", "NULL", "PMPLL"},
+
+	/* Outputs */
+	{"LOUT1", "NULL", "LOUT1 Mixer"},
+	{"ROUT1", "NULL", "ROUT1 Mixer"},
+	{"LOUT2", "NULL", "LOUT2 Mix Amp"},
+	{"ROUT2", "NULL", "ROUT2 Mix Amp"},
+	{"LOUT3", "NULL", "LOUT3 Mixer"},
+	{"ROUT3", "NULL", "ROUT3 Mixer"},
+
+	{"LOUT1 Mixer", "DACL", "DAC Left"},
+	{"ROUT1 Mixer", "DACR", "DAC Right"},
+	{"LOUT2 Mixer", "DACHL", "DAC Left"},
+	{"ROUT2 Mixer", "DACHR", "DAC Right"},
+	{"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"},
+	{"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"},
+	{"LOUT3 Mixer", "DACSL", "DAC Left"},
+	{"ROUT3 Mixer", "DACSR", "DAC Right"},
+
+	/* Inputs */
+	{"LIN MUX", "LIN1", "LIN1"},
+	{"LIN MUX", "LIN2", "LIN2"},
+	{"LIN MUX", "LIN3", "LIN3"},
+	{"LIN MUX", "LIN4", "LIN4"},
+
+	{"RIN MUX", "RIN1", "RIN1"},
+	{"RIN MUX", "RIN2", "RIN2"},
+	{"RIN MUX", "RIN3", "RIN3"},
+	{"RIN MUX", "RIN4", "RIN4"},
+
+	{"LIN1", NULL, "Mic Bias"},
+	{"RIN1", NULL, "Mic Bias"},
+	{"LIN2", NULL, "Mic Bias"},
+	{"RIN2", NULL, "Mic Bias"},
+
+	{"ADC Left", "NULL", "LIN MUX"},
+	{"ADC Right", "NULL", "RIN MUX"},
+
+	/* Analog Loops */
+	{"LIN1 Mixing Circuit", "NULL", "LIN1"},
+	{"RIN1 Mixing Circuit", "NULL", "RIN1"},
+	{"LIN2 Mixing Circuit", "NULL", "LIN2"},
+	{"RIN2 Mixing Circuit", "NULL", "RIN2"},
+	{"LIN3 Mixing Circuit", "NULL", "LIN3"},
+	{"RIN3 Mixing Circuit", "NULL", "RIN3"},
+	{"LIN4 Mixing Circuit", "NULL", "LIN4"},
+	{"RIN4 Mixing Circuit", "NULL", "RIN4"},
+
+	{"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"},
+	{"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"},
+	{"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"},
+	{"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"},
+	{"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"},
+	{"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"},
+
+	{"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"},
+	{"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"},
+	{"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"},
+	{"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"},
+	{"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"},
+	{"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"},
+
+	{"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"},
+	{"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"},
+	{"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"},
+	{"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"},
+	{"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"},
+	{"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"},
+
+	{"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"},
+	{"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"},
+	{"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"},
+	{"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"},
+	{"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"},
+	{"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"},
+};
+
+static int ak4671_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets,
+				  ARRAY_SIZE(ak4671_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	return 0;
+}
+
+static int ak4671_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;
+	u8 fs;
+
+	fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0);
+	fs &= ~AK4671_FS;
+
+	switch (params_rate(params)) {
+	case 8000:
+		fs |= AK4671_FS_8KHZ;
+		break;
+	case 12000:
+		fs |= AK4671_FS_12KHZ;
+		break;
+	case 16000:
+		fs |= AK4671_FS_16KHZ;
+		break;
+	case 24000:
+		fs |= AK4671_FS_24KHZ;
+		break;
+	case 11025:
+		fs |= AK4671_FS_11_025KHZ;
+		break;
+	case 22050:
+		fs |= AK4671_FS_22_05KHZ;
+		break;
+	case 32000:
+		fs |= AK4671_FS_32KHZ;
+		break;
+	case 44100:
+		fs |= AK4671_FS_44_1KHZ;
+		break;
+	case 48000:
+		fs |= AK4671_FS_48KHZ;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs);
+
+	return 0;
+}
+
+static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+		unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 pll;
+
+	pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0);
+	pll &= ~AK4671_PLL;
+
+	switch (freq) {
+	case 11289600:
+		pll |= AK4671_PLL_11_2896MHZ;
+		break;
+	case 12000000:
+		pll |= AK4671_PLL_12MHZ;
+		break;
+	case 12288000:
+		pll |= AK4671_PLL_12_288MHZ;
+		break;
+	case 13000000:
+		pll |= AK4671_PLL_13MHZ;
+		break;
+	case 13500000:
+		pll |= AK4671_PLL_13_5MHZ;
+		break;
+	case 19200000:
+		pll |= AK4671_PLL_19_2MHZ;
+		break;
+	case 24000000:
+		pll |= AK4671_PLL_24MHZ;
+		break;
+	case 26000000:
+		pll |= AK4671_PLL_26MHZ;
+		break;
+	case 27000000:
+		pll |= AK4671_PLL_27MHZ;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll);
+
+	return 0;
+}
+
+static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 mode;
+	u8 format;
+
+	/* set master/slave audio interface */
+	mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1);
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		mode |= AK4671_M_S;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		mode &= ~(AK4671_M_S);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	format = snd_soc_read(codec, AK4671_FORMAT_SELECT);
+	format &= ~AK4671_DIF;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		format |= AK4671_DIF_I2S_MODE;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		format |= AK4671_DIF_MSB_MODE;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		format |= AK4671_DIF_DSP_MODE;
+		format |= AK4671_BCKP;
+		format |= AK4671_MSBS;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set mode and format */
+	snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode);
+	snd_soc_write(codec, AK4671_FORMAT_SELECT, format);
+
+	return 0;
+}
+
+static int ak4671_set_bias_level(struct snd_soc_codec *codec,
+		enum snd_soc_bias_level level)
+{
+	u8 reg;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT);
+		snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT,
+				reg | AK4671_PMVCM);
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define AK4671_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)
+
+#define AK4671_FORMATS		SNDRV_PCM_FMTBIT_S16_LE
+
+static struct snd_soc_dai_ops ak4671_dai_ops = {
+	.hw_params	= ak4671_hw_params,
+	.set_sysclk	= ak4671_set_dai_sysclk,
+	.set_fmt	= ak4671_set_dai_fmt,
+};
+
+struct snd_soc_dai ak4671_dai = {
+	.name = "AK4671",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = AK4671_RATES,
+		.formats = AK4671_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = AK4671_RATES,
+		.formats = AK4671_FORMATS,},
+	.ops = &ak4671_dai_ops,
+};
+EXPORT_SYMBOL_GPL(ak4671_dai);
+
+static int ak4671_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (ak4671_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = ak4671_codec;
+	codec = ak4671_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, ak4671_snd_controls,
+			     ARRAY_SIZE(ak4671_snd_controls));
+	ak4671_add_widgets(codec);
+
+	ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return ret;
+
+pcm_err:
+	return ret;
+}
+
+static int ak4671_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_ak4671 = {
+	.probe = ak4671_probe,
+	.remove = ak4671_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671);
+
+static int ak4671_register(struct ak4671_priv *ak4671,
+		enum snd_soc_control_type control)
+{
+	int ret;
+	struct snd_soc_codec *codec = &ak4671->codec;
+
+	if (ak4671_codec) {
+		dev_err(codec->dev, "Another AK4671 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 = ak4671;
+	codec->name = "AK4671";
+	codec->owner = THIS_MODULE;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = ak4671_set_bias_level;
+	codec->dai = &ak4671_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = AK4671_CACHEREGNUM;
+	codec->reg_cache = &ak4671->reg_cache;
+
+	memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg));
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, control);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		goto err;
+	}
+
+	ak4671_dai.dev = codec->dev;
+	ak4671_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&ak4671_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(ak4671);
+	return ret;
+}
+
+static void ak4671_unregister(struct ak4671_priv *ak4671)
+{
+	ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&ak4671_dai);
+	snd_soc_unregister_codec(&ak4671->codec);
+	kfree(ak4671);
+	ak4671_codec = NULL;
+}
+
+static int __devinit ak4671_i2c_probe(struct i2c_client *client,
+		const struct i2c_device_id *id)
+{
+	struct ak4671_priv *ak4671;
+	struct snd_soc_codec *codec;
+
+	ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL);
+	if (ak4671 == NULL)
+		return -ENOMEM;
+
+	codec = &ak4671->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+
+	i2c_set_clientdata(client, ak4671);
+	codec->control_data = client;
+
+	codec->dev = &client->dev;
+
+	return ak4671_register(ak4671, SND_SOC_I2C);
+}
+
+static __devexit int ak4671_i2c_remove(struct i2c_client *client)
+{
+	struct ak4671_priv *ak4671 = i2c_get_clientdata(client);
+
+	ak4671_unregister(ak4671);
+
+	return 0;
+}
+
+static const struct i2c_device_id ak4671_i2c_id[] = {
+	{ "ak4671", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id);
+
+static struct i2c_driver ak4671_i2c_driver = {
+	.driver = {
+		.name = "ak4671",
+		.owner = THIS_MODULE,
+	},
+	.probe = ak4671_i2c_probe,
+	.remove = __devexit_p(ak4671_i2c_remove),
+	.id_table = ak4671_i2c_id,
+};
+
+static int __init ak4671_modinit(void)
+{
+	return i2c_add_driver(&ak4671_i2c_driver);
+}
+module_init(ak4671_modinit);
+
+static void __exit ak4671_exit(void)
+{
+	i2c_del_driver(&ak4671_i2c_driver);
+}
+module_exit(ak4671_exit);
+
+MODULE_DESCRIPTION("ASoC AK4671 codec driver");
+MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h
new file mode 100644
index 000000000000..e2fad964e88b
--- /dev/null
+++ b/sound/soc/codecs/ak4671.h
@@ -0,0 +1,156 @@
+/*
+ * ak4671.h  --  audio driver for AK4671
+ *
+ * Copyright (C) 2009 Samsung Electronics Co.Ltd
+ * Author: Joonyoung Shim <jy0922.shim@samsung.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef _AK4671_H
+#define _AK4671_H
+
+#define AK4671_AD_DA_POWER_MANAGEMENT		0x00
+#define AK4671_PLL_MODE_SELECT0			0x01
+#define AK4671_PLL_MODE_SELECT1			0x02
+#define AK4671_FORMAT_SELECT			0x03
+#define AK4671_MIC_SIGNAL_SELECT		0x04
+#define AK4671_MIC_AMP_GAIN			0x05
+#define AK4671_MIXING_POWER_MANAGEMENT0		0x06
+#define AK4671_MIXING_POWER_MANAGEMENT1		0x07
+#define AK4671_OUTPUT_VOLUME_CONTROL		0x08
+#define AK4671_LOUT1_SIGNAL_SELECT		0x09
+#define AK4671_ROUT1_SIGNAL_SELECT		0x0a
+#define AK4671_LOUT2_SIGNAL_SELECT		0x0b
+#define AK4671_ROUT2_SIGNAL_SELECT		0x0c
+#define AK4671_LOUT3_SIGNAL_SELECT		0x0d
+#define AK4671_ROUT3_SIGNAL_SELECT		0x0e
+#define AK4671_LOUT1_POWER_MANAGERMENT		0x0f
+#define AK4671_LOUT2_POWER_MANAGERMENT		0x10
+#define AK4671_LOUT3_POWER_MANAGERMENT		0x11
+#define AK4671_LCH_INPUT_VOLUME_CONTROL		0x12
+#define AK4671_RCH_INPUT_VOLUME_CONTROL		0x13
+#define AK4671_ALC_REFERENCE_SELECT		0x14
+#define AK4671_DIGITAL_MIXING_CONTROL		0x15
+#define AK4671_ALC_TIMER_SELECT			0x16
+#define AK4671_ALC_MODE_CONTROL			0x17
+#define AK4671_MODE_CONTROL1			0x18
+#define AK4671_MODE_CONTROL2			0x19
+#define AK4671_LCH_OUTPUT_VOLUME_CONTROL	0x1a
+#define AK4671_RCH_OUTPUT_VOLUME_CONTROL	0x1b
+#define AK4671_SIDETONE_A_CONTROL		0x1c
+#define AK4671_DIGITAL_FILTER_SELECT		0x1d
+#define AK4671_FIL3_COEFFICIENT0		0x1e
+#define AK4671_FIL3_COEFFICIENT1		0x1f
+#define AK4671_FIL3_COEFFICIENT2		0x20
+#define AK4671_FIL3_COEFFICIENT3		0x21
+#define AK4671_EQ_COEFFICIENT0			0x22
+#define AK4671_EQ_COEFFICIENT1			0x23
+#define AK4671_EQ_COEFFICIENT2			0x24
+#define AK4671_EQ_COEFFICIENT3			0x25
+#define AK4671_EQ_COEFFICIENT4			0x26
+#define AK4671_EQ_COEFFICIENT5			0x27
+#define AK4671_FIL1_COEFFICIENT0		0x28
+#define AK4671_FIL1_COEFFICIENT1		0x29
+#define AK4671_FIL1_COEFFICIENT2		0x2a
+#define AK4671_FIL1_COEFFICIENT3		0x2b
+#define AK4671_FIL2_COEFFICIENT0		0x2c
+#define AK4671_FIL2_COEFFICIENT1		0x2d
+#define AK4671_FIL2_COEFFICIENT2		0x2e
+#define AK4671_FIL2_COEFFICIENT3		0x2f
+#define AK4671_DIGITAL_FILTER_SELECT2		0x30
+#define AK4671_E1_COEFFICIENT0			0x32
+#define AK4671_E1_COEFFICIENT1			0x33
+#define AK4671_E1_COEFFICIENT2			0x34
+#define AK4671_E1_COEFFICIENT3			0x35
+#define AK4671_E1_COEFFICIENT4			0x36
+#define AK4671_E1_COEFFICIENT5			0x37
+#define AK4671_E2_COEFFICIENT0			0x38
+#define AK4671_E2_COEFFICIENT1			0x39
+#define AK4671_E2_COEFFICIENT2			0x3a
+#define AK4671_E2_COEFFICIENT3			0x3b
+#define AK4671_E2_COEFFICIENT4			0x3c
+#define AK4671_E2_COEFFICIENT5			0x3d
+#define AK4671_E3_COEFFICIENT0			0x3e
+#define AK4671_E3_COEFFICIENT1			0x3f
+#define AK4671_E3_COEFFICIENT2			0x40
+#define AK4671_E3_COEFFICIENT3			0x41
+#define AK4671_E3_COEFFICIENT4			0x42
+#define AK4671_E3_COEFFICIENT5			0x43
+#define AK4671_E4_COEFFICIENT0			0x44
+#define AK4671_E4_COEFFICIENT1			0x45
+#define AK4671_E4_COEFFICIENT2			0x46
+#define AK4671_E4_COEFFICIENT3			0x47
+#define AK4671_E4_COEFFICIENT4			0x48
+#define AK4671_E4_COEFFICIENT5			0x49
+#define AK4671_E5_COEFFICIENT0			0x4a
+#define AK4671_E5_COEFFICIENT1			0x4b
+#define AK4671_E5_COEFFICIENT2			0x4c
+#define AK4671_E5_COEFFICIENT3			0x4d
+#define AK4671_E5_COEFFICIENT4			0x4e
+#define AK4671_E5_COEFFICIENT5			0x4f
+#define AK4671_EQ_CONTROL_250HZ_100HZ		0x50
+#define AK4671_EQ_CONTROL_3500HZ_1KHZ		0x51
+#define AK4671_EQ_CONTRO_10KHZ			0x52
+#define AK4671_PCM_IF_CONTROL0			0x53
+#define AK4671_PCM_IF_CONTROL1			0x54
+#define AK4671_PCM_IF_CONTROL2			0x55
+#define AK4671_DIGITAL_VOLUME_B_CONTROL		0x56
+#define AK4671_DIGITAL_VOLUME_C_CONTROL		0x57
+#define AK4671_SIDETONE_VOLUME_CONTROL		0x58
+#define AK4671_DIGITAL_MIXING_CONTROL2		0x59
+#define AK4671_SAR_ADC_CONTROL			0x5a
+
+#define AK4671_CACHEREGNUM			(AK4671_SAR_ADC_CONTROL + 1)
+
+/* Bitfield Definitions */
+
+/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */
+#define AK4671_PMVCM				0x01
+
+/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */
+#define AK4671_PLL				0x0f
+#define AK4671_PLL_11_2896MHZ			(4 << 0)
+#define AK4671_PLL_12_288MHZ			(5 << 0)
+#define AK4671_PLL_12MHZ			(6 << 0)
+#define AK4671_PLL_24MHZ			(7 << 0)
+#define AK4671_PLL_19_2MHZ			(8 << 0)
+#define AK4671_PLL_13_5MHZ			(12 << 0)
+#define AK4671_PLL_27MHZ			(13 << 0)
+#define AK4671_PLL_13MHZ			(14 << 0)
+#define AK4671_PLL_26MHZ			(15 << 0)
+#define AK4671_FS				0xf0
+#define AK4671_FS_8KHZ				(0 << 4)
+#define AK4671_FS_12KHZ				(1 << 4)
+#define AK4671_FS_16KHZ				(2 << 4)
+#define AK4671_FS_24KHZ				(3 << 4)
+#define AK4671_FS_11_025KHZ			(5 << 4)
+#define AK4671_FS_22_05KHZ			(7 << 4)
+#define AK4671_FS_32KHZ				(10 << 4)
+#define AK4671_FS_48KHZ				(11 << 4)
+#define AK4671_FS_44_1KHZ			(15 << 4)
+
+/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */
+#define AK4671_PMPLL				0x01
+#define AK4671_M_S				0x02
+
+/* AK4671_FORMAT_SELECT (0x03) Fields */
+#define AK4671_DIF				0x03
+#define AK4671_DIF_DSP_MODE			(0 << 0)
+#define AK4671_DIF_MSB_MODE			(2 << 0)
+#define AK4671_DIF_I2S_MODE			(3 << 0)
+#define AK4671_BCKP				0x04
+#define AK4671_MSBS				0x08
+#define AK4671_SDOD				0x10
+
+/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */
+#define AK4671_MUTEN				0x04
+
+extern struct snd_soc_dai ak4671_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ak4671;
+
+#endif
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index ca1e24a8f12a..ffe122d1cd76 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -520,6 +520,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0),
 	SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0),
 	SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
+	SOC_SINGLE("De-emphasis filter", CS4270_TRANS, 0, 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, 1),
@@ -598,13 +599,6 @@ static int cs4270_probe(struct platform_device *pdev)
 		goto error_free_pcms;
 	}
 
-	/* And finally, register the socdev */
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(codec->dev, "failed to register card\n");
-		goto error_free_pcms;
-	}
-
 	return 0;
 
 error_free_pcms:
@@ -802,22 +796,6 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id);
  * 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;
-
-	return snd_soc_suspend_device(codec->dev);
-}
-
-static int cs4270_i2c_resume(struct i2c_client *client)
-{
-	struct cs4270_private *cs4270 = i2c_get_clientdata(client);
-	struct snd_soc_codec *codec = &cs4270->codec;
-
-	return snd_soc_resume_device(codec->dev);
-}
-
 static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg)
 {
 	struct snd_soc_codec *codec = cs4270_codec;
@@ -853,8 +831,6 @@ static int cs4270_soc_resume(struct platform_device *pdev)
 	return snd_soc_write(codec, CS4270_PWRCTL, reg);
 }
 #else
-#define cs4270_i2c_suspend	NULL
-#define cs4270_i2c_resume	NULL
 #define cs4270_soc_suspend	NULL
 #define cs4270_soc_resume	NULL
 #endif /* CONFIG_PM */
@@ -873,8 +849,6 @@ 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/cx20442.c b/sound/soc/codecs/cx20442.c
index 38eac9c866e1..e000cdfec1ec 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -93,7 +93,6 @@ static int cx20442_add_widgets(struct snd_soc_codec *codec)
 	snd_soc_dapm_add_routes(codec, cx20442_audio_map,
 				ARRAY_SIZE(cx20442_audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -355,17 +354,6 @@ static int cx20442_codec_probe(struct platform_device *pdev)
 
 	cx20442_add_widgets(codec);
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(&pdev->dev, "failed to register card\n");
-		goto card_err;
-	}
-
-	return ret;
-
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	return ret;
 }
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index 5cda9e6b5a74..2afcd0a8669d 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -90,13 +90,6 @@ static int pcm3008_soc_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	/* Register Card. */
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "pcm3008: failed to register card\n");
-		goto card_err;
-	}
-
 	/* DEM1  DEM0  DE-EMPHASIS_MODE
 	 * Low   Low   De-emphasis 44.1 kHz ON
 	 * Low   High  De-emphasis OFF
@@ -136,8 +129,6 @@ static int pcm3008_soc_probe(struct platform_device *pdev)
 
 gpio_err:
 	pcm3008_gpio_free(setup);
-card_err:
-	snd_soc_free_pcms(socdev);
 pcm_err:
 	kfree(socdev->card->codec);
 
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index c550750c79c0..d2ff1cde6883 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -210,7 +210,6 @@ static int ssm2602_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -613,17 +612,9 @@ static int ssm2602_init(struct snd_soc_device *socdev)
 	snd_soc_add_controls(codec, ssm2602_snd_controls,
 				ARRAY_SIZE(ssm2602_snd_controls));
 	ssm2602_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		pr_err("ssm2602: failed to register card\n");
-		goto card_err;
-	}
 
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	kfree(codec->reg_cache);
 	return ret;
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index befc6488c39a..bbc72c2ddfca 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -418,9 +418,6 @@ static int stac9766_codec_probe(struct platform_device *pdev)
 	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:
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 90a0264f7538..a9dc5fb54774 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
 	 * of data into val
 	 */
 
-	if ((reg < 0 || reg > 9) && (reg != 15)) {
+	if (reg > 9 && reg != 15) {
 		printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
 		return -1;
 	}
@@ -395,7 +395,6 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
 	/* set up audio path interconnects */
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -706,17 +705,9 @@ static int tlv320aic23_init(struct snd_soc_device *socdev)
 	snd_soc_add_controls(codec, tlv320aic23_snd_controls,
 				ARRAY_SIZE(tlv320aic23_snd_controls));
 	tlv320aic23_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "tlv320aic23: failed to register card\n");
-		goto card_err;
-	}
 
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	kfree(codec->reg_cache);
 	return ret;
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 3387d9e736ea..357b609196e3 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -356,18 +356,7 @@ static int aic26_probe(struct platform_device *pdev)
 			ARRAY_SIZE(aic26_snd_controls));
 	WARN_ON(err < 0);
 
-	/* CODEC is setup, we can register the card now */
-	dev_dbg(&pdev->dev, "Registering card\n");
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(&pdev->dev, "aic26: failed to register card\n");
-		goto card_err;
-	}
 	return 0;
-
- card_err:
-	snd_soc_free_pcms(socdev);
-	return ret;
 }
 
 static int aic26_remove(struct platform_device *pdev)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 3395cf945d56..2b4dc2b0b017 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -753,7 +753,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec)
 	/* set up audio path interconnects */
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -1405,18 +1404,8 @@ static int aic3x_probe(struct platform_device *pdev)
 
 	aic3x_add_widgets(codec);
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "aic3x: failed to register card\n");
-		goto card_err;
-	}
-
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-
 pcm_err:
 	kfree(codec->reg_cache);
 	return ret;
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
new file mode 100644
index 000000000000..9c8903dbe647
--- /dev/null
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -0,0 +1,1229 @@
+/*
+ * ALSA SoC Texas Instruments TLV320DAC33 codec driver
+ *
+ * Author:	Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *
+ * Copyright:   (C) 2009 Nokia Corporation
+ *
+ * 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/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/gpio.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/tlv320dac33-plat.h>
+#include "tlv320dac33.h"
+
+#define DAC33_BUFFER_SIZE_BYTES		24576	/* bytes, 12288 16 bit words,
+						 * 6144 stereo */
+#define DAC33_BUFFER_SIZE_SAMPLES	6144
+
+#define NSAMPLE_MAX		5700
+
+#define LATENCY_TIME_MS		20
+
+static struct snd_soc_codec *tlv320dac33_codec;
+
+enum dac33_state {
+	DAC33_IDLE = 0,
+	DAC33_PREFILL,
+	DAC33_PLAYBACK,
+	DAC33_FLUSH,
+};
+
+struct tlv320dac33_priv {
+	struct mutex mutex;
+	struct workqueue_struct *dac33_wq;
+	struct work_struct work;
+	struct snd_soc_codec codec;
+	int power_gpio;
+	int chip_power;
+	int irq;
+	unsigned int refclk;
+
+	unsigned int alarm_threshold;	/* set to be half of LATENCY_TIME_MS */
+	unsigned int nsample_min;	/* nsample should not be lower than
+					 * this */
+	unsigned int nsample_max;	/* nsample should not be higher than
+					 * this */
+	unsigned int nsample_switch;	/* Use FIFO or bypass FIFO switch */
+	unsigned int nsample;		/* burst read amount from host */
+
+	enum dac33_state state;
+};
+
+static const u8 dac33_reg[DAC33_CACHEREGNUM] = {
+0x00, 0x00, 0x00, 0x00, /* 0x00 - 0x03 */
+0x00, 0x00, 0x00, 0x00, /* 0x04 - 0x07 */
+0x00, 0x00, 0x00, 0x00, /* 0x08 - 0x0b */
+0x00, 0x00, 0x00, 0x00, /* 0x0c - 0x0f */
+0x00, 0x00, 0x00, 0x00, /* 0x10 - 0x13 */
+0x00, 0x00, 0x00, 0x00, /* 0x14 - 0x17 */
+0x00, 0x00, 0x00, 0x00, /* 0x18 - 0x1b */
+0x00, 0x00, 0x00, 0x00, /* 0x1c - 0x1f */
+0x00, 0x00, 0x00, 0x00, /* 0x20 - 0x23 */
+0x00, 0x00, 0x00, 0x00, /* 0x24 - 0x27 */
+0x00, 0x00, 0x00, 0x00, /* 0x28 - 0x2b */
+0x00, 0x00, 0x00, 0x80, /* 0x2c - 0x2f */
+0x80, 0x00, 0x00, 0x00, /* 0x30 - 0x33 */
+0x00, 0x00, 0x00, 0x00, /* 0x34 - 0x37 */
+0x00, 0x00,             /* 0x38 - 0x39 */
+/* Registers 0x3a - 0x3f are reserved  */
+            0x00, 0x00, /* 0x3a - 0x3b */
+0x00, 0x00, 0x00, 0x00, /* 0x3c - 0x3f */
+
+0x00, 0x00, 0x00, 0x00, /* 0x40 - 0x43 */
+0x00, 0x80,             /* 0x44 - 0x45 */
+/* Registers 0x46 - 0x47 are reserved  */
+            0x80, 0x80, /* 0x46 - 0x47 */
+
+0x80, 0x00, 0x00,       /* 0x48 - 0x4a */
+/* Registers 0x4b - 0x7c are reserved  */
+                  0x00, /* 0x4b        */
+0x00, 0x00, 0x00, 0x00, /* 0x4c - 0x4f */
+0x00, 0x00, 0x00, 0x00, /* 0x50 - 0x53 */
+0x00, 0x00, 0x00, 0x00, /* 0x54 - 0x57 */
+0x00, 0x00, 0x00, 0x00, /* 0x58 - 0x5b */
+0x00, 0x00, 0x00, 0x00, /* 0x5c - 0x5f */
+0x00, 0x00, 0x00, 0x00, /* 0x60 - 0x63 */
+0x00, 0x00, 0x00, 0x00, /* 0x64 - 0x67 */
+0x00, 0x00, 0x00, 0x00, /* 0x68 - 0x6b */
+0x00, 0x00, 0x00, 0x00, /* 0x6c - 0x6f */
+0x00, 0x00, 0x00, 0x00, /* 0x70 - 0x73 */
+0x00, 0x00, 0x00, 0x00, /* 0x74 - 0x77 */
+0x00, 0x00, 0x00, 0x00, /* 0x78 - 0x7b */
+0x00,                   /* 0x7c        */
+
+      0xda, 0x33, 0x03, /* 0x7d - 0x7f */
+};
+
+/* Register read and write */
+static inline unsigned int dac33_read_reg_cache(struct snd_soc_codec *codec,
+						unsigned reg)
+{
+	u8 *cache = codec->reg_cache;
+	if (reg >= DAC33_CACHEREGNUM)
+		return 0;
+
+	return cache[reg];
+}
+
+static inline void dac33_write_reg_cache(struct snd_soc_codec *codec,
+					 u8 reg, u8 value)
+{
+	u8 *cache = codec->reg_cache;
+	if (reg >= DAC33_CACHEREGNUM)
+		return;
+
+	cache[reg] = value;
+}
+
+static int dac33_read(struct snd_soc_codec *codec, unsigned int reg,
+		      u8 *value)
+{
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	int val;
+
+	*value = reg & 0xff;
+
+	/* If powered off, return the cached value */
+	if (dac33->chip_power) {
+		val = i2c_smbus_read_byte_data(codec->control_data, value[0]);
+		if (val < 0) {
+			dev_err(codec->dev, "Read failed (%d)\n", val);
+			value[0] = dac33_read_reg_cache(codec, reg);
+		} else {
+			value[0] = val;
+			dac33_write_reg_cache(codec, reg, val);
+		}
+	} else {
+		value[0] = dac33_read_reg_cache(codec, reg);
+	}
+
+	return 0;
+}
+
+static int dac33_write(struct snd_soc_codec *codec, unsigned int reg,
+		       unsigned int value)
+{
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	u8 data[2];
+	int ret = 0;
+
+	/*
+	 * data is
+	 *   D15..D8 dac33 register offset
+	 *   D7...D0 register data
+	 */
+	data[0] = reg & 0xff;
+	data[1] = value & 0xff;
+
+	dac33_write_reg_cache(codec, data[0], data[1]);
+	if (dac33->chip_power) {
+		ret = codec->hw_write(codec->control_data, data, 2);
+		if (ret != 2)
+			dev_err(codec->dev, "Write failed (%d)\n", ret);
+		else
+			ret = 0;
+	}
+
+	return ret;
+}
+
+static int dac33_write_locked(struct snd_soc_codec *codec, unsigned int reg,
+		       unsigned int value)
+{
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	int ret;
+
+	mutex_lock(&dac33->mutex);
+	ret = dac33_write(codec, reg, value);
+	mutex_unlock(&dac33->mutex);
+
+	return ret;
+}
+
+#define DAC33_I2C_ADDR_AUTOINC	0x80
+static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg,
+		       unsigned int value)
+{
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	u8 data[3];
+	int ret = 0;
+
+	/*
+	 * data is
+	 *   D23..D16 dac33 register offset
+	 *   D15..D8  register data MSB
+	 *   D7...D0  register data LSB
+	 */
+	data[0] = reg & 0xff;
+	data[1] = (value >> 8) & 0xff;
+	data[2] = value & 0xff;
+
+	dac33_write_reg_cache(codec, data[0], data[1]);
+	dac33_write_reg_cache(codec, data[0] + 1, data[2]);
+
+	if (dac33->chip_power) {
+		/* We need to set autoincrement mode for 16 bit writes */
+		data[0] |= DAC33_I2C_ADDR_AUTOINC;
+		ret = codec->hw_write(codec->control_data, data, 3);
+		if (ret != 3)
+			dev_err(codec->dev, "Write failed (%d)\n", ret);
+		else
+			ret = 0;
+	}
+
+	return ret;
+}
+
+static void dac33_restore_regs(struct snd_soc_codec *codec)
+{
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	u8 *cache = codec->reg_cache;
+	u8 data[2];
+	int i, ret;
+
+	if (!dac33->chip_power)
+		return;
+
+	for (i = DAC33_PWR_CTRL; i <= DAC33_INTP_CTRL_B; i++) {
+		data[0] = i;
+		data[1] = cache[i];
+		/* Skip the read only registers */
+		if ((i >= DAC33_INT_OSC_STATUS &&
+				i <= DAC33_INT_OSC_FREQ_RAT_READ_B) ||
+		    (i >= DAC33_FIFO_WPTR_MSB && i <= DAC33_FIFO_IRQ_FLAG) ||
+		    i == DAC33_DAC_STATUS_FLAGS ||
+		    i == DAC33_SRC_EST_REF_CLK_RATIO_A ||
+		    i == DAC33_SRC_EST_REF_CLK_RATIO_B)
+			continue;
+		ret = codec->hw_write(codec->control_data, data, 2);
+		if (ret != 2)
+			dev_err(codec->dev, "Write failed (%d)\n", ret);
+	}
+	for (i = DAC33_LDAC_PWR_CTRL; i <= DAC33_LINEL_TO_LLO_VOL; i++) {
+		data[0] = i;
+		data[1] = cache[i];
+		ret = codec->hw_write(codec->control_data, data, 2);
+		if (ret != 2)
+			dev_err(codec->dev, "Write failed (%d)\n", ret);
+	}
+	for (i = DAC33_LINER_TO_RLO_VOL; i <= DAC33_OSC_TRIM; i++) {
+		data[0] = i;
+		data[1] = cache[i];
+		ret = codec->hw_write(codec->control_data, data, 2);
+		if (ret != 2)
+			dev_err(codec->dev, "Write failed (%d)\n", ret);
+	}
+}
+
+static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
+{
+	u8 reg;
+
+	reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL);
+	if (power)
+		reg |= DAC33_PDNALLB;
+	else
+		reg &= ~DAC33_PDNALLB;
+	dac33_write(codec, DAC33_PWR_CTRL, reg);
+}
+
+static void dac33_hard_power(struct snd_soc_codec *codec, int power)
+{
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+
+	mutex_lock(&dac33->mutex);
+	if (power) {
+		if (dac33->power_gpio >= 0) {
+			gpio_set_value(dac33->power_gpio, 1);
+			dac33->chip_power = 1;
+			/* Restore registers */
+			dac33_restore_regs(codec);
+		}
+		dac33_soft_power(codec, 1);
+	} else {
+		dac33_soft_power(codec, 0);
+		if (dac33->power_gpio >= 0) {
+			gpio_set_value(dac33->power_gpio, 0);
+			dac33->chip_power = 0;
+		}
+	}
+	mutex_unlock(&dac33->mutex);
+
+}
+
+static int dac33_get_nsample(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+
+	ucontrol->value.integer.value[0] = dac33->nsample;
+
+	return 0;
+}
+
+static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	int ret = 0;
+
+	if (dac33->nsample == ucontrol->value.integer.value[0])
+		return 0;
+
+	if (ucontrol->value.integer.value[0] < dac33->nsample_min ||
+	    ucontrol->value.integer.value[0] > dac33->nsample_max)
+		ret = -EINVAL;
+	else
+		dac33->nsample = ucontrol->value.integer.value[0];
+
+	return ret;
+}
+
+static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+
+	ucontrol->value.integer.value[0] = dac33->nsample_switch;
+
+	return 0;
+}
+
+static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	int ret = 0;
+
+	if (dac33->nsample_switch == ucontrol->value.integer.value[0])
+		return 0;
+	/* Do not allow changes while stream is running*/
+	if (codec->active)
+		return -EPERM;
+
+	if (ucontrol->value.integer.value[0] < 0 ||
+	    ucontrol->value.integer.value[0] > 1)
+		ret = -EINVAL;
+	else
+		dac33->nsample_switch = ucontrol->value.integer.value[0];
+
+	return ret;
+}
+
+/*
+ * DACL/R digital volume control:
+ * from 0 dB to -63.5 in 0.5 dB steps
+ * Need to be inverted later on:
+ * 0x00 == 0 dB
+ * 0x7f == -63.5 dB
+ */
+static DECLARE_TLV_DB_SCALE(dac_digivol_tlv, -6350, 50, 0);
+
+static const struct snd_kcontrol_new dac33_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("DAC Digital Playback Volume",
+		DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL,
+		0, 0x7f, 1, dac_digivol_tlv),
+	SOC_DOUBLE_R("DAC Digital Playback Switch",
+		 DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, 7, 1, 1),
+	SOC_DOUBLE_R("Line to Line Out Volume",
+		 DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1),
+};
+
+static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = {
+	SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
+		 dac33_get_nsample, dac33_set_nsample),
+	SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0,
+		 dac33_get_nsample_switch, dac33_set_nsample_switch),
+};
+
+/* Analog bypass */
+static const struct snd_kcontrol_new dac33_dapm_abypassl_control =
+	SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1);
+
+static const struct snd_kcontrol_new dac33_dapm_abypassr_control =
+	SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1);
+
+static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("LEFT_LO"),
+	SND_SOC_DAPM_OUTPUT("RIGHT_LO"),
+
+	SND_SOC_DAPM_INPUT("LINEL"),
+	SND_SOC_DAPM_INPUT("LINER"),
+
+	SND_SOC_DAPM_DAC("DACL", "Left Playback", DAC33_LDAC_PWR_CTRL, 2, 0),
+	SND_SOC_DAPM_DAC("DACR", "Right Playback", DAC33_RDAC_PWR_CTRL, 2, 0),
+
+	/* Analog bypass */
+	SND_SOC_DAPM_SWITCH("Analog Left Bypass", SND_SOC_NOPM, 0, 0,
+				&dac33_dapm_abypassl_control),
+	SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0,
+				&dac33_dapm_abypassr_control),
+
+	SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amp Power",
+			 DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0),
+	SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power",
+			 DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Analog bypass */
+	{"Analog Left Bypass", "Switch", "LINEL"},
+	{"Analog Right Bypass", "Switch", "LINER"},
+
+	{"Output Left Amp Power", NULL, "DACL"},
+	{"Output Right Amp Power", NULL, "DACR"},
+
+	{"Output Left Amp Power", NULL, "Analog Left Bypass"},
+	{"Output Right Amp Power", NULL, "Analog Right Bypass"},
+
+	/* output */
+	{"LEFT_LO", NULL, "Output Left Amp Power"},
+	{"RIGHT_LO", NULL, "Output Right Amp Power"},
+};
+
+static int dac33_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, dac33_dapm_widgets,
+				  ARRAY_SIZE(dac33_dapm_widgets));
+
+	/* set up audio path interconnects */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	return 0;
+}
+
+static int dac33_set_bias_level(struct snd_soc_codec *codec,
+				enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		dac33_soft_power(codec, 1);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF)
+			dac33_hard_power(codec, 1);
+		dac33_soft_power(codec, 0);
+		break;
+	case SND_SOC_BIAS_OFF:
+		dac33_hard_power(codec, 0);
+		break;
+	}
+	codec->bias_level = level;
+
+	return 0;
+}
+
+static void dac33_work(struct work_struct *work)
+{
+	struct snd_soc_codec *codec;
+	struct tlv320dac33_priv *dac33;
+	u8 reg;
+
+	dac33 = container_of(work, struct tlv320dac33_priv, work);
+	codec = &dac33->codec;
+
+	mutex_lock(&dac33->mutex);
+	switch (dac33->state) {
+	case DAC33_PREFILL:
+		dac33->state = DAC33_PLAYBACK;
+		dac33_write16(codec, DAC33_NSAMPLE_MSB,
+				DAC33_THRREG(dac33->nsample));
+		dac33_write16(codec, DAC33_PREFILL_MSB,
+				DAC33_THRREG(dac33->alarm_threshold));
+		break;
+	case DAC33_PLAYBACK:
+		dac33_write16(codec, DAC33_NSAMPLE_MSB,
+				DAC33_THRREG(dac33->nsample));
+		break;
+	case DAC33_IDLE:
+		break;
+	case DAC33_FLUSH:
+		dac33->state = DAC33_IDLE;
+		/* Mask all interrupts from dac33 */
+		dac33_write(codec, DAC33_FIFO_IRQ_MASK, 0);
+
+		/* flush fifo */
+		reg = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A);
+		reg |= DAC33_FIFOFLUSH;
+		dac33_write(codec, DAC33_FIFO_CTRL_A, reg);
+		break;
+	}
+	mutex_unlock(&dac33->mutex);
+}
+
+static irqreturn_t dac33_interrupt_handler(int irq, void *dev)
+{
+	struct snd_soc_codec *codec = dev;
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+
+	queue_work(dac33->dac33_wq, &dac33->work);
+
+	return IRQ_HANDLED;
+}
+
+static void dac33_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;
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	unsigned int pwr_ctrl;
+
+	/* Stop pending workqueue */
+	if (dac33->nsample_switch)
+		cancel_work_sync(&dac33->work);
+
+	mutex_lock(&dac33->mutex);
+	pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL);
+	pwr_ctrl &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB);
+	dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl);
+	mutex_unlock(&dac33->mutex);
+}
+
+static void dac33_oscwait(struct snd_soc_codec *codec)
+{
+	int timeout = 20;
+	u8 reg;
+
+	do {
+		msleep(1);
+		dac33_read(codec, DAC33_INT_OSC_STATUS, &reg);
+	} while (((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) && timeout--);
+	if ((reg & 0x03) != DAC33_OSCSTATUS_NORMAL)
+		dev_err(codec->dev,
+			"internal oscillator calibration failed\n");
+}
+
+static int dac33_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;
+
+	/* Check parameters for validity */
+	switch (params_rate(params)) {
+	case 44100:
+	case 48000:
+		break;
+	default:
+		dev_err(codec->dev, "unsupported rate %d\n",
+			params_rate(params));
+		return -EINVAL;
+	}
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	default:
+		dev_err(codec->dev, "unsupported format %d\n",
+			params_format(params));
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+#define CALC_OSCSET(rate, refclk) ( \
+	((((rate * 10000) / refclk) * 4096) + 5000) / 10000)
+#define CALC_RATIOSET(rate, refclk) ( \
+	((((refclk  * 100000) / rate) * 16384) + 50000) / 100000)
+
+/*
+ * tlv320dac33 is strict on the sequence of the register writes, if the register
+ * writes happens in different order, than dac33 might end up in unknown state.
+ * Use the known, working sequence of register writes to initialize the dac33.
+ */
+static int dac33_prepare_chip(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	unsigned int oscset, ratioset, pwr_ctrl, reg_tmp;
+	u8 aictrl_a, fifoctrl_a;
+
+	switch (substream->runtime->rate) {
+	case 44100:
+	case 48000:
+		oscset = CALC_OSCSET(substream->runtime->rate, dac33->refclk);
+		ratioset = CALC_RATIOSET(substream->runtime->rate,
+					 dac33->refclk);
+		break;
+	default:
+		dev_err(codec->dev, "unsupported rate %d\n",
+			substream->runtime->rate);
+		return -EINVAL;
+	}
+
+
+	aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A);
+	aictrl_a &= ~(DAC33_NCYCL_MASK | DAC33_WLEN_MASK);
+	fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A);
+	fifoctrl_a &= ~DAC33_WIDTH;
+	switch (substream->runtime->format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		aictrl_a |= (DAC33_NCYCL_16 | DAC33_WLEN_16);
+		fifoctrl_a |= DAC33_WIDTH;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported format %d\n",
+			substream->runtime->format);
+		return -EINVAL;
+	}
+
+	mutex_lock(&dac33->mutex);
+	dac33_soft_power(codec, 1);
+
+	reg_tmp = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL);
+	dac33_write(codec, DAC33_INT_OSC_CTRL, reg_tmp);
+
+	/* Write registers 0x08 and 0x09 (MSB, LSB) */
+	dac33_write16(codec, DAC33_INT_OSC_FREQ_RAT_A, oscset);
+
+	/* calib time: 128 is a nice number ;) */
+	dac33_write(codec, DAC33_CALIB_TIME, 128);
+
+	/* adjustment treshold & step */
+	dac33_write(codec, DAC33_INT_OSC_CTRL_B, DAC33_ADJTHRSHLD(2) |
+						 DAC33_ADJSTEP(1));
+
+	/* div=4 / gain=1 / div */
+	dac33_write(codec, DAC33_INT_OSC_CTRL_C, DAC33_REFDIV(4));
+
+	pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL);
+	pwr_ctrl |= DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB;
+	dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl);
+
+	dac33_oscwait(codec);
+
+	if (dac33->nsample_switch) {
+		/* 50-51 : ASRC Control registers */
+		dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */
+		dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */
+
+		/* Write registers 0x34 and 0x35 (MSB, LSB) */
+		dac33_write16(codec, DAC33_SRC_REF_CLK_RATIO_A, ratioset);
+
+		/* Set interrupts to high active */
+		dac33_write(codec, DAC33_INTP_CTRL_A, DAC33_INTPM_AHIGH);
+
+		dac33_write(codec, DAC33_FIFO_IRQ_MODE_B,
+			    DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL));
+		dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT);
+	} else {
+		/* 50-51 : ASRC Control registers */
+		dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCBYP);
+		dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */
+	}
+
+	if (dac33->nsample_switch)
+		fifoctrl_a &= ~DAC33_FBYPAS;
+	else
+		fifoctrl_a |= DAC33_FBYPAS;
+	dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a);
+
+	dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a);
+	reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B);
+	if (dac33->nsample_switch)
+		reg_tmp &= ~DAC33_BCLKON;
+	else
+		reg_tmp |= DAC33_BCLKON;
+	dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp);
+
+	if (dac33->nsample_switch) {
+		/* 20: BCLK divide ratio */
+		dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3);
+
+		dac33_write16(codec, DAC33_ATHR_MSB,
+			      DAC33_THRREG(dac33->alarm_threshold));
+	} else {
+		dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
+	}
+
+	mutex_unlock(&dac33->mutex);
+
+	return 0;
+}
+
+static void dac33_calculate_times(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	struct tlv320dac33_priv *dac33 = codec->private_data;
+	unsigned int nsample_limit;
+
+	/* Number of samples (16bit, stereo) in one period */
+	dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4;
+
+	/* Number of samples (16bit, stereo) in ALSA buffer */
+	dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4;
+	/* Subtract one period from the total */
+	dac33->nsample_max -= dac33->nsample_min;
+
+	/* Number of samples for LATENCY_TIME_MS / 2 */
+	dac33->alarm_threshold = substream->runtime->rate /
+				 (1000 / (LATENCY_TIME_MS / 2));
+
+	/* Find and fix up the lowest nsmaple limit */
+	nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS);
+
+	if (dac33->nsample_min < nsample_limit)
+		dac33->nsample_min = nsample_limit;
+
+	if (dac33->nsample < dac33->nsample_min)
+		dac33->nsample = dac33->nsample_min;
+
+	/*
+	 * Find and fix up the highest nsmaple limit
+	 * In order to not overflow the DAC33 buffer substract the
+	 * alarm_threshold value from the size of the DAC33 buffer
+	 */
+	nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold;
+
+	if (dac33->nsample_max > nsample_limit)
+		dac33->nsample_max = nsample_limit;
+
+	if (dac33->nsample > dac33->nsample_max)
+		dac33->nsample = dac33->nsample_max;
+}
+
+static int dac33_pcm_prepare(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	dac33_calculate_times(substream);
+	dac33_prepare_chip(substream);
+
+	return 0;
+}
+
+static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
+			     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 tlv320dac33_priv *dac33 = codec->private_data;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (dac33->nsample_switch) {
+			dac33->state = DAC33_PREFILL;
+			queue_work(dac33->dac33_wq, &dac33->work);
+		}
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (dac33->nsample_switch) {
+			dac33->state = DAC33_FLUSH;
+			queue_work(dac33->dac33_wq, &dac33->work);
+		}
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int dac33_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 tlv320dac33_priv *dac33 = codec->private_data;
+	u8 ioc_reg, asrcb_reg;
+
+	ioc_reg = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL);
+	asrcb_reg = dac33_read_reg_cache(codec, DAC33_ASRC_CTRL_B);
+	switch (clk_id) {
+	case TLV320DAC33_MCLK:
+		ioc_reg |= DAC33_REFSEL;
+		asrcb_reg |= DAC33_SRCREFSEL;
+		break;
+	case TLV320DAC33_SLEEPCLK:
+		ioc_reg &= ~DAC33_REFSEL;
+		asrcb_reg &= ~DAC33_SRCREFSEL;
+		break;
+	default:
+		dev_err(codec->dev, "Invalid clock ID (%d)\n", clk_id);
+		break;
+	}
+	dac33->refclk = freq;
+
+	dac33_write_reg_cache(codec, DAC33_INT_OSC_CTRL, ioc_reg);
+	dac33_write_reg_cache(codec, DAC33_ASRC_CTRL_B, asrcb_reg);
+
+	return 0;
+}
+
+static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			     unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 aictrl_a, aictrl_b;
+
+	aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A);
+	aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B);
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		/* Codec Master */
+		aictrl_a |= (DAC33_MSBCLK | DAC33_MSWCLK);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		/* Codec Slave */
+		aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	aictrl_a &= ~DAC33_AFMT_MASK;
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		aictrl_a |= DAC33_AFMT_I2S;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		aictrl_a |= DAC33_AFMT_DSP;
+		aictrl_b &= ~DAC33_DATA_DELAY_MASK;
+		aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		aictrl_a |= DAC33_AFMT_DSP;
+		aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		aictrl_a |= DAC33_AFMT_RIGHT_J;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		aictrl_a |= DAC33_AFMT_LEFT_J;
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported format (%u)\n",
+			fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a);
+	dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b);
+
+	return 0;
+}
+
+static void dac33_init_chip(struct snd_soc_codec *codec)
+{
+	/* 44-46: DAC Control Registers */
+	/* A : DAC sample rate Fsref/1.5 */
+	dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1));
+	/* B : DAC src=normal, not muted */
+	dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT |
+					     DAC33_DACSRCL_LEFT);
+	/* C : (defaults) */
+	dac33_write(codec, DAC33_DAC_CTRL_C, 0x00);
+
+	/* 64-65 : L&R DAC power control
+	 Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/
+	dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
+	dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2));
+
+	/* 73 : volume soft stepping control,
+	 clock source = internal osc (?) */
+	dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN);
+
+	/* 66 : LOP/LOM Modes */
+	dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff);
+
+	/* 68 : LOM inverted from LOP */
+	dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2));
+
+	dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB);
+}
+
+static int dac33_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	struct tlv320dac33_priv *dac33;
+	int ret = 0;
+
+	BUG_ON(!tlv320dac33_codec);
+
+	codec = tlv320dac33_codec;
+	socdev->card->codec = codec;
+	dac33 = codec->private_data;
+
+	/* Power up the codec */
+	dac33_hard_power(codec, 1);
+	/* Set default configuration */
+	dac33_init_chip(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\n");
+		goto pcm_err;
+	}
+
+	snd_soc_add_controls(codec, dac33_snd_controls,
+			     ARRAY_SIZE(dac33_snd_controls));
+	/* Only add the nSample controls, if we have valid IRQ number */
+	if (dac33->irq >= 0)
+		snd_soc_add_controls(codec, dac33_nsample_snd_controls,
+				     ARRAY_SIZE(dac33_nsample_snd_controls));
+
+	dac33_add_widgets(codec);
+
+	/* power on device */
+	dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+
+pcm_err:
+	dac33_hard_power(codec, 0);
+	return ret;
+}
+
+static int dac33_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	dac33_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+static int dac33_soc_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;
+
+	dac33_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int dac33_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	dac33_set_bias_level(codec, codec->suspend_bias_level);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_tlv320dac33 = {
+	.probe = dac33_soc_probe,
+	.remove = dac33_soc_remove,
+	.suspend = dac33_soc_suspend,
+	.resume = dac33_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33);
+
+#define DAC33_RATES	(SNDRV_PCM_RATE_44100 | \
+			 SNDRV_PCM_RATE_48000)
+#define DAC33_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
+
+static struct snd_soc_dai_ops dac33_dai_ops = {
+	.shutdown	= dac33_shutdown,
+	.hw_params	= dac33_hw_params,
+	.prepare	= dac33_pcm_prepare,
+	.trigger	= dac33_pcm_trigger,
+	.set_sysclk	= dac33_set_dai_sysclk,
+	.set_fmt	= dac33_set_dai_fmt,
+};
+
+struct snd_soc_dai dac33_dai = {
+	.name = "tlv320dac33",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = DAC33_RATES,
+		.formats = DAC33_FORMATS,},
+	.ops = &dac33_dai_ops,
+};
+EXPORT_SYMBOL_GPL(dac33_dai);
+
+static int dac33_i2c_probe(struct i2c_client *client,
+			   const struct i2c_device_id *id)
+{
+	struct tlv320dac33_platform_data *pdata;
+	struct tlv320dac33_priv *dac33;
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (client->dev.platform_data == NULL) {
+		dev_err(&client->dev, "Platform data not set\n");
+		return -ENODEV;
+	}
+	pdata = client->dev.platform_data;
+
+	dac33 = kzalloc(sizeof(struct tlv320dac33_priv), GFP_KERNEL);
+	if (dac33 == NULL)
+		return -ENOMEM;
+
+	codec = &dac33->codec;
+	codec->private_data = dac33;
+	codec->control_data = client;
+
+	mutex_init(&codec->mutex);
+	mutex_init(&dac33->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->name = "tlv320dac33";
+	codec->owner = THIS_MODULE;
+	codec->read = dac33_read_reg_cache;
+	codec->write = dac33_write_locked;
+	codec->hw_write = (hw_write_t) i2c_master_send;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = dac33_set_bias_level;
+	codec->dai = &dac33_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(dac33_reg);
+	codec->reg_cache = kmemdup(dac33_reg, ARRAY_SIZE(dac33_reg),
+				   GFP_KERNEL);
+	if (codec->reg_cache == NULL) {
+		ret = -ENOMEM;
+		goto error_reg;
+	}
+
+	i2c_set_clientdata(client, dac33);
+
+	dac33->power_gpio = pdata->power_gpio;
+	dac33->irq = client->irq;
+	dac33->nsample = NSAMPLE_MAX;
+	/* Disable FIFO use by default */
+	dac33->nsample_switch = 0;
+
+	tlv320dac33_codec = codec;
+
+	codec->dev = &client->dev;
+	dac33_dai.dev = codec->dev;
+
+	/* Check if the reset GPIO number is valid and request it */
+	if (dac33->power_gpio >= 0) {
+		ret = gpio_request(dac33->power_gpio, "tlv320dac33 reset");
+		if (ret < 0) {
+			dev_err(codec->dev,
+				"Failed to request reset GPIO (%d)\n",
+				dac33->power_gpio);
+			snd_soc_unregister_dai(&dac33_dai);
+			snd_soc_unregister_codec(codec);
+			goto error_gpio;
+		}
+		gpio_direction_output(dac33->power_gpio, 0);
+	} else {
+		dac33->chip_power = 1;
+	}
+
+	/* Check if the IRQ number is valid and request it */
+	if (dac33->irq >= 0) {
+		ret = request_irq(dac33->irq, dac33_interrupt_handler,
+				  IRQF_TRIGGER_RISING | IRQF_DISABLED,
+				  codec->name, codec);
+		if (ret < 0) {
+			dev_err(codec->dev, "Could not request IRQ%d (%d)\n",
+						dac33->irq, ret);
+			dac33->irq = -1;
+		}
+		if (dac33->irq != -1) {
+			/* Setup work queue */
+			dac33->dac33_wq =
+				create_singlethread_workqueue("tlv320dac33");
+			if (dac33->dac33_wq == NULL) {
+				free_irq(dac33->irq, &dac33->codec);
+				ret = -ENOMEM;
+				goto error_wq;
+			}
+
+			INIT_WORK(&dac33->work, dac33_work);
+		}
+	}
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto error_codec;
+	}
+
+	ret = snd_soc_register_dai(&dac33_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		goto error_codec;
+	}
+
+	/* Shut down the codec for now */
+	dac33_hard_power(codec, 0);
+
+	return ret;
+
+error_codec:
+	if (dac33->irq >= 0) {
+		free_irq(dac33->irq, &dac33->codec);
+		destroy_workqueue(dac33->dac33_wq);
+	}
+error_wq:
+	if (dac33->power_gpio >= 0)
+		gpio_free(dac33->power_gpio);
+error_gpio:
+	kfree(codec->reg_cache);
+error_reg:
+	tlv320dac33_codec = NULL;
+	kfree(dac33);
+
+	return ret;
+}
+
+static int dac33_i2c_remove(struct i2c_client *client)
+{
+	struct tlv320dac33_priv *dac33;
+
+	dac33 = i2c_get_clientdata(client);
+	dac33_hard_power(&dac33->codec, 0);
+
+	if (dac33->power_gpio >= 0)
+		gpio_free(dac33->power_gpio);
+	if (dac33->irq >= 0)
+		free_irq(dac33->irq, &dac33->codec);
+
+	destroy_workqueue(dac33->dac33_wq);
+	snd_soc_unregister_dai(&dac33_dai);
+	snd_soc_unregister_codec(&dac33->codec);
+	kfree(dac33->codec.reg_cache);
+	kfree(dac33);
+	tlv320dac33_codec = NULL;
+
+	return 0;
+}
+
+static const struct i2c_device_id tlv320dac33_i2c_id[] = {
+	{
+		.name = "tlv320dac33",
+		.driver_data = 0,
+	},
+	{ },
+};
+
+static struct i2c_driver tlv320dac33_i2c_driver = {
+	.driver = {
+		.name = "tlv320dac33",
+		.owner = THIS_MODULE,
+	},
+	.probe		= dac33_i2c_probe,
+	.remove		= __devexit_p(dac33_i2c_remove),
+	.id_table	= tlv320dac33_i2c_id,
+};
+
+static int __init dac33_module_init(void)
+{
+	int r;
+	r = i2c_add_driver(&tlv320dac33_i2c_driver);
+	if (r < 0) {
+		printk(KERN_ERR "DAC33: driver registration failed\n");
+		return r;
+	}
+	return 0;
+}
+module_init(dac33_module_init);
+
+static void __exit dac33_module_exit(void)
+{
+	i2c_del_driver(&tlv320dac33_i2c_driver);
+}
+module_exit(dac33_module_exit);
+
+
+MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver");
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@nokia.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h
new file mode 100644
index 000000000000..eb8ae07f0bd2
--- /dev/null
+++ b/sound/soc/codecs/tlv320dac33.h
@@ -0,0 +1,267 @@
+/*
+ * ALSA SoC Texas Instruments TLV320DAC33 codec driver
+ *
+ * Author:	Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *
+ * Copyright:   (C) 2009 Nokia Corporation
+ *
+ * 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
+ *
+ */
+
+#ifndef __TLV320DAC33_H
+#define __TLV320DAC33_H
+
+#define DAC33_PAGE_SELECT		0x00
+#define DAC33_PWR_CTRL			0x01
+#define DAC33_PLL_CTRL_A		0x02
+#define DAC33_PLL_CTRL_B		0x03
+#define DAC33_PLL_CTRL_C		0x04
+#define DAC33_PLL_CTRL_D		0x05
+#define DAC33_PLL_CTRL_E		0x06
+#define DAC33_INT_OSC_CTRL		0x07
+#define DAC33_INT_OSC_FREQ_RAT_A	0x08
+#define DAC33_INT_OSC_FREQ_RAT_B	0x09
+#define DAC33_INT_OSC_DAC_RATIO_SET	0x0A
+#define DAC33_CALIB_TIME		0x0B
+#define DAC33_INT_OSC_CTRL_B		0x0C
+#define DAC33_INT_OSC_CTRL_C		0x0D
+#define DAC33_INT_OSC_STATUS		0x0E
+#define DAC33_INT_OSC_DAC_RATIO_READ	0x0F
+#define DAC33_INT_OSC_FREQ_RAT_READ_A	0x10
+#define DAC33_INT_OSC_FREQ_RAT_READ_B	0x11
+#define DAC33_SER_AUDIOIF_CTRL_A	0x12
+#define DAC33_SER_AUDIOIF_CTRL_B	0x13
+#define DAC33_SER_AUDIOIF_CTRL_C	0x14
+#define DAC33_FIFO_CTRL_A		0x15
+#define DAC33_UTHR_MSB			0x16
+#define DAC33_UTHR_LSB			0x17
+#define DAC33_ATHR_MSB			0x18
+#define DAC33_ATHR_LSB			0x19
+#define DAC33_LTHR_MSB			0x1A
+#define DAC33_LTHR_LSB			0x1B
+#define DAC33_PREFILL_MSB		0x1C
+#define DAC33_PREFILL_LSB		0x1D
+#define DAC33_NSAMPLE_MSB		0x1E
+#define DAC33_NSAMPLE_LSB		0x1F
+#define DAC33_FIFO_WPTR_MSB		0x20
+#define DAC33_FIFO_WPTR_LSB		0x21
+#define DAC33_FIFO_RPTR_MSB		0x22
+#define DAC33_FIFO_RPTR_LSB		0x23
+#define DAC33_FIFO_DEPTH_MSB		0x24
+#define DAC33_FIFO_DEPTH_LSB		0x25
+#define DAC33_SAMPLES_REMAINING_MSB	0x26
+#define DAC33_SAMPLES_REMAINING_LSB	0x27
+#define DAC33_FIFO_IRQ_FLAG		0x28
+#define DAC33_FIFO_IRQ_MASK		0x29
+#define DAC33_FIFO_IRQ_MODE_A		0x2A
+#define DAC33_FIFO_IRQ_MODE_B		0x2B
+#define DAC33_DAC_CTRL_A		0x2C
+#define DAC33_DAC_CTRL_B		0x2D
+#define DAC33_DAC_CTRL_C		0x2E
+#define DAC33_LDAC_DIG_VOL_CTRL		0x2F
+#define DAC33_RDAC_DIG_VOL_CTRL		0x30
+#define DAC33_DAC_STATUS_FLAGS		0x31
+#define DAC33_ASRC_CTRL_A		0x32
+#define DAC33_ASRC_CTRL_B		0x33
+#define DAC33_SRC_REF_CLK_RATIO_A	0x34
+#define DAC33_SRC_REF_CLK_RATIO_B	0x35
+#define DAC33_SRC_EST_REF_CLK_RATIO_A	0x36
+#define DAC33_SRC_EST_REF_CLK_RATIO_B	0x37
+#define DAC33_INTP_CTRL_A		0x38
+#define DAC33_INTP_CTRL_B		0x39
+/* Registers 0x3A - 0x3F Reserved */
+#define DAC33_LDAC_PWR_CTRL		0x40
+#define DAC33_RDAC_PWR_CTRL		0x41
+#define DAC33_OUT_AMP_CM_CTRL		0x42
+#define DAC33_OUT_AMP_PWR_CTRL		0x43
+#define DAC33_OUT_AMP_CTRL		0x44
+#define DAC33_LINEL_TO_LLO_VOL		0x45
+/* Registers 0x45 - 0x47 Reserved */
+#define DAC33_LINER_TO_RLO_VOL		0x48
+#define DAC33_ANA_VOL_SOFT_STEP_CTRL	0x49
+#define DAC33_OSC_TRIM			0x4A
+/* Registers 0x4B - 0x7C Reserved */
+#define DAC33_DEVICE_ID_MSB		0x7D
+#define DAC33_DEVICE_ID_LSB		0x7E
+#define DAC33_DEVICE_REV_ID		0x7F
+
+#define DAC33_CACHEREGNUM               128
+
+/* Bit definitions */
+
+/* DAC33_PWR_CTRL (0x01) */
+#define DAC33_DACRPDNB			(0x01 << 0)
+#define DAC33_DACLPDNB			(0x01 << 1)
+#define DAC33_OSCPDNB			(0x01 << 2)
+#define DAC33_PLLPDNB			(0x01 << 3)
+#define DAC33_PDNALLB			(0x01 << 4)
+#define DAC33_SOFT_RESET		(0x01 << 7)
+
+/* DAC33_INT_OSC_CTRL (0x07) */
+#define DAC33_REFSEL			(0x01 << 1)
+
+/* DAC33_INT_OSC_CTRL_B (0x0C) */
+#define DAC33_ADJSTEP(x)		(x << 0)
+#define DAC33_ADJTHRSHLD(x)		(x << 4)
+
+/* DAC33_INT_OSC_CTRL_C (0x0D) */
+#define DAC33_REFDIV(x)			(x << 4)
+
+/* DAC33_INT_OSC_STATUS (0x0E) */
+#define DAC33_OSCSTATUS_IDLE_CALIB	(0x00)
+#define DAC33_OSCSTATUS_NORMAL		(0x01)
+#define DAC33_OSCSTATUS_ADJUSTMENT	(0x03)
+#define DAC33_OSCSTATUS_NOT_USED	(0x02)
+
+/* DAC33_SER_AUDIOIF_CTRL_A (0x12) */
+#define DAC33_MSWCLK			(0x01 << 0)
+#define DAC33_MSBCLK			(0x01 << 1)
+#define DAC33_AFMT_MASK			(0x03 << 2)
+#define DAC33_AFMT_I2S			(0x00 << 2)
+#define DAC33_AFMT_DSP			(0x01 << 2)
+#define DAC33_AFMT_RIGHT_J		(0x02 << 2)
+#define DAC33_AFMT_LEFT_J		(0x03 << 2)
+#define DAC33_WLEN_MASK			(0x03 << 4)
+#define DAC33_WLEN_16			(0x00 << 4)
+#define DAC33_WLEN_20			(0x01 << 4)
+#define DAC33_WLEN_24			(0x02 << 4)
+#define DAC33_WLEN_32			(0x03 << 4)
+#define DAC33_NCYCL_MASK		(0x03 << 6)
+#define DAC33_NCYCL_16			(0x00 << 6)
+#define DAC33_NCYCL_20			(0x01 << 6)
+#define DAC33_NCYCL_24			(0x02 << 6)
+#define DAC33_NCYCL_32			(0x03 << 6)
+
+/* DAC33_SER_AUDIOIF_CTRL_B (0x13) */
+#define DAC33_DATA_DELAY_MASK		(0x03 << 2)
+#define DAC33_DATA_DELAY(x)		(x << 2)
+#define DAC33_BCLKON			(0x01 << 5)
+
+/* DAC33_FIFO_CTRL_A (0x15) */
+#define DAC33_WIDTH				(0x01 << 0)
+#define DAC33_FBYPAS				(0x01 << 1)
+#define DAC33_FAUTO				(0x01 << 2)
+#define DAC33_FIFOFLUSH			(0x01 << 3)
+
+/*
+ * UTHR, ATHR, LTHR, PREFILL, NSAMPLE (0x16 - 0x1F)
+ * 13-bit values
+*/
+#define DAC33_THRREG(x)			(((x) & 0x1FFF) << 3)
+
+/* DAC33_FIFO_IRQ_MASK (0x29) */
+#define DAC33_MNS			(0x01 << 0)
+#define DAC33_MPS			(0x01 << 1)
+#define DAC33_MAT			(0x01 << 2)
+#define DAC33_MLT			(0x01 << 3)
+#define DAC33_MUT			(0x01 << 4)
+#define DAC33_MUF			(0x01 << 5)
+#define DAC33_MOF			(0x01 << 6)
+
+#define DAC33_FIFO_IRQ_MODE_MASK	(0x03)
+#define DAC33_FIFO_IRQ_MODE_RISING	(0x00)
+#define DAC33_FIFO_IRQ_MODE_FALLING	(0x01)
+#define DAC33_FIFO_IRQ_MODE_LEVEL	(0x02)
+#define DAC33_FIFO_IRQ_MODE_EDGE	(0x03)
+
+/* DAC33_FIFO_IRQ_MODE_A (0x2A) */
+#define DAC33_UTM(x)			(x << 0)
+#define DAC33_UFM(x)			(x << 2)
+#define DAC33_OFM(x)			(x << 4)
+
+/* DAC33_FIFO_IRQ_MODE_B (0x2B) */
+#define DAC33_NSM(x)			(x << 0)
+#define DAC33_PSM(x)			(x << 2)
+#define DAC33_ATM(x)			(x << 4)
+#define DAC33_LTM(x)			(x << 6)
+
+/* DAC33_DAC_CTRL_A (0x2C) */
+#define DAC33_DACRATE(x)		(x << 0)
+#define DAC33_DACDUAL			(0x01 << 4)
+#define DAC33_DACLKSEL_MASK		(0x03 << 5)
+#define DAC33_DACLKSEL_INTSOC		(0x00 << 5)
+#define DAC33_DACLKSEL_PLL		(0x01 << 5)
+#define DAC33_DACLKSEL_MCLK		(0x02 << 5)
+#define DAC33_DACLKSEL_BCLK		(0x03 << 5)
+
+/* DAC33_DAC_CTRL_B (0x2D) */
+#define DAC33_DACSRCR_MASK		(0x03 << 0)
+#define DAC33_DACSRCR_MUTE		(0x00 << 0)
+#define DAC33_DACSRCR_RIGHT		(0x01 << 0)
+#define DAC33_DACSRCR_LEFT		(0x02 << 0)
+#define DAC33_DACSRCR_MONOMIX		(0x03 << 0)
+#define DAC33_DACSRCL_MASK		(0x03 << 2)
+#define DAC33_DACSRCL_MUTE		(0x00 << 2)
+#define DAC33_DACSRCL_LEFT		(0x01 << 2)
+#define DAC33_DACSRCL_RIGHT		(0x02 << 2)
+#define DAC33_DACSRCL_MONOMIX		(0x03 << 2)
+#define DAC33_DVOLSTEP_MASK		(0x03 << 4)
+#define DAC33_DVOLSTEP_SS_PERFS		(0x00 << 4)
+#define DAC33_DVOLSTEP_SS_PER2FS	(0x01 << 4)
+#define DAC33_DVOLSTEP_SS_DISABLED	(0x02 << 4)
+#define DAC33_DVOLCTRL_MASK		(0x03 << 6)
+#define DAC33_DVOLCTRL_LR_INDEPENDENT1	(0x00 << 6)
+#define DAC33_DVOLCTRL_LR_RIGHT_CONTROL	(0x01 << 6)
+#define DAC33_DVOLCTRL_LR_LEFT_CONTROL	(0x02 << 6)
+#define DAC33_DVOLCTRL_LR_INDEPENDENT2	(0x03 << 6)
+
+/* DAC33_DAC_CTRL_C (0x2E) */
+#define DAC33_DEEMENR			(0x01 << 0)
+#define DAC33_EFFENR			(0x01 << 1)
+#define DAC33_DEEMENL			(0x01 << 2)
+#define DAC33_EFFENL			(0x01 << 3)
+#define DAC33_EN3D			(0x01 << 4)
+#define DAC33_RESYNMUTE			(0x01 << 5)
+#define DAC33_RESYNEN			(0x01 << 6)
+
+/* DAC33_ASRC_CTRL_A (0x32) */
+#define DAC33_SRCBYP			(0x01 << 0)
+#define DAC33_SRCLKSEL_MASK		(0x03 << 1)
+#define DAC33_SRCLKSEL_INTSOC		(0x00 << 1)
+#define DAC33_SRCLKSEL_PLL		(0x01 << 1)
+#define DAC33_SRCLKSEL_MCLK		(0x02 << 1)
+#define DAC33_SRCLKSEL_BCLK		(0x03 << 1)
+#define DAC33_SRCLKDIV(x)		(x << 3)
+
+/* DAC33_ASRC_CTRL_B (0x33) */
+#define DAC33_SRCSETUP(x)		(x << 0)
+#define DAC33_SRCREFSEL			(0x01 << 4)
+#define DAC33_SRCREFDIV(x)		(x << 5)
+
+/* DAC33_INTP_CTRL_A (0x38) */
+#define DAC33_INTPSEL			(0x01 << 0)
+#define DAC33_INTPM_MASK		(0x03 << 1)
+#define DAC33_INTPM_ALOW_OPENDRAIN	(0x00 << 1)
+#define DAC33_INTPM_ALOW		(0x01 << 1)
+#define DAC33_INTPM_AHIGH		(0x02 << 1)
+
+/* DAC33_LDAC_PWR_CTRL (0x40) */
+/* DAC33_RDAC_PWR_CTRL (0x41) */
+#define DAC33_DACLRNUM			(0x01 << 2)
+#define DAC33_LROUT_GAIN(x)		(x << 0)
+
+/* DAC33_ANA_VOL_SOFT_STEP_CTRL (0x49) */
+#define DAC33_VOLCLKSEL			(0x01 << 0)
+#define DAC33_VOLCLKEN			(0x01 << 1)
+#define DAC33_VOLBYPASS			(0x01 << 2)
+
+#define TLV320DAC33_MCLK		0
+#define TLV320DAC33_SLEEPCLK		1
+
+extern struct snd_soc_dai dac33_dai;
+extern struct snd_soc_codec_device soc_codec_dev_tlv320dac33;
+
+#endif /* __TLV320DAC33_H */
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
new file mode 100644
index 000000000000..6b650c1aa3d1
--- /dev/null
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -0,0 +1,463 @@
+/*
+ * ALSA SoC Texas Instruments TPA6130A2 headset stereo amplifier driver
+ *
+ * Copyright (C) Nokia Corporation
+ *
+ * Author: 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
+ * 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/errno.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <sound/tpa6130a2-plat.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "tpa6130a2.h"
+
+static struct i2c_client *tpa6130a2_client;
+
+/* This struct is used to save the context */
+struct tpa6130a2_data {
+	struct mutex mutex;
+	unsigned char regs[TPA6130A2_CACHEREGNUM];
+	int power_gpio;
+	unsigned char power_state;
+};
+
+static int tpa6130a2_i2c_read(int reg)
+{
+	struct tpa6130a2_data *data;
+	int val;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	/* If powered off, return the cached value */
+	if (data->power_state) {
+		val = i2c_smbus_read_byte_data(tpa6130a2_client, reg);
+		if (val < 0)
+			dev_err(&tpa6130a2_client->dev, "Read failed\n");
+		else
+			data->regs[reg] = val;
+	} else {
+		val = data->regs[reg];
+	}
+
+	return val;
+}
+
+static int tpa6130a2_i2c_write(int reg, u8 value)
+{
+	struct tpa6130a2_data *data;
+	int val = 0;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	if (data->power_state) {
+		val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value);
+		if (val < 0)
+			dev_err(&tpa6130a2_client->dev, "Write failed\n");
+	}
+
+	/* Either powered on or off, we save the context */
+	data->regs[reg] = value;
+
+	return val;
+}
+
+static u8 tpa6130a2_read(int reg)
+{
+	struct tpa6130a2_data *data;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	return data->regs[reg];
+}
+
+static void tpa6130a2_initialize(void)
+{
+	struct tpa6130a2_data *data;
+	int i;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	for (i = 1; i < TPA6130A2_REG_VERSION; i++)
+		tpa6130a2_i2c_write(i, data->regs[i]);
+}
+
+static void tpa6130a2_power(int power)
+{
+	struct	tpa6130a2_data *data;
+	u8	val;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	mutex_lock(&data->mutex);
+	if (power) {
+		/* Power on */
+		if (data->power_gpio >= 0) {
+			gpio_set_value(data->power_gpio, 1);
+			data->power_state = 1;
+			tpa6130a2_initialize();
+		}
+		/* Clear SWS */
+		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
+		val &= ~TPA6130A2_SWS;
+		tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val);
+	} else {
+		/* set SWS */
+		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
+		val |= TPA6130A2_SWS;
+		tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val);
+		/* Power off */
+		if (data->power_gpio >= 0) {
+			gpio_set_value(data->power_gpio, 0);
+			data->power_state = 0;
+		}
+	}
+	mutex_unlock(&data->mutex);
+}
+
+static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct tpa6130a2_data *data;
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	unsigned int mask = mc->max;
+	unsigned int invert = mc->invert;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	mutex_lock(&data->mutex);
+
+	ucontrol->value.integer.value[0] =
+		(tpa6130a2_read(reg) >> shift) & mask;
+
+	if (invert)
+		ucontrol->value.integer.value[0] =
+			mask - ucontrol->value.integer.value[0];
+
+	mutex_unlock(&data->mutex);
+	return 0;
+}
+
+static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct tpa6130a2_data *data;
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	unsigned int mask = mc->max;
+	unsigned int invert = mc->invert;
+	unsigned int val = (ucontrol->value.integer.value[0] & mask);
+	unsigned int val_reg;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	if (invert)
+		val = mask - val;
+
+	mutex_lock(&data->mutex);
+
+	val_reg = tpa6130a2_read(reg);
+	if (((val_reg >> shift) & mask) == val) {
+		mutex_unlock(&data->mutex);
+		return 0;
+	}
+
+	val_reg &= ~(mask << shift);
+	val_reg |= val << shift;
+	tpa6130a2_i2c_write(reg, val_reg);
+
+	mutex_unlock(&data->mutex);
+
+	return 1;
+}
+
+/*
+ * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going
+ * down in gain.
+ */
+static const unsigned int tpa6130_tlv[] = {
+	TLV_DB_RANGE_HEAD(10),
+	0, 1, TLV_DB_SCALE_ITEM(-5950, 600, 0),
+	2, 3, TLV_DB_SCALE_ITEM(-5000, 250, 0),
+	4, 5, TLV_DB_SCALE_ITEM(-4550, 160, 0),
+	6, 7, TLV_DB_SCALE_ITEM(-4140, 190, 0),
+	8, 9, TLV_DB_SCALE_ITEM(-3650, 120, 0),
+	10, 11, TLV_DB_SCALE_ITEM(-3330, 160, 0),
+	12, 13, TLV_DB_SCALE_ITEM(-3040, 180, 0),
+	14, 20, TLV_DB_SCALE_ITEM(-2710, 110, 0),
+	21, 37, TLV_DB_SCALE_ITEM(-1960, 74, 0),
+	38, 63, TLV_DB_SCALE_ITEM(-720, 45, 0),
+};
+
+static const struct snd_kcontrol_new tpa6130a2_controls[] = {
+	SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume",
+		       TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0,
+		       tpa6130a2_get_reg, tpa6130a2_set_reg,
+		       tpa6130_tlv),
+};
+
+/*
+ * Enable or disable channel (left or right)
+ * The bit number for mute and amplifier are the same per channel:
+ * bit 6: Right channel
+ * bit 7: Left channel
+ * in both registers.
+ */
+static void tpa6130a2_channel_enable(u8 channel, int enable)
+{
+	struct	tpa6130a2_data *data;
+	u8	val;
+
+	BUG_ON(tpa6130a2_client == NULL);
+	data = i2c_get_clientdata(tpa6130a2_client);
+
+	if (enable) {
+		/* Enable channel */
+		/* Enable amplifier */
+		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
+		val |= channel;
+		tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val);
+
+		/* Unmute channel */
+		val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE);
+		val &= ~channel;
+		tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val);
+	} else {
+		/* Disable channel */
+		/* Mute channel */
+		val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE);
+		val |= channel;
+		tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val);
+
+		/* Disable amplifier */
+		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
+		val &= ~channel;
+		tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val);
+	}
+}
+
+static int tpa6130a2_left_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 1);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 0);
+		break;
+	}
+	return 0;
+}
+
+static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 1);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 0);
+		break;
+	}
+	return 0;
+}
+
+static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		tpa6130a2_power(1);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		tpa6130a2_power(0);
+		break;
+	}
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = {
+	SND_SOC_DAPM_PGA_E("TPA6130A2 Left", SND_SOC_NOPM,
+			0, 0, NULL, 0, tpa6130a2_left_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("TPA6130A2 Right", SND_SOC_NOPM,
+			0, 0, NULL, 0, tpa6130a2_right_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("TPA6130A2 Enable", SND_SOC_NOPM,
+			0, 0, tpa6130a2_supply_event,
+			SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+	/* Outputs */
+	SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL),
+	SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Left"},
+	{"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Right"},
+
+	{"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Enable"},
+	{"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Enable"},
+};
+
+int tpa6130a2_add_controls(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets,
+				ARRAY_SIZE(tpa6130a2_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	return snd_soc_add_controls(codec, tpa6130a2_controls,
+				ARRAY_SIZE(tpa6130a2_controls));
+
+}
+EXPORT_SYMBOL_GPL(tpa6130a2_add_controls);
+
+static int tpa6130a2_probe(struct i2c_client *client,
+			   const struct i2c_device_id *id)
+{
+	struct device *dev;
+	struct tpa6130a2_data *data;
+	struct tpa6130a2_platform_data *pdata;
+	int ret;
+
+	dev = &client->dev;
+
+	if (client->dev.platform_data == NULL) {
+		dev_err(dev, "Platform data not set\n");
+		dump_stack();
+		return -ENODEV;
+	}
+
+	data = kzalloc(sizeof(*data), GFP_KERNEL);
+	if (data == NULL) {
+		dev_err(dev, "Can not allocate memory\n");
+		return -ENOMEM;
+	}
+
+	tpa6130a2_client = client;
+
+	i2c_set_clientdata(tpa6130a2_client, data);
+
+	pdata = client->dev.platform_data;
+	data->power_gpio = pdata->power_gpio;
+
+	mutex_init(&data->mutex);
+
+	/* Set default register values */
+	data->regs[TPA6130A2_REG_CONTROL] =	TPA6130A2_SWS;
+	data->regs[TPA6130A2_REG_VOL_MUTE] =	TPA6130A2_MUTE_R |
+						TPA6130A2_MUTE_L;
+
+	if (data->power_gpio >= 0) {
+		ret = gpio_request(data->power_gpio, "tpa6130a2 enable");
+		if (ret < 0) {
+			dev_err(dev, "Failed to request power GPIO (%d)\n",
+				data->power_gpio);
+			goto fail;
+		}
+		gpio_direction_output(data->power_gpio, 0);
+	} else {
+		data->power_state = 1;
+		tpa6130a2_initialize();
+	}
+
+	tpa6130a2_power(1);
+
+	/* Read version */
+	ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) &
+				 TPA6130A2_VERSION_MASK;
+	if ((ret != 1) && (ret != 2))
+		dev_warn(dev, "UNTESTED version detected (%d)\n", ret);
+
+	/* Disable the chip */
+	tpa6130a2_power(0);
+
+	return 0;
+fail:
+	kfree(data);
+	i2c_set_clientdata(tpa6130a2_client, NULL);
+	tpa6130a2_client = NULL;
+
+	return ret;
+}
+
+static int tpa6130a2_remove(struct i2c_client *client)
+{
+	struct tpa6130a2_data *data = i2c_get_clientdata(client);
+
+	tpa6130a2_power(0);
+
+	if (data->power_gpio >= 0)
+		gpio_free(data->power_gpio);
+	kfree(data);
+	tpa6130a2_client = NULL;
+
+	return 0;
+}
+
+static const struct i2c_device_id tpa6130a2_id[] = {
+	{ "tpa6130a2", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, tpa6130a2_id);
+
+static struct i2c_driver tpa6130a2_i2c_driver = {
+	.driver = {
+		.name = "tpa6130a2",
+		.owner = THIS_MODULE,
+	},
+	.probe = tpa6130a2_probe,
+	.remove = __devexit_p(tpa6130a2_remove),
+	.id_table = tpa6130a2_id,
+};
+
+static int __init tpa6130a2_init(void)
+{
+	return i2c_add_driver(&tpa6130a2_i2c_driver);
+}
+
+static void __exit tpa6130a2_exit(void)
+{
+	i2c_del_driver(&tpa6130a2_i2c_driver);
+}
+
+MODULE_AUTHOR("Peter Ujfalusi");
+MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver");
+MODULE_LICENSE("GPL");
+
+module_init(tpa6130a2_init);
+module_exit(tpa6130a2_exit);
diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h
new file mode 100644
index 000000000000..57e867fd86d1
--- /dev/null
+++ b/sound/soc/codecs/tpa6130a2.h
@@ -0,0 +1,61 @@
+/*
+ * ALSA SoC TPA6130A2 amplifier driver
+ *
+ * Copyright (C) Nokia Corporation
+ *
+ * Author: 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
+ * 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
+ *
+ */
+
+#ifndef __TPA6130A2_H__
+#define __TPA6130A2_H__
+
+/* Register addresses */
+#define TPA6130A2_REG_CONTROL		0x01
+#define TPA6130A2_REG_VOL_MUTE		0x02
+#define TPA6130A2_REG_OUT_IMPEDANCE	0x03
+#define TPA6130A2_REG_VERSION		0x04
+
+#define TPA6130A2_CACHEREGNUM	(TPA6130A2_REG_VERSION + 1)
+
+/* Register bits */
+/* TPA6130A2_REG_CONTROL (0x01) */
+#define TPA6130A2_SWS			(0x01 << 0)
+#define TPA6130A2_TERMAL		(0x01 << 1)
+#define TPA6130A2_MODE(x)		(x << 4)
+#define TPA6130A2_MODE_STEREO		(0x00)
+#define TPA6130A2_MODE_DUAL_MONO	(0x01)
+#define TPA6130A2_MODE_BRIDGE		(0x02)
+#define TPA6130A2_MODE_MASK		(0x03)
+#define TPA6130A2_HP_EN_R		(0x01 << 6)
+#define TPA6130A2_HP_EN_L		(0x01 << 7)
+
+/* TPA6130A2_REG_VOL_MUTE (0x02) */
+#define TPA6130A2_VOLUME(x)		((x & 0x3f) << 0)
+#define TPA6130A2_MUTE_R		(0x01 << 6)
+#define TPA6130A2_MUTE_L		(0x01 << 7)
+
+/* TPA6130A2_REG_OUT_IMPEDANCE (0x03) */
+#define TPA6130A2_HIZ_R			(0x01 << 0)
+#define TPA6130A2_HIZ_L			(0x01 << 1)
+
+/* TPA6130A2_REG_VERSION (0x04) */
+#define TPA6130A2_VERSION_MASK		(0x0f)
+
+extern int tpa6130a2_add_controls(struct snd_soc_codec *codec);
+
+#endif /* __TPA6130A2_H__ */
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 4df7c6c61c76..5f1681f6ca76 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -120,9 +120,10 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 
 /* codec private data */
 struct twl4030_priv {
-	unsigned int bypass_state;
+	struct snd_soc_codec codec;
+
 	unsigned int codec_powered;
-	unsigned int codec_muted;
+	unsigned int apll_enabled;
 
 	struct snd_pcm_substream *master_substream;
 	struct snd_pcm_substream *slave_substream;
@@ -183,19 +184,20 @@ static int twl4030_write(struct snd_soc_codec *codec,
 static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 {
 	struct twl4030_priv *twl4030 = codec->private_data;
-	u8 mode;
+	int mode;
 
 	if (enable == twl4030->codec_powered)
 		return;
 
-	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
 	if (enable)
-		mode |= TWL4030_CODECPDZ;
+		mode = twl4030_codec_enable_resource(TWL4030_CODEC_RES_POWER);
 	else
-		mode &= ~TWL4030_CODECPDZ;
+		mode = twl4030_codec_disable_resource(TWL4030_CODEC_RES_POWER);
 
-	twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
-	twl4030->codec_powered = enable;
+	if (mode >= 0) {
+		twl4030_write_reg_cache(codec, TWL4030_REG_CODEC_MODE, mode);
+		twl4030->codec_powered = enable;
+	}
 
 	/* REVISIT: this delay is present in TI sample drivers */
 	/* but there seems to be no TRM requirement for it     */
@@ -212,31 +214,30 @@ 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,	cache[i]);
+		if (i != TWL4030_REG_APLL_CTL)
+			twl4030_write(codec, i,	cache[i]);
 
 }
 
-static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute)
+static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable)
 {
 	struct twl4030_priv *twl4030 = codec->private_data;
-	u8 reg_val;
+	int status;
 
-	if (mute == twl4030->codec_muted)
+	if (enable == twl4030->apll_enabled)
 		return;
 
-	if (mute) {
-		/* Disable PLL */
-		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
-		reg_val &= ~TWL4030_APLL_EN;
-		twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
-	} else {
+	if (enable)
 		/* Enable PLL */
-		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
-		reg_val |= TWL4030_APLL_EN;
-		twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
-	}
+		status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL);
+	else
+		/* Disable PLL */
+		status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL);
+
+	if (status >= 0)
+		twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status);
 
-	twl4030->codec_muted = mute;
+	twl4030->apll_enabled = enable;
 }
 
 static void twl4030_power_up(struct snd_soc_codec *codec)
@@ -613,6 +614,27 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int vibramux_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff);
+	return 0;
+}
+
+static int apll_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		twl4030_apll_enable(w->codec, 1);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		twl4030_apll_enable(w->codec, 0);
+		break;
+	}
+	return 0;
+}
+
 static void headset_ramp(struct snd_soc_codec *codec, int ramp)
 {
 	struct snd_soc_device *socdev = codec->socdev;
@@ -724,67 +746,6 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static int bypass_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
-{
-	struct soc_mixer_control *m =
-		(struct soc_mixer_control *)w->kcontrols->private_value;
-	struct twl4030_priv *twl4030 = w->codec->private_data;
-	unsigned char reg, misc;
-
-	reg = twl4030_read_reg_cache(w->codec, m->reg);
-
-	/*
-	 * bypass_state[0:3] - analog HiFi bypass
-	 * bypass_state[4]   - analog voice bypass
-	 * bypass_state[5]   - digital voice bypass
-	 * bypass_state[6:7] - digital HiFi bypass
-	 */
-	if (m->reg == TWL4030_REG_VSTPGA) {
-		/* Voice digital bypass */
-		if (reg)
-			twl4030->bypass_state |= (1 << 5);
-		else
-			twl4030->bypass_state &= ~(1 << 5);
-	} else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) {
-		/* Analog bypass */
-		if (reg & (1 << m->shift))
-			twl4030->bypass_state |=
-				(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
-		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 {
-		/* Digital bypass */
-		if (reg & (0x7 << m->shift))
-			twl4030->bypass_state |= (1 << (m->shift ? 7 : 6));
-		else
-			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);
-		else
-			twl4030_codec_mute(w->codec, 1);
-	}
-	return 0;
-}
-
 /*
  * Some of the gain controls in TWL (mostly those which are associated with
  * the outputs) are implemented in an interesting way:
@@ -1192,32 +1153,28 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 			SND_SOC_NOPM, 0, 0),
 
 	/* Analog bypasses */
-	SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
-			&twl4030_dapm_abypassr1_control, bypass_event,
-			SND_SOC_DAPM_POST_REG),
-	SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0,
-			&twl4030_dapm_abypassl1_control,
-			bypass_event, SND_SOC_DAPM_POST_REG),
-	SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0,
-			&twl4030_dapm_abypassr2_control,
-			bypass_event, SND_SOC_DAPM_POST_REG),
-	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),
+	SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassr1_control),
+	SND_SOC_DAPM_SWITCH("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassl1_control),
+	SND_SOC_DAPM_SWITCH("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassr2_control),
+	SND_SOC_DAPM_SWITCH("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassl2_control),
+	SND_SOC_DAPM_SWITCH("Voice Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassv_control),
+
+	/* Master analog loopback switch */
+	SND_SOC_DAPM_SUPPLY("FM Loop Enable", TWL4030_REG_MISC_SET_1, 5, 0,
+			    NULL, 0),
 
 	/* Digital bypasses */
-	SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
-			&twl4030_dapm_dbypassl_control, bypass_event,
-			SND_SOC_DAPM_POST_REG),
-	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_SWITCH("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassl_control),
+	SND_SOC_DAPM_SWITCH("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassr_control),
+	SND_SOC_DAPM_SWITCH("Voice Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassv_control),
 
 	/* Digital mixers, power control for the physical DACs */
 	SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer",
@@ -1243,6 +1200,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer",
 			TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0),
 
+	SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event,
+			    SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
+
 	/* Output MIXER controls */
 	/* Earpiece */
 	SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
@@ -1308,8 +1268,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 			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_E("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0,
+			   &twl4030_dapm_vibra_control, vibramux_event,
+			   SND_SOC_DAPM_PRE_PMU),
 	SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_vibrapath_control),
 
@@ -1369,6 +1330,13 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Digital R2 Playback Mixer", NULL, "DAC Right2"},
 	{"Digital Voice Playback Mixer", NULL, "DAC Voice"},
 
+	/* Supply for the digital part (APLL) */
+	{"Digital R1 Playback Mixer", NULL, "APLL Enable"},
+	{"Digital L1 Playback Mixer", NULL, "APLL Enable"},
+	{"Digital R2 Playback Mixer", NULL, "APLL Enable"},
+	{"Digital L2 Playback Mixer", NULL, "APLL Enable"},
+	{"Digital Voice Playback Mixer", NULL, "APLL Enable"},
+
 	{"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"},
@@ -1482,6 +1450,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"ADC Virtual Left2", NULL, "TX2 Capture Route"},
 	{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
 
+	{"ADC Virtual Left1", NULL, "APLL Enable"},
+	{"ADC Virtual Right1", NULL, "APLL Enable"},
+	{"ADC Virtual Left2", NULL, "APLL Enable"},
+	{"ADC Virtual Right2", NULL, "APLL Enable"},
+
 	/* Analog bypass routes */
 	{"Right1 Analog Loopback", "Switch", "Analog Right"},
 	{"Left1 Analog Loopback", "Switch", "Analog Left"},
@@ -1489,6 +1462,13 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Left2 Analog Loopback", "Switch", "Analog Left"},
 	{"Voice Analog Loopback", "Switch", "Analog Left"},
 
+	/* Supply for the Analog loopbacks */
+	{"Right1 Analog Loopback", NULL, "FM Loop Enable"},
+	{"Left1 Analog Loopback", NULL, "FM Loop Enable"},
+	{"Right2 Analog Loopback", NULL, "FM Loop Enable"},
+	{"Left2 Analog Loopback", NULL, "FM Loop Enable"},
+	{"Voice Analog Loopback", NULL, "FM Loop Enable"},
+
 	{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
 	{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
 	{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
@@ -1513,32 +1493,20 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
 static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
-	struct twl4030_priv *twl4030 = codec->private_data;
-
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		twl4030_codec_mute(codec, 0);
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		twl4030_power_up(codec);
-		if (twl4030->bypass_state)
-			twl4030_codec_mute(codec, 0);
-		else
-			twl4030_codec_mute(codec, 1);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		twl4030_power_up(codec);
-		if (twl4030->bypass_state)
-			twl4030_codec_mute(codec, 0);
-		else
-			twl4030_codec_mute(codec, 1);
+		if (codec->bias_level == SND_SOC_BIAS_OFF)
+			twl4030_power_up(codec);
 		break;
 	case SND_SOC_BIAS_OFF:
 		twl4030_power_down(codec);
@@ -1785,29 +1753,23 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 {
 	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",
-			freq);
+		dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq);
 		return -EINVAL;
 	}
 
-	infreq |= TWL4030_APLL_EN;
-	twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+	if ((freq / 1000) != twl4030->sysclk) {
+		dev_err(codec->dev,
+			"Mismatch in APLL mclk: %u (configured: %u)\n",
+			freq, twl4030->sysclk * 1000);
+		return -EINVAL;
+	}
 
 	return 0;
 }
@@ -1905,18 +1867,16 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
-	u8 infreq;
+	struct twl4030_priv *twl4030 = codec->private_data;
 	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");
+	if (twl4030->sysclk != 26000) {
+		dev_err(codec->dev, "The board is configured for %u Hz, while"
+			"the Voice interface needs 26MHz APLL mclk\n",
+			twl4030->sysclk * 1000);
 		return -EINVAL;
 	}
 
@@ -1989,21 +1949,19 @@ 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;
+	struct twl4030_priv *twl4030 = codec->private_data;
 
-	switch (freq) {
-	case 26000000:
-		infreq = TWL4030_APLL_INFREQ_26000KHZ;
-		break;
-	default:
-		printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n",
-			freq);
+	if (freq != 26000000) {
+		dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice"
+			"interface needs 26MHz APLL mclk\n", freq);
+		return -EINVAL;
+	}
+	if ((freq / 1000) != twl4030->sysclk) {
+		dev_err(codec->dev,
+			"Mismatch in APLL mclk: %u (configured: %u)\n",
+			freq, twl4030->sysclk * 1000);
 		return -EINVAL;
 	}
-
-	infreq |= TWL4030_APLL_EN;
-	twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
-
 	return 0;
 }
 
@@ -2121,7 +2079,7 @@ struct snd_soc_dai twl4030_dai[] = {
 };
 EXPORT_SYMBOL_GPL(twl4030_dai);
 
-static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
+static int twl4030_soc_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;
@@ -2131,7 +2089,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
 	return 0;
 }
 
-static int twl4030_resume(struct platform_device *pdev)
+static int twl4030_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
@@ -2141,147 +2099,181 @@ static int twl4030_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialize the driver
- * register the mixer and dsp interfaces with the kernel
- */
+static struct snd_soc_codec *twl4030_codec;
 
-static int twl4030_init(struct snd_soc_device *socdev)
+static int twl4030_soc_probe(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->card->codec;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct twl4030_setup_data *setup = socdev->codec_data;
-	struct twl4030_priv *twl4030 = codec->private_data;
-	int ret = 0;
+	struct snd_soc_codec *codec;
+	struct twl4030_priv *twl4030;
+	int ret;
 
-	printk(KERN_INFO "TWL4030 Audio Codec init \n");
+	BUG_ON(!twl4030_codec);
 
-	codec->name = "twl4030";
-	codec->owner = THIS_MODULE;
-	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 = 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;
+	codec = twl4030_codec;
+	twl4030 = codec->private_data;
+	socdev->card->codec = codec;
 
 	/* 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;
+		if (setup->sysclk != twl4030->sysclk)
+			dev_warn(&pdev->dev,
+				 "Mismatch in APLL mclk: %u (configured: %u)\n",
+				 setup->sysclk, twl4030->sysclk);
 
 		hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
 		hs_pop &= ~TWL4030_RAMP_DELAY;
 		hs_pop |= (setup->ramp_delay_value << 2);
 		twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-	} else {
-		twl4030->sysclk = 26000;
 	}
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "twl4030: failed to create pcms\n");
-		goto pcm_err;
+		dev_err(&pdev->dev, "failed to create pcms\n");
+		return ret;
 	}
 
-	twl4030_init_chip(codec);
-
-	/* power on device */
-	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	snd_soc_add_controls(codec, twl4030_snd_controls,
 				ARRAY_SIZE(twl4030_snd_controls));
 	twl4030_add_widgets(codec);
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "twl4030: failed to register card\n");
-		goto card_err;
-	}
+	return 0;
+}
 
-	return ret;
+static int twl4030_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
 
-card_err:
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-pcm_err:
-	kfree(codec->reg_cache);
-	return ret;
-}
+	kfree(codec->private_data);
+	kfree(codec);
 
-static struct snd_soc_device *twl4030_socdev;
+	return 0;
+}
 
-static int twl4030_probe(struct platform_device *pdev)
+static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data;
 	struct snd_soc_codec *codec;
 	struct twl4030_priv *twl4030;
+	int ret;
 
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
+	if (!pdata) {
+		dev_err(&pdev->dev, "platform_data is missing\n");
+		return -EINVAL;
+	}
 
 	twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL);
 	if (twl4030 == NULL) {
-		kfree(codec);
+		dev_err(&pdev->dev, "Can not allocate memroy\n");
 		return -ENOMEM;
 	}
 
+	codec = &twl4030->codec;
 	codec->private_data = twl4030;
-	socdev->card->codec = codec;
+	codec->dev = &pdev->dev;
+	twl4030_dai[0].dev = &pdev->dev;
+	twl4030_dai[1].dev = &pdev->dev;
+
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	twl4030_socdev = socdev;
-	twl4030_init(socdev);
+	codec->name = "twl4030";
+	codec->owner = THIS_MODULE;
+	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 = 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) {
+		ret = -ENOMEM;
+		goto error_cache;
+	}
+
+	platform_set_drvdata(pdev, twl4030);
+	twl4030_codec = codec;
+
+	/* Set the defaults, and power up the codec */
+	twl4030->sysclk = twl4030_codec_get_mclk() / 1000;
+	twl4030_init_chip(codec);
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto error_codec;
+	}
+
+	ret = snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAIs: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		goto error_codec;
+	}
 
 	return 0;
+
+error_codec:
+	twl4030_power_down(codec);
+	kfree(codec->reg_cache);
+error_cache:
+	kfree(twl4030);
+	return ret;
 }
 
-static int twl4030_remove(struct platform_device *pdev)
+static int __devexit twl4030_codec_remove(struct platform_device *pdev)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
+	struct twl4030_priv *twl4030 = platform_get_drvdata(pdev);
 
-	printk(KERN_INFO "TWL4030 Audio Codec remove\n");
-	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-	kfree(codec->private_data);
-	kfree(codec);
+	kfree(twl4030);
 
+	twl4030_codec = NULL;
 	return 0;
 }
 
-struct snd_soc_codec_device soc_codec_dev_twl4030 = {
-	.probe = twl4030_probe,
-	.remove = twl4030_remove,
-	.suspend = twl4030_suspend,
-	.resume = twl4030_resume,
+MODULE_ALIAS("platform:twl4030_codec_audio");
+
+static struct platform_driver twl4030_codec_driver = {
+	.probe		= twl4030_codec_probe,
+	.remove		= __devexit_p(twl4030_codec_remove),
+	.driver		= {
+		.name	= "twl4030_codec_audio",
+		.owner	= THIS_MODULE,
+	},
 };
-EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
 
 static int __init twl4030_modinit(void)
 {
-	return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
+	return platform_driver_register(&twl4030_codec_driver);
 }
 module_init(twl4030_modinit);
 
 static void __exit twl4030_exit(void)
 {
-	snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
+	platform_driver_unregister(&twl4030_codec_driver);
 }
 module_exit(twl4030_exit);
 
+struct snd_soc_codec_device soc_codec_dev_twl4030 = {
+	.probe = twl4030_soc_probe,
+	.remove = twl4030_soc_remove,
+	.suspend = twl4030_soc_suspend,
+	.resume = twl4030_soc_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
+
 MODULE_DESCRIPTION("ASoC TWL4030 codec driver");
 MODULE_AUTHOR("Steve Sakoman");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index 2b4bfa23f985..dd6396ec9c79 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -22,245 +22,13 @@
 #ifndef __TWL4030_AUDIO_H__
 #define __TWL4030_AUDIO_H__
 
-#define TWL4030_REG_CODEC_MODE		0x1
-#define TWL4030_REG_OPTION		0x2
-#define TWL4030_REG_UNKNOWN		0x3
-#define TWL4030_REG_MICBIAS_CTL		0x4
-#define TWL4030_REG_ANAMICL		0x5
-#define TWL4030_REG_ANAMICR		0x6
-#define TWL4030_REG_AVADC_CTL		0x7
-#define TWL4030_REG_ADCMICSEL		0x8
-#define TWL4030_REG_DIGMIXING		0x9
-#define TWL4030_REG_ATXL1PGA		0xA
-#define TWL4030_REG_ATXR1PGA		0xB
-#define TWL4030_REG_AVTXL2PGA		0xC
-#define TWL4030_REG_AVTXR2PGA		0xD
-#define TWL4030_REG_AUDIO_IF		0xE
-#define TWL4030_REG_VOICE_IF		0xF
-#define TWL4030_REG_ARXR1PGA		0x10
-#define TWL4030_REG_ARXL1PGA		0x11
-#define TWL4030_REG_ARXR2PGA		0x12
-#define TWL4030_REG_ARXL2PGA		0x13
-#define TWL4030_REG_VRXPGA		0x14
-#define TWL4030_REG_VSTPGA		0x15
-#define TWL4030_REG_VRX2ARXPGA		0x16
-#define TWL4030_REG_AVDAC_CTL		0x17
-#define TWL4030_REG_ARX2VTXPGA		0x18
-#define TWL4030_REG_ARXL1_APGA_CTL	0x19
-#define TWL4030_REG_ARXR1_APGA_CTL	0x1A
-#define TWL4030_REG_ARXL2_APGA_CTL	0x1B
-#define TWL4030_REG_ARXR2_APGA_CTL	0x1C
-#define TWL4030_REG_ATX2ARXPGA		0x1D
-#define TWL4030_REG_BT_IF		0x1E
-#define TWL4030_REG_BTPGA		0x1F
-#define TWL4030_REG_BTSTPGA		0x20
-#define TWL4030_REG_EAR_CTL		0x21
-#define TWL4030_REG_HS_SEL		0x22
-#define TWL4030_REG_HS_GAIN_SET		0x23
-#define TWL4030_REG_HS_POPN_SET		0x24
-#define TWL4030_REG_PREDL_CTL		0x25
-#define TWL4030_REG_PREDR_CTL		0x26
-#define TWL4030_REG_PRECKL_CTL		0x27
-#define TWL4030_REG_PRECKR_CTL		0x28
-#define TWL4030_REG_HFL_CTL		0x29
-#define TWL4030_REG_HFR_CTL		0x2A
-#define TWL4030_REG_ALC_CTL		0x2B
-#define TWL4030_REG_ALC_SET1		0x2C
-#define TWL4030_REG_ALC_SET2		0x2D
-#define TWL4030_REG_BOOST_CTL		0x2E
-#define TWL4030_REG_SOFTVOL_CTL		0x2F
-#define TWL4030_REG_DTMF_FREQSEL	0x30
-#define TWL4030_REG_DTMF_TONEXT1H	0x31
-#define TWL4030_REG_DTMF_TONEXT1L	0x32
-#define TWL4030_REG_DTMF_TONEXT2H	0x33
-#define TWL4030_REG_DTMF_TONEXT2L	0x34
-#define TWL4030_REG_DTMF_TONOFF		0x35
-#define TWL4030_REG_DTMF_WANONOFF	0x36
-#define TWL4030_REG_I2S_RX_SCRAMBLE_H	0x37
-#define TWL4030_REG_I2S_RX_SCRAMBLE_M	0x38
-#define TWL4030_REG_I2S_RX_SCRAMBLE_L	0x39
-#define TWL4030_REG_APLL_CTL		0x3A
-#define TWL4030_REG_DTMF_CTL		0x3B
-#define TWL4030_REG_DTMF_PGA_CTL2	0x3C
-#define TWL4030_REG_DTMF_PGA_CTL1	0x3D
-#define TWL4030_REG_MISC_SET_1		0x3E
-#define TWL4030_REG_PCMBTMUX		0x3F
-#define TWL4030_REG_RX_PATH_SEL		0x43
-#define TWL4030_REG_VDL_APGA_CTL	0x44
-#define TWL4030_REG_VIBRA_CTL		0x45
-#define TWL4030_REG_VIBRA_SET		0x46
-#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
+/* Register descriptions are here */
+#include <linux/mfd/twl4030-codec.h>
 
+/* Sgadow register used by the audio driver */
+#define TWL4030_REG_SW_SHADOW		0x4A
 #define TWL4030_CACHEREGNUM	(TWL4030_REG_SW_SHADOW + 1)
 
-/* Bitfield Definitions */
-
-/* TWL4030_CODEC_MODE (0x01) Fields */
-
-#define TWL4030_APLL_RATE		0xF0
-#define TWL4030_APLL_RATE_8000		0x00
-#define TWL4030_APLL_RATE_11025		0x10
-#define TWL4030_APLL_RATE_12000		0x20
-#define TWL4030_APLL_RATE_16000		0x40
-#define TWL4030_APLL_RATE_22050		0x50
-#define TWL4030_APLL_RATE_24000		0x60
-#define TWL4030_APLL_RATE_32000		0x80
-#define TWL4030_APLL_RATE_44100		0x90
-#define TWL4030_APLL_RATE_48000		0xA0
-#define TWL4030_APLL_RATE_96000		0xE0
-#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 */
-
-#define TWL4030_MICBIAS2_CTL		0x40
-#define TWL4030_MICBIAS1_CTL		0x20
-#define TWL4030_HSMICBIAS_EN		0x04
-#define TWL4030_MICBIAS2_EN		0x02
-#define TWL4030_MICBIAS1_EN		0x01
-
-/* ANAMICL (0x05) Fields */
-
-#define TWL4030_CNCL_OFFSET_START	0x80
-#define TWL4030_OFFSET_CNCL_SEL		0x60
-#define TWL4030_OFFSET_CNCL_SEL_ARX1	0x00
-#define TWL4030_OFFSET_CNCL_SEL_ARX2	0x20
-#define TWL4030_OFFSET_CNCL_SEL_VRX	0x40
-#define TWL4030_OFFSET_CNCL_SEL_ALL	0x60
-#define TWL4030_MICAMPL_EN		0x10
-#define TWL4030_CKMIC_EN		0x08
-#define TWL4030_AUXL_EN			0x04
-#define TWL4030_HSMIC_EN		0x02
-#define TWL4030_MAINMIC_EN		0x01
-
-/* ANAMICR (0x06) Fields */
-
-#define TWL4030_MICAMPR_EN		0x10
-#define TWL4030_AUXR_EN			0x04
-#define TWL4030_SUBMIC_EN		0x01
-
-/* AVADC_CTL (0x07) Fields */
-
-#define TWL4030_ADCL_EN			0x08
-#define TWL4030_AVADC_CLK_PRIORITY	0x04
-#define TWL4030_ADCR_EN			0x02
-
-/* TWL4030_REG_ADCMICSEL (0x08) Fields */
-
-#define TWL4030_DIGMIC1_EN		0x08
-#define TWL4030_TX2IN_SEL		0x04
-#define TWL4030_DIGMIC0_EN		0x02
-#define TWL4030_TX1IN_SEL		0x01
-
-/* AUDIO_IF (0x0E) Fields */
-
-#define TWL4030_AIF_SLAVE_EN		0x80
-#define TWL4030_DATA_WIDTH		0x60
-#define TWL4030_DATA_WIDTH_16S_16W	0x00
-#define TWL4030_DATA_WIDTH_32S_16W	0x40
-#define TWL4030_DATA_WIDTH_32S_24W	0x60
-#define TWL4030_AIF_FORMAT		0x18
-#define TWL4030_AIF_FORMAT_CODEC	0x00
-#define TWL4030_AIF_FORMAT_LEFT		0x08
-#define TWL4030_AIF_FORMAT_RIGHT	0x10
-#define TWL4030_AIF_FORMAT_TDM		0x18
-#define TWL4030_AIF_TRI_EN		0x04
-#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
-
-/* HS_GAIN_SET (0x23) Fields */
-
-#define TWL4030_HSR_GAIN		0x0C
-#define TWL4030_HSR_GAIN_PWR_DOWN	0x00
-#define TWL4030_HSR_GAIN_PLUS_6DB	0x04
-#define TWL4030_HSR_GAIN_0DB		0x08
-#define TWL4030_HSR_GAIN_MINUS_6DB	0x0C
-#define TWL4030_HSL_GAIN		0x03
-#define TWL4030_HSL_GAIN_PWR_DOWN	0x00
-#define TWL4030_HSL_GAIN_PLUS_6DB	0x01
-#define TWL4030_HSL_GAIN_0DB		0x02
-#define TWL4030_HSL_GAIN_MINUS_6DB	0x03
-
-/* HS_POPN_SET (0x24) Fields */
-
-#define TWL4030_VMID_EN			0x40
-#define	TWL4030_EXTMUTE			0x20
-#define TWL4030_RAMP_DELAY		0x1C
-#define TWL4030_RAMP_DELAY_20MS		0x00
-#define TWL4030_RAMP_DELAY_40MS		0x04
-#define TWL4030_RAMP_DELAY_81MS		0x08
-#define TWL4030_RAMP_DELAY_161MS	0x0C
-#define TWL4030_RAMP_DELAY_323MS	0x10
-#define TWL4030_RAMP_DELAY_645MS	0x14
-#define TWL4030_RAMP_DELAY_1291MS	0x18
-#define TWL4030_RAMP_DELAY_2581MS	0x1C
-#define TWL4030_RAMP_EN			0x02
-
-/* PREDL_CTL (0x25) */
-#define TWL4030_PREDL_GAIN		0x30
-
-/* PREDR_CTL (0x26) */
-#define TWL4030_PREDR_GAIN		0x30
-
-/* PRECKL_CTL (0x27) */
-#define TWL4030_PRECKL_GAIN		0x30
-
-/* PRECKR_CTL (0x28) */
-#define TWL4030_PRECKR_GAIN		0x30
-
-/* HFL_CTL (0x29, 0x2A) Fields */
-#define TWL4030_HF_CTL_HB_EN		0x04
-#define TWL4030_HF_CTL_LOOP_EN		0x08
-#define TWL4030_HF_CTL_RAMP_EN		0x10
-#define TWL4030_HF_CTL_REF_EN		0x20
-
-/* APLL_CTL (0x3A) Fields */
-
-#define TWL4030_APLL_EN			0x10
-#define TWL4030_APLL_INFREQ		0x0F
-#define TWL4030_APLL_INFREQ_19200KHZ	0x05
-#define TWL4030_APLL_INFREQ_26000KHZ	0x06
-#define TWL4030_APLL_INFREQ_38400KHZ	0x0F
-
-/* REG_MISC_SET_1 (0x3E) Fields */
-
-#define TWL4030_CLK64_EN		0x80
-#define TWL4030_SCRAMBLE_EN		0x40
-#define TWL4030_FMLOOP_EN		0x20
-#define TWL4030_SMOOTH_ANAVOL_EN	0x02
-#define TWL4030_DIGMIC_LR_SWAP_EN	0x01
-
 /* TWL4030_REG_SW_SHADOW (0x4A) Fields */
 #define TWL4030_HFL_EN			0x01
 #define TWL4030_HFR_EN			0x02
@@ -279,3 +47,5 @@ struct twl4030_setup_data {
 };
 
 #endif	/* End of __TWL4030_AUDIO_H__ */
+
+
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index c33b92edbded..aa40d985138f 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -562,17 +562,8 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "UDA134X: failed to register card\n");
-		goto card_err;
-	}
-
 	return 0;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	kfree(codec->reg_cache);
 reg_err:
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index 92ec03442154..a2763c2e7348 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -378,7 +378,6 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -713,17 +712,9 @@ static int uda1380_probe(struct platform_device *pdev)
 	snd_soc_add_controls(codec, uda1380_snd_controls,
 				ARRAY_SIZE(uda1380_snd_controls));
 	uda1380_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;
 }
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 593d5b9c9f03..f82125d9e85a 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -800,7 +800,7 @@ static int wm8350_add_widgets(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	return snd_soc_dapm_new_widgets(codec);
+	return 0;
 }
 
 static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
@@ -1101,7 +1101,7 @@ static inline int fll_factors(struct _fll_div *fll_div, unsigned int input,
 }
 
 static int wm8350_set_fll(struct snd_soc_dai *codec_dai,
-			  int pll_id, unsigned int freq_in,
+			  int pll_id, int source, unsigned int freq_in,
 			  unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1501,18 +1501,7 @@ static int wm8350_probe(struct platform_device *pdev)
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(&pdev->dev, "failed to register card\n");
-		goto card_err;
-	}
-
 	return 0;
-
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-	return ret;
 }
 
 static int wm8350_remove(struct platform_device *pdev)
@@ -1680,21 +1669,6 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8350_codec_suspend(struct platform_device *pdev, pm_message_t m)
-{
-	return snd_soc_suspend_device(&pdev->dev);
-}
-
-static int wm8350_codec_resume(struct platform_device *pdev)
-{
-	return snd_soc_resume_device(&pdev->dev);
-}
-#else
-#define wm8350_codec_suspend NULL
-#define wm8350_codec_resume NULL
-#endif
-
 static struct platform_driver wm8350_codec_driver = {
 	.driver = {
 		   .name = "wm8350-codec",
@@ -1702,8 +1676,6 @@ static struct platform_driver wm8350_codec_driver = {
 		   },
 	.probe = wm8350_codec_probe,
 	.remove = __devexit_p(wm8350_codec_remove),
-	.suspend = wm8350_codec_suspend,
-	.resume = wm8350_codec_resume,
 };
 
 static __init int wm8350_init(void)
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index b9ef4d915221..b432f4d4a324 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -915,7 +915,6 @@ static int wm8400_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -1011,7 +1010,8 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
 }
 
 static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
-			      unsigned int freq_in, unsigned int freq_out)
+			      int source, unsigned int freq_in,
+			      unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct wm8400_priv *wm8400 = codec->private_data;
@@ -1399,17 +1399,6 @@ static int wm8400_probe(struct platform_device *pdev)
 	wm8400_add_controls(codec);
 	wm8400_add_widgets(codec);
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(&pdev->dev, "failed to register card\n");
-		goto card_err;
-	}
-
-	return ret;
-
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	return ret;
 }
@@ -1558,21 +1547,6 @@ static int __exit wm8400_codec_remove(struct platform_device *dev)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8400_pdev_suspend(struct platform_device *pdev, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&pdev->dev);
-}
-
-static int wm8400_pdev_resume(struct platform_device *pdev)
-{
-	return snd_soc_resume_device(&pdev->dev);
-}
-#else
-#define wm8400_pdev_suspend NULL
-#define wm8400_pdev_resume NULL
-#endif
-
 static struct platform_driver wm8400_codec_driver = {
 	.driver = {
 		.name = "wm8400-codec",
@@ -1580,8 +1554,6 @@ static struct platform_driver wm8400_codec_driver = {
 	},
 	.probe = wm8400_codec_probe,
 	.remove	= __exit_p(wm8400_codec_remove),
-	.suspend = wm8400_pdev_suspend,
-	.resume = wm8400_pdev_resume,
 };
 
 static int __init wm8400_codec_init(void)
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 060d5d06ba95..265e68c75df8 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -219,7 +219,6 @@ static int wm8510_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -271,8 +270,8 @@ static void pll_factors(unsigned int target, unsigned int source)
 	pll_div.k = K;
 }
 
-static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 reg;
@@ -604,16 +603,9 @@ static int wm8510_init(struct snd_soc_device *socdev,
 	snd_soc_add_controls(codec, wm8510_snd_controls,
 				ARRAY_SIZE(wm8510_snd_controls));
 	wm8510_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8510: failed to register card\n");
-		goto card_err;
-	}
+
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 err:
 	kfree(codec->reg_cache);
 	return ret;
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 25870a4652fb..d3a61d7ea0c5 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -117,7 +117,6 @@ static int wm8523_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -448,17 +447,9 @@ static int wm8523_probe(struct platform_device *pdev)
 	snd_soc_add_controls(codec, wm8523_snd_controls,
 			     ARRAY_SIZE(wm8523_snd_controls));
 	wm8523_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;
 }
@@ -638,21 +629,6 @@ static __devexit int wm8523_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8523_i2c_suspend(struct i2c_client *i2c, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&i2c->dev);
-}
-
-static int wm8523_i2c_resume(struct i2c_client *i2c)
-{
-	return snd_soc_resume_device(&i2c->dev);
-}
-#else
-#define wm8523_i2c_suspend NULL
-#define wm8523_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8523_i2c_id[] = {
 	{ "wm8523", 0 },
 	{ }
@@ -666,8 +642,6 @@ static struct i2c_driver wm8523_i2c_driver = {
 	},
 	.probe =    wm8523_i2c_probe,
 	.remove =   __devexit_p(wm8523_i2c_remove),
-	.suspend =  wm8523_i2c_suspend,
-	.resume =   wm8523_i2c_resume,
 	.id_table = wm8523_i2c_id,
 };
 #endif
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 6bded8c78150..d077df6f5e75 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -315,7 +315,6 @@ static int wm8580_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -407,8 +406,8 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target,
 	return 0;
 }
 
-static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	int offset;
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -800,17 +799,9 @@ static int wm8580_probe(struct platform_device *pdev)
 	snd_soc_add_controls(codec, wm8580_snd_controls,
 			     ARRAY_SIZE(wm8580_snd_controls));
 	wm8580_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;
 }
@@ -961,21 +952,6 @@ static int wm8580_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8580_i2c_suspend(struct i2c_client *client, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm8580_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm8580_i2c_suspend NULL
-#define wm8580_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8580_i2c_id[] = {
 	{ "wm8580", 0 },
 	{ }
@@ -989,8 +965,6 @@ static struct i2c_driver wm8580_i2c_driver = {
 	},
 	.probe =    wm8580_i2c_probe,
 	.remove =   wm8580_i2c_remove,
-	.suspend =  wm8580_i2c_suspend,
-	.resume =   wm8580_i2c_resume,
 	.id_table = wm8580_i2c_id,
 };
 #endif
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
new file mode 100644
index 000000000000..24a35603bcf7
--- /dev/null
+++ b/sound/soc/codecs/wm8711.c
@@ -0,0 +1,633 @@
+/*
+ * wm8711.c  --  WM8711 ALSA SoC Audio driver
+ *
+ * Copyright 2006 Wolfson Microelectronics
+ *
+ * Author: Mike Arthur <linux@wolfsonmicro.com>
+ *
+ * Based on wm8731.c by Richard Purdie
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/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/tlv.h>
+#include <sound/initval.h>
+
+#include "wm8711.h"
+
+static struct snd_soc_codec *wm8711_codec;
+
+/* codec private data */
+struct wm8711_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM8711_CACHEREGNUM];
+	unsigned int sysclk;
+};
+
+/*
+ * wm8711 register cache
+ * We can't read the WM8711 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u16 wm8711_reg[WM8711_CACHEREGNUM] = {
+	0x0079, 0x0079, 0x000a, 0x0008,
+	0x009f, 0x000a, 0x0000, 0x0000
+};
+
+#define wm8711_reset(c)	snd_soc_write(c, WM8711_RESET, 0)
+
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+
+static const struct snd_kcontrol_new wm8711_snd_controls[] = {
+
+SOC_DOUBLE_R_TLV("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V,
+		 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V,
+	7, 1, 0),
+
+};
+
+/* Output Mixer */
+static const struct snd_kcontrol_new wm8711_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1,
+	&wm8711_output_mixer_controls[0],
+	ARRAY_SIZE(wm8711_output_mixer_controls)),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("LHPOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("RHPOUT"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* output mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "HiFi Playback Switch", "DAC"},
+
+	/* outputs */
+	{"RHPOUT", NULL, "Output Mixer"},
+	{"ROUT", NULL, "Output Mixer"},
+	{"LHPOUT", NULL, "Output Mixer"},
+	{"LOUT", NULL, "Output Mixer"},
+};
+
+static int wm8711_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8711_dapm_widgets,
+				  ARRAY_SIZE(wm8711_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	return 0;
+}
+
+struct _coeff_div {
+	u32 mclk;
+	u32 rate;
+	u16 fs;
+	u8 sr:4;
+	u8 bosr:1;
+	u8 usb:1;
+};
+
+/* codec mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+	/* 48k */
+	{12288000, 48000, 256, 0x0, 0x0, 0x0},
+	{18432000, 48000, 384, 0x0, 0x1, 0x0},
+	{12000000, 48000, 250, 0x0, 0x0, 0x1},
+
+	/* 32k */
+	{12288000, 32000, 384, 0x6, 0x0, 0x0},
+	{18432000, 32000, 576, 0x6, 0x1, 0x0},
+	{12000000, 32000, 375, 0x6, 0x0, 0x1},
+
+	/* 8k */
+	{12288000, 8000, 1536, 0x3, 0x0, 0x0},
+	{18432000, 8000, 2304, 0x3, 0x1, 0x0},
+	{11289600, 8000, 1408, 0xb, 0x0, 0x0},
+	{16934400, 8000, 2112, 0xb, 0x1, 0x0},
+	{12000000, 8000, 1500, 0x3, 0x0, 0x1},
+
+	/* 96k */
+	{12288000, 96000, 128, 0x7, 0x0, 0x0},
+	{18432000, 96000, 192, 0x7, 0x1, 0x0},
+	{12000000, 96000, 125, 0x7, 0x0, 0x1},
+
+	/* 44.1k */
+	{11289600, 44100, 256, 0x8, 0x0, 0x0},
+	{16934400, 44100, 384, 0x8, 0x1, 0x0},
+	{12000000, 44100, 272, 0x8, 0x1, 0x1},
+
+	/* 88.2k */
+	{11289600, 88200, 128, 0xf, 0x0, 0x0},
+	{16934400, 88200, 192, 0xf, 0x1, 0x0},
+	{12000000, 88200, 136, 0xf, 0x1, 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 0;
+}
+
+static int wm8711_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 wm8711_priv *wm8711 = codec->private_data;
+	u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfffc;
+	int i = get_coeff(wm8711->sysclk, params_rate(params));
+	u16 srate = (coeff_div[i].sr << 2) |
+		(coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
+	snd_soc_write(codec, WM8711_SRATE, srate);
+
+	/* 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;
+	}
+
+	snd_soc_write(codec, WM8711_IFACE, iface);
+	return 0;
+}
+
+static int wm8711_pcm_prepare(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	/* set active */
+	snd_soc_write(codec, WM8711_ACTIVE, 0x0001);
+
+	return 0;
+}
+
+static void wm8711_shutdown(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	/* deactivate */
+	if (!codec->active) {
+		udelay(50);
+		snd_soc_write(codec, WM8711_ACTIVE, 0x0);
+	}
+}
+
+static int wm8711_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = snd_soc_read(codec, WM8711_APDIGI) & 0xfff7;
+
+	if (mute)
+		snd_soc_write(codec, WM8711_APDIGI, mute_reg | 0x8);
+	else
+		snd_soc_write(codec, WM8711_APDIGI, mute_reg);
+
+	return 0;
+}
+
+static int wm8711_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 wm8711_priv *wm8711 = codec->private_data;
+
+	switch (freq) {
+	case 11289600:
+	case 12000000:
+	case 12288000:
+	case 16934400:
+	case 18432000:
+		wm8711->sysclk = freq;
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int wm8711_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 */
+	snd_soc_write(codec, WM8711_IFACE, iface);
+	return 0;
+}
+
+
+static int wm8711_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_write(codec, WM8711_PWR, reg);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_write(codec, WM8711_PWR, reg | 0x0040);
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_write(codec, WM8711_ACTIVE, 0x0);
+		snd_soc_write(codec, WM8711_PWR, 0xffff);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8711_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8711_ops = {
+	.prepare = wm8711_pcm_prepare,
+	.hw_params = wm8711_hw_params,
+	.shutdown = wm8711_shutdown,
+	.digital_mute = wm8711_mute,
+	.set_sysclk = wm8711_set_dai_sysclk,
+	.set_fmt = wm8711_set_dai_fmt,
+};
+
+struct snd_soc_dai wm8711_dai = {
+	.name = "WM8711",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8711_RATES,
+		.formats = WM8711_FORMATS,
+	},
+	.ops = &wm8711_ops,
+};
+EXPORT_SYMBOL_GPL(wm8711_dai);
+
+static int wm8711_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;
+
+	snd_soc_write(codec, WM8711_ACTIVE, 0x0);
+	wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int wm8711_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(wm8711_reg); i++) {
+		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+		data[1] = cache[i] & 0x00ff;
+		codec->hw_write(codec->control_data, data, 2);
+	}
+	wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	wm8711_set_bias_level(codec, codec->suspend_bias_level);
+	return 0;
+}
+
+static int wm8711_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (wm8711_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = wm8711_codec;
+	codec = wm8711_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, wm8711_snd_controls,
+			     ARRAY_SIZE(wm8711_snd_controls));
+	wm8711_add_widgets(codec);
+
+	return ret;
+
+pcm_err:
+	return ret;
+}
+
+/* power down chip */
+static int wm8711_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_wm8711 = {
+	.probe = 	wm8711_probe,
+	.remove = 	wm8711_remove,
+	.suspend = 	wm8711_suspend,
+	.resume =	wm8711_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711);
+
+static int wm8711_register(struct wm8711_priv *wm8711,
+			   enum snd_soc_control_type control)
+{
+	int ret;
+	struct snd_soc_codec *codec = &wm8711->codec;
+	u16 reg;
+
+	if (wm8711_codec) {
+		dev_err(codec->dev, "Another WM8711 is registered\n");
+		return -EINVAL;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->private_data = wm8711;
+	codec->name = "WM8711";
+	codec->owner = THIS_MODULE;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8711_set_bias_level;
+	codec->dai = &wm8711_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8711_CACHEREGNUM;
+	codec->reg_cache = &wm8711->reg_cache;
+
+	memcpy(codec->reg_cache, wm8711_reg, sizeof(wm8711_reg));
+
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, control);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		goto err;
+	}
+
+	ret = wm8711_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		goto err;
+	}
+
+	wm8711_dai.dev = codec->dev;
+
+	wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* Latch the update bits */
+	reg = snd_soc_read(codec, WM8711_LOUT1V);
+	snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100);
+	reg = snd_soc_read(codec, WM8711_ROUT1V);
+	snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100);
+
+	wm8711_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&wm8711_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(wm8711);
+	return ret;
+}
+
+static void wm8711_unregister(struct wm8711_priv *wm8711)
+{
+	wm8711_set_bias_level(&wm8711->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&wm8711_dai);
+	snd_soc_unregister_codec(&wm8711->codec);
+	kfree(wm8711);
+	wm8711_codec = NULL;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8711_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_codec *codec;
+	struct wm8711_priv *wm8711;
+
+	wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL);
+	if (wm8711 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8711->codec;
+	codec->control_data = spi;
+	codec->dev = &spi->dev;
+
+	dev_set_drvdata(&spi->dev, wm8711);
+
+	return wm8711_register(wm8711, SND_SOC_SPI);
+}
+
+static int __devexit wm8711_spi_remove(struct spi_device *spi)
+{
+	struct wm8711_priv *wm8711 = dev_get_drvdata(&spi->dev);
+
+	wm8711_unregister(wm8711);
+
+	return 0;
+}
+
+static struct spi_driver wm8711_spi_driver = {
+	.driver = {
+		.name	= "wm8711",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8711_spi_probe,
+	.remove		= __devexit_p(wm8711_spi_remove),
+};
+#endif /* CONFIG_SPI_MASTER */
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static __devinit int wm8711_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct wm8711_priv *wm8711;
+	struct snd_soc_codec *codec;
+
+	wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL);
+	if (wm8711 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8711->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+
+	i2c_set_clientdata(i2c, wm8711);
+	codec->control_data = i2c;
+
+	codec->dev = &i2c->dev;
+
+	return wm8711_register(wm8711, SND_SOC_I2C);
+}
+
+static __devexit int wm8711_i2c_remove(struct i2c_client *client)
+{
+	struct wm8711_priv *wm8711 = i2c_get_clientdata(client);
+	wm8711_unregister(wm8711);
+	return 0;
+}
+
+static const struct i2c_device_id wm8711_i2c_id[] = {
+	{ "wm8711", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id);
+
+static struct i2c_driver wm8711_i2c_driver = {
+	.driver = {
+		.name = "WM8711 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8711_i2c_probe,
+	.remove =   __devexit_p(wm8711_i2c_remove),
+	.id_table = wm8711_i2c_id,
+};
+#endif
+
+static int __init wm8711_modinit(void)
+{
+	int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&wm8711_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n",
+		       ret);
+	}
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&wm8711_spi_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8711 SPI driver: %d\n",
+		       ret);
+	}
+#endif
+	return 0;
+}
+module_init(wm8711_modinit);
+
+static void __exit wm8711_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8711_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8711_spi_driver);
+#endif
+}
+module_exit(wm8711_exit);
+
+MODULE_DESCRIPTION("ASoC WM8711 driver");
+MODULE_AUTHOR("Mike Arthur");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8711.h b/sound/soc/codecs/wm8711.h
new file mode 100644
index 000000000000..381e84a43816
--- /dev/null
+++ b/sound/soc/codecs/wm8711.h
@@ -0,0 +1,42 @@
+/*
+ * wm8711.h  --  WM8711 Soc Audio driver
+ *
+ * Copyright 2006 Wolfson Microelectronics
+ *
+ * Author: Mike Arthur <linux@wolfsonmicro.com>
+ *
+ * Based on wm8731.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 _WM8711_H
+#define _WM8711_H
+
+/* WM8711 register space */
+
+#define WM8711_LOUT1V   0x02
+#define WM8711_ROUT1V   0x03
+#define WM8711_APANA    0x04
+#define WM8711_APDIGI   0x05
+#define WM8711_PWR      0x06
+#define WM8711_IFACE    0x07
+#define WM8711_SRATE    0x08
+#define WM8711_ACTIVE   0x09
+#define WM8711_RESET	0x0f
+
+#define WM8711_CACHEREGNUM 	8
+
+#define WM8711_SYSCLK	0
+#define WM8711_DAI		0
+
+struct wm8711_setup_data {
+	unsigned short i2c_address;
+};
+
+extern struct snd_soc_dai wm8711_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8711;
+
+#endif
diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c
new file mode 100644
index 000000000000..d8ffbd641d71
--- /dev/null
+++ b/sound/soc/codecs/wm8727.c
@@ -0,0 +1,135 @@
+/*
+ * wm8727.c
+ *
+ *  Created on: 15-Oct-2009
+ *      Author: neil.jones@imgtec.com
+ *
+ * Copyright (C) 2009 Imagination Technologies Ltd.
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "wm8727.h"
+/*
+ * Note this is a simple chip with no configuration interface, sample rate is
+ * determined automatically by examining the Master clock and Bit clock ratios
+ */
+#define WM8727_RATES  (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+			SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\
+			SNDRV_PCM_RATE_192000)
+
+
+struct snd_soc_dai wm8727_dai = {
+	.name = "WM8727",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = WM8727_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+		},
+};
+EXPORT_SYMBOL_GPL(wm8727_dai);
+
+static int wm8727_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+	mutex_init(&codec->mutex);
+	codec->name = "WM8727";
+	codec->owner = THIS_MODULE;
+	codec->dai = &wm8727_dai;
+	codec->num_dai = 1;
+	socdev->card->codec = codec;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8727: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	return ret;
+
+pcm_err:
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
+	return ret;
+}
+
+static int wm8727_soc_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);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8727 = {
+	.probe = 	wm8727_soc_probe,
+	.remove = 	wm8727_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8727);
+
+
+static __devinit int wm8727_platform_probe(struct platform_device *pdev)
+{
+	wm8727_dai.dev = &pdev->dev;
+	return snd_soc_register_dai(&wm8727_dai);
+}
+
+static int __devexit wm8727_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&wm8727_dai);
+	return 0;
+}
+
+static struct platform_driver wm8727_codec_driver = {
+	.driver = {
+			.name = "wm8727-codec",
+			.owner = THIS_MODULE,
+	},
+
+	.probe = wm8727_platform_probe,
+	.remove = __devexit_p(wm8727_platform_remove),
+};
+
+static int __init wm8727_init(void)
+{
+	return platform_driver_register(&wm8727_codec_driver);
+}
+module_init(wm8727_init);
+
+static void __exit wm8727_exit(void)
+{
+	platform_driver_unregister(&wm8727_codec_driver);
+}
+module_exit(wm8727_exit);
+
+MODULE_DESCRIPTION("ASoC wm8727 driver");
+MODULE_AUTHOR("Neil Jones");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm8727.h b/sound/soc/codecs/wm8727.h
new file mode 100644
index 000000000000..ee19aa71bcdc
--- /dev/null
+++ b/sound/soc/codecs/wm8727.h
@@ -0,0 +1,21 @@
+/*
+ * wm8727.h
+ *
+ *  Created on: 15-Oct-2009
+ *      Author: neil.jones@imgtec.com
+ *
+ * Copyright (C) 2009 Imagination Technologies Ltd.
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#ifndef WM8727_H_
+#define WM8727_H_
+
+extern struct snd_soc_dai wm8727_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8727;
+
+#endif /* WM8727_H_ */
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 16e969a762c3..3fb653ba363a 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -74,8 +74,6 @@ static int wm8728_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
-	snd_soc_dapm_new_widgets(codec);
-
 	return 0;
 }
 
@@ -287,17 +285,9 @@ static int wm8728_init(struct snd_soc_device *socdev,
 	snd_soc_add_controls(codec, wm8728_snd_controls,
 				ARRAY_SIZE(wm8728_snd_controls));
 	wm8728_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8728: failed to register card\n");
-		goto card_err;
-	}
 
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 err:
 	kfree(codec->reg_cache);
 	return ret;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index d3fd4f28d96e..3a497810f939 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -19,6 +19,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
 #include <linux/spi/spi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -33,9 +34,18 @@
 static struct snd_soc_codec *wm8731_codec;
 struct snd_soc_codec_device soc_codec_dev_wm8731;
 
+#define WM8731_NUM_SUPPLIES 4
+static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = {
+	"AVDD",
+	"HPVDD",
+	"DCVDD",
+	"DBVDD",
+};
+
 /* codec private data */
 struct wm8731_priv {
 	struct snd_soc_codec codec;
+	struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES];
 	u16 reg_cache[WM8731_CACHEREGNUM];
 	unsigned int sysclk;
 };
@@ -149,7 +159,6 @@ static int wm8731_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -422,9 +431,12 @@ static int wm8731_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;
+	struct wm8731_priv *wm8731 = codec->private_data;
 
 	snd_soc_write(codec, WM8731_ACTIVE, 0x0);
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
+			       wm8731->supplies);
 	return 0;
 }
 
@@ -432,10 +444,16 @@ static int wm8731_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	int i;
+	struct wm8731_priv *wm8731 = codec->private_data;
+	int i, ret;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
 
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies),
+				    wm8731->supplies);
+	if (ret != 0)
+		return ret;
+
 	/* Sync reg_cache with the hardware */
 	for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) {
 		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
@@ -444,6 +462,7 @@ static int wm8731_resume(struct platform_device *pdev)
 	}
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm8731_set_bias_level(codec, codec->suspend_bias_level);
+
 	return 0;
 }
 #else
@@ -475,17 +494,9 @@ static int wm8731_probe(struct platform_device *pdev)
 	snd_soc_add_controls(codec, wm8731_snd_controls,
 			     ARRAY_SIZE(wm8731_snd_controls));
 	wm8731_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;
 }
@@ -512,7 +523,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 static int wm8731_register(struct wm8731_priv *wm8731,
 			   enum snd_soc_control_type control)
 {
-	int ret;
+	int ret, i;
 	struct snd_soc_codec *codec = &wm8731->codec;
 
 	if (wm8731_codec) {
@@ -543,10 +554,27 @@ static int wm8731_register(struct wm8731_priv *wm8731,
 		goto err;
 	}
 
+	for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++)
+		wm8731->supplies[i].supply = wm8731_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8731->supplies),
+				 wm8731->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		goto err;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies),
+				    wm8731->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_regulator_get;
+	}
+
 	ret = wm8731_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
-		goto err;
+		goto err_regulator_enable;
 	}
 
 	wm8731_dai.dev = codec->dev;
@@ -567,7 +595,7 @@ static int wm8731_register(struct wm8731_priv *wm8731,
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
-		goto err;
+		goto err_regulator_enable;
 	}
 
 	ret = snd_soc_register_dai(&wm8731_dai);
@@ -581,6 +609,10 @@ static int wm8731_register(struct wm8731_priv *wm8731,
 
 err_codec:
 	snd_soc_unregister_codec(codec);
+err_regulator_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
+err_regulator_get:
+	regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
 err:
 	kfree(wm8731);
 	return ret;
@@ -591,6 +623,8 @@ static void wm8731_unregister(struct wm8731_priv *wm8731)
 	wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF);
 	snd_soc_unregister_dai(&wm8731_dai);
 	snd_soc_unregister_codec(&wm8731->codec);
+	regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
+	regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
 	kfree(wm8731);
 	wm8731_codec = NULL;
 }
@@ -623,21 +657,6 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8731_spi_suspend(struct spi_device *spi, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&spi->dev);
-}
-
-static int wm8731_spi_resume(struct spi_device *spi)
-{
-	return snd_soc_resume_device(&spi->dev);
-}
-#else
-#define wm8731_spi_suspend NULL
-#define wm8731_spi_resume NULL
-#endif
-
 static struct spi_driver wm8731_spi_driver = {
 	.driver = {
 		.name	= "wm8731",
@@ -645,8 +664,6 @@ static struct spi_driver wm8731_spi_driver = {
 		.owner	= THIS_MODULE,
 	},
 	.probe		= wm8731_spi_probe,
-	.suspend	= wm8731_spi_suspend,
-	.resume		= wm8731_spi_resume,
 	.remove		= __devexit_p(wm8731_spi_remove),
 };
 #endif /* CONFIG_SPI_MASTER */
@@ -679,21 +696,6 @@ static __devexit int wm8731_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8731_i2c_suspend(struct i2c_client *i2c, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&i2c->dev);
-}
-
-static int wm8731_i2c_resume(struct i2c_client *i2c)
-{
-	return snd_soc_resume_device(&i2c->dev);
-}
-#else
-#define wm8731_i2c_suspend NULL
-#define wm8731_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8731_i2c_id[] = {
 	{ "wm8731", 0 },
 	{ }
@@ -707,8 +709,6 @@ static struct i2c_driver wm8731_i2c_driver = {
 	},
 	.probe =    wm8731_i2c_probe,
 	.remove =   __devexit_p(wm8731_i2c_remove),
-	.suspend =  wm8731_i2c_suspend,
-	.resume =   wm8731_i2c_resume,
 	.id_table = wm8731_i2c_id,
 };
 #endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 4ba1e7e93fb4..475c67ac7818 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -403,7 +403,6 @@ static int wm8750_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -772,16 +771,8 @@ static int wm8750_init(struct snd_soc_device *socdev,
 	snd_soc_add_controls(codec, wm8750_snd_controls,
 				ARRAY_SIZE(wm8750_snd_controls));
 	wm8750_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8750: failed to register card\n");
-		goto card_err;
-	}
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 err:
 	kfree(codec->reg_cache);
 	return ret;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 5ad677ce80da..d6850dacda29 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -673,7 +673,6 @@ static int wm8753_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -724,8 +723,8 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 	pll_div->k = K;
 }
 
-static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	u16 reg, enable;
 	int offset;
@@ -1583,18 +1582,9 @@ static int wm8753_probe(struct platform_device *pdev)
 	snd_soc_add_controls(codec, wm8753_snd_controls,
 			     ARRAY_SIZE(wm8753_snd_controls));
 	wm8753_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8753: failed to register card\n");
-		goto card_err;
-	}
 
 	return 0;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-
 pcm_err:
 	return ret;
 }
@@ -1767,21 +1757,6 @@ static int wm8753_i2c_remove(struct i2c_client *client)
         return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8753_i2c_suspend(struct i2c_client *client, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm8753_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm8753_i2c_suspend NULL
-#define wm8753_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8753_i2c_id[] = {
 	{ "wm8753", 0 },
 	{ }
@@ -1795,8 +1770,6 @@ static struct i2c_driver wm8753_i2c_driver = {
 	},
 	.probe =    wm8753_i2c_probe,
 	.remove =   wm8753_i2c_remove,
-	.suspend =  wm8753_i2c_suspend,
-	.resume =   wm8753_i2c_resume,
 	.id_table = wm8753_i2c_id,
 };
 #endif
@@ -1852,22 +1825,6 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8753_spi_suspend(struct spi_device *spi, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&spi->dev);
-}
-
-static int wm8753_spi_resume(struct spi_device *spi)
-{
-	return snd_soc_resume_device(&spi->dev);
-}
-
-#else
-#define wm8753_spi_suspend NULL
-#define wm8753_spi_resume NULL
-#endif
-
 static struct spi_driver wm8753_spi_driver = {
 	.driver = {
 		.name	= "wm8753",
@@ -1876,8 +1833,6 @@ static struct spi_driver wm8753_spi_driver = {
 	},
 	.probe		= wm8753_spi_probe,
 	.remove		= __devexit_p(wm8753_spi_remove),
-	.suspend	= wm8753_spi_suspend,
-	.resume		= wm8753_spi_resume,
 };
 #endif
 
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index a9829aa26e53..ab2c0da18091 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -447,17 +447,8 @@ static int wm8776_probe(struct platform_device *pdev)
 				  ARRAY_SIZE(wm8776_dapm_widgets));
 	snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes));
 
-	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;
 }
@@ -616,21 +607,6 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&spi->dev);
-}
-
-static int wm8776_spi_resume(struct spi_device *spi)
-{
-	return snd_soc_resume_device(&spi->dev);
-}
-#else
-#define wm8776_spi_suspend NULL
-#define wm8776_spi_resume NULL
-#endif
-
 static struct spi_driver wm8776_spi_driver = {
 	.driver = {
 		.name	= "wm8776",
@@ -638,8 +614,6 @@ static struct spi_driver wm8776_spi_driver = {
 		.owner	= THIS_MODULE,
 	},
 	.probe		= wm8776_spi_probe,
-	.suspend	= wm8776_spi_suspend,
-	.resume		= wm8776_spi_resume,
 	.remove		= __devexit_p(wm8776_spi_remove),
 };
 #endif /* CONFIG_SPI_MASTER */
@@ -673,21 +647,6 @@ static __devexit int wm8776_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&i2c->dev);
-}
-
-static int wm8776_i2c_resume(struct i2c_client *i2c)
-{
-	return snd_soc_resume_device(&i2c->dev);
-}
-#else
-#define wm8776_i2c_suspend NULL
-#define wm8776_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8776_i2c_id[] = {
 	{ "wm8776", 0 },
 	{ }
@@ -701,8 +660,6 @@ static struct i2c_driver wm8776_i2c_driver = {
 	},
 	.probe =    wm8776_i2c_probe,
 	.remove =   __devexit_p(wm8776_i2c_remove),
-	.suspend =  wm8776_i2c_suspend,
-	.resume =   wm8776_i2c_resume,
 	.id_table = wm8776_i2c_id,
 };
 #endif
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 5e9c855c0036..c9438dd62df3 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -618,8 +618,6 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
-
 	return 0;
 }
 
@@ -814,8 +812,8 @@ reenable:
 	return 0;
 }
 
-static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	return wm8900_set_fll(codec_dai->codec, pll_id, freq_in, freq_out);
 }
@@ -1312,21 +1310,6 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8900_i2c_suspend(struct i2c_client *client, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm8900_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm8900_i2c_suspend NULL
-#define wm8900_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8900_i2c_id[] = {
 	{ "wm8900", 0 },
 	{ }
@@ -1340,8 +1323,6 @@ static struct i2c_driver wm8900_i2c_driver = {
 	},
 	.probe = wm8900_i2c_probe,
 	.remove = __devexit_p(wm8900_i2c_remove),
-	.suspend = wm8900_i2c_suspend,
-	.resume = wm8900_i2c_resume,
 	.id_table = wm8900_i2c_id,
 };
 
@@ -1370,17 +1351,6 @@ static int wm8900_probe(struct platform_device *pdev)
 				ARRAY_SIZE(wm8900_snd_controls));
 	wm8900_add_widgets(codec);
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(&pdev->dev, "Failed to register card\n");
-		goto card_err;
-	}
-
-	return ret;
-
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	return ret;
 }
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index fe1307b500cf..b8cae1758642 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -919,8 +919,6 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
-	snd_soc_dapm_new_widgets(codec);
-
 	return 0;
 }
 
@@ -1655,21 +1653,6 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8903_i2c_suspend(struct i2c_client *client, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm8903_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm8903_i2c_suspend NULL
-#define wm8903_i2c_resume NULL
-#endif
-
 /* i2c codec control layer */
 static const struct i2c_device_id wm8903_i2c_id[] = {
        { "wm8903", 0 },
@@ -1684,8 +1667,6 @@ static struct i2c_driver wm8903_i2c_driver = {
 	},
 	.probe    = wm8903_i2c_probe,
 	.remove   = __devexit_p(wm8903_i2c_remove),
-	.suspend  = wm8903_i2c_suspend,
-	.resume   = wm8903_i2c_resume,
 	.id_table = wm8903_i2c_id,
 };
 
@@ -1712,17 +1693,8 @@ static int wm8903_probe(struct platform_device *pdev)
 				ARRAY_SIZE(wm8903_snd_controls));
 	wm8903_add_widgets(socdev->card->codec);
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(&pdev->dev, "wm8903: failed to register card\n");
-		goto card_err;
-	}
-
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 err:
 	return ret;
 }
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index 1ef2454c5205..3d850b97037a 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -298,7 +298,6 @@ static int wm8940_add_widgets(struct snd_soc_codec *codec)
 	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;
@@ -536,8 +535,8 @@ static void pll_factors(unsigned int target, unsigned int source)
 }
 
 /* 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)
+static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 reg;
@@ -731,12 +730,6 @@ static int wm8940_probe(struct platform_device *pdev)
 	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:
@@ -877,21 +870,6 @@ static int __devexit wm8940_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8940_i2c_suspend(struct i2c_client *client, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm8940_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm8940_i2c_suspend NULL
-#define wm8940_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8940_i2c_id[] = {
 	{ "wm8940", 0 },
 	{ }
@@ -905,8 +883,6 @@ static struct i2c_driver wm8940_i2c_driver = {
 	},
 	.probe = wm8940_i2c_probe,
 	.remove = __devexit_p(wm8940_i2c_remove),
-	.suspend = wm8940_i2c_suspend,
-	.resume = wm8940_i2c_resume,
 	.id_table = wm8940_i2c_id,
 };
 
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index f59703be61c8..d07bcc1e1c60 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -307,7 +307,6 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -540,8 +539,8 @@ static int pll_factors(unsigned int source, unsigned int target,
 	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)
+static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 reg;
@@ -713,17 +712,9 @@ static int wm8960_probe(struct platform_device *pdev)
 	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;
 }
@@ -883,21 +874,6 @@ static __devexit int wm8960_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8960_i2c_suspend(struct i2c_client *client, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm8960_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm8960_i2c_suspend NULL
-#define wm8960_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8960_i2c_id[] = {
 	{ "wm8960", 0 },
 	{ }
@@ -911,8 +887,6 @@ static struct i2c_driver wm8960_i2c_driver = {
 	},
 	.probe =    wm8960_i2c_probe,
 	.remove =   __devexit_p(wm8960_i2c_remove),
-	.suspend =  wm8960_i2c_suspend,
-	.resume =   wm8960_i2c_resume,
 	.id_table = wm8960_i2c_id,
 };
 
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 503032085899..a8007d58813f 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -986,19 +986,9 @@ static int wm8961_probe(struct platform_device *pdev)
 	snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets,
 				  ARRAY_SIZE(wm8961_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;
 }
@@ -1206,21 +1196,6 @@ static __devexit int wm8961_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8961_i2c_suspend(struct i2c_client *client, pm_message_t state)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm8961_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm8961_i2c_suspend NULL
-#define wm8961_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8961_i2c_id[] = {
 	{ "wm8961", 0 },
 	{ }
@@ -1234,8 +1209,6 @@ static struct i2c_driver wm8961_i2c_driver = {
 	},
 	.probe =    wm8961_i2c_probe,
 	.remove =   __devexit_p(wm8961_i2c_remove),
-	.suspend =  wm8961_i2c_suspend,
-	.resume =   wm8961_i2c_resume,
 	.id_table = wm8961_i2c_id,
 };
 
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index d66efb0546ea..d9540d55fc89 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -338,8 +338,6 @@ static int wm8971_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
-
 	return 0;
 }
 
@@ -703,16 +701,9 @@ static int wm8971_init(struct snd_soc_device *socdev,
 	snd_soc_add_controls(codec, wm8971_snd_controls,
 				ARRAY_SIZE(wm8971_snd_controls));
 	wm8971_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8971: failed to register card\n");
-		goto card_err;
-	}
+
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 err:
 	kfree(codec->reg_cache);
 	return ret;
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 98d663afc97d..81c57b5c591c 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -276,41 +276,42 @@ static int wm8974_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
 struct pll_ {
-	unsigned int pre_div:4; /* prescale - 1 */
+	unsigned int pre_div:1;
 	unsigned int n:4;
 	unsigned int k;
 };
 
-static struct pll_ pll_div;
-
 /* The size in bits of the pll divide multiplied by 10
  * to allow rounding later */
 #define FIXED_PLL_SIZE ((1 << 24) * 10)
 
-static void pll_factors(unsigned int target, unsigned int source)
+static void pll_factors(struct pll_ *pll_div,
+			unsigned int target, unsigned int source)
 {
 	unsigned long long Kpart;
 	unsigned int K, Ndiv, Nmod;
 
+	/* There is a fixed divide by 4 in the output path */
+	target *= 4;
+
 	Ndiv = target / source;
 	if (Ndiv < 6) {
-		source >>= 1;
-		pll_div.pre_div = 1;
+		source /= 2;
+		pll_div->pre_div = 1;
 		Ndiv = target / source;
 	} else
-		pll_div.pre_div = 0;
+		pll_div->pre_div = 0;
 
 	if ((Ndiv < 6) || (Ndiv > 12))
 		printk(KERN_WARNING
 			"WM8974 N value %u outwith recommended range!\n",
 			Ndiv);
 
-	pll_div.n = Ndiv;
+	pll_div->n = Ndiv;
 	Nmod = target % source;
 	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
 
@@ -325,13 +326,14 @@ static void pll_factors(unsigned int target, unsigned int source)
 	/* Move down to proper range now rounding is done */
 	K /= 10;
 
-	pll_div.k = K;
+	pll_div->k = K;
 }
 
-static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct pll_ pll_div;
 	u16 reg;
 
 	if (freq_in == 0 || freq_out == 0) {
@@ -345,7 +347,7 @@ static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai,
 		return 0;
 	}
 
-	pll_factors(freq_out*4, freq_in);
+	pll_factors(&pll_div, freq_out, freq_in);
 
 	snd_soc_write(codec, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n);
 	snd_soc_write(codec, WM8974_PLLK1, pll_div.k >> 18);
@@ -638,17 +640,9 @@ static int wm8974_probe(struct platform_device *pdev)
 	snd_soc_add_controls(codec, wm8974_snd_controls,
 			     ARRAY_SIZE(wm8974_snd_controls));
 	wm8974_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;
 }
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 3f530f8a972a..2862e4dced27 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -790,19 +790,9 @@ static int wm8988_probe(struct platform_device *pdev)
 	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;
 }
@@ -944,21 +934,6 @@ static int wm8988_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8988_i2c_suspend(struct i2c_client *client, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm8988_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm8988_i2c_suspend NULL
-#define wm8988_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm8988_i2c_id[] = {
 	{ "wm8988", 0 },
 	{ }
@@ -972,8 +947,6 @@ static struct i2c_driver wm8988_i2c_driver = {
 	},
 	.probe = wm8988_i2c_probe,
 	.remove = wm8988_i2c_remove,
-	.suspend = wm8988_i2c_suspend,
-	.resume = wm8988_i2c_resume,
 	.id_table = wm8988_i2c_id,
 };
 #endif
@@ -1006,21 +979,6 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm8988_spi_suspend(struct spi_device *spi, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&spi->dev);
-}
-
-static int wm8988_spi_resume(struct spi_device *spi)
-{
-	return snd_soc_resume_device(&spi->dev);
-}
-#else
-#define wm8988_spi_suspend NULL
-#define wm8988_spi_resume NULL
-#endif
-
 static struct spi_driver wm8988_spi_driver = {
 	.driver = {
 		.name	= "wm8988",
@@ -1029,8 +987,6 @@ static struct spi_driver wm8988_spi_driver = {
 	},
 	.probe		= wm8988_spi_probe,
 	.remove		= __devexit_p(wm8988_spi_remove),
-	.suspend	= wm8988_spi_suspend,
-	.resume		= wm8988_spi_resume,
 };
 #endif
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 2d702db4131d..341481e0e830 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -920,7 +920,6 @@ static int wm8990_add_widgets(struct snd_soc_codec *codec)
 	/* set up the WM8990 audio map */
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -972,8 +971,8 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 	pll_div->k = K;
 }
 
-static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	u16 reg;
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -1409,16 +1408,9 @@ static int wm8990_init(struct snd_soc_device *socdev)
 	snd_soc_add_controls(codec, wm8990_snd_controls,
 				ARRAY_SIZE(wm8990_snd_controls));
 	wm8990_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8990: failed to register card\n");
-		goto card_err;
-	}
+
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 pcm_err:
 	kfree(codec->reg_cache);
 	return ret;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index d9987999e92c..5e32f2ed5fc2 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -422,7 +422,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 	return 0;
 }
 
-static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id,
+static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source,
 			  unsigned int Fref, unsigned int Fout)
 {
 	struct snd_soc_codec *codec = dai->codec;
@@ -1464,19 +1464,8 @@ static int wm8993_probe(struct platform_device *pdev)
 	wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff,
 				    wm8993->pdata.lineout2_diff);
 
-	snd_soc_dapm_new_widgets(codec);
-
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		dev_err(codec->dev, "failed to register card\n");
-		goto card_err;
-	}
-
 	return ret;
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 err:
 	return ret;
 }
@@ -1572,33 +1561,15 @@ static int wm8993_i2c_probe(struct i2c_client *i2c,
 	/* Use automatic clock configuration */
 	snd_soc_update_bits(codec, WM8993_CLOCKING_4, WM8993_SR_MODE, 0);
 
-	if (!wm8993->pdata.lineout1_diff)
-		snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
-				    WM8993_LINEOUT1_MODE,
-				    WM8993_LINEOUT1_MODE);
-	if (!wm8993->pdata.lineout2_diff)
-		snd_soc_update_bits(codec, WM8993_LINE_MIXER2,
-				    WM8993_LINEOUT2_MODE,
-				    WM8993_LINEOUT2_MODE);
-
-	if (wm8993->pdata.lineout1fb)
-		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
-				    WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
-
-	if (wm8993->pdata.lineout2fb)
-		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
-				    WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
-
-	/* Apply the microphone bias/detection configuration - the
-	 * platform data is directly applicable to the register. */
-	snd_soc_update_bits(codec, WM8993_MICBIAS,
-			    WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
-			    WM8993_MICB1_LVL | WM8993_MICB2_LVL,
-			    wm8993->pdata.jd_scthr << WM8993_JD_SCTHR_SHIFT |
-			    wm8993->pdata.jd_thr << WM8993_JD_THR_SHIFT |
-			    wm8993->pdata.micbias1_lvl |
-			    wm8993->pdata.micbias1_lvl << 1);
-
+	wm_hubs_handle_analogue_pdata(codec, wm8993->pdata.lineout1_diff,
+				      wm8993->pdata.lineout2_diff,
+				      wm8993->pdata.lineout1fb,
+				      wm8993->pdata.lineout2fb,
+				      wm8993->pdata.jd_scthr,
+				      wm8993->pdata.jd_thr,
+				      wm8993->pdata.micbias1_lvl,
+				      wm8993->pdata.micbias2_lvl);
+			     
 	ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	if (ret != 0)
 		goto err;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 686e5aa97206..c468497314ba 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1262,19 +1262,9 @@ static int wm9081_probe(struct platform_device *pdev)
 	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;
 }
@@ -1452,21 +1442,6 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int wm9081_i2c_suspend(struct i2c_client *client, pm_message_t msg)
-{
-	return snd_soc_suspend_device(&client->dev);
-}
-
-static int wm9081_i2c_resume(struct i2c_client *client)
-{
-	return snd_soc_resume_device(&client->dev);
-}
-#else
-#define wm9081_i2c_suspend NULL
-#define wm9081_i2c_resume NULL
-#endif
-
 static const struct i2c_device_id wm9081_i2c_id[] = {
 	{ "wm9081", 0 },
 	{ }
@@ -1480,8 +1455,6 @@ static struct i2c_driver wm9081_i2c_driver = {
 	},
 	.probe =    wm9081_i2c_probe,
 	.remove =   __devexit_p(wm9081_i2c_remove),
-	.suspend =  wm9081_i2c_suspend,
-	.resume =   wm9081_i2c_resume,
 	.id_table = wm9081_i2c_id,
 };
 
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index e7d2840d9e59..ec54c6da9856 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -205,7 +205,6 @@ static int wm9705_add_widgets(struct snd_soc_codec *codec)
 	snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets,
 					ARRAY_SIZE(wm9705_dapm_widgets));
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
-	snd_soc_dapm_new_widgets(codec);
 
 	return 0;
 }
@@ -403,12 +402,6 @@ static int wm9705_soc_probe(struct platform_device *pdev)
 				ARRAY_SIZE(wm9705_snd_ac97_controls));
 	wm9705_add_widgets(codec);
 
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm9705: failed to register card\n");
-		goto reset_err;
-	}
-
 	return 0;
 
 reset_err:
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 1fd4e88f50cf..0ac1215dcd9b 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -436,7 +436,6 @@ static int wm9712_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -695,17 +694,11 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	snd_soc_add_controls(codec, wm9712_snd_ac97_controls,
 				ARRAY_SIZE(wm9712_snd_ac97_controls));
 	wm9712_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm9712: failed to register card\n");
-		goto reset_err;
-	}
 
 	return 0;
 
 reset_err:
 	snd_soc_free_pcms(socdev);
-
 pcm_err:
 	snd_soc_free_ac97_codec(codec);
 
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index abed37acf787..c58aab375edb 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -165,9 +165,9 @@ SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
 SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
 SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
 
-SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
-SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
-SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
+SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
+SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
+SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
 
 SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
 SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
@@ -266,7 +266,7 @@ static int mixer_event(struct snd_soc_dapm_widget *w,
 
 /* Left Headphone Mixers */
 static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Beep Playback Switch", HPL_MIXER, 5, 1, 0),
 SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
 SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
 SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
@@ -276,7 +276,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
 
 /* Right Headphone Mixers */
 static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0),
+SOC_DAPM_SINGLE("Beep Playback Switch", HPR_MIXER, 5, 1, 0),
 SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
 SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
 SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
@@ -294,7 +294,7 @@ SOC_DAPM_ENUM("Route", wm9713_enum[0]);
 
 /* Speaker Mixer */
 static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1),
+SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 11, 1, 1),
 SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
 SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
 SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
@@ -304,7 +304,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
 
 /* Mono Mixer */
 static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
-SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1),
+SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 7, 1, 1),
 SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
 SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
 SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
@@ -463,7 +463,7 @@ SND_SOC_DAPM_VMID("VMID"),
 
 static const struct snd_soc_dapm_route audio_map[] = {
 	/* left HP mixer */
-	{"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+	{"Left HP Mixer", "Beep Playback Switch",    "PCBEEP"},
 	{"Left HP Mixer", "Voice Playback Switch",   "Voice DAC"},
 	{"Left HP Mixer", "Aux Playback Switch",     "Aux DAC"},
 	{"Left HP Mixer", "Bypass Playback Switch",  "Left Line In"},
@@ -472,7 +472,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Left HP Mixer", NULL,  "Capture Headphone Mux"},
 
 	/* right HP mixer */
-	{"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
+	{"Right HP Mixer", "Beep Playback Switch",    "PCBEEP"},
 	{"Right HP Mixer", "Voice Playback Switch",   "Voice DAC"},
 	{"Right HP Mixer", "Aux Playback Switch",     "Aux DAC"},
 	{"Right HP Mixer", "Bypass Playback Switch",  "Right Line In"},
@@ -491,7 +491,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Capture Mixer", NULL, "Right Capture Source"},
 
 	/* speaker mixer */
-	{"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"},
+	{"Speaker Mixer", "Beep Playback Switch",    "PCBEEP"},
 	{"Speaker Mixer", "Voice Playback Switch",   "Voice DAC"},
 	{"Speaker Mixer", "Aux Playback Switch",     "Aux DAC"},
 	{"Speaker Mixer", "Bypass Playback Switch",  "Line Mixer"},
@@ -499,7 +499,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Speaker Mixer", "MonoIn Playback Switch",  "Mono In"},
 
 	/* mono mixer */
-	{"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"},
+	{"Mono Mixer", "Beep Playback Switch",    "PCBEEP"},
 	{"Mono Mixer", "Voice Playback Switch",   "Voice DAC"},
 	{"Mono Mixer", "Aux Playback Switch",     "Aux DAC"},
 	{"Mono Mixer", "Bypass Playback Switch",  "Line Mixer"},
@@ -625,7 +625,6 @@ static int wm9713_add_widgets(struct snd_soc_codec *codec)
 
 	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
 
-	snd_soc_dapm_new_widgets(codec);
 	return 0;
 }
 
@@ -800,8 +799,8 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out)
+static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	return wm9713_set_pll(codec, pll_id, freq_in, freq_out);
@@ -1247,14 +1246,11 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	snd_soc_add_controls(codec, wm9713_snd_ac97_controls,
 				ARRAY_SIZE(wm9713_snd_ac97_controls));
 	wm9713_add_widgets(codec);
-	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);
 
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e542027eea89..d73c30536a2c 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -438,11 +438,11 @@ static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("IN1LN"),
 SND_SOC_DAPM_INPUT("IN1LP"),
 SND_SOC_DAPM_INPUT("IN2LN"),
-SND_SOC_DAPM_INPUT("IN2LP/VXRN"),
+SND_SOC_DAPM_INPUT("IN2LP:VXRN"),
 SND_SOC_DAPM_INPUT("IN1RN"),
 SND_SOC_DAPM_INPUT("IN1RP"),
 SND_SOC_DAPM_INPUT("IN2RN"),
-SND_SOC_DAPM_INPUT("IN2RP/VXRP"),
+SND_SOC_DAPM_INPUT("IN2RP:VXRP"),
 
 SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0),
 SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0),
@@ -537,14 +537,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "IN1R PGA", "IN1RP Switch", "IN1RP" },
 	{ "IN1R PGA", "IN1RN Switch", "IN1RN" },
 
-	{ "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" },
+	{ "IN2L PGA", "IN2LP Switch", "IN2LP:VXRN" },
 	{ "IN2L PGA", "IN2LN Switch", "IN2LN" },
 
-	{ "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" },
+	{ "IN2R PGA", "IN2RP Switch", "IN2RP:VXRP" },
 	{ "IN2R PGA", "IN2RN Switch", "IN2RN" },
 
-	{ "Direct Voice", NULL, "IN2LP/VXRN" },
-	{ "Direct Voice", NULL, "IN2RP/VXRP" },
+	{ "Direct Voice", NULL, "IN2LP:VXRN" },
+	{ "Direct Voice", NULL, "IN2RP:VXRP" },
 
 	{ "MIXINL", "IN1L Switch", "IN1L PGA" },
 	{ "MIXINL", "IN2L Switch", "IN2L PGA" },
@@ -565,7 +565,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "Left Output Mixer", "Right Input Switch", "MIXINR" },
 	{ "Left Output Mixer", "IN2RN Switch", "IN2RN" },
 	{ "Left Output Mixer", "IN2LN Switch", "IN2LN" },
-	{ "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" },
+	{ "Left Output Mixer", "IN2LP Switch", "IN2LP:VXRN" },
 	{ "Left Output Mixer", "IN1L Switch", "IN1L PGA" },
 	{ "Left Output Mixer", "IN1R Switch", "IN1R PGA" },
 
@@ -573,7 +573,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "Right Output Mixer", "Right Input Switch", "MIXINR" },
 	{ "Right Output Mixer", "IN2LN Switch", "IN2LN" },
 	{ "Right Output Mixer", "IN2RN Switch", "IN2RN" },
-	{ "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" },
+	{ "Right Output Mixer", "IN2RP Switch", "IN2RP:VXRP" },
 	{ "Right Output Mixer", "IN1L Switch", "IN1L PGA" },
 	{ "Right Output Mixer", "IN1R Switch", "IN1R PGA" },
 
@@ -738,6 +738,41 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
 }
 EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes);
 
+int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec,
+				  int lineout1_diff, int lineout2_diff,
+				  int lineout1fb, int lineout2fb,
+				  int jd_scthr, int jd_thr, int micbias1_lvl,
+				  int micbias2_lvl)
+{
+	if (!lineout1_diff)
+		snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
+				    WM8993_LINEOUT1_MODE,
+				    WM8993_LINEOUT1_MODE);
+	if (!lineout2_diff)
+		snd_soc_update_bits(codec, WM8993_LINE_MIXER2,
+				    WM8993_LINEOUT2_MODE,
+				    WM8993_LINEOUT2_MODE);
+
+	if (lineout1fb)
+		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
+				    WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
+
+	if (lineout2fb)
+		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
+				    WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
+
+	snd_soc_update_bits(codec, WM8993_MICBIAS,
+			    WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
+			    WM8993_MICB1_LVL | WM8993_MICB2_LVL,
+			    jd_scthr << WM8993_JD_SCTHR_SHIFT |
+			    jd_thr << WM8993_JD_THR_SHIFT |
+			    micbias1_lvl |
+			    micbias2_lvl << WM8993_MICB2_LVL_SHIFT);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata);
+
 MODULE_DESCRIPTION("Shared support for Wolfson hubs products");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index ec09cb6a2939..36d3fba1de8b 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -20,5 +20,10 @@ extern const unsigned int wm_hubs_spkmix_tlv[];
 
 extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
 extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int);
+extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *,
+					 int lineout1_diff, int lineout2_diff,
+					 int lineout1fb, int lineout2fb,
+					 int jd_scthr, int jd_thr,
+					 int micbias1_lvl, int micbias2_lvl);
 
 #endif
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index 4dfd4ad9d90e..047ee39418c0 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -13,9 +13,9 @@ config SND_DAVINCI_SOC_MCASP
 	tristate
 
 config SND_DAVINCI_SOC_EVM
-	tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM"
+	tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM"
 	depends on SND_DAVINCI_SOC
-	depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM
+	depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM  || MACH_DAVINCI_DM365_EVM
 	select SND_DAVINCI_SOC_I2S
 	select SND_SOC_TLV320AIC3X
 	help
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 67414f659405..7ccbe6684fc2 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -45,7 +45,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream,
 	unsigned sysclk;
 
 	/* ASP1 on DM355 EVM is clocked by an external oscillator */
-	if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm())
+	if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() ||
+	    machine_is_davinci_dm365_evm())
 		sysclk = 27000000;
 
 	/* ASP0 in DM6446 EVM is clocked by U55, as configured by
@@ -176,7 +177,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
 	.ops = &evm_ops,
 };
 
-/* davinci-evm audio machine driver */
+/* davinci dm6446, dm355 or dm365 evm audio machine driver */
 static struct snd_soc_card snd_soc_card_evm = {
 	.name = "DaVinci EVM",
 	.platform = &davinci_soc_platform,
@@ -243,7 +244,7 @@ static int __init evm_init(void)
 	int index;
 	int ret;
 
-	if (machine_is_davinci_evm()) {
+	if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) {
 		evm_snd_dev_data = &evm_snd_devdata;
 		index = 0;
 	} else if (machine_is_davinci_dm355_evm()) {
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 4ae707048021..6362ca05506e 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -97,12 +97,24 @@ enum {
 	DAVINCI_MCBSP_WORD_32,
 };
 
+static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = {
+	[SNDRV_PCM_FORMAT_S8]		= 1,
+	[SNDRV_PCM_FORMAT_S16_LE]	= 2,
+	[SNDRV_PCM_FORMAT_S32_LE]	= 4,
+};
+
+static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = {
+	[SNDRV_PCM_FORMAT_S8]		= DAVINCI_MCBSP_WORD_8,
+	[SNDRV_PCM_FORMAT_S16_LE]	= DAVINCI_MCBSP_WORD_16,
+	[SNDRV_PCM_FORMAT_S32_LE]	= DAVINCI_MCBSP_WORD_32,
+};
+
+static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = {
+	[SNDRV_PCM_FORMAT_S8]		= SNDRV_PCM_FORMAT_S16_LE,
+	[SNDRV_PCM_FORMAT_S16_LE]	= SNDRV_PCM_FORMAT_S32_LE,
+};
+
 struct davinci_mcbsp_dev {
-	/*
-	 * dma_params must be first because rtd->dai->cpu_dai->private_data
-	 * is cast to a pointer of an array of struct davinci_pcm_dma_params in
-	 * davinci_pcm_open.
-	 */
 	struct davinci_pcm_dma_params	dma_params[2];
 	void __iomem			*base;
 #define MOD_DSP_A	0
@@ -110,6 +122,27 @@ struct davinci_mcbsp_dev {
 	int				mode;
 	u32				pcr;
 	struct clk			*clk;
+	/*
+	 * Combining both channels into 1 element will at least double the
+	 * amount of time between servicing the dma channel, increase
+	 * effiency, and reduce the chance of overrun/underrun. But,
+	 * it will result in the left & right channels being swapped.
+	 *
+	 * If relabeling the left and right channels is not possible,
+	 * you may want to let the codec know to swap them back.
+	 *
+	 * It may allow x10 the amount of time to service dma requests,
+	 * if the codec is master and is using an unnecessarily fast bit clock
+	 * (ie. tlvaic23b), independent of the sample rate. So, having an
+	 * entire frame at once means it can be serviced at the sample rate
+	 * instead of the bit clock rate.
+	 *
+	 * In the now unlikely case that an underrun still
+	 * occurs, both the left and right samples will be repeated
+	 * so that no pops are heard, and the left and right channels
+	 * won't end up being swapped because of the underrun.
+	 */
+	unsigned enable_channel_combine:1;
 };
 
 static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
@@ -349,6 +382,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 	int mcbsp_word_length;
 	unsigned int rcr, xcr, srgr;
 	u32 spcr;
+	snd_pcm_format_t fmt;
+	unsigned element_cnt = 1;
 
 	/* general line settings */
 	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
@@ -378,27 +413,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 		xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
 	}
 	/* Determine xfer data type */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S8:
-		dma_params->data_type = 1;
-		mcbsp_word_length = DAVINCI_MCBSP_WORD_8;
-		break;
-	case SNDRV_PCM_FORMAT_S16_LE:
-		dma_params->data_type = 2;
-		mcbsp_word_length = DAVINCI_MCBSP_WORD_16;
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-		dma_params->data_type = 4;
-		mcbsp_word_length = DAVINCI_MCBSP_WORD_32;
-		break;
-	default:
+	fmt = params_format(params);
+	if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) {
 		printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n");
 		return -EINVAL;
 	}
 
-	dma_params->acnt  = dma_params->data_type;
-	rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1);
-	xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1);
+	if (params_channels(params) == 2) {
+		element_cnt = 2;
+		if (double_fmt[fmt] && dev->enable_channel_combine) {
+			element_cnt = 1;
+			fmt = double_fmt[fmt];
+		}
+	}
+	dma_params->acnt = dma_params->data_type = data_type[fmt];
+	dma_params->fifo_level = 0;
+	mcbsp_word_length = asp_word_length[fmt];
+	rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1);
+	xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1);
 
 	rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
 		DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length);
@@ -513,7 +545,13 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 		ret = -ENOMEM;
 		goto err_release_region;
 	}
-
+	if (pdata) {
+		dev->enable_channel_combine = pdata->enable_channel_combine;
+		dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size =
+			pdata->sram_size_playback;
+		dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size =
+			pdata->sram_size_capture;
+	}
 	dev->clk = clk_get(&pdev->dev, NULL);
 	if (IS_ERR(dev->clk)) {
 		ret = -ENODEV;
@@ -547,6 +585,7 @@ static int davinci_i2s_probe(struct platform_device *pdev)
 	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
 
 	davinci_i2s_dai.private_data = dev;
+	davinci_i2s_dai.dma_data = dev->dma_params;
 	ret = snd_soc_register_dai(&davinci_i2s_dai);
 	if (ret != 0)
 		goto err_free_mem;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 5d1f98a4c978..0a302e1080d9 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -714,16 +714,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	struct davinci_pcm_dma_params *dma_params =
 					&dev->dma_params[substream->stream];
 	int word_length;
-	u8 numevt;
+	u8 fifo_level;
 
 	davinci_hw_common_param(dev, substream->stream);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		numevt = dev->txnumevt;
+		fifo_level = dev->txnumevt;
 	else
-		numevt = dev->rxnumevt;
-
-	if (!numevt)
-		numevt = 1;
+		fifo_level = dev->rxnumevt;
 
 	if (dev->op_mode == DAVINCI_MCASP_DIT_MODE)
 		davinci_hw_dit_param(dev);
@@ -751,12 +748,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	if (dev->version == MCASP_VERSION_2) {
-		dma_params->data_type *= numevt;
-		dma_params->acnt = 4 * numevt;
-	} else
+	if (dev->version == MCASP_VERSION_2 && !fifo_level)
+		dma_params->acnt = 4;
+	else
 		dma_params->acnt = dma_params->data_type;
 
+	dma_params->fifo_level = fifo_level;
 	davinci_config_channel_size(dev, word_length);
 
 	return 0;
@@ -907,6 +904,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	dma_data->channel = res->start;
 	davinci_mcasp_dai[pdata->op_mode].private_data = dev;
+	davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params;
 	davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev;
 	ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]);
 
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 9d179cc88f7b..582c9249ef09 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -39,11 +39,6 @@ enum {
 };
 
 struct davinci_audio_dev {
-	/*
-	 * dma_params must be first because rtd->dai->cpu_dai->private_data
-	 * is cast to a pointer of an array of struct davinci_pcm_dma_params in
-	 * davinci_pcm_open.
-	 */
 	struct davinci_pcm_dma_params dma_params[2];
 	void __iomem *base;
 	int sample_rate;
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index c73a915f233f..ad4d7f47a86b 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -3,6 +3,7 @@
  *
  * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
  * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ * added SRAM ping/pong (C) 2008 Troy Kisky <troy.kisky@boundarydevices.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
@@ -23,10 +24,29 @@
 
 #include <asm/dma.h>
 #include <mach/edma.h>
+#include <mach/sram.h>
 
 #include "davinci-pcm.h"
 
-static struct snd_pcm_hardware davinci_pcm_hardware = {
+#ifdef DEBUG
+static void print_buf_info(int slot, char *name)
+{
+	struct edmacc_param p;
+	if (slot < 0)
+		return;
+	edma_read_slot(slot, &p);
+	printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n",
+			name, slot, p.opt, p.src, p.a_b_cnt, p.dst);
+	printk(KERN_DEBUG "    src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n",
+			p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt);
+}
+#else
+static void print_buf_info(int slot, char *name)
+{
+}
+#endif
+
+static struct snd_pcm_hardware pcm_hardware_playback = {
 	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
 		 SNDRV_PCM_INFO_PAUSE),
@@ -48,102 +68,432 @@ static struct snd_pcm_hardware davinci_pcm_hardware = {
 	.fifo_size = 0,
 };
 
+static struct snd_pcm_hardware pcm_hardware_capture = {
+	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_PAUSE),
+	.formats = (SNDRV_PCM_FMTBIT_S16_LE),
+	.rates = (SNDRV_PCM_RATE_8000 | 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 |
+		  SNDRV_PCM_RATE_KNOT),
+	.rate_min = 8000,
+	.rate_max = 96000,
+	.channels_min = 2,
+	.channels_max = 2,
+	.buffer_bytes_max = 128 * 1024,
+	.period_bytes_min = 32,
+	.period_bytes_max = 8 * 1024,
+	.periods_min = 16,
+	.periods_max = 255,
+	.fifo_size = 0,
+};
+
+/*
+ * How ping/pong works....
+ *
+ * Playback:
+ * ram_params - copys 2*ping_size from start of SDRAM to iram,
+ * 	links to ram_link2
+ * ram_link2 - copys rest of SDRAM to iram in ping_size units,
+ * 	links to ram_link
+ * ram_link - copys entire SDRAM to iram in ping_size uints,
+ * 	links to self
+ *
+ * asp_params - same as asp_link[0]
+ * asp_link[0] - copys from lower half of iram to asp port
+ * 	links to asp_link[1], triggers iram copy event on completion
+ * asp_link[1] - copys from upper half of iram to asp port
+ * 	links to asp_link[0], triggers iram copy event on completion
+ * 	triggers interrupt only needed to let upper SOC levels update position
+ * 	in stream on completion
+ *
+ * When playback is started:
+ * 	ram_params started
+ * 	asp_params started
+ *
+ * Capture:
+ * ram_params - same as ram_link,
+ * 	links to ram_link
+ * ram_link - same as playback
+ * 	links to self
+ *
+ * asp_params - same as playback
+ * asp_link[0] - same as playback
+ * asp_link[1] - same as playback
+ *
+ * When capture is started:
+ * 	asp_params started
+ */
 struct davinci_runtime_data {
 	spinlock_t lock;
 	int period;		/* current DMA period */
-	int master_lch;		/* Master DMA channel */
-	int slave_lch;		/* linked parameter RAM reload slot */
+	int asp_channel;	/* Master DMA channel */
+	int asp_link[2];	/* asp parameter link channel, ping/pong */
 	struct davinci_pcm_dma_params *params;	/* DMA params */
+	int ram_channel;
+	int ram_link;
+	int ram_link2;
+	struct edmacc_param asp_params;
+	struct edmacc_param ram_params;
 };
 
+/*
+ * Not used with ping/pong
+ */
 static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 {
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	int lch = prtd->slave_lch;
+	int link = prtd->asp_link[0];
 	unsigned int period_size;
 	unsigned int dma_offset;
 	dma_addr_t dma_pos;
 	dma_addr_t src, dst;
 	unsigned short src_bidx, dst_bidx;
+	unsigned short src_cidx, dst_cidx;
 	unsigned int data_type;
 	unsigned short acnt;
 	unsigned int count;
+	unsigned int fifo_level;
 
 	period_size = snd_pcm_lib_period_bytes(substream);
 	dma_offset = prtd->period * period_size;
 	dma_pos = runtime->dma_addr + dma_offset;
+	fifo_level = prtd->params->fifo_level;
 
 	pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
-		"dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size);
+		"dma_ptr = %x period_size=%x\n", link, dma_pos, period_size);
 
 	data_type = prtd->params->data_type;
 	count = period_size / data_type;
+	if (fifo_level)
+		count /= fifo_level;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		src = dma_pos;
 		dst = prtd->params->dma_addr;
 		src_bidx = data_type;
 		dst_bidx = 0;
+		src_cidx = data_type * fifo_level;
+		dst_cidx = 0;
 	} else {
 		src = prtd->params->dma_addr;
 		dst = dma_pos;
 		src_bidx = 0;
 		dst_bidx = data_type;
+		src_cidx = 0;
+		dst_cidx = data_type * fifo_level;
 	}
 
 	acnt = prtd->params->acnt;
-	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, acnt, count, 1, 0, ASYNC);
+	edma_set_src(link, src, INCR, W8BIT);
+	edma_set_dest(link, dst, INCR, W8BIT);
+
+	edma_set_src_index(link, src_bidx, src_cidx);
+	edma_set_dest_index(link, dst_bidx, dst_cidx);
+
+	if (!fifo_level)
+		edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC);
+	else
+		edma_set_transfer_params(link, acnt, fifo_level, count,
+							fifo_level, ABSYNC);
 
 	prtd->period++;
 	if (unlikely(prtd->period >= runtime->periods))
 		prtd->period = 0;
 }
 
-static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data)
+static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
 {
 	struct snd_pcm_substream *substream = data;
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
 
-	pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status);
+	print_buf_info(prtd->ram_channel, "i ram_channel");
+	pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status);
 
 	if (unlikely(ch_status != DMA_COMPLETE))
 		return;
 
 	if (snd_pcm_running(substream)) {
+		if (prtd->ram_channel < 0) {
+			/* No ping/pong must fix up link dma data*/
+			spin_lock(&prtd->lock);
+			davinci_pcm_enqueue_dma(substream);
+			spin_unlock(&prtd->lock);
+		}
 		snd_pcm_period_elapsed(substream);
+	}
+}
+
+static int allocate_sram(struct snd_pcm_substream *substream, unsigned size,
+		struct snd_pcm_hardware *ppcm)
+{
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	struct snd_dma_buffer *iram_dma = NULL;
+	dma_addr_t iram_phys = 0;
+	void *iram_virt = NULL;
+
+	if (buf->private_data || !size)
+		return 0;
+
+	ppcm->period_bytes_max = size;
+	iram_virt = sram_alloc(size, &iram_phys);
+	if (!iram_virt)
+		goto exit1;
+	iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL);
+	if (!iram_dma)
+		goto exit2;
+	iram_dma->area = iram_virt;
+	iram_dma->addr = iram_phys;
+	memset(iram_dma->area, 0, size);
+	iram_dma->bytes = size;
+	buf->private_data = iram_dma;
+	return 0;
+exit2:
+	if (iram_virt)
+		sram_free(iram_virt, size);
+exit1:
+	return -ENOMEM;
+}
+
+/*
+ * Only used with ping/pong.
+ * This is called after runtime->dma_addr, period_bytes and data_type are valid
+ */
+static int ping_pong_dma_setup(struct snd_pcm_substream *substream)
+{
+	unsigned short ram_src_cidx, ram_dst_cidx;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct davinci_runtime_data *prtd = runtime->private_data;
+	struct snd_dma_buffer *iram_dma =
+		(struct snd_dma_buffer *)substream->dma_buffer.private_data;
+	struct davinci_pcm_dma_params *params = prtd->params;
+	unsigned int data_type = params->data_type;
+	unsigned int acnt = params->acnt;
+	/* divide by 2 for ping/pong */
+	unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1;
+	int link = prtd->asp_link[1];
+	unsigned int fifo_level = prtd->params->fifo_level;
+	unsigned int count;
+	if ((data_type == 0) || (data_type > 4)) {
+		printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type);
+		return -EINVAL;
+	}
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dma_addr_t asp_src_pong = iram_dma->addr + ping_size;
+		ram_src_cidx = ping_size;
+		ram_dst_cidx = -ping_size;
+		edma_set_src(link, asp_src_pong, INCR, W8BIT);
+
+		link = prtd->asp_link[0];
+		edma_set_src_index(link, data_type, data_type * fifo_level);
+		link = prtd->asp_link[1];
+		edma_set_src_index(link, data_type, data_type * fifo_level);
+
+		link = prtd->ram_link;
+		edma_set_src(link, runtime->dma_addr, INCR, W32BIT);
+	} else {
+		dma_addr_t asp_dst_pong = iram_dma->addr + ping_size;
+		ram_src_cidx = -ping_size;
+		ram_dst_cidx = ping_size;
+		edma_set_dest(link, asp_dst_pong, INCR, W8BIT);
+
+		link = prtd->asp_link[0];
+		edma_set_dest_index(link, data_type, data_type * fifo_level);
+		link = prtd->asp_link[1];
+		edma_set_dest_index(link, data_type, data_type * fifo_level);
+
+		link = prtd->ram_link;
+		edma_set_dest(link, runtime->dma_addr, INCR, W32BIT);
+	}
 
-		spin_lock(&prtd->lock);
-		davinci_pcm_enqueue_dma(substream);
-		spin_unlock(&prtd->lock);
+	if (!fifo_level) {
+		count = ping_size / data_type;
+		edma_set_transfer_params(prtd->asp_link[0], acnt, count,
+				1, 0, ASYNC);
+		edma_set_transfer_params(prtd->asp_link[1], acnt, count,
+				1, 0, ASYNC);
+	} else {
+		count = ping_size / (data_type * fifo_level);
+		edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level,
+				count, fifo_level, ABSYNC);
+		edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level,
+				count, fifo_level, ABSYNC);
 	}
+
+	link = prtd->ram_link;
+	edma_set_src_index(link, ping_size, ram_src_cidx);
+	edma_set_dest_index(link, ping_size, ram_dst_cidx);
+	edma_set_transfer_params(link, ping_size, 2,
+			runtime->periods, 2, ASYNC);
+
+	/* init master params */
+	edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
+	edma_read_slot(prtd->ram_link, &prtd->ram_params);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		struct edmacc_param p_ram;
+		/* Copy entire iram buffer before playback started */
+		prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1);
+		/* 0 dst_bidx */
+		prtd->ram_params.src_dst_bidx = (ping_size << 1);
+		/* 0 dst_cidx */
+		prtd->ram_params.src_dst_cidx = (ping_size << 1);
+		prtd->ram_params.ccnt = 1;
+
+		/* Skip 1st period */
+		edma_read_slot(prtd->ram_link, &p_ram);
+		p_ram.src += (ping_size << 1);
+		p_ram.ccnt -= 1;
+		edma_write_slot(prtd->ram_link2, &p_ram);
+		/*
+		 * When 1st started, ram -> iram dma channel will fill the
+		 * entire iram.  Then, whenever a ping/pong asp buffer finishes,
+		 * 1/2 iram will be filled.
+		 */
+		prtd->ram_params.link_bcntrld =
+			EDMA_CHAN_SLOT(prtd->ram_link2) << 5;
+	}
+	return 0;
+}
+
+/* 1 asp tx or rx channel using 2 parameter channels
+ * 1 ram to/from iram channel using 1 parameter channel
+ *
+ * Playback
+ * ram copy channel kicks off first,
+ * 1st ram copy of entire iram buffer completion kicks off asp channel
+ * asp tcc always kicks off ram copy of 1/2 iram buffer
+ *
+ * Record
+ * asp channel starts, tcc kicks off ram copy
+ */
+static int request_ping_pong(struct snd_pcm_substream *substream,
+		struct davinci_runtime_data *prtd,
+		struct snd_dma_buffer *iram_dma)
+{
+	dma_addr_t asp_src_ping;
+	dma_addr_t asp_dst_ping;
+	int link;
+	struct davinci_pcm_dma_params *params = prtd->params;
+
+	/* Request ram master channel */
+	link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY,
+				  davinci_pcm_dma_irq, substream,
+				  EVENTQ_1);
+	if (link < 0)
+		goto exit1;
+
+	/* Request ram link channel */
+	link = prtd->ram_link = edma_alloc_slot(
+			EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY);
+	if (link < 0)
+		goto exit2;
+
+	link = prtd->asp_link[1] = edma_alloc_slot(
+			EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY);
+	if (link < 0)
+		goto exit3;
+
+	prtd->ram_link2 = -1;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		link = prtd->ram_link2 = edma_alloc_slot(
+			EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY);
+		if (link < 0)
+			goto exit4;
+	}
+	/* circle ping-pong buffers */
+	edma_link(prtd->asp_link[0], prtd->asp_link[1]);
+	edma_link(prtd->asp_link[1], prtd->asp_link[0]);
+	/* circle ram buffers */
+	edma_link(prtd->ram_link, prtd->ram_link);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		asp_src_ping = iram_dma->addr;
+		asp_dst_ping = params->dma_addr;	/* fifo */
+	} else {
+		asp_src_ping = params->dma_addr;	/* fifo */
+		asp_dst_ping = iram_dma->addr;
+	}
+	/* ping */
+	link = prtd->asp_link[0];
+	edma_set_src(link, asp_src_ping, INCR, W16BIT);
+	edma_set_dest(link, asp_dst_ping, INCR, W16BIT);
+	edma_set_src_index(link, 0, 0);
+	edma_set_dest_index(link, 0, 0);
+
+	edma_read_slot(link, &prtd->asp_params);
+	prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN);
+	prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f);
+	edma_write_slot(link, &prtd->asp_params);
+
+	/* pong */
+	link = prtd->asp_link[1];
+	edma_set_src(link, asp_src_ping, INCR, W16BIT);
+	edma_set_dest(link, asp_dst_ping, INCR, W16BIT);
+	edma_set_src_index(link, 0, 0);
+	edma_set_dest_index(link, 0, 0);
+
+	edma_read_slot(link, &prtd->asp_params);
+	prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f));
+	/* interrupt after every pong completion */
+	prtd->asp_params.opt |= TCINTEN | TCCHEN |
+		EDMA_TCC(EDMA_CHAN_SLOT(prtd->ram_channel));
+	edma_write_slot(link, &prtd->asp_params);
+
+	/* ram */
+	link = prtd->ram_link;
+	edma_set_src(link, iram_dma->addr, INCR, W32BIT);
+	edma_set_dest(link, iram_dma->addr, INCR, W32BIT);
+	pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u,"
+		"for asp:%u %u %u\n", __func__,
+		prtd->ram_channel, prtd->ram_link, prtd->ram_link2,
+		prtd->asp_channel, prtd->asp_link[0],
+		prtd->asp_link[1]);
+	return 0;
+exit4:
+	edma_free_channel(prtd->asp_link[1]);
+	prtd->asp_link[1] = -1;
+exit3:
+	edma_free_channel(prtd->ram_link);
+	prtd->ram_link = -1;
+exit2:
+	edma_free_channel(prtd->ram_channel);
+	prtd->ram_channel = -1;
+exit1:
+	return link;
 }
 
 static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 {
+	struct snd_dma_buffer *iram_dma;
 	struct davinci_runtime_data *prtd = substream->runtime->private_data;
-	struct edmacc_param p_ram;
-	int ret;
+	struct davinci_pcm_dma_params *params = prtd->params;
+	int link;
 
-	/* Request master DMA channel */
-	ret = edma_alloc_channel(prtd->params->channel,
-				  davinci_pcm_dma_irq, substream,
-				  EVENTQ_0);
-	if (ret < 0)
-		return ret;
-	prtd->master_lch = ret;
+	if (!params)
+		return -ENODEV;
 
-	/* Request parameter RAM reload slot */
-	ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY);
-	if (ret < 0) {
-		edma_free_channel(prtd->master_lch);
-		return ret;
+	/* Request asp master DMA channel */
+	link = prtd->asp_channel = edma_alloc_channel(params->channel,
+			davinci_pcm_dma_irq, substream, EVENTQ_0);
+	if (link < 0)
+		goto exit1;
+
+	/* Request asp link channels */
+	link = prtd->asp_link[0] = edma_alloc_slot(
+			EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY);
+	if (link < 0)
+		goto exit2;
+
+	iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data;
+	if (iram_dma) {
+		if (request_ping_pong(substream, prtd, iram_dma) == 0)
+			return 0;
+		printk(KERN_WARNING "%s: dma channel allocation failed,"
+				"not using sram\n", __func__);
 	}
-	prtd->slave_lch = ret;
 
 	/* Issue transfer completion IRQ when the channel completes a
 	 * transfer, then always reload from the same slot (by a kind
@@ -154,12 +504,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
 	 * 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(EDMA_CHAN_SLOT(prtd->master_lch));
-	p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5;
-	edma_write_slot(prtd->slave_lch, &p_ram);
-
+	edma_read_slot(link, &prtd->asp_params);
+	prtd->asp_params.opt |= TCINTEN |
+		EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel));
+	prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5;
+	edma_write_slot(link, &prtd->asp_params);
 	return 0;
+exit2:
+	edma_free_channel(prtd->asp_channel);
+	prtd->asp_channel = -1;
+exit1:
+	return link;
 }
 
 static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -173,12 +528,12 @@ 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:
-		edma_start(prtd->master_lch);
+		edma_resume(prtd->asp_channel);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		edma_stop(prtd->master_lch);
+		edma_pause(prtd->asp_channel);
 		break;
 	default:
 		ret = -EINVAL;
@@ -193,15 +548,37 @@ 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 edmacc_param temp;
 
+	if (prtd->ram_channel >= 0) {
+		int ret = ping_pong_dma_setup(substream);
+		if (ret < 0)
+			return ret;
+
+		edma_write_slot(prtd->ram_channel, &prtd->ram_params);
+		edma_write_slot(prtd->asp_channel, &prtd->asp_params);
+
+		print_buf_info(prtd->ram_channel, "ram_channel");
+		print_buf_info(prtd->ram_link, "ram_link");
+		print_buf_info(prtd->ram_link2, "ram_link2");
+		print_buf_info(prtd->asp_channel, "asp_channel");
+		print_buf_info(prtd->asp_link[0], "asp_link[0]");
+		print_buf_info(prtd->asp_link[1], "asp_link[1]");
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			/* copy 1st iram buffer */
+			edma_start(prtd->ram_channel);
+		}
+		edma_start(prtd->asp_channel);
+		return 0;
+	}
 	prtd->period = 0;
 	davinci_pcm_enqueue_dma(substream);
 
 	/* Copy self-linked parameter RAM entry into master channel */
-	edma_read_slot(prtd->slave_lch, &temp);
-	edma_write_slot(prtd->master_lch, &temp);
+	edma_read_slot(prtd->asp_link[0], &prtd->asp_params);
+	edma_write_slot(prtd->asp_channel, &prtd->asp_params);
 	davinci_pcm_enqueue_dma(substream);
+	edma_start(prtd->asp_channel);
 
 	return 0;
 }
@@ -212,20 +589,53 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct davinci_runtime_data *prtd = runtime->private_data;
 	unsigned int offset;
-	dma_addr_t count;
-	dma_addr_t src, dst;
+	int asp_count;
+	dma_addr_t asp_src, asp_dst;
 
 	spin_lock(&prtd->lock);
-
-	edma_get_position(prtd->master_lch, &src, &dst);
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		count = src - runtime->dma_addr;
-	else
-		count = dst - runtime->dma_addr;
-
+	if (prtd->ram_channel >= 0) {
+		int ram_count;
+		int mod_ram;
+		dma_addr_t ram_src, ram_dst;
+		unsigned int period_size = snd_pcm_lib_period_bytes(substream);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			/* reading ram before asp should be safe
+			 * as long as the asp transfers less than a ping size
+			 * of bytes between the 2 reads
+			 */
+			edma_get_position(prtd->ram_channel,
+					&ram_src, &ram_dst);
+			edma_get_position(prtd->asp_channel,
+					&asp_src, &asp_dst);
+			asp_count = asp_src - prtd->asp_params.src;
+			ram_count = ram_src - prtd->ram_params.src;
+			mod_ram = ram_count % period_size;
+			mod_ram -= asp_count;
+			if (mod_ram < 0)
+				mod_ram += period_size;
+			else if (mod_ram == 0) {
+				if (snd_pcm_running(substream))
+					mod_ram += period_size;
+			}
+			ram_count -= mod_ram;
+			if (ram_count < 0)
+				ram_count += period_size * runtime->periods;
+		} else {
+			edma_get_position(prtd->ram_channel,
+					&ram_src, &ram_dst);
+			ram_count = ram_dst - prtd->ram_params.dst;
+		}
+		asp_count = ram_count;
+	} else {
+		edma_get_position(prtd->asp_channel, &asp_src, &asp_dst);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			asp_count = asp_src - runtime->dma_addr;
+		else
+			asp_count = asp_dst - runtime->dma_addr;
+	}
 	spin_unlock(&prtd->lock);
 
-	offset = bytes_to_frames(runtime, count);
+	offset = bytes_to_frames(runtime, asp_count);
 	if (offset >= runtime->buffer_size)
 		offset = 0;
 
@@ -236,14 +646,19 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct davinci_runtime_data *prtd;
+	struct snd_pcm_hardware *ppcm;
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data;
-	struct davinci_pcm_dma_params *params = &pa[substream->stream];
-	if (!params)
+	struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data;
+	struct davinci_pcm_dma_params *params;
+	if (!pa)
 		return -ENODEV;
+	params = &pa[substream->stream];
 
-	snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware);
+	ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+			&pcm_hardware_playback : &pcm_hardware_capture;
+	allocate_sram(substream, params->sram_size, ppcm);
+	snd_soc_set_runtime_hwparams(substream, ppcm);
 	/* ensure that buffer size is a multiple of period size */
 	ret = snd_pcm_hw_constraint_integer(runtime,
 						SNDRV_PCM_HW_PARAM_PERIODS);
@@ -256,6 +671,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream)
 
 	spin_lock_init(&prtd->lock);
 	prtd->params = params;
+	prtd->asp_channel = -1;
+	prtd->asp_link[0] = prtd->asp_link[1] = -1;
+	prtd->ram_channel = -1;
+	prtd->ram_link = -1;
+	prtd->ram_link2 = -1;
 
 	runtime->private_data = prtd;
 
@@ -273,10 +693,29 @@ 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;
 
-	edma_unlink(prtd->slave_lch);
-
-	edma_free_slot(prtd->slave_lch);
-	edma_free_channel(prtd->master_lch);
+	if (prtd->ram_channel >= 0)
+		edma_stop(prtd->ram_channel);
+	if (prtd->asp_channel >= 0)
+		edma_stop(prtd->asp_channel);
+	if (prtd->asp_link[0] >= 0)
+		edma_unlink(prtd->asp_link[0]);
+	if (prtd->asp_link[1] >= 0)
+		edma_unlink(prtd->asp_link[1]);
+	if (prtd->ram_link >= 0)
+		edma_unlink(prtd->ram_link);
+
+	if (prtd->asp_link[0] >= 0)
+		edma_free_slot(prtd->asp_link[0]);
+	if (prtd->asp_link[1] >= 0)
+		edma_free_slot(prtd->asp_link[1]);
+	if (prtd->asp_channel >= 0)
+		edma_free_channel(prtd->asp_channel);
+	if (prtd->ram_link >= 0)
+		edma_free_slot(prtd->ram_link);
+	if (prtd->ram_link2 >= 0)
+		edma_free_slot(prtd->ram_link2);
+	if (prtd->ram_channel >= 0)
+		edma_free_channel(prtd->ram_channel);
 
 	kfree(prtd);
 
@@ -318,11 +757,11 @@ static struct snd_pcm_ops davinci_pcm_ops = {
 	.mmap = 	davinci_pcm_mmap,
 };
 
-static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
+		size_t size)
 {
 	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
 	struct snd_dma_buffer *buf = &substream->dma_buffer;
-	size_t size = davinci_pcm_hardware.buffer_bytes_max;
 
 	buf->dev.type = SNDRV_DMA_TYPE_DEV;
 	buf->dev.dev = pcm->card->dev;
@@ -347,6 +786,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm)
 	int stream;
 
 	for (stream = 0; stream < 2; stream++) {
+		struct snd_dma_buffer *iram_dma;
 		substream = pcm->streams[stream].substream;
 		if (!substream)
 			continue;
@@ -358,6 +798,11 @@ static void davinci_pcm_free(struct snd_pcm *pcm)
 		dma_free_writecombine(pcm->card->dev, buf->bytes,
 				      buf->area, buf->addr);
 		buf->area = NULL;
+		iram_dma = (struct snd_dma_buffer *)buf->private_data;
+		if (iram_dma) {
+			sram_free(iram_dma->area, iram_dma->bytes);
+			kfree(iram_dma);
+		}
 	}
 }
 
@@ -375,14 +820,16 @@ static int davinci_pcm_new(struct snd_card *card,
 
 	if (dai->playback.channels_min) {
 		ret = davinci_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_PLAYBACK);
+			SNDRV_PCM_STREAM_PLAYBACK,
+			pcm_hardware_playback.buffer_bytes_max);
 		if (ret)
 			return ret;
 	}
 
 	if (dai->capture.channels_min) {
 		ret = davinci_pcm_preallocate_dma_buffer(pcm,
-			SNDRV_PCM_STREAM_CAPTURE);
+			SNDRV_PCM_STREAM_CAPTURE,
+			pcm_hardware_capture.buffer_bytes_max);
 		if (ret)
 			return ret;
 	}
diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h
index 8746606efc89..0764944cf10f 100644
--- a/sound/soc/davinci/davinci-pcm.h
+++ b/sound/soc/davinci/davinci-pcm.h
@@ -20,9 +20,11 @@ struct davinci_pcm_dma_params {
 	int channel;			/* sync dma channel ID */
 	unsigned short acnt;
 	dma_addr_t dma_addr;		/* device physical address for DMA */
+	unsigned sram_size;
 	enum dma_event_q eventq_no;	/* event queue number */
 	unsigned char data_type;	/* xfer data type */
 	unsigned char convert_mono_stereo;
+	unsigned int fifo_level;
 };
 
 
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index 6096d22283e6..30ed568afb2e 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -58,47 +58,15 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s)
 	/* 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;
+	bd->data[0] = s->runtime->dma_addr + (s->period_next * s->period_bytes);
 	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)
-{
-	if (s->appl_ptr > s->runtime->control->appl_ptr) {
-		/*
-		 * In this case s->runtime->control->appl_ptr has wrapped around.
-		 * Play the data to the end of the boundary, then wrap our own
-		 * appl_ptr back around.
-		 */
-		while (s->appl_ptr < s->runtime->boundary) {
-			if (bcom_queue_full(s->bcom_task))
-				return;
-
-			s->appl_ptr += s->period_size;
-
-			psc_dma_bcom_enqueue_next_buffer(s);
-		}
-		s->appl_ptr -= s->runtime->boundary;
-	}
-
-	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);
-	}
+	s->period_next = (s->period_next + 1) % s->runtime->periods;
 }
 
 /* Bestcomm DMA irq handler */
-static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream)
+static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream)
 {
 	struct psc_dma_stream *s = _psc_dma_stream;
 
@@ -108,34 +76,8 @@ static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream)
 	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;
+		s->period_current = (s->period_current+1) % s->runtime->periods;
+		s->period_count++;
 
 		psc_dma_bcom_enqueue_next_buffer(s);
 	}
@@ -166,54 +108,38 @@ 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 psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
 	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:
+		dev_dbg(psc_dma->dev, "START: stream=%i fbits=%u ps=%u #p=%u\n",
+			substream->pstr->stream, runtime->frame_bits,
+			(int)runtime->period_size, runtime->periods);
 		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->period_next = 0;
+		s->period_current = 0;
 		s->active = 1;
-
-		/* track appl_ptr so that we have a better chance of detecting
-		 * end of stream and not over running it.
-		 */
+		s->period_count = 0;
 		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) {
+		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 {
+		else
 			bcom_gen_bd_tx_reset(s->bcom_task);
-			psc_dma_bcom_enqueue_tx(s);
-		}
+
+		for (i = 0; i < runtime->periods; i++)
+			if (!bcom_queue_full(s->bcom_task))
+				psc_dma_bcom_enqueue_next_buffer(s);
 
 		bcom_enable(s->bcom_task);
 		spin_unlock_irqrestore(&psc_dma->lock, flags);
@@ -223,6 +149,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
+		dev_dbg(psc_dma->dev, "STOP: stream=%i periods_count=%i\n",
+			substream->pstr->stream, s->period_count);
 		s->active = 0;
 
 		spin_lock_irqsave(&psc_dma->lock, flags);
@@ -236,7 +164,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd)
 		break;
 
 	default:
-		dev_dbg(psc_dma->dev, "invalid command\n");
+		dev_dbg(psc_dma->dev, "unhandled trigger: stream=%i cmd=%i\n",
+			substream->pstr->stream, cmd);
 		return -EINVAL;
 	}
 
@@ -343,7 +272,7 @@ psc_dma_pointer(struct snd_pcm_substream *substream)
 	else
 		s = &psc_dma->playback;
 
-	count = s->period_current_pt - s->period_start;
+	count = s->period_current * s->period_bytes;
 
 	return bytes_to_frames(substream->runtime, count);
 }
@@ -532,11 +461,9 @@ int mpc5200_audio_dma_create(struct of_device *op)
 
 	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,
+	rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED,
 			  "psc-dma-capture", &psc_dma->capture);
-	rc |= request_irq(psc_dma->playback.irq,
-			  &psc_dma_bcom_irq_tx, IRQF_SHARED,
+	rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED,
 			  "psc-dma-playback", &psc_dma->playback);
 	if (rc) {
 		ret = -ENODEV;
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
index 8d396bb9d9fe..22208b373fb9 100644
--- a/sound/soc/fsl/mpc5200_dma.h
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -13,26 +13,25 @@
  * @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
+ * @ac97_slot_bits:	Enable bits for turning on the correct AC97 slot
  */
 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_next;
+	int period_current;
 	int period_bytes;
-	int period_size;
+	int period_count;
+
+	/* AC97 state */
+	u32 ac97_slot_bits;
 };
 
 /**
@@ -73,6 +72,15 @@ struct psc_dma {
 	} stats;
 };
 
+/* Utility for retrieving psc_dma_stream structure from a substream */
+inline struct psc_dma_stream *
+to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma)
+{
+	if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+		return &psc_dma->capture;
+	return &psc_dma->playback;
+}
+
 int mpc5200_audio_dma_create(struct of_device *op);
 int mpc5200_audio_dma_destroy(struct of_device *op);
 
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index c4ae3e096bb9..3dbc7f7cd7b9 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -130,6 +130,7 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *cpu_dai)
 {
 	struct psc_dma *psc_dma = cpu_dai->private_data;
+	struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
 
 	dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
 		" periods=%i buffer_size=%i  buffer_bytes=%i channels=%i"
@@ -140,20 +141,10 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
 		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);
-
+	/* Determine the set of enable bits to turn on */
+	s->ac97_slot_bits = (params_channels(params) == 1) ? 0x100 : 0x300;
+	if (substream->pstr->stream != SNDRV_PCM_STREAM_CAPTURE)
+		s->ac97_slot_bits <<= 16;
 	return 0;
 }
 
@@ -163,6 +154,8 @@ static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream,
 {
 	struct psc_dma *psc_dma = cpu_dai->private_data;
 
+	dev_dbg(psc_dma->dev, "%s(substream=%p)\n", __func__, substream);
+
 	if (params_channels(params) == 1)
 		out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000);
 	else
@@ -176,14 +169,24 @@ static int psc_ac97_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 psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma);
 
 	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		dev_dbg(psc_dma->dev, "AC97 START: stream=%i\n",
+			substream->pstr->stream);
+
+		/* Set the slot enable bits */
+		psc_dma->slots |= s->ac97_slot_bits;
+		out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+		break;
+
 	case SNDRV_PCM_TRIGGER_STOP:
-		if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
-			psc_dma->slots &= 0xFFFF0000;
-		else
-			psc_dma->slots &= 0x0000FFFF;
+		dev_dbg(psc_dma->dev, "AC97 STOP: stream=%i\n",
+			substream->pstr->stream);
 
+		/* Clear the slot enable bits */
+		psc_dma->slots &= ~(s->ac97_slot_bits);
 		out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
 		break;
 	}
diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c
index e4dcb539108a..0267d2d91685 100644
--- a/sound/soc/imx/mx27vis_wm8974.c
+++ b/sound/soc/imx/mx27vis_wm8974.c
@@ -157,7 +157,7 @@ static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream,
 
 
 	/* codec PLL input is 25 MHz */
-	ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG,
+	ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG,
 					25000000, pll_out);
 	if (ret < 0) {
 		printk(KERN_ERR "Error when setting PLL input\n");
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 653a362425df..61952aa6cd5a 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -43,12 +43,13 @@ config SND_OMAP_SOC_OSK5912
 	  Say Y if you want to add support for SoC audio on osk5912.
 
 config SND_OMAP_SOC_OVERO
-	tristate "SoC Audio support for Gumstix Overo"
-	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO
+	tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35"
+	depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35)
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TWL4030
 	help
-	  Say Y if you want to add support for SoC audio on the Gumstix Overo.
+	  Say Y if you want to add support for SoC audio on the
+	  Gumstix Overo or CompuLab CM-T35
 
 config SND_OMAP_SOC_OMAP2EVM
 	tristate "SoC Audio support for OMAP2EVM board"
@@ -66,6 +67,15 @@ config SND_OMAP_SOC_OMAP3EVM
 	help
 	  Say Y if you want to add support for SoC audio on the omap3evm board.
 
+config SND_OMAP_SOC_AM3517EVM
+	tristate "SoC Audio support for OMAP3517 / AM3517 EVM"
+	depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517
+	  EVM.
+
 config SND_OMAP_SOC_SDP3430
 	tristate "SoC Audio support for Texas Instruments SDP3430"
 	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
@@ -99,3 +109,10 @@ config SND_OMAP_SOC_ZOOM2
 	help
 	  Say Y if you want to add support for Soc audio on Zoom2 board.
 
+config SND_OMAP_SOC_IGEP0020
+	tristate "SoC Audio support for IGEP v2"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for Soc audio on IGEP v2 board.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 02d69471dcb5..d49458a29bb7 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -12,10 +12,12 @@ 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-am3517evm-objs := am3517evm.o
 snd-soc-sdp3430-objs := sdp3430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
 snd-soc-omap3beagle-objs := omap3beagle.o
 snd-soc-zoom2-objs := zoom2.o
+snd-soc-igep0020-objs := igep0020.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
@@ -23,7 +25,9 @@ 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-$(CONFIG_MACH_OMAP3517EVM) += snd-soc-am3517evm.o
 obj-$(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
 obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
+obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
new file mode 100644
index 000000000000..135901b2ea11
--- /dev/null
+++ b/sound/soc/omap/am3517evm.c
@@ -0,0 +1,202 @@
+/*
+ * am3517evm.c  -- ALSA SoC support for OMAP3517 / AM3517 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2009 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 <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 	12000000
+
+static int am3517evm_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_DSP_B |
+				  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_DSP_B |
+				  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,
+			CODEC_CLOCK, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0,
+				SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n");
+		return ret;
+	}
+
+	snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
+				SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops am3517evm_ops = {
+	.hw_params = am3517evm_hw_params,
+};
+
+/* am3517evm machine dapm widgets */
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Line Out", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_MIC("Mic In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Line Out connected to LLOUT, RLOUT */
+	{"Line Out", NULL, "LOUT"},
+	{"Line Out", NULL, "ROUT"},
+
+	{"LLINEIN", NULL, "Line In"},
+	{"RLINEIN", NULL, "Line In"},
+
+	{"MICIN", NULL, "Mic In"},
+};
+
+static int am3517evm_aic23_init(struct snd_soc_codec *codec)
+{
+	/* Add am3517-evm specific widgets */
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up davinci-evm specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(codec, "Line Out");
+	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(codec, "Mic In");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link am3517evm_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = am3517evm_aic23_init,
+	.ops = &am3517evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_am3517evm = {
+	.name = "am3517evm",
+	.platform = &omap_soc_platform,
+	.dai_link = &am3517evm_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device am3517evm_snd_devdata = {
+	.card = &snd_soc_am3517evm,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *am3517evm_snd_device;
+
+static int __init am3517evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3517evm()) {
+		pr_err("Not OMAP3517 / AM3517 EVM!\n");
+		return -ENODEV;
+	}
+	pr_info("OMAP3517 / AM3517 EVM SoC init\n");
+
+	am3517evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!am3517evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(am3517evm_snd_device, &am3517evm_snd_devdata);
+	am3517evm_snd_devdata.dev = &am3517evm_snd_device->dev;
+	*(unsigned int *)am3517evm_dai.cpu_dai->private_data = 0; /* McBSP1 */
+
+	ret = platform_device_add(am3517evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(am3517evm_snd_device);
+
+	return ret;
+}
+
+static void __exit am3517evm_soc_exit(void)
+{
+	platform_device_unregister(am3517evm_snd_device);
+}
+
+module_init(am3517evm_soc_init);
+module_exit(am3517evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 5a5166ac7279..ae0fc9b135d4 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -40,7 +40,7 @@
 
 
 /* Board specific DAPM widgets */
- const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
 	/* Handset */
 	SND_SOC_DAPM_MIC("Mouthpiece", NULL),
 	SND_SOC_DAPM_HP("Earpiece", NULL),
@@ -81,7 +81,7 @@ static const char *ams_delta_audio_mode[] =
 						(1 << AMS_DELTA_SPEAKER))
 #define AMS_DELTA_SPEAKERPHONE	(AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
 
-unsigned short ams_delta_audio_mode_pins[] = {
+static const unsigned short ams_delta_audio_mode_pins[] = {
 	AMS_DELTA_MIXED,
 	AMS_DELTA_HANDSET,
 	AMS_DELTA_HANDSFREE,
diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c
new file mode 100644
index 000000000000..3583c429f9be
--- /dev/null
+++ b/sound/soc/omap/igep0020.c
@@ -0,0 +1,148 @@
+/*
+ * igep0020.c  --  SoC audio for IGEP v2
+ *
+ * Based on sound/soc/omap/overo.c by Steve Sakoman
+ *
+ * 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/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 <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int igep2_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 igep2_ops = {
+	.hw_params = igep2_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link igep2_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+	.ops = &igep2_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_igep2 = {
+	.name = "igep2",
+	.platform = &omap_soc_platform,
+	.dai_link = &igep2_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device igep2_snd_devdata = {
+	.card = &snd_soc_card_igep2,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *igep2_snd_device;
+
+static int __init igep2_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_igep0020()) {
+		pr_debug("Not IGEP v2!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "IGEP v2 SoC init\n");
+
+	igep2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!igep2_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(igep2_snd_device, &igep2_snd_devdata);
+	igep2_snd_devdata.dev = &igep2_snd_device->dev;
+	*(unsigned int *)igep2_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(igep2_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(igep2_snd_device);
+
+	return ret;
+}
+module_init(igep2_soc_init);
+
+static void __exit igep2_soc_exit(void)
+{
+	platform_device_unregister(igep2_snd_device);
+}
+module_exit(igep2_soc_exit);
+
+MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo@iseebcn.com>");
+MODULE_DESCRIPTION("ALSA SoC IGEP v2");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 3341f49402ca..45be94201c89 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -49,6 +49,8 @@ struct omap_mcbsp_data {
 	 */
 	int				active;
 	int				configured;
+	unsigned int			in_freq;
+	int				clk_div;
 };
 
 #define to_mcbsp(priv)	container_of((priv), struct omap_mcbsp_data, bus_id)
@@ -257,7 +259,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
 	int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
 	unsigned long port;
-	unsigned int format;
+	unsigned int format, div, framesize, master;
 
 	if (cpu_class_is_omap1()) {
 		dma = omap1_dma_reqs[bus_id][substream->stream];
@@ -294,28 +296,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 
 	format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
 	wpf = channels = params_channels(params);
-	switch (channels) {
-	case 2:
-		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:
-	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 */
-		return -EINVAL;
+	if (channels == 2 && 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);
 	}
 
+	regs->rcr1	|= RFRLEN1(wpf - 1);
+	regs->xcr1	|= XFRLEN1(wpf - 1);
+
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		/* Set word lengths */
@@ -330,15 +323,30 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	/* In McBSP master modes, FRAME (i.e. sample rate) is generated
+	 * by _counting_ BCLKs. Calculate frame size in BCLKs */
+	master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK;
+	if (master ==	SND_SOC_DAIFMT_CBS_CFS) {
+		div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1;
+		framesize = (mcbsp_data->in_freq / div) / params_rate(params);
+
+		if (framesize < wlen * channels) {
+			printk(KERN_ERR "%s: not enough bandwidth for desired rate and "
+					"channels\n", __func__);
+			return -EINVAL;
+		}
+	} else
+		framesize = wlen * channels;
+
 	/* Set FS period and length in terms of bit clock periods */
 	switch (format) {
 	case SND_SOC_DAIFMT_I2S:
-		regs->srgr2	|= FPER(wlen * channels - 1);
-		regs->srgr1	|= FWID(wlen - 1);
+		regs->srgr2	|= FPER(framesize - 1);
+		regs->srgr1	|= FWID((framesize >> 1) - 1);
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 	case SND_SOC_DAIFMT_DSP_B:
-		regs->srgr2	|= FPER(wlen * channels - 1);
+		regs->srgr2	|= FPER(framesize - 1);
 		regs->srgr1	|= FWID(0);
 		break;
 	}
@@ -454,6 +462,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
 	if (div_id != OMAP_MCBSP_CLKGDV)
 		return -ENODEV;
 
+	mcbsp_data->clk_div = div;
 	regs->srgr1	|= CLKGDV(div - 1);
 
 	return 0;
@@ -554,6 +563,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int err = 0;
 
+	mcbsp_data->in_freq = freq;
+
 	switch (clk_id) {
 	case OMAP_MCBSP_SYSCLK_CLK:
 		regs->srgr2	|= CLKSM;
@@ -598,13 +609,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
 	.id = (link_id),					\
 	.playback = {						\
 		.channels_min = 1,				\
-		.channels_max = 4,				\
+		.channels_max = 16,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
 	.capture = {						\
 		.channels_min = 1,				\
-		.channels_max = 4,				\
+		.channels_max = 16,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index 13aa380de162..f484dcd63408 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -93,10 +93,17 @@ static struct snd_soc_card snd_soc_omap3evm = {
 	.num_links = 1,
 };
 
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+	.ramp_delay_value = 4,
+	.sysclk = 26000,
+};
+
 /* Audio subsystem */
 static struct snd_soc_device omap3evm_snd_devdata = {
 	.card = &snd_soc_omap3evm,
 	.codec_dev = &soc_codec_dev_twl4030,
+	.codec_data = &twl4030_setup,
 };
 
 static struct platform_device *omap3evm_snd_device;
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 0cd06f5dd356..71b2c161158d 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -40,9 +40,12 @@
 
 #define PREFIX "ASoC omap3pandora: "
 
-static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
-	struct snd_soc_dai *cpu_dai, unsigned int fmt)
+static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params, unsigned int fmt)
 {
+	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 */
@@ -68,8 +71,9 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
 	}
 
 	/* Set McBSP clock to external */
-	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0,
-					    SND_SOC_CLOCK_IN);
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT,
+				     256 * params_rate(params),
+				     SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		pr_err(PREFIX "can't set cpu system clock\n");
 		return ret;
@@ -87,11 +91,7 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
 static int omap3pandora_out_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;
-
-	return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+	return omap3pandora_cmn_hw_params(substream, params,
 					  SND_SOC_DAIFMT_I2S |
 					  SND_SOC_DAIFMT_IB_NF |
 					  SND_SOC_DAIFMT_CBS_CFS);
@@ -100,11 +100,7 @@ static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream,
 static int omap3pandora_in_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;
-
-	return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+	return omap3pandora_cmn_hw_params(substream, params,
 					  SND_SOC_DAIFMT_I2S |
 					  SND_SOC_DAIFMT_NB_NF |
 					  SND_SOC_DAIFMT_CBS_CFS);
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index ec4f8fd8b3a2..97a4d6308bd6 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -107,8 +107,8 @@ static int __init overo_soc_init(void)
 {
 	int ret;
 
-	if (!machine_is_overo()) {
-		pr_debug("Not Overo!\n");
+	if (!(machine_is_overo() || machine_is_cm_t35())) {
+		pr_debug("Incomatible machine!\n");
 		return -ENODEV;
 	}
 	printk(KERN_INFO "overo SoC init\n");
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index dcb3181bb340..376e14a9c273 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -90,7 +90,8 @@ config SND_PXA2XX_SOC_E800
 
 config SND_PXA2XX_SOC_EM_X270
 	tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
-	depends on SND_PXA2XX_SOC && MACH_EM_X270
+	depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \
+			MACH_CM_X300)
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
@@ -117,6 +118,15 @@ config SND_SOC_ZYLONITE
 	  Say Y if you want to add support for SoC audio on the
 	  Marvell Zylonite reference platform.
 
+config SND_SOC_RAUMFELD
+	tristate "SoC Audio support Raumfeld audio adapter"
+	depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR)
+	select SND_PXA_SOC_SSP
+	select SND_SOC_CS4270
+	select SND_SOC_AK4104
+	help
+	  Say Y if you want to add support for SoC audio on Raumfeld devices
+
 config SND_PXA2XX_SOC_MAGICIAN
 	tristate "SoC Audio support for HTC Magician"
 	depends on SND_PXA2XX_SOC && MACH_MAGICIAN
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 6e096b480335..f3e08fd40ca2 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -23,6 +23,7 @@ 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
+snd-soc-raumfeld-objs := raumfeld.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -37,3 +38,4 @@ 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
+obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 9f7c61e23daf..4c8d99a8d386 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -213,7 +213,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* set SSP audio pll clock */
-	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index d11a6d7e384a..3bd7712f029b 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -305,8 +305,8 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
 /*
  * Configure the PLL frequency pxa27x and (afaik - pxa320 only)
  */
-static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai,
-	int pll_id, unsigned int freq_in, unsigned int freq_out)
+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_device *ssp = priv->dev.ssp;
@@ -760,13 +760,13 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 		.resume = pxa_ssp_resume,
 		.playback = {
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 8,
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		},
 		.capture = {
 			 .channels_min = 1,
-			 .channels_max = 2,
+			 .channels_max = 8,
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
@@ -780,13 +780,13 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 		.resume = pxa_ssp_resume,
 		.playback = {
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 8,
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		},
 		.capture = {
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 8,
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
@@ -801,13 +801,13 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 		.resume = pxa_ssp_resume,
 		.playback = {
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 8,
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		},
 		.capture = {
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 8,
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
@@ -822,13 +822,13 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 		.resume = pxa_ssp_resume,
 		.playback = {
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 8,
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		},
 		.capture = {
 			.channels_min = 1,
-			.channels_max = 2,
+			.channels_max = 8,
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
new file mode 100644
index 000000000000..acfce1c0f1c9
--- /dev/null
+++ b/sound/soc/pxa/raumfeld.c
@@ -0,0 +1,335 @@
+/*
+ * raumfeld_audio.c  --  SoC audio for Raumfeld audio devices
+ *
+ * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de>
+ *
+ * based on code from:
+ *
+ *    Wolfson Microelectronics PLC.
+ *    Openedhand Ltd.
+ *    Liam Girdwood <lrg@slimlogic.co.uk>
+ *    Richard Purdie <richard@openedhand.com>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#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)
+#define GPIO_MCLK_RESET		(111)
+#define GPIO_CODEC_RESET	(120)
+
+static struct i2c_client *max9486_client;
+static struct i2c_board_info max9486_hwmon_info = {
+	I2C_BOARD_INFO("max9485", 0x63),
+};
+
+#define MAX9485_MCLK_FREQ_112896 0x22
+#define	MAX9485_MCLK_FREQ_122880 0x23
+
+static void set_max9485_clk(char clk)
+{
+	i2c_master_send(max9486_client, &clk, 1);
+}
+
+static void raumfeld_enable_audio(bool en)
+{
+	if (en) {
+		gpio_set_value(GPIO_MCLK_RESET, 1);
+
+		/* wait some time to let the clocks become stable */
+		msleep(100);
+
+		gpio_set_value(GPIO_SPDIF_RESET, 1);
+		gpio_set_value(GPIO_CODEC_RESET, 1);
+	} else {
+		gpio_set_value(GPIO_MCLK_RESET, 0);
+		gpio_set_value(GPIO_SPDIF_RESET, 0);
+		gpio_set_value(GPIO_CODEC_RESET, 0);
+	}
+}
+
+/* CS4270 */
+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;
+
+	set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+
+	return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0);
+}
+
+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;
+	unsigned int fmt, clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+		clk = 11289600;
+		break;
+	}
+
+	fmt = SND_SOC_DAIFMT_I2S |
+	      SND_SOC_DAIFMT_NB_NF |
+	      SND_SOC_DAIFMT_CBS_CFS;
+
+	/* setup the CODEC DAI */
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0);
+	if (ret < 0)
+		return ret;
+
+	/* setup the CPU DAI */
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops raumfeld_cs4270_ops = {
+	.startup = raumfeld_cs4270_startup,
+	.hw_params = raumfeld_cs4270_hw_params,
+};
+
+static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	raumfeld_enable_audio(false);
+	return 0;
+}
+
+static int raumfeld_line_resume(struct platform_device *pdev)
+{
+	raumfeld_enable_audio(true);
+	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;
+	int fmt, ret = 0, clk = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+		set_max9485_clk(MAX9485_MCLK_FREQ_112896);
+		clk = 11289600;
+		break;
+	}
+
+	fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF;
+
+	/* setup the CODEC DAI */
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* setup the CPU DAI */
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops raumfeld_ak4104_ops = {
+	.hw_params = raumfeld_ak4104_hw_params,
+};
+
+static struct snd_soc_dai_link raumfeld_spdif_dai = {
+	.name		= "ak4104",
+	.stream_name	= "Playback",
+	.cpu_dai	= &pxa_ssp_dai[PXA_DAI_SSP2],
+	.codec_dai	= &ak4104_dai,
+	.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,
+};
+
+static struct snd_soc_device raumfeld_spdif_devdata = {
+	.card = &snd_soc_spdif_raumfeld,
+	.codec_dev = &soc_codec_device_ak4104,
+};
+
+static struct platform_device *raumfeld_audio_line_device;
+static struct platform_device *raumfeld_audio_spdif_device;
+
+static int __init raumfeld_audio_init(void)
+{
+	int ret;
+
+	if (!machine_is_raumfeld_speaker() &&
+	    !machine_is_raumfeld_connector())
+		return 0;
+
+	max9486_client = i2c_new_device(i2c_get_adapter(0),
+					&max9486_hwmon_info);
+
+	if (!max9486_client)
+		return -ENOMEM;
+
+	set_max9485_clk(MAX9485_MCLK_FREQ_122880);
+
+	/* LINE */
+	raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0);
+	if (!raumfeld_audio_line_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);
+
+	/* 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;
+}
+
+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);
+
+	i2c_unregister_device(max9486_client);
+
+	gpio_free(GPIO_MCLK_RESET);
+	gpio_free(GPIO_CODEC_RESET);
+	gpio_free(GPIO_SPDIF_RESET);
+}
+
+module_init(raumfeld_audio_init);
+module_exit(raumfeld_audio_exit);
+
+/* Module information */
+MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
+MODULE_DESCRIPTION("Raumfeld audio SoC");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index 9a386b4c4ed1..dd678ae24398 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -74,7 +74,8 @@ static const struct snd_soc_dapm_route audio_map[] = {
 static int zylonite_wm9713_init(struct snd_soc_codec *codec)
 {
 	if (clk_pout)
-		snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0);
+		snd_soc_dai_set_pll(&codec->dai[0], 0, 0,
+				    clk_get_rate(pout), 0);
 
 	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
 				  ARRAY_SIZE(zylonite_dapm_widgets));
@@ -128,7 +129,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out);
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out);
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 923428fc1adb..b489f1ae103d 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -24,6 +24,9 @@ config SND_S3C64XX_SOC_I2S
 	select SND_S3C_I2SV2_SOC
 	select S3C64XX_DMA
 
+config SND_S3C_SOC_PCM
+	tristate
+
 config SND_S3C2443_SOC_AC97
 	tristate
 	select S3C2410_DMA
@@ -56,6 +59,15 @@ config SND_S3C24XX_SOC_JIVE_WM8750
 	help
 	  Sat Y if you want to add support for SoC audio on the Jive.
 
+config SND_S3C64XX_SOC_WM8580
+	tristate "SoC I2S Audio support for WM8580 on SMDK64XX"
+	depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410)
+	depends on BROKEN
+	select SND_SOC_WM8580
+	select SND_S3C64XX_SOC_I2S
+	help
+	  Sat Y if you want to add support for SoC audio on the SMDK64XX.
+
 config SND_S3C24XX_SOC_SMDK2443_WM9710
 	tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
 	depends on SND_S3C24XX_SOC && MACH_SMDK2443
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 99f5a7dd3fc6..b744657733d7 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -1,10 +1,11 @@
 # S3c24XX Platform Support
-snd-soc-s3c24xx-objs := s3c24xx-pcm.o
+snd-soc-s3c24xx-objs := s3c-dma.o
 snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
 snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
 snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o
 snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o
 snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o
+snd-soc-s3c-pcm-objs := s3c-pcm.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
 obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
@@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o
 obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
 obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o
 obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
+obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o
 
 # S3C24XX Machine Support
 snd-soc-jive-wm8750-objs := jive_wm8750.o
@@ -23,6 +25,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
 snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o
 snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
 snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
+snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -33,4 +36,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
+obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o
 
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 93e6c87b7399..59dc2c6b56d9 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -25,7 +25,7 @@
 
 #include <asm/mach-types.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c2412-i2s.h"
 
 #include "../codecs/wm8750.h"
diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c
index 12c71482d258..d00d359a03e6 100644
--- a/sound/soc/s3c24xx/ln2440sbc_alc650.c
+++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c
@@ -24,7 +24,7 @@
 #include <sound/soc-dapm.h>
 
 #include "../codecs/ac97.h"
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-ac97.h"
 
 static struct snd_soc_card ln2440sbc;
diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
index 0c52e36ddd87..dea83d30a5c9 100644
--- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -32,7 +32,7 @@
 #include <asm/io.h>
 #include <mach/gta02.h>
 #include "../codecs/wm8753.h"
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-i2s.h"
 
 static struct snd_soc_card neo1973_gta02;
@@ -119,7 +119,7 @@ static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* codec PLL input is PCLK/4 */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
 		iis_clkrate / 4, pll_out);
 	if (ret < 0)
 		return ret;
@@ -133,7 +133,7 @@ static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
 	/* disable the PLL */
-	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
 }
 
 /*
@@ -183,7 +183,7 @@ static int neo1973_gta02_voice_hw_params(
 		return ret;
 
 	/* configue and enable PLL for 12.288MHz output */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0,
 		iis_clkrate / 4, 12288000);
 	if (ret < 0)
 		return ret;
@@ -197,7 +197,7 @@ static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
 	/* disable the PLL */
-	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
 }
 
 static struct snd_soc_ops neo1973_gta02_voice_ops = {
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 906709e6dd5f..0cb4f86f6d1e 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -29,7 +29,6 @@
 #include <mach/regs-clock.h>
 #include <mach/regs-gpio.h>
 #include <mach/hardware.h>
-#include <plat/audio.h>
 #include <linux/io.h>
 #include <mach/spi-gpio.h>
 
@@ -37,7 +36,7 @@
 
 #include "../codecs/wm8753.h"
 #include "lm4857.h"
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-i2s.h"
 
 /* define the scenarios */
@@ -137,7 +136,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* codec PLL input is PCLK/4 */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
+	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
 		iis_clkrate / 4, pll_out);
 	if (ret < 0)
 		return ret;
@@ -153,7 +152,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
 	pr_debug("Entered %s\n", __func__);
 
 	/* disable the PLL */
-	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
+	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
 }
 
 /*
@@ -203,7 +202,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* configue and enable PLL for 12.288MHz output */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
+	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0,
 		iis_clkrate / 4, 12288000);
 	if (ret < 0)
 		return ret;
@@ -219,7 +218,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
 	pr_debug("Entered %s\n", __func__);
 
 	/* disable the PLL */
-	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
+	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
 }
 
 static struct snd_soc_ops neo1973_voice_ops = {
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c-dma.c
index 1f35c6fcf5fd..7725e26d6c91 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c-dma.c
@@ -1,5 +1,5 @@
 /*
- * s3c24xx-pcm.c  --  ALSA Soc Audio Layer
+ * s3c-dma.c  --  ALSA Soc Audio Layer
  *
  * (c) 2006 Wolfson Microelectronics PLC.
  * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
@@ -29,11 +29,10 @@
 #include <asm/dma.h>
 #include <mach/hardware.h>
 #include <mach/dma.h>
-#include <plat/audio.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 
-static const struct snd_pcm_hardware s3c24xx_pcm_hardware = {
+static const struct snd_pcm_hardware s3c_dma_hardware = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
 				    SNDRV_PCM_INFO_BLOCK_TRANSFER |
 				    SNDRV_PCM_INFO_MMAP |
@@ -63,15 +62,15 @@ struct s3c24xx_runtime_data {
 	dma_addr_t dma_start;
 	dma_addr_t dma_pos;
 	dma_addr_t dma_end;
-	struct s3c24xx_pcm_dma_params *params;
+	struct s3c_dma_params *params;
 };
 
-/* s3c24xx_pcm_enqueue
+/* s3c_dma_enqueue
  *
  * place a dma buffer onto the queue for the dma system
  * to handle.
 */
-static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
+static void s3c_dma_enqueue(struct snd_pcm_substream *substream)
 {
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	dma_addr_t pos = prtd->dma_pos;
@@ -80,12 +79,13 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
 
 	pr_debug("Entered %s\n", __func__);
 
-	if (s3c_dma_has_circular()) {
+	if (s3c_dma_has_circular())
 		limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period;
-	} else
+	else
 		limit = prtd->dma_limit;
 
-	pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit);
+	pr_debug("%s: loaded %d, limit %d\n",
+				__func__, prtd->dma_loaded, limit);
 
 	while (prtd->dma_loaded < limit) {
 		unsigned long len = prtd->dma_period;
@@ -133,19 +133,19 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
 	spin_lock(&prtd->lock);
 	if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) {
 		prtd->dma_loaded--;
-		s3c24xx_pcm_enqueue(substream);
+		s3c_dma_enqueue(substream);
 	}
 
 	spin_unlock(&prtd->lock);
 }
 
-static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
+static int s3c_dma_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd = runtime->private_data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+	struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data;
 	unsigned long totbytes = params_buffer_bytes(params);
 	int ret = 0;
 
@@ -198,7 +198,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream)
+static int s3c_dma_hw_free(struct snd_pcm_substream *substream)
 {
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 
@@ -215,7 +215,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
+static int s3c_dma_prepare(struct snd_pcm_substream *substream)
 {
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	int ret = 0;
@@ -248,12 +248,12 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
 	prtd->dma_pos = prtd->dma_start;
 
 	/* enqueue dma buffers */
-	s3c24xx_pcm_enqueue(substream);
+	s3c_dma_enqueue(substream);
 
 	return ret;
 }
 
-static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	int ret = 0;
@@ -288,7 +288,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 }
 
 static snd_pcm_uframes_t
-s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
+s3c_dma_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd = runtime->private_data;
@@ -323,7 +323,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
 	return bytes_to_frames(substream->runtime, res);
 }
 
-static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
+static int s3c_dma_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd;
@@ -331,7 +331,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
 	pr_debug("Entered %s\n", __func__);
 
 	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
-	snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
+	snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware);
 
 	prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL);
 	if (prtd == NULL)
@@ -343,7 +343,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
+static int s3c_dma_close(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd = runtime->private_data;
@@ -351,14 +351,14 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
 	pr_debug("Entered %s\n", __func__);
 
 	if (!prtd)
-		pr_debug("s3c24xx_pcm_close called with prtd == NULL\n");
+		pr_debug("s3c_dma_close called with prtd == NULL\n");
 
 	kfree(prtd);
 
 	return 0;
 }
 
-static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
+static int s3c_dma_mmap(struct snd_pcm_substream *substream,
 	struct vm_area_struct *vma)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
@@ -371,23 +371,23 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
 				     runtime->dma_bytes);
 }
 
-static struct snd_pcm_ops s3c24xx_pcm_ops = {
-	.open		= s3c24xx_pcm_open,
-	.close		= s3c24xx_pcm_close,
+static struct snd_pcm_ops s3c_dma_ops = {
+	.open		= s3c_dma_open,
+	.close		= s3c_dma_close,
 	.ioctl		= snd_pcm_lib_ioctl,
-	.hw_params	= s3c24xx_pcm_hw_params,
-	.hw_free	= s3c24xx_pcm_hw_free,
-	.prepare	= s3c24xx_pcm_prepare,
-	.trigger	= s3c24xx_pcm_trigger,
-	.pointer	= s3c24xx_pcm_pointer,
-	.mmap		= s3c24xx_pcm_mmap,
+	.hw_params	= s3c_dma_hw_params,
+	.hw_free	= s3c_dma_hw_free,
+	.prepare	= s3c_dma_prepare,
+	.trigger	= s3c_dma_trigger,
+	.pointer	= s3c_dma_pointer,
+	.mmap		= s3c_dma_mmap,
 };
 
-static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 {
 	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
 	struct snd_dma_buffer *buf = &substream->dma_buffer;
-	size_t size = s3c24xx_pcm_hardware.buffer_bytes_max;
+	size_t size = s3c_dma_hardware.buffer_bytes_max;
 
 	pr_debug("Entered %s\n", __func__);
 
@@ -402,7 +402,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	return 0;
 }
 
-static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
+static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm)
 {
 	struct snd_pcm_substream *substream;
 	struct snd_dma_buffer *buf;
@@ -425,9 +425,9 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 	}
 }
 
-static u64 s3c24xx_pcm_dmamask = DMA_BIT_MASK(32);
+static u64 s3c_dma_mask = DMA_BIT_MASK(32);
 
-static int s3c24xx_pcm_new(struct snd_card *card,
+static int s3c_dma_new(struct snd_card *card,
 	struct snd_soc_dai *dai, struct snd_pcm *pcm)
 {
 	int ret = 0;
@@ -435,19 +435,19 @@ static int s3c24xx_pcm_new(struct snd_card *card,
 	pr_debug("Entered %s\n", __func__);
 
 	if (!card->dev->dma_mask)
-		card->dev->dma_mask = &s3c24xx_pcm_dmamask;
+		card->dev->dma_mask = &s3c_dma_mask;
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = 0xffffffff;
 
 	if (dai->playback.channels_min) {
-		ret = s3c24xx_pcm_preallocate_dma_buffer(pcm,
+		ret = s3c_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
 	if (dai->capture.channels_min) {
-		ret = s3c24xx_pcm_preallocate_dma_buffer(pcm,
+		ret = s3c_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
 			goto out;
@@ -458,9 +458,9 @@ static int s3c24xx_pcm_new(struct snd_card *card,
 
 struct snd_soc_platform s3c24xx_soc_platform = {
 	.name		= "s3c24xx-audio",
-	.pcm_ops 	= &s3c24xx_pcm_ops,
-	.pcm_new	= s3c24xx_pcm_new,
-	.pcm_free	= s3c24xx_pcm_free_dma_buffers,
+	.pcm_ops 	= &s3c_dma_ops,
+	.pcm_new	= s3c_dma_new,
+	.pcm_free	= s3c_dma_free_dma_buffers,
 };
 EXPORT_SYMBOL_GPL(s3c24xx_soc_platform);
 
@@ -477,5 +477,5 @@ static void __exit s3c24xx_soc_platform_exit(void)
 module_exit(s3c24xx_soc_platform_exit);
 
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
-MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module");
+MODULE_DESCRIPTION("Samsung S3C Audio DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c-dma.h
index 0088c79822ea..69bb6bf6fc1c 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.h
+++ b/sound/soc/s3c24xx/s3c-dma.h
@@ -1,5 +1,5 @@
 /*
- *  s3c24xx-pcm.h --
+ *  s3c-dma.h --
  *
  *  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
@@ -9,13 +9,13 @@
  *  ALSA PCM interface for the Samsung S3C24xx CPU
  */
 
-#ifndef _S3C24XX_PCM_H
-#define _S3C24XX_PCM_H
+#ifndef _S3C_AUDIO_H
+#define _S3C_AUDIO_H
 
 #define ST_RUNNING		(1<<0)
 #define ST_OPENED		(1<<1)
 
-struct s3c24xx_pcm_dma_params {
+struct s3c_dma_params {
 	struct s3c2410_dma_client *client;	/* stream identifier */
 	int channel;				/* Channel ID */
 	dma_addr_t dma_addr;
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 9bc4aa35caab..e994d8374fe6 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -32,11 +32,10 @@
 
 #include <plat/regs-s3c2412-iis.h>
 
-#include <plat/audio.h>
 #include <mach/dma.h>
 
 #include "s3c-i2s-v2.h"
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 
 #undef S3C_IIS_V2_SUPPORTED
 
@@ -312,12 +311,15 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_RIGHT_J:
+		iismod |= S3C2412_IISMOD_LR_RLOW;
 		iismod |= S3C2412_IISMOD_SDF_MSB;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
+		iismod |= S3C2412_IISMOD_LR_RLOW;
 		iismod |= S3C2412_IISMOD_SDF_LSB;
 		break;
 	case SND_SOC_DAIFMT_I2S:
+		iismod &= ~S3C2412_IISMOD_LR_RLOW;
 		iismod |= S3C2412_IISMOD_SDF_IIS;
 		break;
 	default:
@@ -392,7 +394,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
 	unsigned long irqs;
 	int ret = 0;
-	int channel = ((struct s3c24xx_pcm_dma_params *)
+	int channel = ((struct s3c_dma_params *)
 		  rtd->dai->cpu_dai->dma_data)->channel;
 
 	pr_debug("Entered %s\n", __func__);
@@ -467,6 +469,31 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 
 	switch (div_id) {
 	case S3C_I2SV2_DIV_BCLK:
+		if (div > 3) {
+			/* convert value to bit field */
+
+			switch (div) {
+			case 16:
+				div = S3C2412_IISMOD_BCLK_16FS;
+				break;
+
+			case 32:
+				div = S3C2412_IISMOD_BCLK_32FS;
+				break;
+
+			case 24:
+				div = S3C2412_IISMOD_BCLK_24FS;
+				break;
+
+			case 48:
+				div = S3C2412_IISMOD_BCLK_48FS;
+				break;
+
+			default:
+				return -EINVAL;
+			}
+		}
+
 		reg = readl(i2s->regs + S3C2412_IISMOD);
 		reg &= ~S3C2412_IISMOD_BCLK_MASK;
 		writel(reg | div, i2s->regs + S3C2412_IISMOD);
@@ -626,7 +653,7 @@ int s3c_i2sv2_probe(struct platform_device *pdev,
 	}
 
 	i2s->iis_pclk = clk_get(dev, "iis");
-	if (i2s->iis_pclk == NULL) {
+	if (IS_ERR(i2s->iis_pclk)) {
 		dev_err(dev, "failed to get iis_clock\n");
 		iounmap(i2s->regs);
 		return -ENOENT;
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h
index f66854a77fb2..ecf8eaaed1db 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.h
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.h
@@ -49,8 +49,8 @@ struct s3c_i2sv2_info {
 
 	unsigned char	 master;
 
-	struct s3c24xx_pcm_dma_params	*dma_playback;
-	struct s3c24xx_pcm_dma_params	*dma_capture;
+	struct s3c_dma_params	*dma_playback;
+	struct s3c_dma_params	*dma_capture;
 
 	u32		 suspend_iismod;
 	u32		 suspend_iiscon;
diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c
new file mode 100644
index 000000000000..9e61a7c2d9ac
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c-pcm.c
@@ -0,0 +1,552 @@
+/* sound/soc/s3c24xx/s3c-pcm.c
+ *
+ * ALSA SoC Audio Layer - S3C PCM-Controller driver
+ *
+ * Copyright (c) 2009 Samsung Electronics Co. Ltd
+ * Author: Jaswinder Singh <jassi.brar@samsung.com>
+ * based upon I2S drivers by Ben Dooks.
+ *
+ * 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/kernel.h>
+#include <linux/gpio.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 <plat/audio.h>
+#include <plat/dma.h>
+
+#include "s3c-dma.h"
+#include "s3c-pcm.h"
+
+static struct s3c2410_dma_client s3c_pcm_dma_client_out = {
+	.name		= "PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c_pcm_dma_client_in = {
+	.name		= "PCM Stereo in"
+};
+
+static struct s3c_dma_params s3c_pcm_stereo_out[] = {
+	[0] = {
+		.client		= &s3c_pcm_dma_client_out,
+		.dma_size	= 4,
+	},
+	[1] = {
+		.client		= &s3c_pcm_dma_client_out,
+		.dma_size	= 4,
+	},
+};
+
+static struct s3c_dma_params s3c_pcm_stereo_in[] = {
+	[0] = {
+		.client		= &s3c_pcm_dma_client_in,
+		.dma_size	= 4,
+	},
+	[1] = {
+		.client		= &s3c_pcm_dma_client_in,
+		.dma_size	= 4,
+	},
+};
+
+static struct s3c_pcm_info s3c_pcm[2];
+
+static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+	return cpu_dai->private_data;
+}
+
+static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on)
+{
+	void __iomem *regs = pcm->regs;
+	u32 ctl, clkctl;
+
+	clkctl = readl(regs + S3C_PCM_CLKCTL);
+	ctl = readl(regs + S3C_PCM_CTL);
+	ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK
+			 << S3C_PCM_CTL_TXDIPSTICK_SHIFT);
+
+	if (on) {
+		ctl |= S3C_PCM_CTL_TXDMA_EN;
+		ctl |= S3C_PCM_CTL_TXFIFO_EN;
+		ctl |= S3C_PCM_CTL_ENABLE;
+		ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT);
+		clkctl |= S3C_PCM_CLKCTL_SERCLK_EN;
+	} else {
+		ctl &= ~S3C_PCM_CTL_TXDMA_EN;
+		ctl &= ~S3C_PCM_CTL_TXFIFO_EN;
+
+		if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) {
+			ctl &= ~S3C_PCM_CTL_ENABLE;
+			if (!pcm->idleclk)
+				clkctl |= S3C_PCM_CLKCTL_SERCLK_EN;
+		}
+	}
+
+	writel(clkctl, regs + S3C_PCM_CLKCTL);
+	writel(ctl, regs + S3C_PCM_CTL);
+}
+
+static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on)
+{
+	void __iomem *regs = pcm->regs;
+	u32 ctl, clkctl;
+
+	ctl = readl(regs + S3C_PCM_CTL);
+	clkctl = readl(regs + S3C_PCM_CLKCTL);
+
+	if (on) {
+		ctl |= S3C_PCM_CTL_RXDMA_EN;
+		ctl |= S3C_PCM_CTL_RXFIFO_EN;
+		ctl |= S3C_PCM_CTL_ENABLE;
+		clkctl |= S3C_PCM_CLKCTL_SERCLK_EN;
+	} else {
+		ctl &= ~S3C_PCM_CTL_RXDMA_EN;
+		ctl &= ~S3C_PCM_CTL_RXFIFO_EN;
+
+		if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) {
+			ctl &= ~S3C_PCM_CTL_ENABLE;
+			if (!pcm->idleclk)
+				clkctl |= S3C_PCM_CLKCTL_SERCLK_EN;
+		}
+	}
+
+	writel(clkctl, regs + S3C_PCM_CLKCTL);
+	writel(ctl, regs + S3C_PCM_CTL);
+}
+
+static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai);
+	unsigned long flags;
+
+	dev_dbg(pcm->dev, "Entered %s\n", __func__);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		spin_lock_irqsave(&pcm->lock, flags);
+
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+			s3c_pcm_snd_rxctrl(pcm, 1);
+		else
+			s3c_pcm_snd_txctrl(pcm, 1);
+
+		spin_unlock_irqrestore(&pcm->lock, flags);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		spin_lock_irqsave(&pcm->lock, flags);
+
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+			s3c_pcm_snd_rxctrl(pcm, 0);
+		else
+			s3c_pcm_snd_txctrl(pcm, 0);
+
+		spin_unlock_irqrestore(&pcm->lock, flags);
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int s3c_pcm_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *socdai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai_link *dai = rtd->dai;
+	struct s3c_pcm_info *pcm = to_info(dai->cpu_dai);
+	void __iomem *regs = pcm->regs;
+	struct clk *clk;
+	int sclk_div, sync_div;
+	unsigned long flags;
+	u32 clkctl;
+
+	dev_dbg(pcm->dev, "Entered %s\n", __func__);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dai->cpu_dai->dma_data = pcm->dma_playback;
+	else
+		dai->cpu_dai->dma_data = pcm->dma_capture;
+
+	/* Strictly check for sample size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	spin_lock_irqsave(&pcm->lock, flags);
+
+	/* Get hold of the PCMSOURCE_CLK */
+	clkctl = readl(regs + S3C_PCM_CLKCTL);
+	if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK)
+		clk = pcm->pclk;
+	else
+		clk = pcm->cclk;
+
+	/* Set the SCLK divider */
+	sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs /
+					params_rate(params) / 2 - 1;
+
+	clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK
+			<< S3C_PCM_CLKCTL_SCLKDIV_SHIFT);
+	clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK)
+			<< S3C_PCM_CLKCTL_SCLKDIV_SHIFT);
+
+	/* Set the SYNC divider */
+	sync_div = pcm->sclk_per_fs - 1;
+
+	clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK
+				<< S3C_PCM_CLKCTL_SYNCDIV_SHIFT);
+	clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK)
+				<< S3C_PCM_CLKCTL_SYNCDIV_SHIFT);
+
+	writel(clkctl, regs + S3C_PCM_CLKCTL);
+
+	spin_unlock_irqrestore(&pcm->lock, flags);
+
+	dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs \
+				SCLK_DIV=%d SYNC_DIV=%d\n",
+				clk_get_rate(clk), pcm->sclk_per_fs,
+				sclk_div, sync_div);
+
+	return 0;
+}
+
+static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai,
+			       unsigned int fmt)
+{
+	struct s3c_pcm_info *pcm = to_info(cpu_dai);
+	void __iomem *regs = pcm->regs;
+	unsigned long flags;
+	int ret = 0;
+	u32 ctl;
+
+	dev_dbg(pcm->dev, "Entered %s\n", __func__);
+
+	spin_lock_irqsave(&pcm->lock, flags);
+
+	ctl = readl(regs + S3C_PCM_CTL);
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		/* Nothing to do, NB_NF by default */
+		break;
+	default:
+		dev_err(pcm->dev, "Unsupported clock inversion!\n");
+		ret = -EINVAL;
+		goto exit;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		/* Nothing to do, Master by default */
+		break;
+	default:
+		dev_err(pcm->dev, "Unsupported master/slave format!\n");
+		ret = -EINVAL;
+		goto exit;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
+	case SND_SOC_DAIFMT_CONT:
+		pcm->idleclk = 1;
+		break;
+	case SND_SOC_DAIFMT_GATED:
+		pcm->idleclk = 0;
+		break;
+	default:
+		dev_err(pcm->dev, "Invalid Clock gating request!\n");
+		ret = -EINVAL;
+		goto exit;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC;
+		ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC;
+		ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC;
+		break;
+	default:
+		dev_err(pcm->dev, "Unsupported data format!\n");
+		ret = -EINVAL;
+		goto exit;
+	}
+
+	writel(ctl, regs + S3C_PCM_CTL);
+
+exit:
+	spin_unlock_irqrestore(&pcm->lock, flags);
+
+	return ret;
+}
+
+static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai,
+						int div_id, int div)
+{
+	struct s3c_pcm_info *pcm = to_info(cpu_dai);
+
+	switch (div_id) {
+	case S3C_PCM_SCLK_PER_FS:
+		pcm->sclk_per_fs = div;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct s3c_pcm_info *pcm = to_info(cpu_dai);
+	void __iomem *regs = pcm->regs;
+	u32 clkctl = readl(regs + S3C_PCM_CLKCTL);
+
+	switch (clk_id) {
+	case S3C_PCM_CLKSRC_PCLK:
+		clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK;
+		break;
+
+	case S3C_PCM_CLKSRC_MUX:
+		clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK;
+
+		if (clk_get_rate(pcm->cclk) != freq)
+			clk_set_rate(pcm->cclk, freq);
+
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	writel(clkctl, regs + S3C_PCM_CLKCTL);
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops s3c_pcm_dai_ops = {
+	.set_sysclk	= s3c_pcm_set_sysclk,
+	.set_clkdiv	= s3c_pcm_set_clkdiv,
+	.trigger	= s3c_pcm_trigger,
+	.hw_params	= s3c_pcm_hw_params,
+	.set_fmt	= s3c_pcm_set_fmt,
+};
+
+#define S3C_PCM_RATES  SNDRV_PCM_RATE_8000_96000
+
+#define S3C_PCM_DECLARE(n)			\
+{								\
+	.name		 = "samsung-pcm",			\
+	.id		 = (n),				\
+	.symmetric_rates = 1,					\
+	.ops = &s3c_pcm_dai_ops,				\
+	.playback = {						\
+		.channels_min	= 2,				\
+		.channels_max	= 2,				\
+		.rates		= S3C_PCM_RATES,		\
+		.formats	= SNDRV_PCM_FMTBIT_S16_LE,	\
+	},							\
+	.capture = {						\
+		.channels_min	= 2,				\
+		.channels_max	= 2,				\
+		.rates		= S3C_PCM_RATES,		\
+		.formats	= SNDRV_PCM_FMTBIT_S16_LE,	\
+	},							\
+}
+
+struct snd_soc_dai s3c_pcm_dai[] = {
+	S3C_PCM_DECLARE(0),
+	S3C_PCM_DECLARE(1),
+};
+EXPORT_SYMBOL_GPL(s3c_pcm_dai);
+
+static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev)
+{
+	struct s3c_pcm_info *pcm;
+	struct snd_soc_dai *dai;
+	struct resource *mem_res, *dmatx_res, *dmarx_res;
+	struct s3c_audio_pdata *pcm_pdata;
+	int ret;
+
+	/* Check for valid device index */
+	if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) {
+		dev_err(&pdev->dev, "id %d out of range\n", pdev->id);
+		return -EINVAL;
+	}
+
+	pcm_pdata = pdev->dev.platform_data;
+
+	/* Check for availability of necessary resource */
+	dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!dmatx_res) {
+		dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n");
+		return -ENXIO;
+	}
+
+	dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!dmarx_res) {
+		dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n");
+		return -ENXIO;
+	}
+
+	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!mem_res) {
+		dev_err(&pdev->dev, "Unable to get register resource\n");
+		return -ENXIO;
+	}
+
+	if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) {
+		dev_err(&pdev->dev, "Unable to configure gpio\n");
+		return -EINVAL;
+	}
+
+	pcm = &s3c_pcm[pdev->id];
+	pcm->dev = &pdev->dev;
+
+	spin_lock_init(&pcm->lock);
+
+	dai = &s3c_pcm_dai[pdev->id];
+	dai->dev = &pdev->dev;
+
+	/* Default is 128fs */
+	pcm->sclk_per_fs = 128;
+
+	pcm->cclk = clk_get(&pdev->dev, "audio-bus");
+	if (IS_ERR(pcm->cclk)) {
+		dev_err(&pdev->dev, "failed to get audio-bus\n");
+		ret = PTR_ERR(pcm->cclk);
+		goto err1;
+	}
+	clk_enable(pcm->cclk);
+
+	/* record our pcm structure for later use in the callbacks */
+	dai->private_data = pcm;
+
+	if (!request_mem_region(mem_res->start,
+				resource_size(mem_res), "samsung-pcm")) {
+		dev_err(&pdev->dev, "Unable to request register region\n");
+		ret = -EBUSY;
+		goto err2;
+	}
+
+	pcm->regs = ioremap(mem_res->start, 0x100);
+	if (pcm->regs == NULL) {
+		dev_err(&pdev->dev, "cannot ioremap registers\n");
+		ret = -ENXIO;
+		goto err3;
+	}
+
+	pcm->pclk = clk_get(&pdev->dev, "pcm");
+	if (IS_ERR(pcm->pclk)) {
+		dev_err(&pdev->dev, "failed to get pcm_clock\n");
+		ret = -ENOENT;
+		goto err4;
+	}
+	clk_enable(pcm->pclk);
+
+	ret = snd_soc_register_dai(dai);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "failed to get pcm_clock\n");
+		goto err5;
+	}
+
+	s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start
+							+ S3C_PCM_RXFIFO;
+	s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start
+							+ S3C_PCM_TXFIFO;
+
+	s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start;
+	s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start;
+
+	pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id];
+	pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id];
+
+	return 0;
+
+err5:
+	clk_disable(pcm->pclk);
+	clk_put(pcm->pclk);
+err4:
+	iounmap(pcm->regs);
+err3:
+	release_mem_region(mem_res->start, resource_size(mem_res));
+err2:
+	clk_disable(pcm->cclk);
+	clk_put(pcm->cclk);
+err1:
+	return ret;
+}
+
+static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev)
+{
+	struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id];
+	struct resource *mem_res;
+
+	iounmap(pcm->regs);
+
+	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	release_mem_region(mem_res->start, resource_size(mem_res));
+
+	clk_disable(pcm->cclk);
+	clk_disable(pcm->pclk);
+	clk_put(pcm->pclk);
+	clk_put(pcm->cclk);
+
+	return 0;
+}
+
+static struct platform_driver s3c_pcm_driver = {
+	.probe  = s3c_pcm_dev_probe,
+	.remove = s3c_pcm_dev_remove,
+	.driver = {
+		.name = "samsung-pcm",
+		.owner = THIS_MODULE,
+	},
+};
+
+static int __init s3c_pcm_init(void)
+{
+	return platform_driver_register(&s3c_pcm_driver);
+}
+module_init(s3c_pcm_init);
+
+static void __exit s3c_pcm_exit(void)
+{
+	platform_driver_unregister(&s3c_pcm_driver);
+}
+module_exit(s3c_pcm_exit);
+
+/* Module information */
+MODULE_AUTHOR("Jaswinder Singh, <jassi.brar@samsung.com>");
+MODULE_DESCRIPTION("S3C PCM Controller Driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c-pcm.h b/sound/soc/s3c24xx/s3c-pcm.h
new file mode 100644
index 000000000000..69ff9971692f
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c-pcm.h
@@ -0,0 +1,123 @@
+/*  sound/soc/s3c24xx/s3c-pcm.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 __S3C_PCM_H
+#define __S3C_PCM_H __FILE__
+
+/*Register Offsets */
+#define S3C_PCM_CTL	(0x00)
+#define S3C_PCM_CLKCTL	(0x04)
+#define S3C_PCM_TXFIFO	(0x08)
+#define S3C_PCM_RXFIFO	(0x0C)
+#define S3C_PCM_IRQCTL	(0x10)
+#define S3C_PCM_IRQSTAT	(0x14)
+#define S3C_PCM_FIFOSTAT	(0x18)
+#define S3C_PCM_CLRINT	(0x20)
+
+/* PCM_CTL Bit-Fields */
+#define S3C_PCM_CTL_TXDIPSTICK_MASK		(0x3f)
+#define S3C_PCM_CTL_TXDIPSTICK_SHIFT	(13)
+#define S3C_PCM_CTL_RXDIPSTICK_MSK		(0x3f<<7)
+#define S3C_PCM_CTL_TXDMA_EN		(0x1<<6)
+#define S3C_PCM_CTL_RXDMA_EN		(0x1<<5)
+#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC	(0x1<<4)
+#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC	(0x1<<3)
+#define S3C_PCM_CTL_TXFIFO_EN		(0x1<<2)
+#define S3C_PCM_CTL_RXFIFO_EN		(0x1<<1)
+#define S3C_PCM_CTL_ENABLE			(0x1<<0)
+
+/* PCM_CLKCTL Bit-Fields */
+#define S3C_PCM_CLKCTL_SERCLK_EN		(0x1<<19)
+#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK	(0x1<<18)
+#define S3C_PCM_CLKCTL_SCLKDIV_MASK		(0x1ff)
+#define S3C_PCM_CLKCTL_SYNCDIV_MASK		(0x1ff)
+#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT	(9)
+#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT	(0)
+
+/* PCM_TXFIFO Bit-Fields */
+#define S3C_PCM_TXFIFO_DVALID	(0x1<<16)
+#define S3C_PCM_TXFIFO_DATA_MSK	(0xffff<<0)
+
+/* PCM_RXFIFO Bit-Fields */
+#define S3C_PCM_RXFIFO_DVALID	(0x1<<16)
+#define S3C_PCM_RXFIFO_DATA_MSK	(0xffff<<0)
+
+/* PCM_IRQCTL Bit-Fields */
+#define S3C_PCM_IRQCTL_IRQEN		(0x1<<14)
+#define S3C_PCM_IRQCTL_WRDEN		(0x1<<12)
+#define S3C_PCM_IRQCTL_TXEMPTYEN		(0x1<<11)
+#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN	(0x1<<10)
+#define S3C_PCM_IRQCTL_TXFULLEN		(0x1<<9)
+#define S3C_PCM_IRQCTL_TXALMSTFULLEN	(0x1<<8)
+#define S3C_PCM_IRQCTL_TXSTARVEN		(0x1<<7)
+#define S3C_PCM_IRQCTL_TXERROVRFLEN		(0x1<<6)
+#define S3C_PCM_IRQCTL_RXEMPTEN		(0x1<<5)
+#define S3C_PCM_IRQCTL_RXALMSTEMPTEN	(0x1<<4)
+#define S3C_PCM_IRQCTL_RXFULLEN		(0x1<<3)
+#define S3C_PCM_IRQCTL_RXALMSTFULLEN	(0x1<<2)
+#define S3C_PCM_IRQCTL_RXSTARVEN		(0x1<<1)
+#define S3C_PCM_IRQCTL_RXERROVRFLEN		(0x1<<0)
+
+/* PCM_IRQSTAT Bit-Fields */
+#define S3C_PCM_IRQSTAT_IRQPND		(0x1<<13)
+#define S3C_PCM_IRQSTAT_WRD_XFER		(0x1<<12)
+#define S3C_PCM_IRQSTAT_TXEMPTY		(0x1<<11)
+#define S3C_PCM_IRQSTAT_TXALMSTEMPTY	(0x1<<10)
+#define S3C_PCM_IRQSTAT_TXFULL		(0x1<<9)
+#define S3C_PCM_IRQSTAT_TXALMSTFULL		(0x1<<8)
+#define S3C_PCM_IRQSTAT_TXSTARV		(0x1<<7)
+#define S3C_PCM_IRQSTAT_TXERROVRFL		(0x1<<6)
+#define S3C_PCM_IRQSTAT_RXEMPT		(0x1<<5)
+#define S3C_PCM_IRQSTAT_RXALMSTEMPT		(0x1<<4)
+#define S3C_PCM_IRQSTAT_RXFULL		(0x1<<3)
+#define S3C_PCM_IRQSTAT_RXALMSTFULL		(0x1<<2)
+#define S3C_PCM_IRQSTAT_RXSTARV		(0x1<<1)
+#define S3C_PCM_IRQSTAT_RXERROVRFL		(0x1<<0)
+
+/* PCM_FIFOSTAT Bit-Fields */
+#define S3C_PCM_FIFOSTAT_TXCNT_MSK		(0x3f<<14)
+#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY	(0x1<<13)
+#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY	(0x1<<12)
+#define S3C_PCM_FIFOSTAT_TXFIFOFULL		(0x1<<11)
+#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL	(0x1<<10)
+#define S3C_PCM_FIFOSTAT_RXCNT_MSK		(0x3f<<4)
+#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY	(0x1<<3)
+#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY	(0x1<<2)
+#define S3C_PCM_FIFOSTAT_RXFIFOFULL		(0x1<<1)
+#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL	(0x1<<0)
+
+#define S3C_PCM_CLKSRC_PCLK	0
+#define S3C_PCM_CLKSRC_MUX	1
+
+#define S3C_PCM_SCLK_PER_FS	0
+
+/**
+ * struct s3c_pcm_info - S3C PCM Controller information
+ * @dev: The parent device passed to use from the probe.
+ * @regs: The pointer to the device register block.
+ * @dma_playback: DMA information for playback channel.
+ * @dma_capture: DMA information for capture channel.
+ */
+struct s3c_pcm_info {
+	spinlock_t lock;
+	struct device	*dev;
+	void __iomem	*regs;
+
+	unsigned int sclk_per_fs;
+
+	/* Whether to keep PCMSCLK enabled even when idle(no active xfer) */
+	unsigned int idleclk;
+
+	struct clk	*pclk;
+	struct clk	*cclk;
+
+	struct s3c_dma_params	*dma_playback;
+	struct s3c_dma_params	*dma_capture;
+};
+
+#endif /* __S3C_PCM_H */
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index a587ec40b449..359e59346ba2 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -34,11 +34,10 @@
 
 #include <plat/regs-s3c2412-iis.h>
 
-#include <plat/audio.h>
 #include <mach/regs-gpio.h>
 #include <mach/dma.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c2412-i2s.h"
 
 #define S3C2412_I2S_DEBUG 0
@@ -51,14 +50,14 @@ static struct s3c2410_dma_client s3c2412_dma_client_in = {
 	.name		= "I2S PCM Stereo in"
 };
 
-static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = {
+static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = {
 	.client		= &s3c2412_dma_client_out,
 	.channel	= DMACH_I2S_OUT,
 	.dma_addr	= S3C2410_PA_IIS + S3C2412_IISTXD,
 	.dma_size	= 4,
 };
 
-static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = {
+static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = {
 	.client		= &s3c2412_dma_client_in,
 	.channel	= DMACH_I2S_IN,
 	.dma_addr	= S3C2410_PA_IIS + S3C2412_IISRXD,
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index fc1beb0930b9..0191e3acb0b4 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -32,11 +32,10 @@
 #include <plat/regs-ac97.h>
 #include <mach/regs-gpio.h>
 #include <mach/regs-clock.h>
-#include <plat/audio.h>
 #include <asm/dma.h>
 #include <mach/dma.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-ac97.h"
 
 struct s3c24xx_ac97_info {
@@ -189,21 +188,21 @@ static struct s3c2410_dma_client s3c2443_dma_client_micin = {
 	.name = "AC97 Mic Mono in"
 };
 
-static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = {
+static struct s3c_dma_params s3c2443_ac97_pcm_stereo_out = {
 	.client		= &s3c2443_dma_client_out,
 	.channel	= DMACH_PCM_OUT,
 	.dma_addr	= S3C2440_PA_AC97 + S3C_AC97_PCM_DATA,
 	.dma_size	= 4,
 };
 
-static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = {
+static struct s3c_dma_params s3c2443_ac97_pcm_stereo_in = {
 	.client		= &s3c2443_dma_client_in,
 	.channel	= DMACH_PCM_IN,
 	.dma_addr	= S3C2440_PA_AC97 + S3C_AC97_PCM_DATA,
 	.dma_size	= 4,
 };
 
-static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = {
+static struct s3c_dma_params s3c2443_ac97_mic_mono_in = {
 	.client		= &s3c2443_dma_client_micin,
 	.channel	= DMACH_MIC_IN,
 	.dma_addr	= S3C2440_PA_AC97 + S3C_AC97_MIC_DATA,
@@ -291,7 +290,7 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	u32 ac_glbctrl;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c24xx_pcm_dma_params *)
+	int channel = ((struct s3c_dma_params *)
 		  rtd->dai->cpu_dai->dma_data)->channel;
 
 	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
@@ -340,7 +339,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
 {
 	u32 ac_glbctrl;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c24xx_pcm_dma_params *)
+	int channel = ((struct s3c_dma_params *)
 		  rtd->dai->cpu_dai->dma_data)->channel;
 
 	ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 40e2c4790f0d..0bc5950b9f02 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -32,13 +32,13 @@
 #include <mach/hardware.h>
 #include <mach/regs-gpio.h>
 #include <mach/regs-clock.h>
-#include <plat/audio.h>
+
 #include <asm/dma.h>
 #include <mach/dma.h>
 
 #include <plat/regs-iis.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-i2s.h"
 
 static struct s3c2410_dma_client s3c24xx_dma_client_out = {
@@ -49,14 +49,14 @@ static struct s3c2410_dma_client s3c24xx_dma_client_in = {
 	.name = "I2S PCM Stereo in"
 };
 
-static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = {
+static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = {
 	.client		= &s3c24xx_dma_client_out,
 	.channel	= DMACH_I2S_OUT,
 	.dma_addr	= S3C2410_PA_IIS + S3C2410_IISFIFO,
 	.dma_size	= 2,
 };
 
-static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = {
+static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = {
 	.client		= &s3c24xx_dma_client_in,
 	.channel	= DMACH_I2S_IN,
 	.dma_addr	= S3C2410_PA_IIS + S3C2410_IISFIFO,
@@ -258,12 +258,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
 		iismod &= ~S3C2410_IISMOD_16BIT;
-		((struct s3c24xx_pcm_dma_params *)
+		((struct s3c_dma_params *)
 		  rtd->dai->cpu_dai->dma_data)->dma_size = 1;
 		break;
 	case SNDRV_PCM_FORMAT_S16_LE:
 		iismod |= S3C2410_IISMOD_16BIT;
-		((struct s3c24xx_pcm_dma_params *)
+		((struct s3c_dma_params *)
 		  rtd->dai->cpu_dai->dma_data)->dma_size = 2;
 		break;
 	default:
@@ -280,7 +280,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	int channel = ((struct s3c24xx_pcm_dma_params *)
+	int channel = ((struct s3c_dma_params *)
 		  rtd->dai->cpu_dai->dma_data)->channel;
 
 	pr_debug("Entered %s\n", __func__);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
index 1966e0d5652d..507b2ed5d58b 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -21,7 +21,7 @@
 
 #include <plat/audio-simtec.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-i2s.h"
 #include "s3c24xx_simtec.h"
 
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
index 8346bd96eaf5..bdf8951af8e3 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
@@ -18,7 +18,7 @@
 
 #include <plat/audio-simtec.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-i2s.h"
 #include "s3c24xx_simtec.h"
 
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
index 25797e096175..185c0acb5ce6 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
@@ -18,7 +18,7 @@
 
 #include <plat/audio-simtec.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-i2s.h"
 #include "s3c24xx_simtec.h"
 
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index c215d32d6322..052d59659c29 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -24,7 +24,7 @@
 
 #include <plat/regs-iis.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-i2s.h"
 #include "../codecs/uda134x.h"
 
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 105a77eeded0..cc7edb5f792d 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -31,12 +31,11 @@
 #include <plat/gpio-bank-d.h>
 #include <plat/gpio-bank-e.h>
 #include <plat/gpio-cfg.h>
-#include <plat/audio.h>
 
 #include <mach/map.h>
 #include <mach/dma.h>
 
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c64xx-i2s.h"
 
 static struct s3c2410_dma_client s3c64xx_dma_client_out = {
@@ -47,7 +46,7 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = {
 	.name		= "I2S PCM Stereo in"
 };
 
-static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = {
+static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[2] = {
 	[0] = {
 		.channel	= DMACH_I2S0_OUT,
 		.client		= &s3c64xx_dma_client_out,
@@ -62,7 +61,7 @@ static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = {
 	},
 };
 
-static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = {
+static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[2] = {
 	[0] = {
 		.channel	= DMACH_I2S0_IN,
 		.client		= &s3c64xx_dma_client_in,
@@ -99,6 +98,19 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 		iismod |= S3C64XX_IISMOD_IMS_SYSMUX;
 		break;
 
+	case S3C64XX_CLKSRC_CDCLK:
+		switch (dir) {
+		case SND_SOC_CLOCK_IN:
+			iismod |= S3C64XX_IISMOD_CDCLKCON;
+			break;
+		case SND_SOC_CLOCK_OUT:
+			iismod &= ~S3C64XX_IISMOD_CDCLKCON;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+
 	default:
 		return -EINVAL;
 	}
@@ -111,8 +123,12 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai)
 {
 	struct s3c_i2sv2_info *i2s = to_info(dai);
+	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
 
-	return i2s->iis_cclk;
+	if (iismod & S3C64XX_IISMOD_IMS_SYSMUX)
+		return i2s->iis_cclk;
+	else
+		return i2s->iis_pclk;
 }
 EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock);
 
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
index 02148cee2613..abe7253b55fc 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.h
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -25,6 +25,7 @@ struct clk;
 
 #define S3C64XX_CLKSRC_PCLK	(0)
 #define S3C64XX_CLKSRC_MUX	(1)
+#define S3C64XX_CLKSRC_CDCLK    (2)
 
 extern struct snd_soc_dai s3c64xx_i2s_dai[];
 
diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c
index a2a4f5323c17..12b783b12fcb 100644
--- a/sound/soc/s3c24xx/smdk2443_wm9710.c
+++ b/sound/soc/s3c24xx/smdk2443_wm9710.c
@@ -20,7 +20,7 @@
 #include <sound/soc-dapm.h>
 
 #include "../codecs/ac97.h"
-#include "s3c24xx-pcm.h"
+#include "s3c-dma.h"
 #include "s3c24xx-ac97.h"
 
 static struct snd_soc_card smdk2443;
diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c
new file mode 100644
index 000000000000..efe4901213a3
--- /dev/null
+++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c
@@ -0,0 +1,268 @@
+/*
+ *  smdk64xx_wm8580.c
+ *
+ *  Copyright (c) 2009 Samsung Electronics Co. Ltd
+ *  Author: Jaswinder Singh <jassi.brar@samsung.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "../codecs/wm8580.h"
+#include "s3c-dma.h"
+#include "s3c64xx-i2s.h"
+
+#define S3C64XX_I2S_V4 2
+
+/* SMDK64XX has a 12MHZ crystal attached to WM8580 */
+#define SMDK64XX_WM8580_FREQ 12000000
+
+static int smdk64xx_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 *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	unsigned int pll_out;
+	int bfs, rfs, ret;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_U8:
+	case SNDRV_PCM_FORMAT_S8:
+		bfs = 16;
+		break;
+	case SNDRV_PCM_FORMAT_U16_LE:
+	case SNDRV_PCM_FORMAT_S16_LE:
+		bfs = 32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* The Fvco for WM8580 PLLs must fall within [90,100]MHz.
+	 * This criterion can't be met if we request PLL output
+	 * as {8000x256, 64000x256, 11025x256}Hz.
+	 * As a wayout, we rather change rfs to a minimum value that
+	 * results in (params_rate(params) * rfs), and itself, acceptable
+	 * to both - the CODEC and the CPU.
+	 */
+	switch (params_rate(params)) {
+	case 16000:
+	case 22050:
+	case 32000:
+	case 44100:
+	case 48000:
+	case 88200:
+	case 96000:
+		rfs = 256;
+		break;
+	case 64000:
+		rfs = 384;
+		break;
+	case 8000:
+	case 11025:
+		rfs = 512;
+		break;
+	default:
+		return -EINVAL;
+	}
+	pll_out = params_rate(params) * rfs;
+
+	/* Set the 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)
+		return ret;
+
+	/* Set the AP 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)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK,
+					0, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* We use PCLK for basic ops in SoC-Slave mode */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK,
+					0, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* Set WM8580 to drive MCLK from its PLLA */
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK,
+					WM8580_CLKSRC_PLLA);
+	if (ret < 0)
+		return ret;
+
+	/* Explicitly set WM8580-DAC to source from MCLK */
+	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL,
+					WM8580_CLKSRC_MCLK);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0,
+					SMDK64XX_WM8580_FREQ, pll_out);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+/*
+ * SMDK64XX WM8580 DAI operations.
+ */
+static struct snd_soc_ops smdk64xx_ops = {
+	.hw_params = smdk64xx_hw_params,
+};
+
+/* SMDK64xx Playback widgets */
+static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = {
+	SND_SOC_DAPM_HP("Front-L/R", NULL),
+	SND_SOC_DAPM_HP("Center/Sub", NULL),
+	SND_SOC_DAPM_HP("Rear-L/R", NULL),
+};
+
+/* SMDK64xx Capture widgets */
+static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = {
+	SND_SOC_DAPM_MIC("MicIn", NULL),
+	SND_SOC_DAPM_LINE("LineIn", NULL),
+};
+
+/* SMDK-PAIFTX connections */
+static const struct snd_soc_dapm_route audio_map_tx[] = {
+	/* MicIn feeds AINL */
+	{"AINL", NULL, "MicIn"},
+
+	/* LineIn feeds AINL/R */
+	{"AINL", NULL, "LineIn"},
+	{"AINR", NULL, "LineIn"},
+};
+
+/* SMDK-PAIFRX connections */
+static const struct snd_soc_dapm_route audio_map_rx[] = {
+	/* Front Left/Right are fed VOUT1L/R */
+	{"Front-L/R", NULL, "VOUT1L"},
+	{"Front-L/R", NULL, "VOUT1R"},
+
+	/* Center/Sub are fed VOUT2L/R */
+	{"Center/Sub", NULL, "VOUT2L"},
+	{"Center/Sub", NULL, "VOUT2R"},
+
+	/* Rear Left/Right are fed VOUT3L/R */
+	{"Rear-L/R", NULL, "VOUT3L"},
+	{"Rear-L/R", NULL, "VOUT3R"},
+};
+
+static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec)
+{
+	/* Add smdk64xx specific Capture widgets */
+	snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt,
+				  ARRAY_SIZE(wm8580_dapm_widgets_cpt));
+
+	/* Set up PAIFTX audio path */
+	snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx));
+
+	/* Enabling the microphone requires the fitting of a 0R
+	 * resistor to connect the line from the microphone jack.
+	 */
+	snd_soc_dapm_disable_pin(codec, "MicIn");
+
+	/* signal a DAPM event */
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec)
+{
+	/* Add smdk64xx specific Playback widgets */
+	snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk,
+				  ARRAY_SIZE(wm8580_dapm_widgets_pbk));
+
+	/* Set up PAIFRX audio path */
+	snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx));
+
+	/* signal a DAPM event */
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link smdk64xx_dai[] = {
+{ /* Primary Playback i/f */
+	.name = "WM8580 PAIF RX",
+	.stream_name = "Playback",
+	.cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4],
+	.codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX],
+	.init = smdk64xx_wm8580_init_paifrx,
+	.ops = &smdk64xx_ops,
+},
+{ /* Primary Capture i/f */
+	.name = "WM8580 PAIF TX",
+	.stream_name = "Capture",
+	.cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4],
+	.codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX],
+	.init = smdk64xx_wm8580_init_paiftx,
+	.ops = &smdk64xx_ops,
+},
+};
+
+static struct snd_soc_card smdk64xx = {
+	.name = "smdk64xx",
+	.platform = &s3c24xx_soc_platform,
+	.dai_link = smdk64xx_dai,
+	.num_links = ARRAY_SIZE(smdk64xx_dai),
+};
+
+static struct snd_soc_device smdk64xx_snd_devdata = {
+	.card = &smdk64xx,
+	.codec_dev = &soc_codec_dev_wm8580,
+};
+
+static struct platform_device *smdk64xx_snd_device;
+
+static int __init smdk64xx_audio_init(void)
+{
+	int ret;
+
+	smdk64xx_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!smdk64xx_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata);
+	smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev;
+	ret = platform_device_add(smdk64xx_snd_device);
+
+	if (ret)
+		platform_device_put(smdk64xx_snd_device);
+
+	return ret;
+}
+module_init(smdk64xx_audio_init);
+
+MODULE_AUTHOR("Jaswinder Singh, jassi.brar@samsung.com");
+MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 83b8028e209d..0eb1722f6581 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -423,7 +423,7 @@ static void s6000_pcm_free(struct snd_pcm *pcm)
 	snd_pcm_lib_preallocate_free_for_all(pcm);
 }
 
-static u64 s6000_pcm_dmamask = DMA_32BIT_MASK;
+static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32);
 
 static int s6000_pcm_new(struct snd_card *card,
 			 struct snd_soc_dai *dai, struct snd_pcm *pcm)
@@ -435,7 +435,7 @@ static int s6000_pcm_new(struct snd_card *card,
 	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;
+		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
 
 	if (params->dma_in) {
 		s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in),
diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig
index 9154b4363db3..9e6976586554 100644
--- a/sound/soc/sh/Kconfig
+++ b/sound/soc/sh/Kconfig
@@ -23,7 +23,6 @@ config SND_SOC_SH4_SSI
 config SND_SOC_SH4_FSI
 	tristate "SH4 FSI support"
 	depends on CPU_SUBTYPE_SH7724
-        select SH_DMA
 	help
 	  This option enables FSI sound support
 
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 44123248b630..9c49c11c43ce 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -17,7 +17,7 @@
 #include <linux/platform_device.h>
 #include <linux/delay.h>
 #include <linux/list.h>
-#include <linux/clk.h>
+#include <linux/pm_runtime.h>
 #include <linux/io.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -26,8 +26,6 @@
 #include <sound/pcm_params.h>
 #include <sound/sh_fsi.h>
 #include <asm/atomic.h>
-#include <asm/dma.h>
-#include <asm/dma-sh.h>
 
 #define DO_FMT		0x0000
 #define DOFF_CTL	0x0004
@@ -97,7 +95,6 @@ struct fsi_priv {
 
 	int fifo_max;
 	int chan;
-	int dma_chan;
 
 	int byte_offset;
 	int period_len;
@@ -108,7 +105,6 @@ struct fsi_priv {
 struct fsi_master {
 	void __iomem *base;
 	int irq;
-	struct clk *clk;
 	struct fsi_priv fsia;
 	struct fsi_priv fsib;
 	struct sh_fsi_platform_info *info;
@@ -308,62 +304,6 @@ static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play)
 	return residue;
 }
 
-static int fsi_get_residue(struct fsi_priv *fsi, int is_play)
-{
-	int residue;
-	int width;
-	struct snd_pcm_runtime *runtime;
-
-	runtime = fsi->substream->runtime;
-
-	/* get 1 channel data width */
-	width = frames_to_bytes(runtime, 1) / fsi->chan;
-
-	if (2 == width)
-		residue = fsi_get_fifo_residue(fsi, is_play);
-	else
-		residue = get_dma_residue(fsi->dma_chan);
-
-	return residue;
-}
-
-/************************************************************************
-
-
-		basic dma function
-
-
-************************************************************************/
-#define PORTA_DMA 0
-#define PORTB_DMA 1
-
-static int fsi_get_dma_chan(void)
-{
-	if (0 != request_dma(PORTA_DMA, "fsia"))
-		return -EIO;
-
-	if (0 != request_dma(PORTB_DMA, "fsib")) {
-		free_dma(PORTA_DMA);
-		return -EIO;
-	}
-
-	master->fsia.dma_chan = PORTA_DMA;
-	master->fsib.dma_chan = PORTB_DMA;
-
-	return 0;
-}
-
-static void fsi_free_dma_chan(void)
-{
-	dma_wait_for_completion(PORTA_DMA);
-	dma_wait_for_completion(PORTB_DMA);
-	free_dma(PORTA_DMA);
-	free_dma(PORTB_DMA);
-
-	master->fsia.dma_chan = -1;
-	master->fsib.dma_chan = -1;
-}
-
 /************************************************************************
 
 
@@ -435,44 +375,6 @@ static void fsi_soft_all_reset(void)
 	mdelay(10);
 }
 
-static void fsi_16data_push(struct fsi_priv *fsi,
-			   struct snd_pcm_runtime *runtime,
-			   int send)
-{
-	u16 *dma_start;
-	u32 snd;
-	int i;
-
-	/* get dma start position for FSI */
-	dma_start = (u16 *)runtime->dma_area;
-	dma_start += fsi->byte_offset / 2;
-
-	/*
-	 * soft dma
-	 * FSI can not use DMA when 16bpp
-	 */
-	for (i = 0; i < send; i++) {
-		snd = (u32)dma_start[i];
-		fsi_reg_write(fsi, DODT, snd << 8);
-	}
-}
-
-static void fsi_32data_push(struct fsi_priv *fsi,
-			   struct snd_pcm_runtime *runtime,
-			   int send)
-{
-	u32 *dma_start;
-
-	/* get dma start position for FSI */
-	dma_start = (u32 *)runtime->dma_area;
-	dma_start += fsi->byte_offset / 4;
-
-	dma_wait_for_completion(fsi->dma_chan);
-	dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR));
-	dma_write(fsi->dma_chan, (u32)dma_start,
-		  (u32)(fsi->base + DODT), send * 4);
-}
-
 /* playback interrupt */
 static int fsi_data_push(struct fsi_priv *fsi)
 {
@@ -481,6 +383,8 @@ static int fsi_data_push(struct fsi_priv *fsi)
 	int send;
 	int fifo_free;
 	int width;
+	u8 *start;
+	int i;
 
 	if (!fsi			||
 	    !fsi->substream		||
@@ -515,12 +419,22 @@ static int fsi_data_push(struct fsi_priv *fsi)
 	if (fifo_free < send)
 		send = fifo_free;
 
-	if (2 == width)
-		fsi_16data_push(fsi, runtime, send);
-	else if (4 == width)
-		fsi_32data_push(fsi, runtime, send);
-	else
+	start = runtime->dma_area;
+	start += fsi->byte_offset;
+
+	switch (width) {
+	case 2:
+		for (i = 0; i < send; i++)
+			fsi_reg_write(fsi, DODT,
+				      ((u32)*((u16 *)start + i) << 8));
+		break;
+	case 4:
+		for (i = 0; i < send; i++)
+			fsi_reg_write(fsi, DODT, *((u32 *)start + i));
+		break;
+	default:
 		return -EINVAL;
+	}
 
 	fsi->byte_offset += send * width;
 
@@ -532,6 +446,75 @@ static int fsi_data_push(struct fsi_priv *fsi)
 	return 0;
 }
 
+static int fsi_data_pop(struct fsi_priv *fsi)
+{
+	struct snd_pcm_runtime *runtime;
+	struct snd_pcm_substream *substream = NULL;
+	int free;
+	int fifo_fill;
+	int width;
+	u8 *start;
+	int i;
+
+	if (!fsi			||
+	    !fsi->substream		||
+	    !fsi->substream->runtime)
+		return -EINVAL;
+
+	runtime = fsi->substream->runtime;
+
+	/* FSI FIFO has limit.
+	 * So, this driver can not send periods data at a time
+	 */
+	if (fsi->byte_offset >=
+	    fsi->period_len * (fsi->periods + 1)) {
+
+		substream = fsi->substream;
+		fsi->periods = (fsi->periods + 1) % runtime->periods;
+
+		if (0 == fsi->periods)
+			fsi->byte_offset = 0;
+	}
+
+	/* get 1 channel data width */
+	width = frames_to_bytes(runtime, 1) / fsi->chan;
+
+	/* get free space for alsa */
+	free = (fsi->buffer_len - fsi->byte_offset) / width;
+
+	/* get recv size */
+	fifo_fill = fsi_get_fifo_residue(fsi, 0);
+
+	if (free < fifo_fill)
+		fifo_fill = free;
+
+	start = runtime->dma_area;
+	start += fsi->byte_offset;
+
+	switch (width) {
+	case 2:
+		for (i = 0; i < fifo_fill; i++)
+			*((u16 *)start + i) =
+				(u16)(fsi_reg_read(fsi, DIDT) >> 8);
+		break;
+	case 4:
+		for (i = 0; i < fifo_fill; i++)
+			*((u32 *)start + i) = fsi_reg_read(fsi, DIDT);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	fsi->byte_offset += fifo_fill * width;
+
+	fsi_irq_enable(fsi, 0);
+
+	if (substream)
+		snd_pcm_period_elapsed(substream);
+
+	return 0;
+}
+
 static irqreturn_t fsi_interrupt(int irq, void *data)
 {
 	u32 status = fsi_master_read(SOFT_RST) & ~0x00000010;
@@ -545,6 +528,10 @@ static irqreturn_t fsi_interrupt(int irq, void *data)
 		fsi_data_push(&master->fsia);
 	if (int_st & INT_B_OUT)
 		fsi_data_push(&master->fsib);
+	if (int_st & INT_A_IN)
+		fsi_data_pop(&master->fsia);
+	if (int_st & INT_B_IN)
+		fsi_data_pop(&master->fsib);
 
 	fsi_master_write(INT_ST, 0x0000000);
 
@@ -571,7 +558,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 	int is_master;
 	int ret = 0;
 
-	clk_enable(master->clk);
+	pm_runtime_get_sync(dai->dev);
 
 	/* CKG1 */
 	data = is_play ? (1 << 0) : (1 << 4);
@@ -664,8 +651,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 	}
 
 	fsi_reg_write(fsi, reg, data);
-	dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n",
-		msg, fsi->chan, fsi->dma_chan);
 
 	/*
 	 * clear clk reset if master mode
@@ -688,7 +673,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
 	fsi_irq_disable(fsi, is_play);
 	fsi_clk_ctrl(fsi, 0);
 
-	clk_disable(master->clk);
+	pm_runtime_put_sync(dai->dev);
 }
 
 static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -699,16 +684,12 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	int ret = 0;
 
-	/* capture not supported */
-	if (!is_play)
-		return -ENODEV;
-
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		fsi_stream_push(fsi, substream,
 				frames_to_bytes(runtime, runtime->buffer_size),
 				frames_to_bytes(runtime, runtime->period_size));
-		ret = fsi_data_push(fsi);
+		ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi);
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		fsi_irq_disable(fsi, is_play);
@@ -780,10 +761,9 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsi_priv *fsi = fsi_get(substream);
-	int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	long location;
 
-	location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play);
+	location = (fsi->byte_offset - 1);
 	if (location < 0)
 		location = 0;
 
@@ -845,7 +825,12 @@ struct snd_soc_dai fsi_soc_dai[] = {
 			.channels_min	= 1,
 			.channels_max	= 8,
 		},
-		/* capture not supported */
+		.capture = {
+			.rates		= FSI_RATES,
+			.formats	= FSI_FMTS,
+			.channels_min	= 1,
+			.channels_max	= 8,
+		},
 		.ops = &fsi_dai_ops,
 	},
 	{
@@ -857,7 +842,12 @@ struct snd_soc_dai fsi_soc_dai[] = {
 			.channels_min	= 1,
 			.channels_max	= 8,
 		},
-		/* capture not supported */
+		.capture = {
+			.rates		= FSI_RATES,
+			.formats	= FSI_FMTS,
+			.channels_min	= 1,
+			.channels_max	= 8,
+		},
 		.ops = &fsi_dai_ops,
 	},
 };
@@ -881,7 +871,6 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform);
 static int fsi_probe(struct platform_device *pdev)
 {
 	struct resource *res;
-	char clk_name[8];
 	unsigned int irq;
 	int ret;
 
@@ -912,23 +901,8 @@ static int fsi_probe(struct platform_device *pdev)
 	master->fsia.base	= master->base;
 	master->fsib.base	= master->base + 0x40;
 
-	master->fsia.dma_chan = -1;
-	master->fsib.dma_chan = -1;
-
-	ret = fsi_get_dma_chan();
-	if (ret < 0) {
-		dev_err(&pdev->dev, "cannot get dma api\n");
-		goto exit_iounmap;
-	}
-
-	/* FSI is based on SPU mstp */
-	snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id);
-	master->clk = clk_get(NULL, clk_name);
-	if (IS_ERR(master->clk)) {
-		dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name);
-		ret = -EIO;
-		goto exit_free_dma;
-	}
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_resume(&pdev->dev);
 
 	fsi_soc_dai[0].dev		= &pdev->dev;
 	fsi_soc_dai[1].dev		= &pdev->dev;
@@ -938,7 +912,7 @@ static int fsi_probe(struct platform_device *pdev)
 	ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master);
 	if (ret) {
 		dev_err(&pdev->dev, "irq request err\n");
-		goto exit_free_dma;
+		goto exit_iounmap;
 	}
 
 	ret = snd_soc_register_platform(&fsi_soc_platform);
@@ -951,10 +925,9 @@ static int fsi_probe(struct platform_device *pdev)
 
 exit_free_irq:
 	free_irq(irq, master);
-exit_free_dma:
-	fsi_free_dma_chan();
 exit_iounmap:
 	iounmap(master->base);
+	pm_runtime_disable(&pdev->dev);
 exit_kfree:
 	kfree(master);
 	master = NULL;
@@ -967,9 +940,7 @@ static int fsi_remove(struct platform_device *pdev)
 	snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai));
 	snd_soc_unregister_platform(&fsi_soc_platform);
 
-	clk_put(master->clk);
-
-	fsi_free_dma_chan();
+	pm_runtime_disable(&pdev->dev);
 
 	free_irq(master->irq, master);
 
@@ -979,9 +950,27 @@ static int fsi_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static int fsi_runtime_nop(struct device *dev)
+{
+	/* Runtime PM callback shared between ->runtime_suspend()
+	 * and ->runtime_resume(). Simply returns success.
+	 *
+	 * This driver re-initializes all registers after
+	 * pm_runtime_get_sync() anyway so there is no need
+	 * to save and restore registers here.
+	 */
+	return 0;
+}
+
+static struct dev_pm_ops fsi_pm_ops = {
+	.runtime_suspend	= fsi_runtime_nop,
+	.runtime_resume		= fsi_runtime_nop,
+};
+
 static struct platform_driver fsi_driver = {
 	.driver 	= {
 		.name	= "sh_fsi",
+		.pm	= &fsi_pm_ops,
 	},
 	.probe		= fsi_probe,
 	.remove		= fsi_remove,
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index c8ceddc2a26c..d2505e8b06c9 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -77,6 +77,35 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data,
 #define snd_soc_7_9_spi_write NULL
 #endif
 
+static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
+			     unsigned int value)
+{
+	u8 *cache = codec->reg_cache;
+	u8 data[2];
+
+	BUG_ON(codec->volatile_register);
+
+	data[0] = reg & 0xff;
+	data[1] = value & 0xff;
+
+	if (reg < codec->reg_cache_size)
+		cache[reg] = value;
+
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
+				     unsigned int reg)
+{
+	u8 *cache = codec->reg_cache;
+	if (reg >= codec->reg_cache_size)
+		return -1;
+	return cache[reg];
+}
+
 static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
 			      unsigned int value)
 {
@@ -150,9 +179,20 @@ static struct {
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
 } io_types[] = {
-	{ 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read },
-	{ 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read,
-	  snd_soc_8_16_read_i2c },
+	{
+		.addr_bits = 7, .data_bits = 9,
+		.write = snd_soc_7_9_write, .read = snd_soc_7_9_read,
+		.spi_write = snd_soc_7_9_spi_write 
+	},
+	{
+		.addr_bits = 8, .data_bits = 8,
+		.write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
+	},
+	{
+		.addr_bits = 8, .data_bits = 16,
+		.write = snd_soc_8_16_write, .read = snd_soc_8_16_read,
+		.i2c_read = snd_soc_8_16_read_i2c,
+	},
 };
 
 /**
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 0a1b2f64bbee..ef8f28284cb9 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -37,7 +37,6 @@
 #include <sound/initval.h>
 
 static DEFINE_MUTEX(pcm_mutex);
-static DEFINE_MUTEX(io_mutex);
 static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
 
 #ifdef CONFIG_DEBUG_FS
@@ -81,6 +80,173 @@ static int run_delayed_work(struct delayed_work *dwork)
 	return ret;
 }
 
+/* codec register dump */
+static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
+{
+	int i, step = 1, count = 0;
+
+	if (!codec->reg_cache_size)
+		return 0;
+
+	if (codec->reg_cache_step)
+		step = codec->reg_cache_step;
+
+	count += sprintf(buf, "%s registers\n", codec->name);
+	for (i = 0; i < codec->reg_cache_size; i += step) {
+		if (codec->readable_register && !codec->readable_register(i))
+			continue;
+
+		count += sprintf(buf + count, "%2x: ", i);
+		if (count >= PAGE_SIZE - 1)
+			break;
+
+		if (codec->display_register)
+			count += codec->display_register(codec, buf + count,
+							 PAGE_SIZE - count, i);
+		else
+			count += snprintf(buf + count, PAGE_SIZE - count,
+					  "%4x", codec->read(codec, i));
+
+		if (count >= PAGE_SIZE - 1)
+			break;
+
+		count += snprintf(buf + count, PAGE_SIZE - count, "\n");
+		if (count >= PAGE_SIZE - 1)
+			break;
+	}
+
+	/* Truncate count; min() would cause a warning */
+	if (count >= PAGE_SIZE)
+		count = PAGE_SIZE - 1;
+
+	return count;
+}
+static ssize_t codec_reg_show(struct device *dev,
+	struct device_attribute *attr, char *buf)
+{
+	struct snd_soc_device *devdata = dev_get_drvdata(dev);
+	return soc_codec_reg_show(devdata->card->codec, buf);
+}
+
+static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
+
+#ifdef CONFIG_DEBUG_FS
+static int codec_reg_open_file(struct inode *inode, struct file *file)
+{
+	file->private_data = inode->i_private;
+	return 0;
+}
+
+static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
+			       size_t count, loff_t *ppos)
+{
+	ssize_t ret;
+	struct snd_soc_codec *codec = file->private_data;
+	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
+	ret = soc_codec_reg_show(codec, buf);
+	if (ret >= 0)
+		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+	kfree(buf);
+	return ret;
+}
+
+static ssize_t codec_reg_write_file(struct file *file,
+		const char __user *user_buf, size_t count, loff_t *ppos)
+{
+	char buf[32];
+	int buf_size;
+	char *start = buf;
+	unsigned long reg, value;
+	int step = 1;
+	struct snd_soc_codec *codec = file->private_data;
+
+	buf_size = min(count, (sizeof(buf)-1));
+	if (copy_from_user(buf, user_buf, buf_size))
+		return -EFAULT;
+	buf[buf_size] = 0;
+
+	if (codec->reg_cache_step)
+		step = codec->reg_cache_step;
+
+	while (*start == ' ')
+		start++;
+	reg = simple_strtoul(start, &start, 16);
+	if ((reg >= codec->reg_cache_size) || (reg % step))
+		return -EINVAL;
+	while (*start == ' ')
+		start++;
+	if (strict_strtoul(start, 16, &value))
+		return -EINVAL;
+	codec->write(codec, reg, value);
+	return buf_size;
+}
+
+static const struct file_operations codec_reg_fops = {
+	.open = codec_reg_open_file,
+	.read = codec_reg_read_file,
+	.write = codec_reg_write_file,
+};
+
+static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+{
+	char codec_root[128];
+
+	if (codec->dev)
+		snprintf(codec_root, sizeof(codec_root),
+			"%s.%s", codec->name, dev_name(codec->dev));
+	else
+		snprintf(codec_root, sizeof(codec_root),
+			"%s", codec->name);
+
+	codec->debugfs_codec_root = debugfs_create_dir(codec_root,
+						       debugfs_root);
+	if (!codec->debugfs_codec_root) {
+		printk(KERN_WARNING
+		       "ASoC: Failed to create codec debugfs directory\n");
+		return;
+	}
+
+	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
+						 codec->debugfs_codec_root,
+						 codec, &codec_reg_fops);
+	if (!codec->debugfs_reg)
+		printk(KERN_WARNING
+		       "ASoC: Failed to create codec register debugfs file\n");
+
+	codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
+						     codec->debugfs_codec_root,
+						     &codec->pop_time);
+	if (!codec->debugfs_pop_time)
+		printk(KERN_WARNING
+		       "Failed to create pop time debugfs file\n");
+
+	codec->debugfs_dapm = debugfs_create_dir("dapm",
+						 codec->debugfs_codec_root);
+	if (!codec->debugfs_dapm)
+		printk(KERN_WARNING
+		       "Failed to create DAPM debugfs directory\n");
+
+	snd_soc_dapm_debugfs_init(codec);
+}
+
+static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
+{
+	debugfs_remove_recursive(codec->debugfs_codec_root);
+}
+
+#else
+
+static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
+{
+}
+
+static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
+{
+}
+#endif
+
 #ifdef CONFIG_SND_SOC_AC97_BUS
 /* unregister ac97 codec */
 static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
@@ -790,45 +956,6 @@ static int soc_resume(struct device *dev)
 
 	return 0;
 }
-
-/**
- * snd_soc_suspend_device: Notify core of device suspend
- *
- * @dev: Device being suspended.
- *
- * In order to ensure that the entire audio subsystem is suspended in a
- * coordinated fashion ASoC devices should suspend themselves when
- * called by ASoC.  When the standard kernel suspend process asks the
- * device to suspend it should call this function to initiate a suspend
- * of the entire ASoC card.
- *
- * \note Currently this function is stubbed out.
- */
-int snd_soc_suspend_device(struct device *dev)
-{
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_suspend_device);
-
-/**
- * snd_soc_resume_device: Notify core of device resume
- *
- * @dev: Device being resumed.
- *
- * In order to ensure that the entire audio subsystem is resumed in a
- * coordinated fashion ASoC devices should resume themselves when called
- * by ASoC.  When the standard kernel resume process asks the device
- * to resume it should call this function.  Once all the components of
- * the card have notified that they are ready to be resumed the card
- * will be resumed.
- *
- * \note Currently this function is stubbed out.
- */
-int snd_soc_resume_device(struct device *dev)
-{
-	return 0;
-}
-EXPORT_SYMBOL_GPL(snd_soc_resume_device);
 #else
 #define soc_suspend	NULL
 #define soc_resume	NULL
@@ -843,6 +970,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 						    struct platform_device,
 						    dev);
 	struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
+	struct snd_soc_codec *codec;
 	struct snd_soc_platform *platform;
 	struct snd_soc_dai *dai;
 	int i, found, ret, ac97;
@@ -931,6 +1059,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 		if (ret < 0)
 			goto cpu_dai_err;
 	}
+	codec = card->codec;
 
 	if (platform->probe) {
 		ret = platform->probe(pdev);
@@ -945,10 +1074,69 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
 #endif
 
+	for (i = 0; i < card->num_links; i++) {
+		if (card->dai_link[i].init) {
+			ret = card->dai_link[i].init(codec);
+			if (ret < 0) {
+				printk(KERN_ERR "asoc: failed to init %s\n",
+					card->dai_link[i].stream_name);
+				continue;
+			}
+		}
+		if (card->dai_link[i].codec_dai->ac97_control)
+			ac97 = 1;
+	}
+
+	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
+		 "%s",  card->name);
+	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",
+				codec->name);
+		goto card_err;
+	}
+
+	mutex_lock(&codec->mutex);
+#ifdef CONFIG_SND_SOC_AC97_BUS
+	/* Only instantiate AC97 if not already done by the adaptor
+	 * for the generic AC97 subsystem.
+	 */
+	if (ac97 && strcmp(codec->name, "AC97") != 0) {
+		ret = soc_ac97_dev_register(codec);
+		if (ret < 0) {
+			printk(KERN_ERR "asoc: AC97 device register failed\n");
+			snd_card_free(codec->card);
+			mutex_unlock(&codec->mutex);
+			goto card_err;
+		}
+	}
+#endif
+
+	ret = snd_soc_dapm_sys_add(card->socdev->dev);
+	if (ret < 0)
+		printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
+
+	ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg);
+	if (ret < 0)
+		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
+
+	soc_init_codec_debugfs(codec);
+	mutex_unlock(&codec->mutex);
+
 	card->instantiated = 1;
 
 	return;
 
+card_err:
+	if (platform->remove)
+		platform->remove(pdev);
+
 platform_err:
 	if (codec_dev->remove)
 		codec_dev->remove(pdev);
@@ -1151,157 +1339,6 @@ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg)
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register);
 
-/* codec register dump */
-static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
-{
-	int i, step = 1, count = 0;
-
-	if (!codec->reg_cache_size)
-		return 0;
-
-	if (codec->reg_cache_step)
-		step = codec->reg_cache_step;
-
-	count += sprintf(buf, "%s registers\n", codec->name);
-	for (i = 0; i < codec->reg_cache_size; i += step) {
-		if (codec->readable_register && !codec->readable_register(i))
-			continue;
-
-		count += sprintf(buf + count, "%2x: ", i);
-		if (count >= PAGE_SIZE - 1)
-			break;
-
-		if (codec->display_register)
-			count += codec->display_register(codec, buf + count,
-							 PAGE_SIZE - count, i);
-		else
-			count += snprintf(buf + count, PAGE_SIZE - count,
-					  "%4x", codec->read(codec, i));
-
-		if (count >= PAGE_SIZE - 1)
-			break;
-
-		count += snprintf(buf + count, PAGE_SIZE - count, "\n");
-		if (count >= PAGE_SIZE - 1)
-			break;
-	}
-
-	/* Truncate count; min() would cause a warning */
-	if (count >= PAGE_SIZE)
-		count = PAGE_SIZE - 1;
-
-	return count;
-}
-static ssize_t codec_reg_show(struct device *dev,
-	struct device_attribute *attr, char *buf)
-{
-	struct snd_soc_device *devdata = dev_get_drvdata(dev);
-	return soc_codec_reg_show(devdata->card->codec, buf);
-}
-
-static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
-
-#ifdef CONFIG_DEBUG_FS
-static int codec_reg_open_file(struct inode *inode, struct file *file)
-{
-	file->private_data = inode->i_private;
-	return 0;
-}
-
-static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
-			       size_t count, loff_t *ppos)
-{
-	ssize_t ret;
-	struct snd_soc_codec *codec = file->private_data;
-	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
-	if (!buf)
-		return -ENOMEM;
-	ret = soc_codec_reg_show(codec, buf);
-	if (ret >= 0)
-		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
-	kfree(buf);
-	return ret;
-}
-
-static ssize_t codec_reg_write_file(struct file *file,
-		const char __user *user_buf, size_t count, loff_t *ppos)
-{
-	char buf[32];
-	int buf_size;
-	char *start = buf;
-	unsigned long reg, value;
-	int step = 1;
-	struct snd_soc_codec *codec = file->private_data;
-
-	buf_size = min(count, (sizeof(buf)-1));
-	if (copy_from_user(buf, user_buf, buf_size))
-		return -EFAULT;
-	buf[buf_size] = 0;
-
-	if (codec->reg_cache_step)
-		step = codec->reg_cache_step;
-
-	while (*start == ' ')
-		start++;
-	reg = simple_strtoul(start, &start, 16);
-	if ((reg >= codec->reg_cache_size) || (reg % step))
-		return -EINVAL;
-	while (*start == ' ')
-		start++;
-	if (strict_strtoul(start, 16, &value))
-		return -EINVAL;
-	codec->write(codec, reg, value);
-	return buf_size;
-}
-
-static const struct file_operations codec_reg_fops = {
-	.open = codec_reg_open_file,
-	.read = codec_reg_read_file,
-	.write = codec_reg_write_file,
-};
-
-static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
-{
-	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
-						 debugfs_root, codec,
-						 &codec_reg_fops);
-	if (!codec->debugfs_reg)
-		printk(KERN_WARNING
-		       "ASoC: Failed to create codec register debugfs file\n");
-
-	codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
-						     debugfs_root,
-						     &codec->pop_time);
-	if (!codec->debugfs_pop_time)
-		printk(KERN_WARNING
-		       "Failed to create pop time debugfs file\n");
-
-	codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root);
-	if (!codec->debugfs_dapm)
-		printk(KERN_WARNING
-		       "Failed to create DAPM debugfs directory\n");
-
-	snd_soc_dapm_debugfs_init(codec);
-}
-
-static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
-{
-	debugfs_remove_recursive(codec->debugfs_dapm);
-	debugfs_remove(codec->debugfs_pop_time);
-	debugfs_remove(codec->debugfs_reg);
-}
-
-#else
-
-static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
-{
-}
-
-static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
-{
-}
-#endif
-
 /**
  * snd_soc_new_ac97_codec - initailise AC97 device
  * @codec: audio codec
@@ -1369,19 +1406,41 @@ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
 	int change;
 	unsigned int old, new;
 
-	mutex_lock(&io_mutex);
 	old = snd_soc_read(codec, reg);
 	new = (old & ~mask) | value;
 	change = old != new;
 	if (change)
 		snd_soc_write(codec, reg, new);
 
-	mutex_unlock(&io_mutex);
 	return change;
 }
 EXPORT_SYMBOL_GPL(snd_soc_update_bits);
 
 /**
+ * snd_soc_update_bits_locked - update codec register bits
+ * @codec: audio codec
+ * @reg: codec register
+ * @mask: register mask
+ * @value: new value
+ *
+ * Writes new register value, and takes the codec mutex.
+ *
+ * Returns 1 for change else 0.
+ */
+static int snd_soc_update_bits_locked(struct snd_soc_codec *codec,
+				unsigned short reg, unsigned int mask,
+				unsigned int value)
+{
+	int change;
+
+	mutex_lock(&codec->mutex);
+	change = snd_soc_update_bits(codec, reg, mask, value);
+	mutex_unlock(&codec->mutex);
+
+	return change;
+}
+
+/**
  * snd_soc_test_bits - test register for change
  * @codec: audio codec
  * @reg: codec register
@@ -1399,11 +1458,9 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
 	int change;
 	unsigned int old, new;
 
-	mutex_lock(&io_mutex);
 	old = snd_soc_read(codec, reg);
 	new = (old & ~mask) | value;
 	change = old != new;
-	mutex_unlock(&io_mutex);
 
 	return change;
 }
@@ -1450,89 +1507,16 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 			mutex_unlock(&codec->mutex);
 			return ret;
 		}
-	}
-
-	mutex_unlock(&codec->mutex);
-	return ret;
-}
-EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
-
-/**
- * snd_soc_init_card - register sound card
- * @socdev: the SoC audio device
- *
- * Register a SoC sound card. Also registers an AC97 device if the
- * codec is AC97 for ad hoc devices.
- *
- * Returns 0 for success, else error.
- */
-int snd_soc_init_card(struct snd_soc_device *socdev)
-{
-	struct snd_soc_card *card = socdev->card;
-	struct snd_soc_codec *codec = card->codec;
-	int ret = 0, i, ac97 = 0, err = 0;
-
-	for (i = 0; i < card->num_links; i++) {
-		if (card->dai_link[i].init) {
-			err = card->dai_link[i].init(codec);
-			if (err < 0) {
-				printk(KERN_ERR "asoc: failed to init %s\n",
-					card->dai_link[i].stream_name);
-				continue;
-			}
-		}
 		if (card->dai_link[i].codec_dai->ac97_control) {
-			ac97 = 1;
 			snd_ac97_dev_add_pdata(codec->ac97,
 				card->dai_link[i].cpu_dai->ac97_pdata);
 		}
 	}
-	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
-		 "%s",  card->name);
-	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",
-				codec->name);
-		goto out;
-	}
-
-	mutex_lock(&codec->mutex);
-#ifdef CONFIG_SND_SOC_AC97_BUS
-	/* Only instantiate AC97 if not already done by the adaptor
-	 * for the generic AC97 subsystem.
-	 */
-	if (ac97 && strcmp(codec->name, "AC97") != 0) {
-		ret = soc_ac97_dev_register(codec);
-		if (ret < 0) {
-			printk(KERN_ERR "asoc: AC97 device register failed\n");
-			snd_card_free(codec->card);
-			mutex_unlock(&codec->mutex);
-			goto out;
-		}
-	}
-#endif
-
-	err = snd_soc_dapm_sys_add(socdev->dev);
-	if (err < 0)
-		printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
-
-	err = device_create_file(socdev->dev, &dev_attr_codec_reg);
-	if (err < 0)
-		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
-	soc_init_codec_debugfs(codec);
 	mutex_unlock(&codec->mutex);
-
-out:
 	return ret;
 }
-EXPORT_SYMBOL_GPL(snd_soc_init_card);
+EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
 
 /**
  * snd_soc_free_pcms - free sound card and pcms
@@ -1734,7 +1718,7 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
 		mask |= (bitmask - 1) << e->shift_r;
 	}
 
-	return snd_soc_update_bits(codec, e->reg, mask, val);
+	return snd_soc_update_bits_locked(codec, e->reg, mask, val);
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
 
@@ -1808,7 +1792,7 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
 		mask |= e->mask << e->shift_r;
 	}
 
-	return snd_soc_update_bits(codec, e->reg, mask, val);
+	return snd_soc_update_bits_locked(codec, e->reg, mask, val);
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double);
 
@@ -1969,7 +1953,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 		val_mask |= mask << rshift;
 		val |= val2 << rshift;
 	}
-	return snd_soc_update_bits(codec, reg, val_mask, val);
+	return snd_soc_update_bits_locked(codec, reg, val_mask, val);
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
 
@@ -2075,11 +2059,11 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
 	val = val << shift;
 	val2 = val2 << shift;
 
-	err = snd_soc_update_bits(codec, reg, val_mask, val);
+	err = snd_soc_update_bits_locked(codec, reg, val_mask, val);
 	if (err < 0)
 		return err;
 
-	err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+	err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2);
 	return err;
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);
@@ -2158,7 +2142,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 	val = (ucontrol->value.integer.value[0]+min) & 0xff;
 	val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;
 
-	return snd_soc_update_bits(codec, reg, 0xffff, val);
+	return snd_soc_update_bits_locked(codec, reg, 0xffff, val);
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 
@@ -2205,16 +2189,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
  * snd_soc_dai_set_pll - configure DAI PLL.
  * @dai: DAI
  * @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
  * @freq_in: PLL input clock frequency in Hz
  * @freq_out: requested PLL output clock frequency in Hz
  *
  * Configures and enables PLL to generate output clock based on input clock.
  */
-int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
-	int pll_id, unsigned int freq_in, unsigned int freq_out)
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source,
+	unsigned int freq_in, unsigned int freq_out)
 {
 	if (dai->ops && dai->ops->set_pll)
-		return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
+		return dai->ops->set_pll(dai, pll_id, source,
+					 freq_in, freq_out);
 	else
 		return -EINVAL;
 }
@@ -2259,6 +2245,30 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
 
 /**
+ * snd_soc_dai_set_channel_map - configure DAI audio channel map
+ * @dai: DAI
+ * @tx_num: how many TX channels
+ * @tx_slot: pointer to an array which imply the TX slot number channel
+ *           0~num-1 uses
+ * @rx_num: how many RX channels
+ * @rx_slot: pointer to an array which imply the RX slot number channel
+ *           0~num-1 uses
+ *
+ * configure the relationship between channel number and TDM slot number.
+ */
+int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
+	unsigned int tx_num, unsigned int *tx_slot,
+	unsigned int rx_num, unsigned int *rx_slot)
+{
+	if (dai->ops && dai->ops->set_channel_map)
+		return dai->ops->set_channel_map(dai, tx_num, tx_slot,
+			rx_num, rx_slot);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map);
+
+/**
  * snd_soc_dai_set_tristate - configure DAI system or master clock.
  * @dai: DAI
  * @tristate: tristate enable
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 66d4c165f99b..0d294ef72590 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -719,6 +719,10 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w)
 
 	/* Check if one of our outputs is connected */
 	list_for_each_entry(path, &w->sinks, list_source) {
+		if (path->connected &&
+		    !path->connected(path->source, path->sink))
+			continue;
+
 		if (path->sink && path->sink->power_check &&
 		    path->sink->power_check(path->sink)) {
 			power = 1;
@@ -1152,6 +1156,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 				w->active ? "active" : "inactive");
 
 	list_for_each_entry(p, &w->sources, list_sink) {
+		if (p->connected && !p->connected(w, p->sink))
+			continue;
+
 		if (p->connect)
 			ret += snprintf(buf + ret, PAGE_SIZE - ret,
 					" in  %s %s\n",
@@ -1159,6 +1166,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 					p->source->name);
 	}
 	list_for_each_entry(p, &w->sinks, list_source) {
+		if (p->connected && !p->connected(w, p->sink))
+			continue;
+
 		if (p->connect)
 			ret += snprintf(buf + ret, PAGE_SIZE - ret,
 					" out %s %s\n",
@@ -1206,8 +1216,8 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec)
 
 /* test and update the power status of a mux widget */
 static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
-				 struct snd_kcontrol *kcontrol, int mask,
-				 int mux, int val, struct soc_enum *e)
+				 struct snd_kcontrol *kcontrol, int change,
+				 int mux, struct soc_enum *e)
 {
 	struct snd_soc_dapm_path *path;
 	int found = 0;
@@ -1216,7 +1226,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
 	    widget->id != snd_soc_dapm_value_mux)
 		return -ENODEV;
 
-	if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
+	if (!change)
 		return 0;
 
 	/* find dapm widget path assoc with kcontrol */
@@ -1401,10 +1411,13 @@ int snd_soc_dapm_sync(struct snd_soc_codec *codec)
 EXPORT_SYMBOL_GPL(snd_soc_dapm_sync);
 
 static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
-	const char *sink, const char *control, const char *source)
+				  const struct snd_soc_dapm_route *route)
 {
 	struct snd_soc_dapm_path *path;
 	struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
+	const char *sink = route->sink;
+	const char *control = route->control;
+	const char *source = route->source;
 	int ret = 0;
 
 	/* find src and dest widgets */
@@ -1428,6 +1441,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
 
 	path->source = wsource;
 	path->sink = wsink;
+	path->connected = route->connected;
 	INIT_LIST_HEAD(&path->list);
 	INIT_LIST_HEAD(&path->list_source);
 	INIT_LIST_HEAD(&path->list_sink);
@@ -1528,8 +1542,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
 	int i, ret;
 
 	for (i = 0; i < num; i++) {
-		ret = snd_soc_dapm_add_route(codec, route->sink,
-					     route->control, route->source);
+		ret = snd_soc_dapm_add_route(codec, route);
 		if (ret < 0) {
 			printk(KERN_ERR "Failed to add route %s->%s\n",
 			       route->source,
@@ -1766,7 +1779,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-	unsigned int val, mux;
+	unsigned int val, mux, change;
 	unsigned int mask, bitmask;
 	int ret = 0;
 
@@ -1786,20 +1799,21 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 
 	mutex_lock(&widget->codec->mutex);
 	widget->value = val;
-	dapm_mux_update_power(widget, kcontrol, mask, mux, val, e);
-	if (widget->event) {
-		if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
-			ret = widget->event(widget,
-				kcontrol, SND_SOC_DAPM_PRE_REG);
-			if (ret < 0)
-				goto out;
-		}
-		ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
-		if (widget->event_flags & SND_SOC_DAPM_POST_REG)
-			ret = widget->event(widget,
-				kcontrol, SND_SOC_DAPM_POST_REG);
-	} else
-		ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+	change = snd_soc_test_bits(widget->codec, e->reg, mask, val);
+	dapm_mux_update_power(widget, kcontrol, change, mux, e);
+
+	if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+		ret = widget->event(widget,
+				    kcontrol, SND_SOC_DAPM_PRE_REG);
+		if (ret < 0)
+			goto out;
+	}
+
+	ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+
+	if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+		ret = widget->event(widget,
+				    kcontrol, SND_SOC_DAPM_POST_REG);
 
 out:
 	mutex_unlock(&widget->codec->mutex);
@@ -1808,6 +1822,54 @@ out:
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
 
 /**
+ * snd_soc_dapm_get_enum_virt - Get virtual DAPM mux
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.enumerated.item[0] = widget->value;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt);
+
+/**
+ * snd_soc_dapm_put_enum_virt - Set virtual DAPM mux
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *e =
+		(struct soc_enum *)kcontrol->private_value;
+	int change;
+	int ret = 0;
+
+	if (ucontrol->value.enumerated.item[0] >= e->max)
+		return -EINVAL;
+
+	mutex_lock(&widget->codec->mutex);
+
+	change = widget->value != ucontrol->value.enumerated.item[0];
+	widget->value = ucontrol->value.enumerated.item[0];
+	dapm_mux_update_power(widget, kcontrol, change, widget->value, e);
+
+	mutex_unlock(&widget->codec->mutex);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt);
+
+/**
  * snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get
  *					callback
  * @kcontrol: mixer control
@@ -1865,7 +1927,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
-	unsigned int val, mux;
+	unsigned int val, mux, change;
 	unsigned int mask;
 	int ret = 0;
 
@@ -1883,20 +1945,21 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol,
 
 	mutex_lock(&widget->codec->mutex);
 	widget->value = val;
-	dapm_mux_update_power(widget, kcontrol, mask, mux, val, e);
-	if (widget->event) {
-		if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
-			ret = widget->event(widget,
-				kcontrol, SND_SOC_DAPM_PRE_REG);
-			if (ret < 0)
-				goto out;
-		}
-		ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
-		if (widget->event_flags & SND_SOC_DAPM_POST_REG)
-			ret = widget->event(widget,
-				kcontrol, SND_SOC_DAPM_POST_REG);
-	} else
-		ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+	change = snd_soc_test_bits(widget->codec, e->reg, mask, val);
+	dapm_mux_update_power(widget, kcontrol, change, mux, e);
+
+	if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
+		ret = widget->event(widget,
+				    kcontrol, SND_SOC_DAPM_PRE_REG);
+		if (ret < 0)
+			goto out;
+	}
+
+	ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
+
+	if (widget->event_flags & SND_SOC_DAPM_POST_REG)
+		ret = widget->event(widget,
+				    kcontrol, SND_SOC_DAPM_POST_REG);
 
 out:
 	mutex_unlock(&widget->codec->mutex);
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 1d455ab79490..3c07a94c2e30 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -58,7 +58,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new);
  */
 void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 {
-	struct snd_soc_codec *codec = jack->card->codec;
+	struct snd_soc_codec *codec;
 	struct snd_soc_jack_pin *pin;
 	int enable;
 	int oldstatus;
@@ -67,6 +67,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
 		WARN_ON_ONCE(!jack);
 		return;
 	}
+	codec = jack->card->codec;
 
 	mutex_lock(&codec->mutex);
 
@@ -162,6 +163,9 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
 	else
 		report = 0;
 
+	if (gpio->jack_status_check)
+		report = gpio->jack_status_check();
+
 	snd_soc_jack_report(jack, report, gpio->report);
 }
 
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
new file mode 100644
index 000000000000..1d07b931f3d8
--- /dev/null
+++ b/sound/soc/soc-utils.c
@@ -0,0 +1,74 @@
+/*
+ * soc-util.c  --  ALSA SoC Audio Layer utility functions
+ *
+ * Copyright 2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *         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 as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots)
+{
+	return sample_size * channels * tdm_slots;
+}
+EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size);
+
+int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params)
+{
+	int sample_size;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+	case SNDRV_PCM_FORMAT_S16_BE:
+		sample_size = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+	case SNDRV_PCM_FORMAT_S20_3BE:
+		sample_size = 20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+	case SNDRV_PCM_FORMAT_S24_BE:
+		sample_size = 24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+	case SNDRV_PCM_FORMAT_S32_BE:
+		sample_size = 32;
+		break;
+	default:
+		return -ENOTSUPP;
+	}
+
+	return snd_soc_calc_frame_size(sample_size, params_channels(params),
+				       1);
+}
+EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size);
+
+int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots)
+{
+	return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots);
+}
+EXPORT_SYMBOL_GPL(snd_soc_calc_bclk);
+
+int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
+{
+	int ret;
+
+	ret = snd_soc_params_to_frame_size(params);
+
+	if (ret > 0)
+		return ret * params_rate(params);
+	else
+		return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);