Merge branch 'topic/remove-irqf_disable' into for-linus

author: Takashi Iwai <tiwai@suse.de> 2011-10-26 23:51:48 +0200
committer: Takashi Iwai <tiwai@suse.de> 2011-10-26 23:51:48 +0200
commit: 9430148d800dd929ad73da4c6afb67f793f8af43 (patch)
tree: 437d0aec41428cbb310a202100ba581c63fbe89e /sound
parent: d22665702226e9c40bc331098559e3d55e7cd43d (diff)
parent: 88e24c3a4b30a6bd361f2b5ce602667a8161b2e8 (diff)
123 files changed, 6060 insertions, 1177 deletions
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index d0cead38d5f..e518d38b1c7 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -443,7 +443,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream)
 	mutex_lock(&aaci->irq_lock);
 	if (!aaci->users++) {
 		ret = request_irq(aaci->dev->irq[0], aaci_irq,
-			   IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci);
+			   IRQF_SHARED, DRIVER_NAME, aaci);
 		if (ret != 0)
 			aaci->users--;
 	}
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 88eec3847df..8ad65352bf9 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -359,7 +359,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
 	if (ret)
 		goto err_clk2;
 
-	ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL);
+	ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL);
 	if (ret < 0)
 		goto err_irq;
 
diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c
index 5cfcb908c43..2c7a7636f47 100644
--- a/sound/drivers/ml403-ac97cr.c
+++ b/sound/drivers/ml403-ac97cr.c
@@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
 		   "0x%x done\n", (unsigned int)ml403_ac97cr->port);
 	/* get irq */
 	irq = platform_get_irq(pfdev, 0);
-	if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
+	if (request_irq(irq, snd_ml403_ac97cr_irq, 0,
 			dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
 			   "unable to grab IRQ %d\n",
@@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev,
 		   "request (playback) irq %d done\n",
 		   ml403_ac97cr->irq);
 	irq = platform_get_irq(pfdev, 1);
-	if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED,
+	if (request_irq(irq, snd_ml403_ac97cr_irq, 0,
 			dev_name(&pfdev->dev), (void *)ml403_ac97cr)) {
 		snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": "
 			   "unable to grab IRQ %d\n",
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 9d01c181fec..e91698a634b 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -577,7 +577,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device,
 	else
 		mpu->cport = port + 1;
 	if (irq >= 0) {
-		if (request_irq(irq, snd_mpu401_uart_interrupt, IRQF_DISABLED,
+		if (request_irq(irq, snd_mpu401_uart_interrupt, 0,
 				"MPU401 UART", (void *) mpu)) {
 			snd_printk(KERN_ERR "mpu401_uart: "
 				   "unable to grab IRQ %d\n", irq);
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 5c426df8767..1eef4ccebe4 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -589,7 +589,7 @@ static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard)
 		return -EBUSY;
 	}
 	mcard->port = port;
-	if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) {
+	if (request_irq(irq, snd_mtpav_irqh, 0, "MOTU MTPAV", mcard)) {
 		snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq);
 		return -EBUSY;
 	}
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index a25fb7b1f44..fc1d822802c 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -816,7 +816,7 @@ static int __devinit snd_uart16550_create(struct snd_card *card,
 
 	if (irq >= 0 && irq != SNDRV_AUTO_IRQ) {
 		if (request_irq(irq, snd_uart16550_interrupt,
-				IRQF_DISABLED, "Serial MIDI", uart)) {
+				0, "Serial MIDI", uart)) {
 			snd_printk(KERN_WARNING
 				   "irq %d busy. Using Polling.\n", irq);
 		} else {
diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c
index 05aef8b97e9..177eed3271b 100644
--- a/sound/isa/ad1816a/ad1816a_lib.c
+++ b/sound/isa/ad1816a/ad1816a_lib.c
@@ -595,7 +595,7 @@ int __devinit snd_ad1816a_create(struct snd_card *card,
 		snd_ad1816a_free(chip);
 		return -EBUSY;
 	}
-	if (request_irq(irq, snd_ad1816a_interrupt, IRQF_DISABLED, "AD1816A", (void *) chip)) {
+	if (request_irq(irq, snd_ad1816a_interrupt, 0, "AD1816A", (void *) chip)) {
 		snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq);
 		snd_ad1816a_free(chip);
 		return -EBUSY;
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index 07676200496..d3eab6fb086 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -661,7 +661,7 @@ int snd_es1688_create(struct snd_card *card,
 		snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4);
 		return -EBUSY;
 	}
-	if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) {
+	if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) {
 		snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq);
 		return -EBUSY;
 	}
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index aeee8f8bf5e..bf6ad0bf51c 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -1805,7 +1805,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card,
 		return -EBUSY;
 	}
 
-	if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx",
+	if (request_irq(irq, snd_es18xx_interrupt, 0, "ES18xx",
 			(void *) card)) {
 		snd_es18xx_free(card);
 		snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq);
diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
index 12eb98f2f93..3167e5ac369 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -180,7 +180,7 @@ int snd_gus_create(struct snd_card *card,
 		snd_gus_free(gus);
 		return -EBUSY;
 	}
-	if (irq >= 0 && request_irq(irq, snd_gus_interrupt, IRQF_DISABLED, "GUS GF1", (void *) gus)) {
+	if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) {
 		snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq);
 		snd_gus_free(gus);
 		return -EBUSY;
diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c
index 3e4a58b7291..c43faa057ff 100644
--- a/sound/isa/gus/gusmax.c
+++ b/sound/isa/gus/gusmax.c
@@ -291,7 +291,7 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev)
 		goto _err;
 	}
 
-	if (request_irq(xirq, snd_gusmax_interrupt, IRQF_DISABLED, "GUS MAX", (void *)maxcard)) {
+	if (request_irq(xirq, snd_gusmax_interrupt, 0, "GUS MAX", (void *)maxcard)) {
 		snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
 		err = -EBUSY;
 		goto _err;
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index c7b80e4730f..5f869a32b48 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -684,7 +684,7 @@ static int __devinit snd_interwave_probe(struct snd_card *card, int dev)
 	if ((err = snd_gus_initialize(gus)) < 0)
 		return err;
 
-	if (request_irq(xirq, snd_interwave_interrupt, IRQF_DISABLED,
+	if (request_irq(xirq, snd_interwave_interrupt, 0,
 			"InterWave", iwcard)) {
 		snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq);
 		return -EBUSY;
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index de99f47770b..bbafb0b543e 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -667,7 +667,7 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev)
 	err = snd_opl3sa2_detect(card);
 	if (err < 0)
 		return err;
-	err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED,
+	err = request_irq(xirq, snd_opl3sa2_interrupt, 0,
 			  "OPL3-SA2", card);
 	if (err) {
 		snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq);
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 346e12baa98..6dbbfa76b44 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -892,7 +892,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
 #endif
 #ifdef OPTi93X
 	error = request_irq(irq, snd_opti93x_interrupt,
-			    IRQF_DISABLED, DEV_NAME" - WSS", chip);
+			    0, DEV_NAME" - WSS", chip);
 	if (error < 0) {
 		snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq);
 		return error;
diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c
index eae6c1c0eff..d2e19215813 100644
--- a/sound/isa/sb/sb_common.c
+++ b/sound/isa/sb/sb_common.c
@@ -240,7 +240,7 @@ int snd_sbdsp_create(struct snd_card *card,
 	if (request_irq(irq, irq_handler,
 			(hardware == SB_HW_ALS4000 ||
 			 hardware == SB_HW_CS5530) ?
-			IRQF_SHARED : IRQF_DISABLED,
+			IRQF_SHARED : 0,
 			"SoundBlaster", (void *) chip)) {
 		snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq);
 		snd_sbdsp_free(chip);
diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c
index 83f291d89a9..87142977335 100644
--- a/sound/isa/wavefront/wavefront.c
+++ b/sound/isa/wavefront/wavefront.c
@@ -418,7 +418,7 @@ snd_wavefront_probe (struct snd_card *card, int dev)
 		return -EBUSY;
 	}
 	if (request_irq(ics2115_irq[dev], snd_wavefront_ics2115_interrupt,
-			IRQF_DISABLED, "ICS2115", acard)) {
+			0, "ICS2115", acard)) {
 		snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]);
 		return -EBUSY;
 	}
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 2a42cc37795..7277c5b7df6 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1833,7 +1833,7 @@ int snd_wss_create(struct snd_card *card,
 	}
 	chip->cport = cport;
 	if (!(hwshare & WSS_HWSHARE_IRQ))
-		if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED,
+		if (request_irq(irq, snd_wss_interrupt, 0,
 				"WSS", (void *) chip)) {
 			snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq);
 			snd_wss_free(chip);
diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig
index a9823fad85c..77dd0a13aec 100644
--- a/sound/mips/Kconfig
+++ b/sound/mips/Kconfig
@@ -23,12 +23,15 @@ config SND_SGI_HAL2
 
 
 config SND_AU1X00
-	tristate "Au1x00 AC97 Port Driver"
+	tristate "Au1x00 AC97 Port Driver (DEPRECATED)"
 	depends on SOC_AU1000 || SOC_AU1100 || SOC_AU1500
 	select SND_PCM
 	select SND_AC97_CODEC
 	help
 	  ALSA Sound driver for the Au1x00's AC97 port.
 
+	  Newer drivers for ASoC are available, please do not use
+	  this driver as it will be removed in the future.
+
 endif	# SND_MIPS
 
diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c
index 446cf974866..7567ebd7191 100644
--- a/sound/mips/au1x00.c
+++ b/sound/mips/au1x00.c
@@ -465,13 +465,13 @@ snd_au1000_pcm_new(struct snd_au1000 *au1000)
 
 	flags = claim_dma_lock();
 	if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX,
-			"AC97 TX", au1000_dma_interrupt, IRQF_DISABLED,
+			"AC97 TX", au1000_dma_interrupt, 0,
 			au1000->stream[PLAYBACK])) < 0) {
 		release_dma_lock(flags);
 		return -EBUSY;
 	}
 	if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX,
-			"AC97 RX", au1000_dma_interrupt, IRQF_DISABLED,
+			"AC97 RX", au1000_dma_interrupt, 0,
 			au1000->stream[CAPTURE])) < 0){
 		release_dma_lock(flags);
 		return -EBUSY;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index bcf61524a13..5ffb20b1878 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1234,7 +1234,7 @@ static int sis_resume(struct pci_dev *pci)
 		goto error;
 	}
 
-	if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
+	if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED,
 			KBUILD_MODNAME, sis)) {
 		printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq);
 		goto error;
@@ -1340,7 +1340,7 @@ static int __devinit sis_chip_create(struct snd_card *card,
 	if (rc)
 		goto error_out_cleanup;
 
-	if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED,
+	if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED,
 			KBUILD_MODNAME, sis)) {
 		printk(KERN_ERR "unable to allocate irq %d\n", sis->irq);
 		goto error_out_cleanup;
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index bc823a54755..775bd95d4be 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -845,7 +845,7 @@ static int __devinit snd_ps3_allocate_irq(void)
 		return ret;
 	}
 
-	ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED,
+	ret = request_irq(the_card.irq_no, snd_ps3_interrupt, 0,
 			  SND_PS3_DRIVER_NAME, &the_card);
 	if (ret) {
 		pr_info("%s: request_irq failed (%d)\n", __func__, ret);
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 8224db5f043..1381db853ef 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -7,6 +7,8 @@ menuconfig SND_SOC
 	select SND_PCM
 	select AC97_BUS if SND_SOC_AC97_BUS
 	select SND_JACK if INPUT=y || INPUT=SND
+	select REGMAP_I2C if I2C
+	select REGMAP_SPI if SPI_MASTER
 	---help---
 
 	  If you want ASoC support, you should say Y here and also to the
@@ -51,6 +53,7 @@ source "sound/soc/nuc900/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/kirkwood/Kconfig"
 source "sound/soc/mid-x86/Kconfig"
+source "sound/soc/mxs/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/samsung/Kconfig"
 source "sound/soc/s6000/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 4f913876f33..9ea8ac827ad 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -12,6 +12,7 @@ obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)   += imx/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
 obj-$(CONFIG_SND_SOC)	+= mid-x86/
+obj-$(CONFIG_SND_SOC)	+= mxs/
 obj-$(CONFIG_SND_SOC)	+= nuc900/
 obj-$(CONFIG_SND_SOC)	+= omap/
 obj-$(CONFIG_SND_SOC)	+= kirkwood/
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 1aac2f4dbcf..2909bfaed26 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -383,14 +383,17 @@ static int __init playpaq_asoc_init(void)
 	_gclk0 = clk_get(NULL, "gclk0");
 	if (IS_ERR(_gclk0)) {
 		_gclk0 = NULL;
+		ret = PTR_ERR(_gclk0);
 		goto err_gclk0;
 	}
 	_pll0 = clk_get(NULL, "pll0");
 	if (IS_ERR(_pll0)) {
 		_pll0 = NULL;
+		ret = PTR_ERR(_pll0);
 		goto err_pll0;
 	}
-	if (clk_set_parent(_gclk0, _pll0)) {
+	ret = clk_set_parent(_gclk0, _pll0);
+	if (ret) {
 		pr_warning("snd-soc-playpaq: "
 			   "Failed to set PLL0 as parent for DAC clock\n");
 		goto err_set_clk;
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig
index 4b67140fdec..6d592546e8f 100644
--- a/sound/soc/au1x/Kconfig
+++ b/sound/soc/au1x/Kconfig
@@ -18,10 +18,38 @@ config SND_SOC_AU1XPSC_AC97
 	select SND_AC97_CODEC
 	select SND_SOC_AC97_BUS
 
+##
+## Au1000/1500/1100 DMA + AC97C/I2SC
+##
+config SND_SOC_AU1XAUDIO
+	tristate "SoC Audio for Au1000/Au1500/Au1100"
+	depends on MIPS_ALCHEMY
+	help
+	  This is a driver set for the AC97 unit and the
+	  old DMA controller as found on the Au1000/Au1500/Au1100 chips.
+
+config SND_SOC_AU1XAC97C
+	tristate
+	select AC97_BUS
+	select SND_AC97_CODEC
+	select SND_SOC_AC97_BUS
+
+config SND_SOC_AU1XI2SC
+	tristate
+
 
 ##
 ## Boards
 ##
+config SND_SOC_DB1000
+	tristate "DB1000 Audio support"
+	depends on SND_SOC_AU1XAUDIO
+	select SND_SOC_AU1XAC97C
+	select SND_SOC_AC97_CODEC
+	help
+	  Select this option to enable AC97 audio on the early DB1x00 series
+	  of boards (DB1000/DB1500/DB1100).
+
 config SND_SOC_DB1200
 	tristate "DB1200 AC97+I2S audio support"
 	depends on SND_SOC_AU1XPSC
diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile
index 16873076e8c..920710514ea 100644
--- a/sound/soc/au1x/Makefile
+++ b/sound/soc/au1x/Makefile
@@ -3,11 +3,21 @@ snd-soc-au1xpsc-dbdma-objs := dbdma2.o
 snd-soc-au1xpsc-i2s-objs := psc-i2s.o
 snd-soc-au1xpsc-ac97-objs := psc-ac97.o
 
+# Au1000/1500/1100 Audio units
+snd-soc-au1x-dma-objs := dma.o
+snd-soc-au1x-ac97c-objs := ac97c.o
+snd-soc-au1x-i2sc-objs := i2sc.o
+
 obj-$(CONFIG_SND_SOC_AU1XPSC) += snd-soc-au1xpsc-dbdma.o
 obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o
 obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o
+obj-$(CONFIG_SND_SOC_AU1XAUDIO) += snd-soc-au1x-dma.o
+obj-$(CONFIG_SND_SOC_AU1XAC97C) += snd-soc-au1x-ac97c.o
+obj-$(CONFIG_SND_SOC_AU1XI2SC) += snd-soc-au1x-i2sc.o
 
 # Boards
+snd-soc-db1000-objs := db1000.o
 snd-soc-db1200-objs := db1200.o
 
+obj-$(CONFIG_SND_SOC_DB1000) += snd-soc-db1000.o
 obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c
new file mode 100644
index 00000000000..13802ff7cf0
--- /dev/null
+++ b/sound/soc/au1x/ac97c.c
@@ -0,0 +1,363 @@
+/*
+ * Au1000/Au1500/Au1100 AC97C controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * based on the old ALSA driver originally written by
+ *			Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/mutex.h>
+#include <linux/platform_device.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+/* register offsets and bits */
+#define AC97_CONFIG	0x00
+#define AC97_STATUS	0x04
+#define AC97_DATA	0x08
+#define AC97_CMDRESP	0x0c
+#define AC97_ENABLE	0x10
+
+#define CFG_RC(x)	(((x) & 0x3ff) << 13)	/* valid rx slots mask */
+#define CFG_XS(x)	(((x) & 0x3ff) << 3)	/* valid tx slots mask */
+#define CFG_SG		(1 << 2)	/* sync gate */
+#define CFG_SN		(1 << 1)	/* sync control */
+#define CFG_RS		(1 << 0)	/* acrst# control */
+#define STAT_XU		(1 << 11)	/* tx underflow */
+#define STAT_XO		(1 << 10)	/* tx overflow */
+#define STAT_RU		(1 << 9)	/* rx underflow */
+#define STAT_RO		(1 << 8)	/* rx overflow */
+#define STAT_RD		(1 << 7)	/* codec ready */
+#define STAT_CP		(1 << 6)	/* command pending */
+#define STAT_TE		(1 << 4)	/* tx fifo empty */
+#define STAT_TF		(1 << 3)	/* tx fifo full */
+#define STAT_RE		(1 << 1)	/* rx fifo empty */
+#define STAT_RF		(1 << 0)	/* rx fifo full */
+#define CMD_SET_DATA(x)	(((x) & 0xffff) << 16)
+#define CMD_GET_DATA(x)	((x) & 0xffff)
+#define CMD_READ	(1 << 7)
+#define CMD_WRITE	(0 << 7)
+#define CMD_IDX(x)	((x) & 0x7f)
+#define EN_D		(1 << 1)	/* DISable bit */
+#define EN_CE		(1 << 0)	/* clock enable bit */
+
+/* how often to retry failed codec register reads/writes */
+#define AC97_RW_RETRIES	5
+
+#define AC97_RATES	\
+	SNDRV_PCM_RATE_CONTINUOUS
+
+#define AC97_FMTS	\
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE)
+
+/* instance data. There can be only one, MacLeod!!!!, fortunately there IS only
+ * once AC97C on early Alchemy chips. The newer ones aren't so lucky.
+ */
+static struct au1xpsc_audio_data *ac97c_workdata;
+#define ac97_to_ctx(x)		ac97c_workdata
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+	return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+	__raw_writel(v, ctx->mmio + reg);
+	wmb();
+}
+
+static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97,
+					  unsigned short r)
+{
+	struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+	unsigned int tmo, retry;
+	unsigned long data;
+
+	data = ~0;
+	retry = AC97_RW_RETRIES;
+	do {
+		mutex_lock(&ctx->lock);
+
+		tmo = 5;
+		while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+			udelay(21);	/* wait an ac97 frame time */
+		if (!tmo) {
+			pr_debug("ac97rd timeout #1\n");
+			goto next;
+		}
+
+		WR(ctx, AC97_CMDRESP, CMD_IDX(r) | CMD_READ);
+
+		/* stupid errata: data is only valid for 21us, so
+		 * poll, Forrest, poll...
+		 */
+		tmo = 0x10000;
+		while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--)
+			asm volatile ("nop");
+		data = RD(ctx, AC97_CMDRESP);
+
+		if (!tmo)
+			pr_debug("ac97rd timeout #2\n");
+
+next:
+		mutex_unlock(&ctx->lock);
+	} while (--retry && !tmo);
+
+	pr_debug("AC97RD %04x %04lx %d\n", r, data, retry);
+
+	return retry ? data & 0xffff : 0xffff;
+}
+
+static void au1xac97c_ac97_write(struct snd_ac97 *ac97, unsigned short r,
+				 unsigned short v)
+{
+	struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+	unsigned int tmo, retry;
+
+	retry = AC97_RW_RETRIES;
+	do {
+		mutex_lock(&ctx->lock);
+
+		for (tmo = 5; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+			udelay(21);
+		if (!tmo) {
+			pr_debug("ac97wr timeout #1\n");
+			goto next;
+		}
+
+		WR(ctx, AC97_CMDRESP, CMD_WRITE | CMD_IDX(r) | CMD_SET_DATA(v));
+
+		for (tmo = 10; (RD(ctx, AC97_STATUS) & STAT_CP) && tmo; tmo--)
+			udelay(21);
+		if (!tmo)
+			pr_debug("ac97wr timeout #2\n");
+next:
+		mutex_unlock(&ctx->lock);
+	} while (--retry && !tmo);
+
+	pr_debug("AC97WR %04x %04x %d\n", r, v, retry);
+}
+
+static void au1xac97c_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+	struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+
+	WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG | CFG_SN);
+	msleep(20);
+	WR(ctx, AC97_CONFIG, ctx->cfg | CFG_SG);
+	WR(ctx, AC97_CONFIG, ctx->cfg);
+}
+
+static void au1xac97c_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+	struct au1xpsc_audio_data *ctx = ac97_to_ctx(ac97);
+	int i;
+
+	WR(ctx, AC97_CONFIG, ctx->cfg | CFG_RS);
+	msleep(500);
+	WR(ctx, AC97_CONFIG, ctx->cfg);
+
+	/* wait for codec ready */
+	i = 50;
+	while (((RD(ctx, AC97_STATUS) & STAT_RD) == 0) && --i)
+		msleep(20);
+	if (!i)
+		printk(KERN_ERR "ac97c: codec not ready after cold reset\n");
+}
+
+/* AC97 controller operations */
+struct snd_ac97_bus_ops soc_ac97_ops = {
+	.read		= au1xac97c_ac97_read,
+	.write		= au1xac97c_ac97_write,
+	.reset		= au1xac97c_ac97_cold_reset,
+	.warm_reset	= au1xac97c_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);	/* globals be gone! */
+
+static int alchemy_ac97c_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+	snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+	return 0;
+}
+
+static struct snd_soc_dai_ops alchemy_ac97c_ops = {
+	.startup		= alchemy_ac97c_startup,
+};
+
+static int au1xac97c_dai_probe(struct snd_soc_dai *dai)
+{
+	return ac97c_workdata ? 0 : -ENODEV;
+}
+
+static struct snd_soc_dai_driver au1xac97c_dai_driver = {
+	.name			= "alchemy-ac97c",
+	.ac97_control		= 1,
+	.probe			= au1xac97c_dai_probe,
+	.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			= &alchemy_ac97c_ops,
+};
+
+static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
+{
+	int ret;
+	struct resource *r;
+	struct au1xpsc_audio_data *ctx;
+
+	ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+	if (!ctx)
+		return -ENOMEM;
+
+	mutex_init(&ctx->lock);
+
+	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!r) {
+		ret = -ENODEV;
+		goto out0;
+	}
+
+	ret = -EBUSY;
+	if (!request_mem_region(r->start, resource_size(r), pdev->name))
+		goto out0;
+
+	ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+	if (!ctx->mmio)
+		goto out1;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!r)
+		goto out1;
+	ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!r)
+		goto out1;
+	ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+	/* switch it on */
+	WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+	WR(ctx, AC97_ENABLE, EN_CE);
+
+	ctx->cfg = CFG_RC(3) | CFG_XS(3);
+	WR(ctx, AC97_CONFIG, ctx->cfg);
+
+	platform_set_drvdata(pdev, ctx);
+
+	ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver);
+	if (ret)
+		goto out1;
+
+	ac97c_workdata = ctx;
+	return 0;
+
+out1:
+	release_mem_region(r->start, resource_size(r));
+out0:
+	kfree(ctx);
+	return ret;
+}
+
+static int __devexit au1xac97c_drvremove(struct platform_device *pdev)
+{
+	struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+	struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+	snd_soc_unregister_dai(&pdev->dev);
+
+	WR(ctx, AC97_ENABLE, EN_D);	/* clock off, disable */
+
+	iounmap(ctx->mmio);
+	release_mem_region(r->start, resource_size(r));
+	kfree(ctx);
+
+	ac97c_workdata = NULL;	/* MDEV */
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xac97c_drvsuspend(struct device *dev)
+{
+	struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+	WR(ctx, AC97_ENABLE, EN_D);	/* clock off, disable */
+
+	return 0;
+}
+
+static int au1xac97c_drvresume(struct device *dev)
+{
+	struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+	WR(ctx, AC97_ENABLE, EN_D | EN_CE);
+	WR(ctx, AC97_ENABLE, EN_CE);
+	WR(ctx, AC97_CONFIG, ctx->cfg);
+
+	return 0;
+}
+
+static const struct dev_pm_ops au1xpscac97_pmops = {
+	.suspend	= au1xac97c_drvsuspend,
+	.resume		= au1xac97c_drvresume,
+};
+
+#define AU1XPSCAC97_PMOPS (&au1xpscac97_pmops)
+
+#else
+
+#define AU1XPSCAC97_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xac97c_driver = {
+	.driver	= {
+		.name	= "alchemy-ac97c",
+		.owner	= THIS_MODULE,
+		.pm	= AU1XPSCAC97_PMOPS,
+	},
+	.probe		= au1xac97c_drvprobe,
+	.remove		= __devexit_p(au1xac97c_drvremove),
+};
+
+static int __init au1xac97c_load(void)
+{
+	ac97c_workdata = NULL;
+	return platform_driver_register(&au1xac97c_driver);
+}
+
+static void __exit au1xac97c_unload(void)
+{
+	platform_driver_unregister(&au1xac97c_driver);
+}
+
+module_init(au1xac97c_load);
+module_exit(au1xac97c_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
new file mode 100644
index 00000000000..127477a5e0c
--- /dev/null
+++ b/sound/soc/au1x/db1000.c
@@ -0,0 +1,75 @@
+/*
+ * DB1000/DB1500/DB1100 ASoC audio fabric support code.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-db1x00/bcsr.h>
+
+#include "psc.h"
+
+static struct snd_soc_dai_link db1000_ac97_dai = {
+	.name		= "AC97",
+	.stream_name	= "AC97 HiFi",
+	.codec_dai_name	= "ac97-hifi",
+	.cpu_dai_name	= "alchemy-ac97c",
+	.platform_name	= "alchemy-pcm-dma.0",
+	.codec_name	= "ac97-codec",
+};
+
+static struct snd_soc_card db1000_ac97 = {
+	.name		= "DB1000_AC97",
+	.dai_link	= &db1000_ac97_dai,
+	.num_links	= 1,
+};
+
+static int __devinit db1000_audio_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &db1000_ac97;
+	card->dev = &pdev->dev;
+	return snd_soc_register_card(card);
+}
+
+static int __devexit db1000_audio_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	snd_soc_unregister_card(card);
+	return 0;
+}
+
+static struct platform_driver db1000_audio_driver = {
+	.driver	= {
+		.name	= "db1000-audio",
+		.owner	= THIS_MODULE,
+		.pm	= &snd_soc_pm_ops,
+	},
+	.probe		= db1000_audio_probe,
+	.remove		= __devexit_p(db1000_audio_remove),
+};
+
+static int __init db1000_audio_load(void)
+{
+	return platform_driver_register(&db1000_audio_driver);
+}
+
+static void __exit db1000_audio_unload(void)
+{
+	platform_driver_unregister(&db1000_audio_driver);
+}
+
+module_init(db1000_audio_load);
+module_exit(db1000_audio_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("DB1000/DB1500/DB1100 ASoC audio");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 1d3e258c9ea..289312c14b9 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -1,7 +1,7 @@
 /*
  * DB1200 ASoC audio fabric support code.
  *
- * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com>
+ * (c) 2008-2011 Manuel Lauss <manuel.lauss@googlemail.com>
  *
  */
 
@@ -21,6 +21,17 @@
 #include "../codecs/wm8731.h"
 #include "psc.h"
 
+static struct platform_device_id db1200_pids[] = {
+	{
+		.name		= "db1200-ac97",
+		.driver_data	= 0,
+	}, {
+		.name		= "db1200-i2s",
+		.driver_data	= 1,
+	},
+	{},
+};
+
 /*-------------------------  AC97 PART  ---------------------------*/
 
 static struct snd_soc_dai_link db1200_ac97_dai = {
@@ -89,36 +100,47 @@ static struct snd_soc_card db1200_i2s_machine = {
 
 /*-------------------------  COMMON PART  ---------------------------*/
 
-static struct platform_device *db1200_asoc_dev;
+static struct snd_soc_card *db1200_cards[] __devinitdata = {
+	&db1200_ac97_machine,
+	&db1200_i2s_machine,
+};
 
-static int __init db1200_audio_load(void)
+static int __devinit db1200_audio_probe(struct platform_device *pdev)
 {
-	int ret;
+	const struct platform_device_id *pid = platform_get_device_id(pdev);
+	struct snd_soc_card *card;
 
-	ret = -ENOMEM;
-	db1200_asoc_dev = platform_device_alloc("soc-audio", 1); /* PSC1 */
-	if (!db1200_asoc_dev)
-		goto out;
+	card = db1200_cards[pid->driver_data];
+	card->dev = &pdev->dev;
+	return snd_soc_register_card(card);
+}
 
-	/* DB1200 board setup set PSC1MUX to preferred audio device */
-	if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX)
-		platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_machine);
-	else
-		platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_machine);
+static int __devexit db1200_audio_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	snd_soc_unregister_card(card);
+	return 0;
+}
 
-	ret = platform_device_add(db1200_asoc_dev);
+static struct platform_driver db1200_audio_driver = {
+	.driver	= {
+		.name	= "db1200-ac97",
+		.owner	= THIS_MODULE,
+		.pm	= &snd_soc_pm_ops,
+	},
+	.id_table	= db1200_pids,
+	.probe		= db1200_audio_probe,
+	.remove		= __devexit_p(db1200_audio_remove),
+};
 
-	if (ret) {
-		platform_device_put(db1200_asoc_dev);
-		db1200_asoc_dev = NULL;
-	}
-out:
-	return ret;
+static int __init db1200_audio_load(void)
+{
+	return platform_driver_register(&db1200_audio_driver);
 }
 
 static void __exit db1200_audio_unload(void)
 {
-	platform_device_unregister(db1200_asoc_dev);
+	platform_driver_unregister(&db1200_audio_driver);
 }
 
 module_init(db1200_audio_load);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 20bb53a837b..d7d04e26eee 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -169,7 +169,7 @@ static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
 
 	au1x_pcm_dbdma_free(pcd);
 
-	if (stype == PCM_RX)
+	if (stype == SNDRV_PCM_STREAM_CAPTURE)
 		pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id,
 					DSCR_CMD0_ALWAYS,
 					au1x_pcm_dmarx_cb, (void *)pcd);
@@ -198,7 +198,7 @@ static inline struct au1xpsc_audio_dmadata *to_dmadata(struct snd_pcm_substream
 	struct snd_soc_pcm_runtime *rtd = ss->private_data;
 	struct au1xpsc_audio_dmadata *pcd =
 				snd_soc_platform_get_drvdata(rtd->platform);
-	return &pcd[SUBSTREAM_TYPE(ss)];
+	return &pcd[ss->stream];
 }
 
 static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
@@ -212,7 +212,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		goto out;
 
-	stype = SUBSTREAM_TYPE(substream);
+	stype = substream->stream;
 	pcd = to_dmadata(substream);
 
 	DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d "
@@ -255,7 +255,7 @@ static int au1xpsc_pcm_prepare(struct snd_pcm_substream *substream)
 
 	au1xxx_dbdma_reset(pcd->ddma_chan);
 
-	if (SUBSTREAM_TYPE(substream) == PCM_RX) {
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
 		au1x_pcm_queue_rx(pcd);
 		au1x_pcm_queue_rx(pcd);
 	} else {
@@ -293,6 +293,16 @@ au1xpsc_pcm_pointer(struct snd_pcm_substream *substream)
 
 static int au1xpsc_pcm_open(struct snd_pcm_substream *substream)
 {
+	struct au1xpsc_audio_dmadata *pcd = to_dmadata(substream);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int stype = substream->stream, *dmaids;
+
+	dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	if (!dmaids)
+		return -ENODEV;	/* whoa, has ordering changed? */
+
+	pcd->ddma_id = dmaids[stype];
+
 	snd_soc_set_runtime_hwparams(substream, &au1xpsc_pcm_hardware);
 	return 0;
 }
@@ -340,36 +350,18 @@ struct snd_soc_platform_driver au1xpsc_soc_platform = {
 static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev)
 {
 	struct au1xpsc_audio_dmadata *dmadata;
-	struct resource *r;
 	int ret;
 
 	dmadata = kzalloc(2 * sizeof(struct au1xpsc_audio_dmadata), GFP_KERNEL);
 	if (!dmadata)
 		return -ENOMEM;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r) {
-		ret = -ENODEV;
-		goto out1;
-	}
-	dmadata[PCM_TX].ddma_id = r->start;
-
-	/* RX DMA */
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!r) {
-		ret = -ENODEV;
-		goto out1;
-	}
-	dmadata[PCM_RX].ddma_id = r->start;
-
 	platform_set_drvdata(pdev, dmadata);
 
 	ret = snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform);
-	if (!ret)
-		return ret;
+	if (ret)
+		kfree(dmadata);
 
-out1:
-	kfree(dmadata);
 	return ret;
 }
 
@@ -405,57 +397,6 @@ static void __exit au1xpsc_audio_dbdma_unload(void)
 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", pdev->id);
-	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");
diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c
new file mode 100644
index 00000000000..177f7137a9c
--- /dev/null
+++ b/sound/soc/au1x/dma.c
@@ -0,0 +1,377 @@
+/*
+ * Au1000/Au1500/Au1100 Audio DMA support.
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * copied almost verbatim from the old ALSA driver, written by
+ *			Charles Eidsness <charles@cooper-street.com>
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
+
+#include "psc.h"
+
+#define ALCHEMY_PCM_FMTS					\
+	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_U8 |	\
+	 SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |	\
+	 SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |	\
+	 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE |	\
+	 SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE |	\
+	 0)
+
+struct pcm_period {
+	u32 start;
+	u32 relative_end;	/* relative to start of buffer */
+	struct pcm_period *next;
+};
+
+struct audio_stream {
+	struct snd_pcm_substream *substream;
+	int dma;
+	struct pcm_period *buffer;
+	unsigned int period_size;
+	unsigned int periods;
+};
+
+struct alchemy_pcm_ctx {
+	struct audio_stream stream[2];	/* playback & capture */
+};
+
+static void au1000_release_dma_link(struct audio_stream *stream)
+{
+	struct pcm_period *pointer;
+	struct pcm_period *pointer_next;
+
+	stream->period_size = 0;
+	stream->periods = 0;
+	pointer = stream->buffer;
+	if (!pointer)
+		return;
+	do {
+		pointer_next = pointer->next;
+		kfree(pointer);
+		pointer = pointer_next;
+	} while (pointer != stream->buffer);
+	stream->buffer = NULL;
+}
+
+static int au1000_setup_dma_link(struct audio_stream *stream,
+				 unsigned int period_bytes,
+				 unsigned int periods)
+{
+	struct snd_pcm_substream *substream = stream->substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct pcm_period *pointer;
+	unsigned long dma_start;
+	int i;
+
+	dma_start = virt_to_phys(runtime->dma_area);
+
+	if (stream->period_size == period_bytes &&
+	    stream->periods == periods)
+		return 0; /* not changed */
+
+	au1000_release_dma_link(stream);
+
+	stream->period_size = period_bytes;
+	stream->periods = periods;
+
+	stream->buffer = kmalloc(sizeof(struct pcm_period), GFP_KERNEL);
+	if (!stream->buffer)
+		return -ENOMEM;
+	pointer = stream->buffer;
+	for (i = 0; i < periods; i++) {
+		pointer->start = (u32)(dma_start + (i * period_bytes));
+		pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1);
+		if (i < periods - 1) {
+			pointer->next = kmalloc(sizeof(struct pcm_period),
+						GFP_KERNEL);
+			if (!pointer->next) {
+				au1000_release_dma_link(stream);
+				return -ENOMEM;
+			}
+			pointer = pointer->next;
+		}
+	}
+	pointer->next = stream->buffer;
+	return 0;
+}
+
+static void au1000_dma_stop(struct audio_stream *stream)
+{
+	if (stream->buffer)
+		disable_dma(stream->dma);
+}
+
+static void au1000_dma_start(struct audio_stream *stream)
+{
+	if (!stream->buffer)
+		return;
+
+	init_dma(stream->dma);
+	if (get_dma_active_buffer(stream->dma) == 0) {
+		clear_dma_done0(stream->dma);
+		set_dma_addr0(stream->dma, stream->buffer->start);
+		set_dma_count0(stream->dma, stream->period_size >> 1);
+		set_dma_addr1(stream->dma, stream->buffer->next->start);
+		set_dma_count1(stream->dma, stream->period_size >> 1);
+	} else {
+		clear_dma_done1(stream->dma);
+		set_dma_addr1(stream->dma, stream->buffer->start);
+		set_dma_count1(stream->dma, stream->period_size >> 1);
+		set_dma_addr0(stream->dma, stream->buffer->next->start);
+		set_dma_count0(stream->dma, stream->period_size >> 1);
+	}
+	enable_dma_buffers(stream->dma);
+	start_dma(stream->dma);
+}
+
+static irqreturn_t au1000_dma_interrupt(int irq, void *ptr)
+{
+	struct audio_stream *stream = (struct audio_stream *)ptr;
+	struct snd_pcm_substream *substream = stream->substream;
+
+	switch (get_dma_buffer_done(stream->dma)) {
+	case DMA_D0:
+		stream->buffer = stream->buffer->next;
+		clear_dma_done0(stream->dma);
+		set_dma_addr0(stream->dma, stream->buffer->next->start);
+		set_dma_count0(stream->dma, stream->period_size >> 1);
+		enable_dma_buffer0(stream->dma);
+		break;
+	case DMA_D1:
+		stream->buffer = stream->buffer->next;
+		clear_dma_done1(stream->dma);
+		set_dma_addr1(stream->dma, stream->buffer->next->start);
+		set_dma_count1(stream->dma, stream->period_size >> 1);
+		enable_dma_buffer1(stream->dma);
+		break;
+	case (DMA_D0 | DMA_D1):
+		pr_debug("DMA %d missed interrupt.\n", stream->dma);
+		au1000_dma_stop(stream);
+		au1000_dma_start(stream);
+		break;
+	case (~DMA_D0 & ~DMA_D1):
+		pr_debug("DMA %d empty irq.\n", stream->dma);
+	}
+	snd_pcm_period_elapsed(substream);
+	return IRQ_HANDLED;
+}
+
+static const struct snd_pcm_hardware alchemy_pcm_hardware = {
+	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+			    SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
+	.formats	  = ALCHEMY_PCM_FMTS,
+	.rates		  = SNDRV_PCM_RATE_8000_192000,
+	.rate_min	  = SNDRV_PCM_RATE_8000,
+	.rate_max	  = SNDRV_PCM_RATE_192000,
+	.channels_min	  = 2,
+	.channels_max	  = 2,
+	.period_bytes_min = 1024,
+	.period_bytes_max = 16 * 1024 - 1,
+	.periods_min	  = 4,
+	.periods_max	  = 255,
+	.buffer_bytes_max = 128 * 1024,
+	.fifo_size	  = 16,
+};
+
+static inline struct alchemy_pcm_ctx *ss_to_ctx(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_pcm_runtime *rtd = ss->private_data;
+	return snd_soc_platform_get_drvdata(rtd->platform);
+}
+
+static inline struct audio_stream *ss_to_as(struct snd_pcm_substream *ss)
+{
+	struct alchemy_pcm_ctx *ctx = ss_to_ctx(ss);
+	return &(ctx->stream[ss->stream]);
+}
+
+static int alchemy_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int *dmaids, s = substream->stream;
+	char *name;
+
+	dmaids = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	if (!dmaids)
+		return -ENODEV;	/* whoa, has ordering changed? */
+
+	/* DMA setup */
+	name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx";
+	ctx->stream[s].dma = request_au1000_dma(dmaids[s], name,
+					au1000_dma_interrupt, 0,
+					&ctx->stream[s]);
+	set_dma_mode(ctx->stream[s].dma,
+		     get_dma_mode(ctx->stream[s].dma) & ~DMA_NC);
+
+	ctx->stream[s].substream = substream;
+	ctx->stream[s].buffer = NULL;
+	snd_soc_set_runtime_hwparams(substream, &alchemy_pcm_hardware);
+
+	return 0;
+}
+
+static int alchemy_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct alchemy_pcm_ctx *ctx = ss_to_ctx(substream);
+	int stype = substream->stream;
+
+	ctx->stream[stype].substream = NULL;
+	free_au1000_dma(ctx->stream[stype].dma);
+
+	return 0;
+}
+
+static int alchemy_pcm_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *hw_params)
+{
+	struct audio_stream *stream = ss_to_as(substream);
+	int err;
+
+	err = snd_pcm_lib_malloc_pages(substream,
+				       params_buffer_bytes(hw_params));
+	if (err < 0)
+		return err;
+	err = au1000_setup_dma_link(stream,
+				    params_period_bytes(hw_params),
+				    params_periods(hw_params));
+	if (err)
+		snd_pcm_lib_free_pages(substream);
+
+	return err;
+}
+
+static int alchemy_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct audio_stream *stream = ss_to_as(substream);
+	au1000_release_dma_link(stream);
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int alchemy_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct audio_stream *stream = ss_to_as(substream);
+	int err = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		au1000_dma_start(stream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		au1000_dma_stop(stream);
+		break;
+	default:
+		err = -EINVAL;
+		break;
+	}
+	return err;
+}
+
+static snd_pcm_uframes_t alchemy_pcm_pointer(struct snd_pcm_substream *ss)
+{
+	struct audio_stream *stream = ss_to_as(ss);
+	long location;
+
+	location = get_dma_residue(stream->dma);
+	location = stream->buffer->relative_end - location;
+	if (location == -1)
+		location = 0;
+	return bytes_to_frames(ss->runtime, location);
+}
+
+static struct snd_pcm_ops alchemy_pcm_ops = {
+	.open			= alchemy_pcm_open,
+	.close			= alchemy_pcm_close,
+	.ioctl			= snd_pcm_lib_ioctl,
+	.hw_params	        = alchemy_pcm_hw_params,
+	.hw_free	        = alchemy_pcm_hw_free,
+	.trigger		= alchemy_pcm_trigger,
+	.pointer		= alchemy_pcm_pointer,
+};
+
+static void alchemy_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	snd_pcm_lib_preallocate_free_for_all(pcm);
+}
+
+static int alchemy_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm *pcm = rtd->pcm;
+
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+		snd_dma_continuous_data(GFP_KERNEL), 65536, (4096 * 1024) - 1);
+
+	return 0;
+}
+
+struct snd_soc_platform_driver alchemy_pcm_soc_platform = {
+	.ops		= &alchemy_pcm_ops,
+	.pcm_new	= alchemy_pcm_new,
+	.pcm_free	= alchemy_pcm_free_dma_buffers,
+};
+
+static int __devinit alchemy_pcm_drvprobe(struct platform_device *pdev)
+{
+	struct alchemy_pcm_ctx *ctx;
+	int ret;
+
+	ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+	if (!ctx)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, ctx);
+
+	ret = snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform);
+	if (ret)
+		kfree(ctx);
+
+	return ret;
+}
+
+static int __devexit alchemy_pcm_drvremove(struct platform_device *pdev)
+{
+	struct alchemy_pcm_ctx *ctx = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_platform(&pdev->dev);
+	kfree(ctx);
+
+	return 0;
+}
+
+static struct platform_driver alchemy_pcmdma_driver = {
+	.driver	= {
+		.name	= "alchemy-pcm-dma",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= alchemy_pcm_drvprobe,
+	.remove		= __devexit_p(alchemy_pcm_drvremove),
+};
+
+static int __init alchemy_pcmdma_load(void)
+{
+	return platform_driver_register(&alchemy_pcmdma_driver);
+}
+
+static void __exit alchemy_pcmdma_unload(void)
+{
+	platform_driver_unregister(&alchemy_pcmdma_driver);
+}
+
+module_init(alchemy_pcmdma_load);
+module_exit(alchemy_pcmdma_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/Au1500/Au1100 Audio DMA driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c
new file mode 100644
index 00000000000..19e0d2a9c82
--- /dev/null
+++ b/sound/soc/au1x/i2sc.c
@@ -0,0 +1,346 @@
+/*
+ * Au1000/Au1500/Au1100 I2S controller driver for ASoC
+ *
+ * (c) 2011 Manuel Lauss <manuel.lauss@googlemail.com>
+ *
+ * Note: clock supplied to the I2S controller must be 256x samplerate.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/suspend.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <asm/mach-au1x00/au1000.h>
+
+#include "psc.h"
+
+#define I2S_RXTX	0x00
+#define I2S_CFG		0x04
+#define I2S_ENABLE	0x08
+
+#define CFG_XU		(1 << 25)	/* tx underflow */
+#define CFG_XO		(1 << 24)
+#define CFG_RU		(1 << 23)
+#define CFG_RO		(1 << 22)
+#define CFG_TR		(1 << 21)
+#define CFG_TE		(1 << 20)
+#define CFG_TF		(1 << 19)
+#define CFG_RR		(1 << 18)
+#define CFG_RF		(1 << 17)
+#define CFG_ICK		(1 << 12)	/* clock invert */
+#define CFG_PD		(1 << 11)	/* set to make I2SDIO INPUT */
+#define CFG_LB		(1 << 10)	/* loopback */
+#define CFG_IC		(1 << 9)	/* word select invert */
+#define CFG_FM_I2S	(0 << 7)	/* I2S format */
+#define CFG_FM_LJ	(1 << 7)	/* left-justified */
+#define CFG_FM_RJ	(2 << 7)	/* right-justified */
+#define CFG_FM_MASK	(3 << 7)
+#define CFG_TN		(1 << 6)	/* tx fifo en */
+#define CFG_RN		(1 << 5)	/* rx fifo en */
+#define CFG_SZ_8	(0x08)
+#define CFG_SZ_16	(0x10)
+#define CFG_SZ_18	(0x12)
+#define CFG_SZ_20	(0x14)
+#define CFG_SZ_24	(0x18)
+#define CFG_SZ_MASK	(0x1f)
+#define EN_D		(1 << 1)	/* DISable */
+#define EN_CE		(1 << 0)	/* clock enable */
+
+/* only limited by clock generator and board design */
+#define AU1XI2SC_RATES \
+	SNDRV_PCM_RATE_CONTINUOUS
+
+#define AU1XI2SC_FMTS \
+	(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |		\
+	SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |	\
+	SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE |	\
+	SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE |	\
+	SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE |	\
+	SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE |	\
+	SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE |	\
+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |	\
+	SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE |	\
+	0)
+
+static inline unsigned long RD(struct au1xpsc_audio_data *ctx, int reg)
+{
+	return __raw_readl(ctx->mmio + reg);
+}
+
+static inline void WR(struct au1xpsc_audio_data *ctx, int reg, unsigned long v)
+{
+	__raw_writel(v, ctx->mmio + reg);
+	wmb();
+}
+
+static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned long c;
+	int ret;
+
+	ret = -EINVAL;
+	c = ctx->cfg;
+
+	c &= ~CFG_FM_MASK;
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		c |= CFG_FM_I2S;
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		c |= CFG_FM_RJ;
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		c |= CFG_FM_LJ;
+		break;
+	default:
+		goto out;
+	}
+
+	c &= ~(CFG_IC | CFG_ICK);		/* IB-IF */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		c |= CFG_IC | CFG_ICK;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		c |= CFG_IC;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		c |= CFG_ICK;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		break;
+	default:
+		goto out;
+	}
+
+	/* I2S controller only supports master */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:	/* CODEC slave */
+		break;
+	default:
+		goto out;
+	}
+
+	ret = 0;
+	ctx->cfg = c;
+out:
+	return ret;
+}
+
+static int au1xi2s_trigger(struct snd_pcm_substream *substream,
+			   int cmd, struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+	int stype = SUBSTREAM_TYPE(substream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		/* power up */
+		WR(ctx, I2S_ENABLE, EN_D | EN_CE);
+		WR(ctx, I2S_ENABLE, EN_CE);
+		ctx->cfg |= (stype == PCM_TX) ? CFG_TN : CFG_RN;
+		WR(ctx, I2S_CFG, ctx->cfg);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		ctx->cfg &= ~((stype == PCM_TX) ? CFG_TN : CFG_RN);
+		WR(ctx, I2S_CFG, ctx->cfg);
+		WR(ctx, I2S_ENABLE, EN_D);		/* power off */
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static unsigned long msbits_to_reg(int msbits)
+{
+	switch (msbits) {
+	case 8:
+		return CFG_SZ_8;
+	case 16:
+		return CFG_SZ_16;
+	case 18:
+		return CFG_SZ_18;
+	case 20:
+		return CFG_SZ_20;
+	case 24:
+		return CFG_SZ_24;
+	}
+	return 0;
+}
+
+static int au1xi2s_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+	unsigned long v;
+
+	v = msbits_to_reg(params->msbits);
+	if (!v)
+		return -EINVAL;
+
+	ctx->cfg &= ~CFG_SZ_MASK;
+	ctx->cfg |= v;
+	return 0;
+}
+
+static int au1xi2s_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *ctx = snd_soc_dai_get_drvdata(dai);
+	snd_soc_dai_set_dma_data(dai, substream, &ctx->dmaids[0]);
+	return 0;
+}
+
+static const struct snd_soc_dai_ops au1xi2s_dai_ops = {
+	.startup	= au1xi2s_startup,
+	.trigger	= au1xi2s_trigger,
+	.hw_params	= au1xi2s_hw_params,
+	.set_fmt	= au1xi2s_set_fmt,
+};
+
+static struct snd_soc_dai_driver au1xi2s_dai_driver = {
+	.symmetric_rates	= 1,
+	.playback = {
+		.rates		= AU1XI2SC_RATES,
+		.formats	= AU1XI2SC_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 2,
+	},
+	.capture = {
+		.rates		= AU1XI2SC_RATES,
+		.formats	= AU1XI2SC_FMTS,
+		.channels_min	= 2,
+		.channels_max	= 2,
+	},
+	.ops = &au1xi2s_dai_ops,
+};
+
+static int __devinit au1xi2s_drvprobe(struct platform_device *pdev)
+{
+	int ret;
+	struct resource *r;
+	struct au1xpsc_audio_data *ctx;
+
+	ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
+	if (!ctx)
+		return -ENOMEM;
+
+	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!r) {
+		ret = -ENODEV;
+		goto out0;
+	}
+
+	ret = -EBUSY;
+	if (!request_mem_region(r->start, resource_size(r), pdev->name))
+		goto out0;
+
+	ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+	if (!ctx->mmio)
+		goto out1;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!r)
+		goto out1;
+	ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!r)
+		goto out1;
+	ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
+	platform_set_drvdata(pdev, ctx);
+
+	ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver);
+	if (ret)
+		goto out1;
+
+	return 0;
+
+out1:
+	release_mem_region(r->start, resource_size(r));
+out0:
+	kfree(ctx);
+	return ret;
+}
+
+static int __devexit au1xi2s_drvremove(struct platform_device *pdev)
+{
+	struct au1xpsc_audio_data *ctx = platform_get_drvdata(pdev);
+	struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+	snd_soc_unregister_dai(&pdev->dev);
+
+	WR(ctx, I2S_ENABLE, EN_D);	/* clock off, disable */
+
+	iounmap(ctx->mmio);
+	release_mem_region(r->start, resource_size(r));
+	kfree(ctx);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int au1xi2s_drvsuspend(struct device *dev)
+{
+	struct au1xpsc_audio_data *ctx = dev_get_drvdata(dev);
+
+	WR(ctx, I2S_ENABLE, EN_D);	/* clock off, disable */
+
+	return 0;
+}
+
+static int au1xi2s_drvresume(struct device *dev)
+{
+	return 0;
+}
+
+static const struct dev_pm_ops au1xi2sc_pmops = {
+	.suspend	= au1xi2s_drvsuspend,
+	.resume		= au1xi2s_drvresume,
+};
+
+#define AU1XI2SC_PMOPS (&au1xi2sc_pmops)
+
+#else
+
+#define AU1XI2SC_PMOPS NULL
+
+#endif
+
+static struct platform_driver au1xi2s_driver = {
+	.driver	= {
+		.name	= "alchemy-i2sc",
+		.owner	= THIS_MODULE,
+		.pm	= AU1XI2SC_PMOPS,
+	},
+	.probe		= au1xi2s_drvprobe,
+	.remove		= __devexit_p(au1xi2s_drvremove),
+};
+
+static int __init au1xi2s_load(void)
+{
+	return platform_driver_register(&au1xi2s_driver);
+}
+
+static void __exit au1xi2s_unload(void)
+{
+	platform_driver_unregister(&au1xi2s_driver);
+}
+
+module_init(au1xi2s_load);
+module_exit(au1xi2s_unload);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Au1000/1500/1100 I2S ASoC driver");
+MODULE_AUTHOR("Manuel Lauss");
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index d0db66f24a0..172eefd38b2 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -41,14 +41,14 @@
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
 
 #define AC97PCR_START(stype)	\
-	((stype) == PCM_TX ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS)
 #define AC97PCR_STOP(stype)	\
-	((stype) == PCM_TX ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP)
 #define AC97PCR_CLRFIFO(stype)	\
-	((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
 
 #define AC97STAT_BUSY(stype)	\
-	((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
 
 /* instance data. There can be only one, MacLeod!!!! */
 static struct au1xpsc_audio_data *au1xpsc_ac97_workdata;
@@ -215,7 +215,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 {
 	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
 	unsigned long r, ro, stat;
-	int chans, t, stype = SUBSTREAM_TYPE(substream);
+	int chans, t, stype = substream->stream;
 
 	chans = params_channels(params);
 
@@ -235,7 +235,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 		r |= PSC_AC97CFG_SET_LEN(params->msbits);
 
 		/* channels: enable slots for front L/R channel */
-		if (stype == PCM_TX) {
+		if (stype == SNDRV_PCM_STREAM_PLAYBACK) {
 			r &= ~PSC_AC97CFG_TXSLOT_MASK;
 			r |= PSC_AC97CFG_TXSLOT_ENA(3);
 			r |= PSC_AC97CFG_TXSLOT_ENA(4);
@@ -294,7 +294,7 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
 				int cmd, struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
-	int ret, stype = SUBSTREAM_TYPE(substream);
+	int ret, stype = substream->stream;
 
 	ret = 0;
 
@@ -324,12 +324,21 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int au1xpsc_ac97_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+	snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+	return 0;
+}
+
 static int au1xpsc_ac97_probe(struct snd_soc_dai *dai)
 {
 	return au1xpsc_ac97_workdata ? 0 : -ENODEV;
 }
 
 static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.startup	= au1xpsc_ac97_startup,
 	.trigger	= au1xpsc_ac97_trigger,
 	.hw_params	= au1xpsc_ac97_hw_params,
 };
@@ -379,6 +388,16 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 	if (!wd->mmio)
 		goto out1;
 
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!r)
+		goto out2;
+	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!r)
+		goto out2;
+	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
 	/* configuration: max dma trigger threshold, enable ac97 */
 	wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
 		  PSC_AC97CFG_DE_ENABLE;
@@ -401,15 +420,13 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 
 	ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
 	if (ret)
-		goto out1;
+		goto out2;
 
-	wd->dmapd = au1xpsc_pcm_add(pdev);
-	if (wd->dmapd) {
-		au1xpsc_ac97_workdata = wd;
-		return 0;
-	}
+	au1xpsc_ac97_workdata = wd;
+	return 0;
 
-	snd_soc_unregister_dai(&pdev->dev);
+out2:
+	iounmap(wd->mmio);
 out1:
 	release_mem_region(r->start, resource_size(r));
 out0:
@@ -422,9 +439,6 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev)
 	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
 	struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 
-	if (wd->dmapd)
-		au1xpsc_pcm_destroy(wd->dmapd);
-
 	snd_soc_unregister_dai(&pdev->dev);
 
 	/* disable PSC completely */
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index fca09127632..7c5ae920544 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -42,13 +42,13 @@
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
 #define I2SSTAT_BUSY(stype)	\
-	((stype) == PCM_TX ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB)
 #define I2SPCR_START(stype)	\
-	((stype) == PCM_TX ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS)
 #define I2SPCR_STOP(stype)	\
-	((stype) == PCM_TX ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP)
 #define I2SPCR_CLRFIFO(stype)	\
-	((stype) == PCM_TX ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
+	((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
 
 
 static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
@@ -240,7 +240,7 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 			       struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
-	int ret, stype = SUBSTREAM_TYPE(substream);
+	int ret, stype = substream->stream;
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -257,7 +257,16 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
+static int au1xpsc_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct au1xpsc_audio_data *pscdata = snd_soc_dai_get_drvdata(dai);
+	snd_soc_dai_set_dma_data(dai, substream, &pscdata->dmaids[0]);
+	return 0;
+}
+
 static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+	.startup	= au1xpsc_i2s_startup,
 	.trigger	= au1xpsc_i2s_trigger,
 	.hw_params	= au1xpsc_i2s_hw_params,
 	.set_fmt	= au1xpsc_i2s_set_fmt,
@@ -304,6 +313,16 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 	if (!wd->mmio)
 		goto out1;
 
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!r)
+		goto out2;
+	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!r)
+		goto out2;
+	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+
 	/* preserve PSC clock source set up by platform (dev.platform_data
 	 * is already occupied by soc layer)
 	 */
@@ -330,15 +349,11 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 	platform_set_drvdata(pdev, wd);
 
 	ret = snd_soc_register_dai(&pdev->dev, &wd->dai_drv);
-	if (ret)
-		goto out1;
-
-	/* finally add the DMA device for this PSC */
-	wd->dmapd = au1xpsc_pcm_add(pdev);
-	if (wd->dmapd)
+	if (!ret)
 		return 0;
 
-	snd_soc_unregister_dai(&pdev->dev);
+out2:
+	iounmap(wd->mmio);
 out1:
 	release_mem_region(r->start, resource_size(r));
 out0:
@@ -351,9 +366,6 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev)
 	struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev);
 	struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 
-	if (wd->dmapd)
-		au1xpsc_pcm_destroy(wd->dmapd);
-
 	snd_soc_unregister_dai(&pdev->dev);
 
 	au_writel(0, I2S_CFG(wd));
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index b30eadd422a..b16b2e02e0c 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -1,7 +1,7 @@
 /*
- * Au12x0/Au1550 PSC ALSA ASoC audio support.
+ * Alchemy ALSA ASoC audio support.
  *
- * (c) 2007-2008 MSC Vertriebsges.m.b.H.,
+ * (c) 2007-2011 MSC Vertriebsges.m.b.H.,
  *	Manuel Lauss <manuel.lauss@gmail.com>
  *
  * This program is free software; you can redistribute it and/or modify
@@ -13,10 +13,6 @@
 #ifndef _AU1X_PCM_H
 #define _AU1X_PCM_H
 
-/* 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;
 
@@ -27,15 +23,9 @@ struct au1xpsc_audio_data {
 
 	unsigned long pm[2];
 	struct mutex lock;
-	struct platform_device *dmapd;
+	int dmaids[2];
 };
 
-#define PCM_TX	0
-#define PCM_RX	1
-
-#define SUBSTREAM_TYPE(substream) \
-	((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX)
-
 /* easy access macros */
 #define PSC_CTRL(x)	((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET)
 #define PSC_SEL(x)	((unsigned long)((x)->mmio) + PSC_SEL_OFFSET)
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index fe9d548a683..9f6bc55fc39 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -27,6 +27,19 @@ config SND_SOC_BFIN_EVAL_ADAU1701
 	  board connected to one of the Blackfin evaluation boards like the
 	  BF5XX-STAMP or BF5XX-EZKIT.
 
+config SND_SOC_BFIN_EVAL_ADAU1373
+	tristate "Support for the EVAL-ADAU1373 board on Blackfin eval boards"
+	depends on SND_BF5XX_I2S && I2C
+	select SND_BF5XX_SOC_I2S
+	select SND_SOC_ADAU1373
+	help
+	  Say Y if you want to add support for the Analog Devices EVAL-ADAU1373
+	  board connected to one of the Blackfin evaluation boards like the
+	  BF5XX-STAMP or BF5XX-EZKIT.
+
+	  Note: This driver assumes that first ADAU1373 DAI is connected to the
+	  first SPORT port on the BF5XX board.
+
 config SND_SOC_BFIN_EVAL_ADAV80X
 	tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
 	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 6018bf52a23..1bf86ccaa8d 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -21,6 +21,7 @@ snd-ad1980-objs := bf5xx-ad1980.o
 snd-ssm2602-objs := bf5xx-ssm2602.o
 snd-ad73311-objs := bf5xx-ad73311.o
 snd-ad193x-objs := bf5xx-ad193x.o
+snd-soc-bfin-eval-adau1373-objs := bfin-eval-adau1373.o
 snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o
 snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o
 
@@ -29,5 +30,6 @@ obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
 obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1373) += snd-soc-bfin-eval-adau1373.o
 obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o
 obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
new file mode 100644
index 00000000000..8df2a3b0cb3
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -0,0 +1,202 @@
+/*
+ * Machine driver for EVAL-ADAU1373 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/adau1373.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line In1", NULL),
+	SND_SOC_DAPM_LINE("Line In2", NULL),
+	SND_SOC_DAPM_LINE("Line In3", NULL),
+	SND_SOC_DAPM_LINE("Line In4", NULL),
+
+	SND_SOC_DAPM_LINE("Line Out1", NULL),
+	SND_SOC_DAPM_LINE("Line Out2", NULL),
+	SND_SOC_DAPM_LINE("Stereo Out", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_HP("Earpiece", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = {
+	{ "AIN1L", NULL, "Line In1" },
+	{ "AIN1R", NULL, "Line In1" },
+	{ "AIN2L", NULL, "Line In2" },
+	{ "AIN2R", NULL, "Line In2" },
+	{ "AIN3L", NULL, "Line In3" },
+	{ "AIN3R", NULL, "Line In3" },
+	{ "AIN4L", NULL, "Line In4" },
+	{ "AIN4R", NULL, "Line In4" },
+
+	/* MICBIAS can be connected via a jumper to the line-in jack, since w
+	   don't know which one is going to be used, just power both. */
+	{ "Line In1", NULL, "MICBIAS1" },
+	{ "Line In2", NULL, "MICBIAS1" },
+	{ "Line In3", NULL, "MICBIAS1" },
+	{ "Line In4", NULL, "MICBIAS1" },
+	{ "Line In1", NULL, "MICBIAS2" },
+	{ "Line In2", NULL, "MICBIAS2" },
+	{ "Line In3", NULL, "MICBIAS2" },
+	{ "Line In4", NULL, "MICBIAS2" },
+
+	{ "Line Out1", NULL, "LOUT1L" },
+	{ "Line Out1", NULL, "LOUT1R" },
+	{ "Line Out2", NULL, "LOUT2L" },
+	{ "Line Out2", NULL, "LOUT2R" },
+	{ "Headphone", NULL, "HPL" },
+	{ "Headphone", NULL, "HPR" },
+	{ "Earpiece", NULL, "EP" },
+	{ "Speaker", NULL, "SPKL" },
+	{ "Stereo Out", NULL, "SPKR" },
+};
+
+static int bfin_eval_adau1373_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->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int ret;
+	int pll_rate;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret)
+		return ret;
+
+	switch (params_rate(params)) {
+	case 48000:
+	case 8000:
+	case 12000:
+	case 16000:
+	case 24000:
+	case 32000:
+		pll_rate = 48000 * 1024;
+		break;
+	case 44100:
+	case 7350:
+	case 11025:
+	case 14700:
+	case 22050:
+	case 29400:
+		pll_rate = 44100 * 1024;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1,
+			ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate,
+			SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+
+static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	unsigned int pll_rate = 48000 * 1024;
+	int ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1,
+			ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate,
+			SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+static struct snd_soc_ops bfin_eval_adau1373_ops = {
+	.hw_params = bfin_eval_adau1373_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adau1373_dai = {
+	.name = "adau1373",
+	.stream_name = "adau1373",
+	.cpu_dai_name = "bfin-i2s.0",
+	.codec_dai_name = "adau1373-aif1",
+	.platform_name = "bfin-i2s-pcm-audio",
+	.codec_name = "adau1373.0-001a",
+	.ops = &bfin_eval_adau1373_ops,
+	.init = bfin_eval_adau1373_codec_init,
+};
+
+static struct snd_soc_card bfin_eval_adau1373 = {
+	.name = "bfin-eval-adau1373",
+	.dai_link = &bfin_eval_adau1373_dai,
+	.num_links = 1,
+
+	.dapm_widgets		= bfin_eval_adau1373_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets),
+	.dapm_routes		= bfin_eval_adau1373_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(bfin_eval_adau1373_dapm_routes),
+};
+
+static int bfin_eval_adau1373_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &bfin_eval_adau1373;
+
+	card->dev = &pdev->dev;
+
+	return snd_soc_register_card(&bfin_eval_adau1373);
+}
+
+static int __devexit bfin_eval_adau1373_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver bfin_eval_adau1373_driver = {
+	.driver = {
+		.name = "bfin-eval-adau1373",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = bfin_eval_adau1373_probe,
+	.remove = __devexit_p(bfin_eval_adau1373_remove),
+};
+
+static int __init bfin_eval_adau1373_init(void)
+{
+	return platform_driver_register(&bfin_eval_adau1373_driver);
+}
+module_init(bfin_eval_adau1373_init);
+
+static void __exit bfin_eval_adau1373_exit(void)
+{
+	platform_driver_unregister(&bfin_eval_adau1373_driver);
+}
+module_exit(bfin_eval_adau1373_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bfin-eval-adau1373");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 665d9240c4a..71b46c8f70d 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_AD73311
+	select SND_SOC_ADAU1373 if I2C
 	select SND_SOC_ADAV80X
 	select SND_SOC_ADS117X
 	select SND_SOC_AK4104 if SPI_MASTER
@@ -139,6 +140,9 @@ config SND_SOC_ADAU1701
 	select SIGMA
 	tristate
 
+config SND_SOC_ADAU1373
+	tristate
+
 config SND_SOC_ADAV80X
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 5119a7e2c1a..70c1769acd1 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
 snd-soc-adau1701-objs := adau1701.o
+snd-soc-adau1373-objs := adau1373.o
 snd-soc-adav80x-objs := adav80x.o
 snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
@@ -100,6 +101,7 @@ obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
 obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_ADAU1373)	+= snd-soc-adau1373.o
 obj-$(CONFIG_SND_SOC_ADAU1701)  += snd-soc-adau1701.o
 obj-$(CONFIG_SND_SOC_ADAV80X)  += snd-soc-adav80x.o
 obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index eedb6f5e582..f934670199a 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -23,7 +23,7 @@
 
 /* codec private data */
 struct ad193x_priv {
-	enum snd_soc_control_type control_type;
+	struct regmap *regmap;
 	int sysclk;
 };
 
@@ -349,10 +349,8 @@ static int ad193x_probe(struct snd_soc_codec *codec)
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
-	if (ad193x->control_type == SND_SOC_I2C)
-		ret = snd_soc_codec_set_cache_io(codec, 8, 8, ad193x->control_type);
-	else
-		ret = snd_soc_codec_set_cache_io(codec, 16, 8, ad193x->control_type);
+	codec->control_data = ad193x->regmap;
+	ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
 	if (ret < 0) {
 		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
 		return ret;
@@ -388,6 +386,14 @@ static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
 };
 
 #if defined(CONFIG_SPI_MASTER)
+
+static const struct regmap_config ad193x_spi_regmap_config = {
+	.val_bits = 8,
+	.reg_bits = 16,
+	.read_flag_mask = 0x09,
+	.write_flag_mask = 0x08,
+};
+
 static int __devinit ad193x_spi_probe(struct spi_device *spi)
 {
 	struct ad193x_priv *ad193x;
@@ -397,20 +403,36 @@ static int __devinit ad193x_spi_probe(struct spi_device *spi)
 	if (ad193x == NULL)
 		return -ENOMEM;
 
+	ad193x->regmap = regmap_init_spi(spi, &ad193x_spi_regmap_config);
+	if (IS_ERR(ad193x->regmap)) {
+		ret = PTR_ERR(ad193x->regmap);
+		goto err_free;
+	}
+
 	spi_set_drvdata(spi, ad193x);
-	ad193x->control_type = SND_SOC_SPI;
 
 	ret = snd_soc_register_codec(&spi->dev,
 			&soc_codec_dev_ad193x, &ad193x_dai, 1);
 	if (ret < 0)
-		kfree(ad193x);
+		goto err_regmap_exit;
+
+	return 0;
+
+err_regmap_exit:
+	regmap_exit(ad193x->regmap);
+err_free:
+	kfree(ad193x);
+
 	return ret;
 }
 
 static int __devexit ad193x_spi_remove(struct spi_device *spi)
 {
+	struct ad193x_priv *ad193x = spi_get_drvdata(spi);
+
 	snd_soc_unregister_codec(&spi->dev);
-	kfree(spi_get_drvdata(spi));
+	regmap_exit(ad193x->regmap);
+	kfree(ad193x);
 	return 0;
 }
 
@@ -425,6 +447,12 @@ static struct spi_driver ad193x_spi_driver = {
 #endif
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+
+static const struct regmap_config ad193x_i2c_regmap_config = {
+	.val_bits = 8,
+	.reg_bits = 8,
+};
+
 static const struct i2c_device_id ad193x_id[] = {
 	{ "ad1936", 0 },
 	{ "ad1937", 0 },
@@ -442,20 +470,35 @@ static int __devinit ad193x_i2c_probe(struct i2c_client *client,
 	if (ad193x == NULL)
 		return -ENOMEM;
 
+	ad193x->regmap = regmap_init_i2c(client, &ad193x_i2c_regmap_config);
+	if (IS_ERR(ad193x->regmap)) {
+		ret = PTR_ERR(ad193x->regmap);
+		goto err_free;
+	}
+
 	i2c_set_clientdata(client, ad193x);
-	ad193x->control_type = SND_SOC_I2C;
 
 	ret =  snd_soc_register_codec(&client->dev,
 			&soc_codec_dev_ad193x, &ad193x_dai, 1);
 	if (ret < 0)
-		kfree(ad193x);
+		goto err_regmap_exit;
+
+	return 0;
+
+err_regmap_exit:
+	regmap_exit(ad193x->regmap);
+err_free:
+	kfree(ad193x);
 	return ret;
 }
 
 static int __devexit ad193x_i2c_remove(struct i2c_client *client)
 {
+	struct ad193x_priv *ad193x = i2c_get_clientdata(client);
+
 	snd_soc_unregister_codec(&client->dev);
-	kfree(i2c_get_clientdata(client));
+	regmap_exit(ad193x->regmap);
+	kfree(ad193x);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index cccc2e8e5fb..536e5f2b136 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -9,20 +9,20 @@
 #ifndef __AD193X_H__
 #define __AD193X_H__
 
-#define AD193X_PLL_CLK_CTRL0    0x800
+#define AD193X_PLL_CLK_CTRL0    0x00
 #define AD193X_PLL_POWERDOWN           0x01
 #define AD193X_PLL_INPUT_MASK   (~0x6)
 #define AD193X_PLL_INPUT_256    (0 << 1)
 #define AD193X_PLL_INPUT_384    (1 << 1)
 #define AD193X_PLL_INPUT_512    (2 << 1)
 #define AD193X_PLL_INPUT_768    (3 << 1)
-#define AD193X_PLL_CLK_CTRL1    0x801
-#define AD193X_DAC_CTRL0        0x802
+#define AD193X_PLL_CLK_CTRL1    0x01
+#define AD193X_DAC_CTRL0        0x02
 #define AD193X_DAC_POWERDOWN           0x01
 #define AD193X_DAC_SERFMT_MASK		0xC0
 #define AD193X_DAC_SERFMT_STEREO	(0 << 6)
 #define AD193X_DAC_SERFMT_TDM		(1 << 6)
-#define AD193X_DAC_CTRL1        0x803
+#define AD193X_DAC_CTRL1        0x03
 #define AD193X_DAC_2_CHANNELS   0
 #define AD193X_DAC_4_CHANNELS   1
 #define AD193X_DAC_8_CHANNELS   2
@@ -33,11 +33,11 @@
 #define AD193X_DAC_BCLK_MASTER  (1 << 5)
 #define AD193X_DAC_LEFT_HIGH    (1 << 3)
 #define AD193X_DAC_BCLK_INV     (1 << 7)
-#define AD193X_DAC_CTRL2        0x804
+#define AD193X_DAC_CTRL2        0x04
 #define AD193X_DAC_WORD_LEN_SHFT        3
 #define AD193X_DAC_WORD_LEN_MASK        0x18
 #define AD193X_DAC_MASTER_MUTE  1
-#define AD193X_DAC_CHNL_MUTE    0x805
+#define AD193X_DAC_CHNL_MUTE    0x05
 #define AD193X_DACL1_MUTE       0
 #define AD193X_DACR1_MUTE       1
 #define AD193X_DACL2_MUTE       2
@@ -46,28 +46,28 @@
 #define AD193X_DACR3_MUTE       5
 #define AD193X_DACL4_MUTE       6
 #define AD193X_DACR4_MUTE       7
-#define AD193X_DAC_L1_VOL       0x806
-#define AD193X_DAC_R1_VOL       0x807
-#define AD193X_DAC_L2_VOL       0x808
-#define AD193X_DAC_R2_VOL       0x809
-#define AD193X_DAC_L3_VOL       0x80a
-#define AD193X_DAC_R3_VOL       0x80b
-#define AD193X_DAC_L4_VOL       0x80c
-#define AD193X_DAC_R4_VOL       0x80d
-#define AD193X_ADC_CTRL0        0x80e
+#define AD193X_DAC_L1_VOL       0x06
+#define AD193X_DAC_R1_VOL       0x07
+#define AD193X_DAC_L2_VOL       0x08
+#define AD193X_DAC_R2_VOL       0x09
+#define AD193X_DAC_L3_VOL       0x0a
+#define AD193X_DAC_R3_VOL       0x0b
+#define AD193X_DAC_L4_VOL       0x0c
+#define AD193X_DAC_R4_VOL       0x0d
+#define AD193X_ADC_CTRL0        0x0e
 #define AD193X_ADC_POWERDOWN           0x01
 #define AD193X_ADC_HIGHPASS_FILTER	1
 #define AD193X_ADCL1_MUTE 		2
 #define AD193X_ADCR1_MUTE 		3
 #define AD193X_ADCL2_MUTE 		4
 #define AD193X_ADCR2_MUTE 		5
-#define AD193X_ADC_CTRL1        0x80f
+#define AD193X_ADC_CTRL1        0x0f
 #define AD193X_ADC_SERFMT_MASK		0x60
 #define AD193X_ADC_SERFMT_STEREO	(0 << 5)
 #define AD193X_ADC_SERFMT_TDM		(1 << 5)
 #define AD193X_ADC_SERFMT_AUX		(2 << 5)
 #define AD193X_ADC_WORD_LEN_MASK	0x3
-#define AD193X_ADC_CTRL2        0x810
+#define AD193X_ADC_CTRL2        0x10
 #define AD193X_ADC_2_CHANNELS   0
 #define AD193X_ADC_4_CHANNELS   1
 #define AD193X_ADC_8_CHANNELS   2
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 923b364a3e4..4c0fc30a4cc 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -200,18 +200,22 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec)
 	}
 
 	/* Read out vendor ID to make sure it is ad1980 */
-	if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144)
+	if (ac97_read(codec, AC97_VENDOR_ID1) != 0x4144) {
+		ret = -ENODEV;
 		goto reset_err;
+	}
 
 	vendor_id2 = ac97_read(codec, AC97_VENDOR_ID2);
 
 	if (vendor_id2 != 0x5370) {
-		if (vendor_id2 != 0x5374)
+		if (vendor_id2 != 0x5374) {
+			ret = -ENODEV;
 			goto reset_err;
-		else
+		} else {
 			printk(KERN_WARNING "ad1980: "
 				"Found AD1981 - only 2/2 IN/OUT Channels "
 				"supported\n");
+		}
 	}
 
 	/* unmute captures and playbacks volume */
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
new file mode 100644
index 00000000000..2aa40c3731d
--- /dev/null
+++ b/sound/soc/codecs/adau1373.c
@@ -0,0 +1,1414 @@
+/*
+ * Analog Devices ADAU1373 Audio Codec drive
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/gcd.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/adau1373.h>
+
+#include "adau1373.h"
+
+struct adau1373_dai {
+	unsigned int clk_src;
+	unsigned int sysclk;
+	bool enable_src;
+	bool master;
+};
+
+struct adau1373 {
+	struct adau1373_dai dais[3];
+};
+
+#define ADAU1373_INPUT_MODE	0x00
+#define ADAU1373_AINL_CTRL(x)	(0x01 + (x) * 2)
+#define ADAU1373_AINR_CTRL(x)	(0x02 + (x) * 2)
+#define ADAU1373_LLINE_OUT(x)	(0x9 + (x) * 2)
+#define ADAU1373_RLINE_OUT(x)	(0xa + (x) * 2)
+#define ADAU1373_LSPK_OUT	0x0d
+#define ADAU1373_RSPK_OUT	0x0e
+#define ADAU1373_LHP_OUT	0x0f
+#define ADAU1373_RHP_OUT	0x10
+#define ADAU1373_ADC_GAIN	0x11
+#define ADAU1373_LADC_MIXER	0x12
+#define ADAU1373_RADC_MIXER	0x13
+#define ADAU1373_LLINE1_MIX	0x14
+#define ADAU1373_RLINE1_MIX	0x15
+#define ADAU1373_LLINE2_MIX	0x16
+#define ADAU1373_RLINE2_MIX	0x17
+#define ADAU1373_LSPK_MIX	0x18
+#define ADAU1373_RSPK_MIX	0x19
+#define ADAU1373_LHP_MIX	0x1a
+#define ADAU1373_RHP_MIX	0x1b
+#define ADAU1373_EP_MIX		0x1c
+#define ADAU1373_HP_CTRL	0x1d
+#define ADAU1373_HP_CTRL2	0x1e
+#define ADAU1373_LS_CTRL	0x1f
+#define ADAU1373_EP_CTRL	0x21
+#define ADAU1373_MICBIAS_CTRL1	0x22
+#define ADAU1373_MICBIAS_CTRL2	0x23
+#define ADAU1373_OUTPUT_CTRL	0x24
+#define ADAU1373_PWDN_CTRL1	0x25
+#define ADAU1373_PWDN_CTRL2	0x26
+#define ADAU1373_PWDN_CTRL3	0x27
+#define ADAU1373_DPLL_CTRL(x)	(0x28 + (x) * 7)
+#define ADAU1373_PLL_CTRL1(x)	(0x29 + (x) * 7)
+#define ADAU1373_PLL_CTRL2(x)	(0x2a + (x) * 7)
+#define ADAU1373_PLL_CTRL3(x)	(0x2b + (x) * 7)
+#define ADAU1373_PLL_CTRL4(x)	(0x2c + (x) * 7)
+#define ADAU1373_PLL_CTRL5(x)	(0x2d + (x) * 7)
+#define ADAU1373_PLL_CTRL6(x)	(0x2e + (x) * 7)
+#define ADAU1373_PLL_CTRL7(x)	(0x2f + (x) * 7)
+#define ADAU1373_HEADDECT	0x36
+#define ADAU1373_ADC_DAC_STATUS	0x37
+#define ADAU1373_ADC_CTRL	0x3c
+#define ADAU1373_DAI(x)		(0x44 + (x))
+#define ADAU1373_CLK_SRC_DIV(x)	(0x40 + (x) * 2)
+#define ADAU1373_BCLKDIV(x)	(0x47 + (x))
+#define ADAU1373_SRC_RATIOA(x)	(0x4a + (x) * 2)
+#define ADAU1373_SRC_RATIOB(x)	(0x4b + (x) * 2)
+#define ADAU1373_DEEMP_CTRL	0x50
+#define ADAU1373_SRC_DAI_CTRL(x) (0x51 + (x))
+#define ADAU1373_DIN_MIX_CTRL(x) (0x56 + (x))
+#define ADAU1373_DOUT_MIX_CTRL(x) (0x5b + (x))
+#define ADAU1373_DAI_PBL_VOL(x)	(0x62 + (x) * 2)
+#define ADAU1373_DAI_PBR_VOL(x)	(0x63 + (x) * 2)
+#define ADAU1373_DAI_RECL_VOL(x) (0x68 + (x) * 2)
+#define ADAU1373_DAI_RECR_VOL(x) (0x69 + (x) * 2)
+#define ADAU1373_DAC1_PBL_VOL	0x6e
+#define ADAU1373_DAC1_PBR_VOL	0x6f
+#define ADAU1373_DAC2_PBL_VOL	0x70
+#define ADAU1373_DAC2_PBR_VOL	0x71
+#define ADAU1373_ADC_RECL_VOL	0x72
+#define ADAU1373_ADC_RECR_VOL	0x73
+#define ADAU1373_DMIC_RECL_VOL	0x74
+#define ADAU1373_DMIC_RECR_VOL	0x75
+#define ADAU1373_VOL_GAIN1	0x76
+#define ADAU1373_VOL_GAIN2	0x77
+#define ADAU1373_VOL_GAIN3	0x78
+#define ADAU1373_HPF_CTRL	0x7d
+#define ADAU1373_BASS1		0x7e
+#define ADAU1373_BASS2		0x7f
+#define ADAU1373_DRC(x)		(0x80 + (x) * 0x10)
+#define ADAU1373_3D_CTRL1	0xc0
+#define ADAU1373_3D_CTRL2	0xc1
+#define ADAU1373_FDSP_SEL1	0xdc
+#define ADAU1373_FDSP_SEL2	0xdd
+#define ADAU1373_FDSP_SEL3	0xde
+#define ADAU1373_FDSP_SEL4	0xdf
+#define ADAU1373_DIGMICCTRL	0xe2
+#define ADAU1373_DIGEN		0xeb
+#define ADAU1373_SOFT_RESET	0xff
+
+
+#define ADAU1373_PLL_CTRL6_DPLL_BYPASS	BIT(1)
+#define ADAU1373_PLL_CTRL6_PLL_EN	BIT(0)
+
+#define ADAU1373_DAI_INVERT_BCLK	BIT(7)
+#define ADAU1373_DAI_MASTER		BIT(6)
+#define ADAU1373_DAI_INVERT_LRCLK	BIT(4)
+#define ADAU1373_DAI_WLEN_16		0x0
+#define ADAU1373_DAI_WLEN_20		0x4
+#define ADAU1373_DAI_WLEN_24		0x8
+#define ADAU1373_DAI_WLEN_32		0xc
+#define ADAU1373_DAI_WLEN_MASK		0xc
+#define ADAU1373_DAI_FORMAT_RIGHT_J	0x0
+#define ADAU1373_DAI_FORMAT_LEFT_J	0x1
+#define ADAU1373_DAI_FORMAT_I2S		0x2
+#define ADAU1373_DAI_FORMAT_DSP		0x3
+
+#define ADAU1373_BCLKDIV_SOURCE		BIT(5)
+#define ADAU1373_BCLKDIV_32		0x03
+#define ADAU1373_BCLKDIV_64		0x02
+#define ADAU1373_BCLKDIV_128		0x01
+#define ADAU1373_BCLKDIV_256		0x00
+
+#define ADAU1373_ADC_CTRL_PEAK_DETECT	BIT(0)
+#define ADAU1373_ADC_CTRL_RESET		BIT(1)
+#define ADAU1373_ADC_CTRL_RESET_FORCE	BIT(2)
+
+#define ADAU1373_OUTPUT_CTRL_LDIFF	BIT(3)
+#define ADAU1373_OUTPUT_CTRL_LNFBEN	BIT(2)
+
+#define ADAU1373_PWDN_CTRL3_PWR_EN BIT(0)
+
+#define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4
+#define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2
+
+static const uint8_t adau1373_default_regs[] = {
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */
+	0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */
+	0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
+	0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */
+	0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
+	0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */
+	0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */
+	0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */
+	0x00, 0x1f, 0x0f, 0x00, 0x00,
+};
+
+static const unsigned int adau1373_out_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1),
+	8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0),
+	16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0),
+	24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0),
+};
+
+static const DECLARE_TLV_DB_MINMAX(adau1373_digital_tlv, -9563, 0);
+static const DECLARE_TLV_DB_SCALE(adau1373_in_pga_tlv, -1300, 100, 1);
+static const DECLARE_TLV_DB_SCALE(adau1373_ep_tlv, -600, 600, 1);
+
+static const DECLARE_TLV_DB_SCALE(adau1373_input_boost_tlv, 0, 2000, 0);
+static const DECLARE_TLV_DB_SCALE(adau1373_gain_boost_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adau1373_speaker_boost_tlv, 1200, 600, 0);
+
+static const char *adau1373_fdsp_sel_text[] = {
+	"None",
+	"Channel 1",
+	"Channel 2",
+	"Channel 3",
+	"Channel 4",
+	"Channel 5",
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum,
+	ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum,
+	ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum,
+	ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum,
+	ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum,
+	ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text);
+
+static const char *adau1373_hpf_cutoff_text[] = {
+	"3.7Hz", "50Hz", "100Hz", "150Hz", "200Hz", "250Hz", "300Hz", "350Hz",
+	"400Hz", "450Hz", "500Hz", "550Hz", "600Hz", "650Hz", "700Hz", "750Hz",
+	"800Hz",
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum,
+	ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text);
+
+static const char *adau1373_bass_lpf_cutoff_text[] = {
+	"801Hz", "1001Hz",
+};
+
+static const char *adau1373_bass_clip_level_text[] = {
+	"0.125", "0.250", "0.370", "0.500", "0.625", "0.750", "0.875",
+};
+
+static const unsigned int adau1373_bass_clip_level_values[] = {
+	1, 2, 3, 4, 5, 6, 7,
+};
+
+static const char *adau1373_bass_hpf_cutoff_text[] = {
+	"158Hz", "232Hz", "347Hz", "520Hz",
+};
+
+static const unsigned int adau1373_bass_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1),
+	3, 4, TLV_DB_SCALE_ITEM(950, 250, 0),
+	5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0),
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum,
+	ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text);
+
+static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum,
+	ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text,
+	adau1373_bass_clip_level_values);
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum,
+	ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text);
+
+static const char *adau1373_3d_level_text[] = {
+	"0%", "6.67%", "13.33%", "20%", "26.67%", "33.33%",
+	"40%", "46.67%", "53.33%", "60%", "66.67%", "73.33%",
+	"80%", "86.67", "99.33%", "100%"
+};
+
+static const char *adau1373_3d_cutoff_text[] = {
+	"No 3D", "0.03125 fs", "0.04583 fs", "0.075 fs", "0.11458 fs",
+	"0.16875 fs", "0.27083 fs"
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum,
+	ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum,
+	ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text);
+
+static const unsigned int adau1373_3d_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+	1, 7, TLV_DB_LINEAR_ITEM(-1800, -120),
+};
+
+static const char *adau1373_lr_mux_text[] = {
+	"Mute",
+	"Right Channel (L+R)",
+	"Left Channel (L+R)",
+	"Stereo",
+};
+
+static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum,
+	ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum,
+	ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text);
+static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum,
+	ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text);
+
+static const struct snd_kcontrol_new adau1373_controls[] = {
+	SOC_DOUBLE_R_TLV("AIF1 Capture Volume", ADAU1373_DAI_RECL_VOL(0),
+		ADAU1373_DAI_RECR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv),
+	SOC_DOUBLE_R_TLV("AIF2 Capture Volume", ADAU1373_DAI_RECL_VOL(1),
+		ADAU1373_DAI_RECR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv),
+	SOC_DOUBLE_R_TLV("AIF3 Capture Volume", ADAU1373_DAI_RECL_VOL(2),
+		ADAU1373_DAI_RECR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv),
+
+	SOC_DOUBLE_R_TLV("ADC Capture Volume", ADAU1373_ADC_RECL_VOL,
+		ADAU1373_ADC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv),
+	SOC_DOUBLE_R_TLV("DMIC Capture Volume", ADAU1373_DMIC_RECL_VOL,
+		ADAU1373_DMIC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv),
+
+	SOC_DOUBLE_R_TLV("AIF1 Playback Volume", ADAU1373_DAI_PBL_VOL(0),
+		ADAU1373_DAI_PBR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv),
+	SOC_DOUBLE_R_TLV("AIF2 Playback Volume", ADAU1373_DAI_PBL_VOL(1),
+		ADAU1373_DAI_PBR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv),
+	SOC_DOUBLE_R_TLV("AIF3 Playback Volume", ADAU1373_DAI_PBL_VOL(2),
+		ADAU1373_DAI_PBR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv),
+
+	SOC_DOUBLE_R_TLV("DAC1 Playback Volume", ADAU1373_DAC1_PBL_VOL,
+		ADAU1373_DAC1_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv),
+	SOC_DOUBLE_R_TLV("DAC2 Playback Volume", ADAU1373_DAC2_PBL_VOL,
+		ADAU1373_DAC2_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv),
+
+	SOC_DOUBLE_R_TLV("Lineout1 Playback Volume", ADAU1373_LLINE_OUT(0),
+		ADAU1373_RLINE_OUT(0), 0, 0x1f, 0, adau1373_out_tlv),
+	SOC_DOUBLE_R_TLV("Speaker Playback Volume", ADAU1373_LSPK_OUT,
+		ADAU1373_RSPK_OUT, 0, 0x1f, 0, adau1373_out_tlv),
+	SOC_DOUBLE_R_TLV("Headphone Playback Volume", ADAU1373_LHP_OUT,
+		ADAU1373_RHP_OUT, 0, 0x1f, 0, adau1373_out_tlv),
+
+	SOC_DOUBLE_R_TLV("Input 1 Capture Volume", ADAU1373_AINL_CTRL(0),
+		ADAU1373_AINR_CTRL(0), 0, 0x1f, 0, adau1373_in_pga_tlv),
+	SOC_DOUBLE_R_TLV("Input 2 Capture Volume", ADAU1373_AINL_CTRL(1),
+		ADAU1373_AINR_CTRL(1), 0, 0x1f, 0, adau1373_in_pga_tlv),
+	SOC_DOUBLE_R_TLV("Input 3 Capture Volume", ADAU1373_AINL_CTRL(2),
+		ADAU1373_AINR_CTRL(2), 0, 0x1f, 0, adau1373_in_pga_tlv),
+	SOC_DOUBLE_R_TLV("Input 4 Capture Volume", ADAU1373_AINL_CTRL(3),
+		ADAU1373_AINR_CTRL(3), 0, 0x1f, 0, adau1373_in_pga_tlv),
+
+	SOC_SINGLE_TLV("Earpiece Playback Volume", ADAU1373_EP_CTRL, 0, 3, 0,
+		adau1373_ep_tlv),
+
+	SOC_DOUBLE_TLV("AIF3 Boost Playback Volume", ADAU1373_VOL_GAIN1, 4, 5,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("AIF2 Boost Playback Volume", ADAU1373_VOL_GAIN1, 2, 3,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("AIF1 Boost Playback Volume", ADAU1373_VOL_GAIN1, 0, 1,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("AIF3 Boost Capture Volume", ADAU1373_VOL_GAIN2, 4, 5,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("AIF2 Boost Capture Volume", ADAU1373_VOL_GAIN2, 2, 3,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("AIF1 Boost Capture Volume", ADAU1373_VOL_GAIN2, 0, 1,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("DMIC Boost Capture Volume", ADAU1373_VOL_GAIN3, 6, 7,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("ADC Boost Capture Volume", ADAU1373_VOL_GAIN3, 4, 5,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("DAC2 Boost Playback Volume", ADAU1373_VOL_GAIN3, 2, 3,
+		1, 0, adau1373_gain_boost_tlv),
+	SOC_DOUBLE_TLV("DAC1 Boost Playback Volume", ADAU1373_VOL_GAIN3, 0, 1,
+		1, 0, adau1373_gain_boost_tlv),
+
+	SOC_DOUBLE_TLV("Input 1 Boost Capture Volume", ADAU1373_ADC_GAIN, 0, 4,
+		1, 0, adau1373_input_boost_tlv),
+	SOC_DOUBLE_TLV("Input 2 Boost Capture Volume", ADAU1373_ADC_GAIN, 1, 5,
+		1, 0, adau1373_input_boost_tlv),
+	SOC_DOUBLE_TLV("Input 3 Boost Capture Volume", ADAU1373_ADC_GAIN, 2, 6,
+		1, 0, adau1373_input_boost_tlv),
+	SOC_DOUBLE_TLV("Input 4 Boost Capture Volume", ADAU1373_ADC_GAIN, 3, 7,
+		1, 0, adau1373_input_boost_tlv),
+
+	SOC_DOUBLE_TLV("Speaker Boost Playback Volume", ADAU1373_LS_CTRL, 2, 3,
+		1, 0, adau1373_speaker_boost_tlv),
+
+	SOC_ENUM("Lineout1 LR Mux", adau1373_lineout1_lr_mux_enum),
+	SOC_ENUM("Speaker LR Mux", adau1373_speaker_lr_mux_enum),
+
+	SOC_ENUM("HPF Cutoff", adau1373_hpf_cutoff_enum),
+	SOC_DOUBLE("HPF Switch", ADAU1373_HPF_CTRL, 1, 0, 1, 0),
+	SOC_ENUM("HPF Channel", adau1373_hpf_channel_enum),
+
+	SOC_ENUM("Bass HPF Cutoff", adau1373_bass_hpf_cutoff_enum),
+	SOC_VALUE_ENUM("Bass Clip Level Threshold",
+	    adau1373_bass_clip_level_enum),
+	SOC_ENUM("Bass LPF Cutoff", adau1373_bass_lpf_cutoff_enum),
+	SOC_DOUBLE("Bass Playback Switch", ADAU1373_BASS2, 0, 1, 1, 0),
+	SOC_SINGLE_TLV("Bass Playback Volume", ADAU1373_BASS2, 2, 7, 0,
+	    adau1373_bass_tlv),
+	SOC_ENUM("Bass Channel", adau1373_bass_channel_enum),
+
+	SOC_ENUM("3D Freq", adau1373_3d_cutoff_enum),
+	SOC_ENUM("3D Level", adau1373_3d_level_enum),
+	SOC_SINGLE("3D Playback Switch", ADAU1373_3D_CTRL2, 0, 1, 0),
+	SOC_SINGLE_TLV("3D Playback Volume", ADAU1373_3D_CTRL2, 2, 7, 0,
+		adau1373_3d_tlv),
+	SOC_ENUM("3D Channel", adau1373_bass_channel_enum),
+
+	SOC_SINGLE("Zero Cross Switch", ADAU1373_PWDN_CTRL3, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new adau1373_lineout2_controls[] = {
+	SOC_DOUBLE_R_TLV("Lineout2 Playback Volume", ADAU1373_LLINE_OUT(1),
+		ADAU1373_RLINE_OUT(1), 0, 0x1f, 0, adau1373_out_tlv),
+	SOC_ENUM("Lineout2 LR Mux", adau1373_lineout2_lr_mux_enum),
+};
+
+static const struct snd_kcontrol_new adau1373_drc_controls[] = {
+	SOC_ENUM("DRC1 Channel", adau1373_drc1_channel_enum),
+	SOC_ENUM("DRC2 Channel", adau1373_drc2_channel_enum),
+	SOC_ENUM("DRC3 Channel", adau1373_drc3_channel_enum),
+};
+
+static int adau1373_pll_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	unsigned int pll_id = w->name[3] - '1';
+	unsigned int val;
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		val = ADAU1373_PLL_CTRL6_PLL_EN;
+	else
+		val = 0;
+
+	snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+		ADAU1373_PLL_CTRL6_PLL_EN, val);
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		mdelay(5);
+
+	return 0;
+}
+
+static const char *adau1373_decimator_text[] = {
+	"ADC",
+	"DMIC1",
+};
+
+static const struct soc_enum adau1373_decimator_enum =
+	SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text);
+
+static const struct snd_kcontrol_new adau1373_decimator_mux =
+	SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum);
+
+static const struct snd_kcontrol_new adau1373_left_adc_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_LADC_MIXER, 4, 1, 0),
+	SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_LADC_MIXER, 3, 1, 0),
+	SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_LADC_MIXER, 2, 1, 0),
+	SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_LADC_MIXER, 1, 1, 0),
+	SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_LADC_MIXER, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new adau1373_right_adc_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_RADC_MIXER, 4, 1, 0),
+	SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_RADC_MIXER, 3, 1, 0),
+	SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_RADC_MIXER, 2, 1, 0),
+	SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_RADC_MIXER, 1, 1, 0),
+	SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_RADC_MIXER, 0, 1, 0),
+};
+
+#define DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(_name, _reg) \
+const struct snd_kcontrol_new _name[] = { \
+	SOC_DAPM_SINGLE("Left DAC2 Switch", _reg, 7, 1, 0), \
+	SOC_DAPM_SINGLE("Right DAC2 Switch", _reg, 6, 1, 0), \
+	SOC_DAPM_SINGLE("Left DAC1 Switch", _reg, 5, 1, 0), \
+	SOC_DAPM_SINGLE("Right DAC1 Switch", _reg, 4, 1, 0), \
+	SOC_DAPM_SINGLE("Input 4 Bypass Switch", _reg, 3, 1, 0), \
+	SOC_DAPM_SINGLE("Input 3 Bypass Switch", _reg, 2, 1, 0), \
+	SOC_DAPM_SINGLE("Input 2 Bypass Switch", _reg, 1, 1, 0), \
+	SOC_DAPM_SINGLE("Input 1 Bypass Switch", _reg, 0, 1, 0), \
+}
+
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line1_mixer_controls,
+	ADAU1373_LLINE1_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line1_mixer_controls,
+	ADAU1373_RLINE1_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line2_mixer_controls,
+	ADAU1373_LLINE2_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line2_mixer_controls,
+	ADAU1373_RLINE2_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_spk_mixer_controls,
+	ADAU1373_LSPK_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_spk_mixer_controls,
+	ADAU1373_RSPK_MIX);
+static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_ep_mixer_controls,
+	ADAU1373_EP_MIX);
+
+static const struct snd_kcontrol_new adau1373_left_hp_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left DAC1 Switch", ADAU1373_LHP_MIX, 5, 1, 0),
+	SOC_DAPM_SINGLE("Left DAC2 Switch", ADAU1373_LHP_MIX, 4, 1, 0),
+	SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_LHP_MIX, 3, 1, 0),
+	SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_LHP_MIX, 2, 1, 0),
+	SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_LHP_MIX, 1, 1, 0),
+	SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_LHP_MIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new adau1373_right_hp_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Right DAC1 Switch", ADAU1373_RHP_MIX, 5, 1, 0),
+	SOC_DAPM_SINGLE("Right DAC2 Switch", ADAU1373_RHP_MIX, 4, 1, 0),
+	SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_RHP_MIX, 3, 1, 0),
+	SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_RHP_MIX, 2, 1, 0),
+	SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_RHP_MIX, 1, 1, 0),
+	SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_RHP_MIX, 0, 1, 0),
+};
+
+#define DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(_name, _reg) \
+const struct snd_kcontrol_new _name[] = { \
+	SOC_DAPM_SINGLE("DMIC2 Swapped Switch", _reg, 6, 1, 0), \
+	SOC_DAPM_SINGLE("DMIC2 Switch", _reg, 5, 1, 0), \
+	SOC_DAPM_SINGLE("ADC/DMIC1 Swapped Switch", _reg, 4, 1, 0), \
+	SOC_DAPM_SINGLE("ADC/DMIC1 Switch", _reg, 3, 1, 0), \
+	SOC_DAPM_SINGLE("AIF3 Switch", _reg, 2, 1, 0), \
+	SOC_DAPM_SINGLE("AIF2 Switch", _reg, 1, 1, 0), \
+	SOC_DAPM_SINGLE("AIF1 Switch", _reg, 0, 1, 0), \
+}
+
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel1_mixer_controls,
+	ADAU1373_DIN_MIX_CTRL(0));
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel2_mixer_controls,
+	ADAU1373_DIN_MIX_CTRL(1));
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel3_mixer_controls,
+	ADAU1373_DIN_MIX_CTRL(2));
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel4_mixer_controls,
+	ADAU1373_DIN_MIX_CTRL(3));
+static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel5_mixer_controls,
+	ADAU1373_DIN_MIX_CTRL(4));
+
+#define DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(_name, _reg) \
+const struct snd_kcontrol_new _name[] = { \
+	SOC_DAPM_SINGLE("DSP Channel5 Switch", _reg, 4, 1, 0), \
+	SOC_DAPM_SINGLE("DSP Channel4 Switch", _reg, 3, 1, 0), \
+	SOC_DAPM_SINGLE("DSP Channel3 Switch", _reg, 2, 1, 0), \
+	SOC_DAPM_SINGLE("DSP Channel2 Switch", _reg, 1, 1, 0), \
+	SOC_DAPM_SINGLE("DSP Channel1 Switch", _reg, 0, 1, 0), \
+}
+
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif1_mixer_controls,
+	ADAU1373_DOUT_MIX_CTRL(0));
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif2_mixer_controls,
+	ADAU1373_DOUT_MIX_CTRL(1));
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif3_mixer_controls,
+	ADAU1373_DOUT_MIX_CTRL(2));
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac1_mixer_controls,
+	ADAU1373_DOUT_MIX_CTRL(3));
+static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac2_mixer_controls,
+	ADAU1373_DOUT_MIX_CTRL(4));
+
+static const struct snd_soc_dapm_widget adau1373_dapm_widgets[] = {
+	/* Datasheet claims Left ADC is bit 6 and Right ADC is bit 7, but that
+	 * doesn't seem to be the case. */
+	SND_SOC_DAPM_ADC("Left ADC", NULL, ADAU1373_PWDN_CTRL1, 7, 0),
+	SND_SOC_DAPM_ADC("Right ADC", NULL, ADAU1373_PWDN_CTRL1, 6, 0),
+
+	SND_SOC_DAPM_ADC("DMIC1", NULL, ADAU1373_DIGMICCTRL, 0, 0),
+	SND_SOC_DAPM_ADC("DMIC2", NULL, ADAU1373_DIGMICCTRL, 2, 0),
+
+	SND_SOC_DAPM_VIRT_MUX("Decimator Mux", SND_SOC_NOPM, 0, 0,
+		&adau1373_decimator_mux),
+
+	SND_SOC_DAPM_SUPPLY("MICBIAS2", ADAU1373_PWDN_CTRL1, 5, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS1", ADAU1373_PWDN_CTRL1, 4, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("IN4PGA", ADAU1373_PWDN_CTRL1, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IN3PGA", ADAU1373_PWDN_CTRL1, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IN2PGA", ADAU1373_PWDN_CTRL1, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("IN1PGA", ADAU1373_PWDN_CTRL1, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_DAC("Left DAC2", NULL, ADAU1373_PWDN_CTRL2, 7, 0),
+	SND_SOC_DAPM_DAC("Right DAC2", NULL, ADAU1373_PWDN_CTRL2, 6, 0),
+	SND_SOC_DAPM_DAC("Left DAC1", NULL, ADAU1373_PWDN_CTRL2, 5, 0),
+	SND_SOC_DAPM_DAC("Right DAC1", NULL, ADAU1373_PWDN_CTRL2, 4, 0),
+
+	SOC_MIXER_ARRAY("Left ADC Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_left_adc_mixer_controls),
+	SOC_MIXER_ARRAY("Right ADC Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_right_adc_mixer_controls),
+
+	SOC_MIXER_ARRAY("Left Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 3, 0,
+		adau1373_left_line2_mixer_controls),
+	SOC_MIXER_ARRAY("Right Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 2, 0,
+		adau1373_right_line2_mixer_controls),
+	SOC_MIXER_ARRAY("Left Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 1, 0,
+		adau1373_left_line1_mixer_controls),
+	SOC_MIXER_ARRAY("Right Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 0, 0,
+		adau1373_right_line1_mixer_controls),
+
+	SOC_MIXER_ARRAY("Earpiece Mixer", ADAU1373_PWDN_CTRL3, 4, 0,
+		adau1373_ep_mixer_controls),
+	SOC_MIXER_ARRAY("Left Speaker Mixer", ADAU1373_PWDN_CTRL3, 3, 0,
+		adau1373_left_spk_mixer_controls),
+	SOC_MIXER_ARRAY("Right Speaker Mixer", ADAU1373_PWDN_CTRL3, 2, 0,
+		adau1373_right_spk_mixer_controls),
+	SOC_MIXER_ARRAY("Left Headphone Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_left_hp_mixer_controls),
+	SOC_MIXER_ARRAY("Right Headphone Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_right_hp_mixer_controls),
+	SND_SOC_DAPM_SUPPLY("Headphone Enable", ADAU1373_PWDN_CTRL3, 1, 0,
+		NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("AIF1 CLK", ADAU1373_SRC_DAI_CTRL(0), 0, 0,
+	    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF2 CLK", ADAU1373_SRC_DAI_CTRL(1), 0, 0,
+	    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF3 CLK", ADAU1373_SRC_DAI_CTRL(2), 0, 0,
+	    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF1 IN SRC", ADAU1373_SRC_DAI_CTRL(0), 2, 0,
+	    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF1 OUT SRC", ADAU1373_SRC_DAI_CTRL(0), 1, 0,
+	    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF2 IN SRC", ADAU1373_SRC_DAI_CTRL(1), 2, 0,
+	    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF2 OUT SRC", ADAU1373_SRC_DAI_CTRL(1), 1, 0,
+	    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF3 IN SRC", ADAU1373_SRC_DAI_CTRL(2), 2, 0,
+	    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("AIF3 OUT SRC", ADAU1373_SRC_DAI_CTRL(2), 1, 0,
+	    NULL, 0),
+
+	SND_SOC_DAPM_AIF_IN("AIF1 IN", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF1 OUT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF2 IN", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF2 OUT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIF3 IN", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AIF3 OUT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0),
+
+	SOC_MIXER_ARRAY("DSP Channel1 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_dsp_channel1_mixer_controls),
+	SOC_MIXER_ARRAY("DSP Channel2 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_dsp_channel2_mixer_controls),
+	SOC_MIXER_ARRAY("DSP Channel3 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_dsp_channel3_mixer_controls),
+	SOC_MIXER_ARRAY("DSP Channel4 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_dsp_channel4_mixer_controls),
+	SOC_MIXER_ARRAY("DSP Channel5 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_dsp_channel5_mixer_controls),
+
+	SOC_MIXER_ARRAY("AIF1 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_aif1_mixer_controls),
+	SOC_MIXER_ARRAY("AIF2 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_aif2_mixer_controls),
+	SOC_MIXER_ARRAY("AIF3 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_aif3_mixer_controls),
+	SOC_MIXER_ARRAY("DAC1 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_dac1_mixer_controls),
+	SOC_MIXER_ARRAY("DAC2 Mixer", SND_SOC_NOPM, 0, 0,
+		adau1373_dac2_mixer_controls),
+
+	SND_SOC_DAPM_SUPPLY("DSP", ADAU1373_DIGEN, 4, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Recording Engine B", ADAU1373_DIGEN, 3, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Recording Engine A", ADAU1373_DIGEN, 2, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Playback Engine B", ADAU1373_DIGEN, 1, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Playback Engine A", ADAU1373_DIGEN, 0, 0, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("PLL1", SND_SOC_NOPM, 0, 0, adau1373_pll_event,
+		SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("PLL2", SND_SOC_NOPM, 0, 0, adau1373_pll_event,
+		SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("SYSCLK1", ADAU1373_CLK_SRC_DIV(0), 7, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("SYSCLK2", ADAU1373_CLK_SRC_DIV(1), 7, 0, NULL, 0),
+
+	SND_SOC_DAPM_INPUT("AIN1L"),
+	SND_SOC_DAPM_INPUT("AIN1R"),
+	SND_SOC_DAPM_INPUT("AIN2L"),
+	SND_SOC_DAPM_INPUT("AIN2R"),
+	SND_SOC_DAPM_INPUT("AIN3L"),
+	SND_SOC_DAPM_INPUT("AIN3R"),
+	SND_SOC_DAPM_INPUT("AIN4L"),
+	SND_SOC_DAPM_INPUT("AIN4R"),
+
+	SND_SOC_DAPM_INPUT("DMIC1DAT"),
+	SND_SOC_DAPM_INPUT("DMIC2DAT"),
+
+	SND_SOC_DAPM_OUTPUT("LOUT1L"),
+	SND_SOC_DAPM_OUTPUT("LOUT1R"),
+	SND_SOC_DAPM_OUTPUT("LOUT2L"),
+	SND_SOC_DAPM_OUTPUT("LOUT2R"),
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+	SND_SOC_DAPM_OUTPUT("SPKL"),
+	SND_SOC_DAPM_OUTPUT("SPKR"),
+	SND_SOC_DAPM_OUTPUT("EP"),
+};
+
+static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source,
+	struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_codec *codec = source->codec;
+	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+	unsigned int dai;
+	const char *clk;
+
+	dai = sink->name[3] - '1';
+
+	if (!adau1373->dais[dai].master)
+		return 0;
+
+	if (adau1373->dais[dai].clk_src == ADAU1373_CLK_SRC_PLL1)
+		clk = "SYSCLK1";
+	else
+		clk = "SYSCLK2";
+
+	return strcmp(source->name, clk) == 0;
+}
+
+static int adau1373_check_src(struct snd_soc_dapm_widget *source,
+	struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_codec *codec = source->codec;
+	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+	unsigned int dai;
+
+	dai = sink->name[3] - '1';
+
+	return adau1373->dais[dai].enable_src;
+}
+
+#define DSP_CHANNEL_MIXER_ROUTES(_sink) \
+	{ _sink, "DMIC2 Swapped Switch", "DMIC2" }, \
+	{ _sink, "DMIC2 Switch", "DMIC2" }, \
+	{ _sink, "ADC/DMIC1 Swapped Switch", "Decimator Mux" }, \
+	{ _sink, "ADC/DMIC1 Switch", "Decimator Mux" }, \
+	{ _sink, "AIF1 Switch", "AIF1 IN" }, \
+	{ _sink, "AIF2 Switch", "AIF2 IN" }, \
+	{ _sink, "AIF3 Switch", "AIF3 IN" }
+
+#define DSP_OUTPUT_MIXER_ROUTES(_sink) \
+	{ _sink, "DSP Channel1 Switch", "DSP Channel1 Mixer" }, \
+	{ _sink, "DSP Channel2 Switch", "DSP Channel2 Mixer" }, \
+	{ _sink, "DSP Channel3 Switch", "DSP Channel3 Mixer" }, \
+	{ _sink, "DSP Channel4 Switch", "DSP Channel4 Mixer" }, \
+	{ _sink, "DSP Channel5 Switch", "DSP Channel5 Mixer" }
+
+#define LEFT_OUTPUT_MIXER_ROUTES(_sink) \
+	{ _sink, "Right DAC2 Switch", "Right DAC2" }, \
+	{ _sink, "Left DAC2 Switch", "Left DAC2" }, \
+	{ _sink, "Right DAC1 Switch", "Right DAC1" }, \
+	{ _sink, "Left DAC1 Switch", "Left DAC1" }, \
+	{ _sink, "Input 1 Bypass Switch", "IN1PGA" }, \
+	{ _sink, "Input 2 Bypass Switch", "IN2PGA" }, \
+	{ _sink, "Input 3 Bypass Switch", "IN3PGA" }, \
+	{ _sink, "Input 4 Bypass Switch", "IN4PGA" }
+
+#define RIGHT_OUTPUT_MIXER_ROUTES(_sink) \
+	{ _sink, "Right DAC2 Switch", "Right DAC2" }, \
+	{ _sink, "Left DAC2 Switch", "Left DAC2" }, \
+	{ _sink, "Right DAC1 Switch", "Right DAC1" }, \
+	{ _sink, "Left DAC1 Switch", "Left DAC1" }, \
+	{ _sink, "Input 1 Bypass Switch", "IN1PGA" }, \
+	{ _sink, "Input 2 Bypass Switch", "IN2PGA" }, \
+	{ _sink, "Input 3 Bypass Switch", "IN3PGA" }, \
+	{ _sink, "Input 4 Bypass Switch", "IN4PGA" }
+
+static const struct snd_soc_dapm_route adau1373_dapm_routes[] = {
+	{ "Left ADC Mixer", "DAC1 Switch", "Left DAC1" },
+	{ "Left ADC Mixer", "Input 1 Switch", "IN1PGA" },
+	{ "Left ADC Mixer", "Input 2 Switch", "IN2PGA" },
+	{ "Left ADC Mixer", "Input 3 Switch", "IN3PGA" },
+	{ "Left ADC Mixer", "Input 4 Switch", "IN4PGA" },
+
+	{ "Right ADC Mixer", "DAC1 Switch", "Right DAC1" },
+	{ "Right ADC Mixer", "Input 1 Switch", "IN1PGA" },
+	{ "Right ADC Mixer", "Input 2 Switch", "IN2PGA" },
+	{ "Right ADC Mixer", "Input 3 Switch", "IN3PGA" },
+	{ "Right ADC Mixer", "Input 4 Switch", "IN4PGA" },
+
+	{ "Left ADC", NULL, "Left ADC Mixer" },
+	{ "Right ADC", NULL, "Right ADC Mixer" },
+
+	{ "Decimator Mux", "ADC", "Left ADC" },
+	{ "Decimator Mux", "ADC", "Right ADC" },
+	{ "Decimator Mux", "DMIC1", "DMIC1" },
+
+	DSP_CHANNEL_MIXER_ROUTES("DSP Channel1 Mixer"),
+	DSP_CHANNEL_MIXER_ROUTES("DSP Channel2 Mixer"),
+	DSP_CHANNEL_MIXER_ROUTES("DSP Channel3 Mixer"),
+	DSP_CHANNEL_MIXER_ROUTES("DSP Channel4 Mixer"),
+	DSP_CHANNEL_MIXER_ROUTES("DSP Channel5 Mixer"),
+
+	DSP_OUTPUT_MIXER_ROUTES("AIF1 Mixer"),
+	DSP_OUTPUT_MIXER_ROUTES("AIF2 Mixer"),
+	DSP_OUTPUT_MIXER_ROUTES("AIF3 Mixer"),
+	DSP_OUTPUT_MIXER_ROUTES("DAC1 Mixer"),
+	DSP_OUTPUT_MIXER_ROUTES("DAC2 Mixer"),
+
+	{ "AIF1 OUT", NULL, "AIF1 Mixer" },
+	{ "AIF2 OUT", NULL, "AIF2 Mixer" },
+	{ "AIF3 OUT", NULL, "AIF3 Mixer" },
+	{ "Left DAC1", NULL, "DAC1 Mixer" },
+	{ "Right DAC1", NULL, "DAC1 Mixer" },
+	{ "Left DAC2", NULL, "DAC2 Mixer" },
+	{ "Right DAC2", NULL, "DAC2 Mixer" },
+
+	LEFT_OUTPUT_MIXER_ROUTES("Left Lineout1 Mixer"),
+	RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout1 Mixer"),
+	LEFT_OUTPUT_MIXER_ROUTES("Left Lineout2 Mixer"),
+	RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout2 Mixer"),
+	LEFT_OUTPUT_MIXER_ROUTES("Left Speaker Mixer"),
+	RIGHT_OUTPUT_MIXER_ROUTES("Right Speaker Mixer"),
+
+	{ "Left Headphone Mixer", "Left DAC2 Switch", "Left DAC2" },
+	{ "Left Headphone Mixer", "Left DAC1 Switch", "Left DAC1" },
+	{ "Left Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" },
+	{ "Left Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" },
+	{ "Left Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" },
+	{ "Left Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" },
+	{ "Right Headphone Mixer", "Right DAC2 Switch", "Right DAC2" },
+	{ "Right Headphone Mixer", "Right DAC1 Switch", "Right DAC1" },
+	{ "Right Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" },
+	{ "Right Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" },
+	{ "Right Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" },
+	{ "Right Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" },
+
+	{ "Left Headphone Mixer", NULL, "Headphone Enable" },
+	{ "Right Headphone Mixer", NULL, "Headphone Enable" },
+
+	{ "Earpiece Mixer", "Right DAC2 Switch", "Right DAC2" },
+	{ "Earpiece Mixer", "Left DAC2 Switch", "Left DAC2" },
+	{ "Earpiece Mixer", "Right DAC1 Switch", "Right DAC1" },
+	{ "Earpiece Mixer", "Left DAC1 Switch", "Left DAC1" },
+	{ "Earpiece Mixer", "Input 1 Bypass Switch", "IN1PGA" },
+	{ "Earpiece Mixer", "Input 2 Bypass Switch", "IN2PGA" },
+	{ "Earpiece Mixer", "Input 3 Bypass Switch", "IN3PGA" },
+	{ "Earpiece Mixer", "Input 4 Bypass Switch", "IN4PGA" },
+
+	{ "LOUT1L", NULL, "Left Lineout1 Mixer" },
+	{ "LOUT1R", NULL, "Right Lineout1 Mixer" },
+	{ "LOUT2L", NULL, "Left Lineout2 Mixer" },
+	{ "LOUT2R", NULL, "Right Lineout2 Mixer" },
+	{ "SPKL", NULL, "Left Speaker Mixer" },
+	{ "SPKR", NULL, "Right Speaker Mixer" },
+	{ "HPL", NULL, "Left Headphone Mixer" },
+	{ "HPR", NULL, "Right Headphone Mixer" },
+	{ "EP", NULL, "Earpiece Mixer" },
+
+	{ "IN1PGA", NULL, "AIN1L" },
+	{ "IN2PGA", NULL, "AIN2L" },
+	{ "IN3PGA", NULL, "AIN3L" },
+	{ "IN4PGA", NULL, "AIN4L" },
+	{ "IN1PGA", NULL, "AIN1R" },
+	{ "IN2PGA", NULL, "AIN2R" },
+	{ "IN3PGA", NULL, "AIN3R" },
+	{ "IN4PGA", NULL, "AIN4R" },
+
+	{ "SYSCLK1", NULL, "PLL1" },
+	{ "SYSCLK2", NULL, "PLL2" },
+
+	{ "Left DAC1", NULL, "SYSCLK1" },
+	{ "Right DAC1", NULL, "SYSCLK1" },
+	{ "Left DAC2", NULL, "SYSCLK1" },
+	{ "Right DAC2", NULL, "SYSCLK1" },
+	{ "Left ADC", NULL, "SYSCLK1" },
+	{ "Right ADC", NULL, "SYSCLK1" },
+
+	{ "DSP", NULL, "SYSCLK1" },
+
+	{ "AIF1 Mixer", NULL, "DSP" },
+	{ "AIF2 Mixer", NULL, "DSP" },
+	{ "AIF3 Mixer", NULL, "DSP" },
+	{ "DAC1 Mixer", NULL, "DSP" },
+	{ "DAC2 Mixer", NULL, "DSP" },
+	{ "DAC1 Mixer", NULL, "Playback Engine A" },
+	{ "DAC2 Mixer", NULL, "Playback Engine B" },
+	{ "Left ADC Mixer", NULL, "Recording Engine A" },
+	{ "Right ADC Mixer", NULL, "Recording Engine A" },
+
+	{ "AIF1 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk },
+	{ "AIF2 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk },
+	{ "AIF3 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk },
+	{ "AIF1 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk },
+	{ "AIF2 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk },
+	{ "AIF3 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk },
+
+	{ "AIF1 IN", NULL, "AIF1 CLK" },
+	{ "AIF1 OUT", NULL, "AIF1 CLK" },
+	{ "AIF2 IN", NULL, "AIF2 CLK" },
+	{ "AIF2 OUT", NULL, "AIF2 CLK" },
+	{ "AIF3 IN", NULL, "AIF3 CLK" },
+	{ "AIF3 OUT", NULL, "AIF3 CLK" },
+	{ "AIF1 IN", NULL, "AIF1 IN SRC", adau1373_check_src },
+	{ "AIF1 OUT", NULL, "AIF1 OUT SRC", adau1373_check_src },
+	{ "AIF2 IN", NULL, "AIF2 IN SRC", adau1373_check_src },
+	{ "AIF2 OUT", NULL, "AIF2 OUT SRC", adau1373_check_src },
+	{ "AIF3 IN", NULL, "AIF3 IN SRC", adau1373_check_src },
+	{ "AIF3 OUT", NULL, "AIF3 OUT SRC", adau1373_check_src },
+
+	{ "DMIC1", NULL, "DMIC1DAT" },
+	{ "DMIC1", NULL, "SYSCLK1" },
+	{ "DMIC1", NULL, "Recording Engine A" },
+	{ "DMIC2", NULL, "DMIC2DAT" },
+	{ "DMIC2", NULL, "SYSCLK1" },
+	{ "DMIC2", NULL, "Recording Engine B" },
+};
+
+static int adau1373_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 adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+	struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id];
+	unsigned int div;
+	unsigned int freq;
+	unsigned int ctrl;
+
+	freq = adau1373_dai->sysclk;
+
+	if (freq % params_rate(params) != 0)
+		return -EINVAL;
+
+	switch (freq / params_rate(params)) {
+	case 1024: /* sysclk / 256 */
+		div = 0;
+		break;
+	case 1536: /* 2/3 sysclk / 256 */
+		div = 1;
+		break;
+	case 2048: /* 1/2 sysclk / 256 */
+		div = 2;
+		break;
+	case 3072: /* 1/3 sysclk / 256 */
+		div = 3;
+		break;
+	case 4096: /* 1/4 sysclk / 256 */
+		div = 4;
+		break;
+	case 6144: /* 1/6 sysclk / 256 */
+		div = 5;
+		break;
+	case 5632: /* 2/11 sysclk / 256 */
+		div = 6;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	adau1373_dai->enable_src = (div != 0);
+
+	snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id),
+		~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		ctrl = ADAU1373_DAI_WLEN_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		ctrl = ADAU1373_DAI_WLEN_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		ctrl = ADAU1373_DAI_WLEN_24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		ctrl = ADAU1373_DAI_WLEN_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id),
+			ADAU1373_DAI_WLEN_MASK, ctrl);
+}
+
+static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec);
+	struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id];
+	unsigned int ctrl;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		ctrl = ADAU1373_DAI_MASTER;
+		adau1373_dai->master = true;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		ctrl = 0;
+		adau1373_dai->master = true;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		ctrl |= ADAU1373_DAI_FORMAT_I2S;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		ctrl |= ADAU1373_DAI_FORMAT_LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		ctrl |= ADAU1373_DAI_FORMAT_RIGHT_J;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		ctrl |= ADAU1373_DAI_FORMAT_DSP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		ctrl |= ADAU1373_DAI_INVERT_BCLK;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		ctrl |= ADAU1373_DAI_INVERT_LRCLK;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		ctrl |= ADAU1373_DAI_INVERT_LRCLK | ADAU1373_DAI_INVERT_BCLK;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ADAU1373_DAI(dai->id),
+		~ADAU1373_DAI_WLEN_MASK, ctrl);
+
+	return 0;
+}
+
+static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai,
+	int clk_id, unsigned int freq, int dir)
+{
+	struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(dai->codec);
+	struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id];
+
+	switch (clk_id) {
+	case ADAU1373_CLK_SRC_PLL1:
+	case ADAU1373_CLK_SRC_PLL2:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	adau1373_dai->sysclk = freq;
+	adau1373_dai->clk_src = clk_id;
+
+	snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id),
+		ADAU1373_BCLKDIV_SOURCE, clk_id << 5);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops adau1373_dai_ops = {
+	.hw_params	= adau1373_hw_params,
+	.set_sysclk	= adau1373_set_dai_sysclk,
+	.set_fmt	= adau1373_set_dai_fmt,
+};
+
+#define ADAU1373_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver adau1373_dai_driver[] = {
+	{
+		.id = 0,
+		.name = "adau1373-aif1",
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_48000,
+			.formats = ADAU1373_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF1 Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_48000,
+			.formats = ADAU1373_FORMATS,
+		},
+		.ops = &adau1373_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.id = 1,
+		.name = "adau1373-aif2",
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_48000,
+			.formats = ADAU1373_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF2 Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_48000,
+			.formats = ADAU1373_FORMATS,
+		},
+		.ops = &adau1373_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.id = 2,
+		.name = "adau1373-aif3",
+		.playback = {
+			.stream_name = "AIF3 Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_48000,
+			.formats = ADAU1373_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIF3 Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_48000,
+			.formats = ADAU1373_FORMATS,
+		},
+		.ops = &adau1373_dai_ops,
+		.symmetric_rates = 1,
+	},
+};
+
+static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id,
+	int source, unsigned int freq_in, unsigned int freq_out)
+{
+	unsigned int dpll_div = 0;
+	unsigned int x, r, n, m, i, j, mode;
+
+	switch (pll_id) {
+	case ADAU1373_PLL1:
+	case ADAU1373_PLL2:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (source) {
+	case ADAU1373_PLL_SRC_BCLK1:
+	case ADAU1373_PLL_SRC_BCLK2:
+	case ADAU1373_PLL_SRC_BCLK3:
+	case ADAU1373_PLL_SRC_LRCLK1:
+	case ADAU1373_PLL_SRC_LRCLK2:
+	case ADAU1373_PLL_SRC_LRCLK3:
+	case ADAU1373_PLL_SRC_MCLK1:
+	case ADAU1373_PLL_SRC_MCLK2:
+	case ADAU1373_PLL_SRC_GPIO1:
+	case ADAU1373_PLL_SRC_GPIO2:
+	case ADAU1373_PLL_SRC_GPIO3:
+	case ADAU1373_PLL_SRC_GPIO4:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (freq_in < 7813 || freq_in > 27000000)
+		return -EINVAL;
+
+	if (freq_out < 45158000 || freq_out > 49152000)
+		return -EINVAL;
+
+	/* APLL input needs to be >= 8Mhz, so in case freq_in is less we use the
+	 * DPLL to get it there. DPLL_out = (DPLL_in / div) * 1024 */
+	while (freq_in < 8000000) {
+		freq_in *= 2;
+		dpll_div++;
+	}
+
+	if (freq_out % freq_in != 0) {
+		/* fout = fin * (r + (n/m)) / x */
+		x = DIV_ROUND_UP(freq_in, 13500000);
+		freq_in /= x;
+		r = freq_out / freq_in;
+		i = freq_out % freq_in;
+		j = gcd(i, freq_in);
+		n = i / j;
+		m = freq_in / j;
+		x--;
+		mode = 1;
+	} else {
+		/* fout = fin / r */
+		r = freq_out / freq_in;
+		n = 0;
+		m = 0;
+		x = 0;
+		mode = 0;
+	}
+
+	if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff)
+		return -EINVAL;
+
+	if (dpll_div) {
+		dpll_div = 11 - dpll_div;
+		snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+			ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0);
+	} else {
+		snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id),
+			ADAU1373_PLL_CTRL6_DPLL_BYPASS,
+			ADAU1373_PLL_CTRL6_DPLL_BYPASS);
+	}
+
+	snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id),
+		(source << 4) | dpll_div);
+	snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff);
+	snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff);
+	snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff);
+	snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff);
+	snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id),
+		(r << 3) | (x << 1) | mode);
+
+	/* Set sysclk to pll_rate / 4 */
+	snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09);
+
+	return 0;
+}
+
+static void adau1373_load_drc_settings(struct snd_soc_codec *codec,
+	unsigned int nr, uint8_t *drc)
+{
+	unsigned int i;
+
+	for (i = 0; i < ADAU1373_DRC_SIZE; ++i)
+		snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]);
+}
+
+static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias)
+{
+	switch (micbias) {
+	case ADAU1373_MICBIAS_2_9V:
+	case ADAU1373_MICBIAS_2_2V:
+	case ADAU1373_MICBIAS_2_6V:
+	case ADAU1373_MICBIAS_1_8V:
+		return true;
+	default:
+		break;
+	}
+	return false;
+}
+
+static int adau1373_probe(struct snd_soc_codec *codec)
+{
+	struct adau1373_platform_data *pdata = codec->dev->platform_data;
+	bool lineout_differential = false;
+	unsigned int val;
+	int ret;
+	int i;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	if (ret) {
+		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	codec->dapm.idle_bias_off = true;
+
+	if (pdata) {
+		if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting))
+			return -EINVAL;
+
+		if (!adau1373_valid_micbias(pdata->micbias1) ||
+			!adau1373_valid_micbias(pdata->micbias2))
+			return -EINVAL;
+
+		for (i = 0; i < pdata->num_drc; ++i) {
+			adau1373_load_drc_settings(codec, i,
+				pdata->drc_setting[i]);
+		}
+
+		snd_soc_add_controls(codec, adau1373_drc_controls,
+			pdata->num_drc);
+
+		val = 0;
+		for (i = 0; i < 4; ++i) {
+			if (pdata->input_differential[i])
+				val |= BIT(i);
+		}
+		snd_soc_write(codec, ADAU1373_INPUT_MODE, val);
+
+		val = 0;
+		if (pdata->lineout_differential)
+			val |= ADAU1373_OUTPUT_CTRL_LDIFF;
+		if (pdata->lineout_ground_sense)
+			val |= ADAU1373_OUTPUT_CTRL_LNFBEN;
+		snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val);
+
+		lineout_differential = pdata->lineout_differential;
+
+		snd_soc_write(codec, ADAU1373_EP_CTRL,
+			(pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) |
+			(pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET));
+	}
+
+	if (!lineout_differential) {
+		snd_soc_add_controls(codec, adau1373_lineout2_controls,
+			ARRAY_SIZE(adau1373_lineout2_controls));
+	}
+
+	snd_soc_write(codec, ADAU1373_ADC_CTRL,
+	    ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT);
+
+	return 0;
+}
+
+static int adau1373_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3,
+			ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN);
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3,
+			ADAU1373_PWDN_CTRL3_PWR_EN, 0);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static int adau1373_remove(struct snd_soc_codec *codec)
+{
+	adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int adau1373_resume(struct snd_soc_codec *codec)
+{
+	adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_cache_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver adau1373_codec_driver = {
+	.probe =	adau1373_probe,
+	.remove =	adau1373_remove,
+	.suspend =	adau1373_suspend,
+	.resume =	adau1373_resume,
+	.set_bias_level = adau1373_set_bias_level,
+	.reg_cache_size = ARRAY_SIZE(adau1373_default_regs),
+	.reg_cache_default = adau1373_default_regs,
+	.reg_word_size = sizeof(uint8_t),
+
+	.set_pll = adau1373_set_pll,
+
+	.controls = adau1373_controls,
+	.num_controls = ARRAY_SIZE(adau1373_controls),
+	.dapm_widgets = adau1373_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(adau1373_dapm_widgets),
+	.dapm_routes = adau1373_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(adau1373_dapm_routes),
+};
+
+static int __devinit adau1373_i2c_probe(struct i2c_client *client,
+	const struct i2c_device_id *id)
+{
+	struct adau1373 *adau1373;
+	int ret;
+
+	adau1373 = kzalloc(sizeof(*adau1373), GFP_KERNEL);
+	if (!adau1373)
+		return -ENOMEM;
+
+	dev_set_drvdata(&client->dev, adau1373);
+
+	ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver,
+			adau1373_dai_driver, ARRAY_SIZE(adau1373_dai_driver));
+	if (ret < 0)
+		kfree(adau1373);
+
+	return ret;
+}
+
+static int __devexit adau1373_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	kfree(dev_get_drvdata(&client->dev));
+	return 0;
+}
+
+static const struct i2c_device_id adau1373_i2c_id[] = {
+	{ "adau1373", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id);
+
+static struct i2c_driver adau1373_i2c_driver = {
+	.driver = {
+		.name = "adau1373",
+		.owner = THIS_MODULE,
+	},
+	.probe = adau1373_i2c_probe,
+	.remove = __devexit_p(adau1373_i2c_remove),
+	.id_table = adau1373_i2c_id,
+};
+
+static int __init adau1373_init(void)
+{
+	return i2c_add_driver(&adau1373_i2c_driver);
+}
+module_init(adau1373_init);
+
+static void __exit adau1373_exit(void)
+{
+	i2c_del_driver(&adau1373_i2c_driver);
+}
+module_exit(adau1373_exit);
+
+MODULE_DESCRIPTION("ASoC ADAU1373 driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adau1373.h b/sound/soc/codecs/adau1373.h
new file mode 100644
index 00000000000..c6ab5530760
--- /dev/null
+++ b/sound/soc/codecs/adau1373.h
@@ -0,0 +1,29 @@
+#ifndef __ADAU1373_H__
+#define __ADAU1373_H__
+
+enum adau1373_pll_src {
+	ADAU1373_PLL_SRC_MCLK1 = 0,
+	ADAU1373_PLL_SRC_BCLK1 = 1,
+	ADAU1373_PLL_SRC_BCLK2 = 2,
+	ADAU1373_PLL_SRC_BCLK3 = 3,
+	ADAU1373_PLL_SRC_LRCLK1 = 4,
+	ADAU1373_PLL_SRC_LRCLK2 = 5,
+	ADAU1373_PLL_SRC_LRCLK3 = 6,
+	ADAU1373_PLL_SRC_GPIO1 = 7,
+	ADAU1373_PLL_SRC_GPIO2 = 8,
+	ADAU1373_PLL_SRC_GPIO3 = 9,
+	ADAU1373_PLL_SRC_GPIO4 = 10,
+	ADAU1373_PLL_SRC_MCLK2 = 11,
+};
+
+enum adau1373_pll {
+	ADAU1373_PLL1 = 0,
+	ADAU1373_PLL2 = 1,
+};
+
+enum adau1373_clk_src {
+	ADAU1373_CLK_SRC_PLL1 = 0,
+	ADAU1373_CLK_SRC_PLL2 = 1,
+};
+
+#endif
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 300c04b70e7..f9f08948e5e 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -523,7 +523,8 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int adav80x_set_sysclk(struct snd_soc_codec *codec,
-		int clk_id, unsigned int freq, int dir)
+			      int clk_id, int source,
+			      unsigned int freq, int dir)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index eecffb54894..05173159507 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -41,7 +41,6 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
 struct alc5623_priv {
 	enum snd_soc_control_type control_type;
 	void *control_data;
-	struct mutex mutex;
 	u8 id;
 	unsigned int sysclk;
 	u16 reg_cache[ALC5623_VENDOR_ID2+2];
@@ -1052,7 +1051,6 @@ static int alc5623_i2c_probe(struct i2c_client *client,
 	i2c_set_clientdata(client, alc5623);
 	alc5623->control_data = client;
 	alc5623->control_type = SND_SOC_I2C;
-	mutex_init(&alc5623->mutex);
 
 	ret =  snd_soc_register_codec(&client->dev,
 		&soc_codec_device_alc5623, &alc5623_dai, 1);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 7e4066e131e..91130fbc691 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -20,6 +20,7 @@
 #include <linux/regulator/driver.h>
 #include <linux/regulator/machine.h>
 #include <linux/regulator/consumer.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/tlv.h>
 #include <sound/pcm.h>
@@ -1436,10 +1437,17 @@ static const struct i2c_device_id sgtl5000_id[] = {
 
 MODULE_DEVICE_TABLE(i2c, sgtl5000_id);
 
+static const struct of_device_id sgtl5000_dt_ids[] = {
+	{ .compatible = "fsl,sgtl5000", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, sgtl5000_dt_ids);
+
 static struct i2c_driver sgtl5000_i2c_driver = {
 	.driver = {
 		   .name = "sgtl5000",
 		   .owner = THIS_MODULE,
+		   .of_match_table = sgtl5000_dt_ids,
 		   },
 	.probe = sgtl5000_i2c_probe,
 	.remove = __devexit_p(sgtl5000_i2c_remove),
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 84ffdebb8a8..29945b00413 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -79,7 +79,7 @@ static void configure_adc(struct snd_soc_codec *sn95031_codec, int val)
  */
 static int find_free_channel(struct snd_soc_codec *sn95031_codec)
 {
-	int ret = 0, i, value;
+	int i, value;
 
 	/* check whether ADC is enabled */
 	value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
@@ -91,12 +91,10 @@ static int find_free_channel(struct snd_soc_codec *sn95031_codec)
 	for (i = 0; i <	SN95031_ADC_CHANLS_MAX; i++) {
 		value = snd_soc_read(sn95031_codec,
 				SN95031_ADC_CHNL_START_ADDR + i);
-		if (value & SN95031_STOPBIT_MASK) {
-			ret = i;
+		if (value & SN95031_STOPBIT_MASK)
 			break;
-		}
 	}
-	return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret;
+	return (i == SN95031_ADC_CHANLS_MAX) ? (-EINVAL) : i;
 }
 
 /* Initialize the ADC for reading micbias values. Can sleep. */
@@ -660,7 +658,7 @@ static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute)
 	return 0;
 }
 
-int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
+static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
 {
 	unsigned int format, rate;
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 9801cd7cfcb..32d6c514186 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -294,7 +294,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
-	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
 
 	/* The DAI has shared clocks so if we already have a playback or
@@ -303,7 +302,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 	 */
 	if (ssm2602->master_substream) {
 		master_runtime = ssm2602->master_substream->runtime;
-		dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
+		dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n",
 			master_runtime->sample_bits,
 			master_runtime->rate);
 
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index fbd7eb9e61c..5c7def3979c 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -524,13 +524,17 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream,
 	rate = params_rate(params);
 	pr_debug("rate: %u\n", rate);
 	for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++)
-		if (interpolation_ratios[i].fs == rate)
+		if (interpolation_ratios[i].fs == rate) {
 			ir = interpolation_ratios[i].ir;
+			break;
+		}
 	if (ir < 0)
 		return -EINVAL;
 	for (i = 0; mclk_ratios[ir][i].ratio; i++)
-		if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk)
+		if (mclk_ratios[ir][i].ratio * rate == sta32x->mclk) {
 			mcs = mclk_ratios[ir][i].mcs;
+			break;
+		}
 	if (mcs < 0)
 		return -EINVAL;
 
@@ -808,6 +812,7 @@ static int sta32x_remove(struct snd_soc_codec *codec)
 {
 	struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
 
+	sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
 	regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
 
@@ -867,18 +872,8 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c,
 static __devexit int sta32x_i2c_remove(struct i2c_client *client)
 {
 	struct sta32x_priv *sta32x = i2c_get_clientdata(client);
-	struct snd_soc_codec *codec = sta32x->codec;
-
-	if (codec)
-		sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
-
-	if (codec) {
-		snd_soc_unregister_codec(&client->dev);
-		snd_soc_codec_set_drvdata(codec, NULL);
-	}
 
+	snd_soc_unregister_codec(&client->dev);
 	kfree(sta32x);
 	return 0;
 }
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index faa5e9fb147..243d1771121 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1431,7 +1431,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
 	/* 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,
+				  IRQF_TRIGGER_RISING,
 				  codec->name, codec);
 		if (ret < 0) {
 			dev_err(codec->dev, "Could not request IRQ%d (%d)\n",
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 239e0c46106..b2572c451c3 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -446,7 +446,6 @@ err_regulator:
 		gpio_free(data->power_gpio);
 err_gpio:
 	kfree(data);
-	i2c_set_clientdata(tpa6130a2_client, NULL);
 	tpa6130a2_client = NULL;
 
 	return ret;
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 443032b3b32..81645c63244 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -118,8 +118,8 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = {
 	0x4A, /* TWL6040_LPPLLDIV	0x09	*/
 	0x00, /* TWL6040_AMICBCTL	0x0A	*/
 	0x00, /* TWL6040_DMICBCTL	0x0B	*/
-	0x18, /* TWL6040_MICLCTL	0x0C	- No input selected on Left Mic */
-	0x18, /* TWL6040_MICRCTL	0x0D	- No input selected on Right Mic */
+	0x00, /* TWL6040_MICLCTL	0x0C	*/
+	0x00, /* TWL6040_MICRCTL	0x0D	*/
 	0x00, /* TWL6040_MICGAIN	0x0E	*/
 	0x1B, /* TWL6040_LINEGAIN	0x0F	*/
 	0x00, /* TWL6040_HSLCTL		0x10	*/
@@ -155,41 +155,8 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = {
 	0x00, /* TWL6040_STATUS (ro)	0x2E	*/
 };
 
-/*
- * twl6040 vio/gnd registers:
- * registers under vio/gnd supply can be accessed
- * before the power-up sequence, after NRESPWRON goes high
- */
-static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = {
-	TWL6040_REG_ASICID,
-	TWL6040_REG_ASICREV,
-	TWL6040_REG_INTID,
-	TWL6040_REG_INTMR,
-	TWL6040_REG_NCPCTL,
-	TWL6040_REG_LDOCTL,
-	TWL6040_REG_AMICBCTL,
-	TWL6040_REG_DMICBCTL,
-	TWL6040_REG_HKCTL1,
-	TWL6040_REG_HKCTL2,
-	TWL6040_REG_GPOCTL,
-	TWL6040_REG_TRIM1,
-	TWL6040_REG_TRIM2,
-	TWL6040_REG_TRIM3,
-	TWL6040_REG_HSOTRIM,
-	TWL6040_REG_HFOTRIM,
-	TWL6040_REG_ACCCTL,
-	TWL6040_REG_STATUS,
-};
-
-/*
- * twl6040 vdd/vss registers:
- * registers under vdd/vss supplies can only be accessed
- * after the power-up sequence
- */
-static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = {
-	TWL6040_REG_HPPLLCTL,
-	TWL6040_REG_LPPLLCTL,
-	TWL6040_REG_LPPLLDIV,
+/* List of registers to be restored after power up */
+static const int twl6040_restore_list[] = {
 	TWL6040_REG_MICLCTL,
 	TWL6040_REG_MICRCTL,
 	TWL6040_REG_MICGAIN,
@@ -202,12 +169,6 @@ static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = {
 	TWL6040_REG_HFLGAIN,
 	TWL6040_REG_HFRCTL,
 	TWL6040_REG_HFRGAIN,
-	TWL6040_REG_VIBCTLL,
-	TWL6040_REG_VIBDATL,
-	TWL6040_REG_VIBCTLR,
-	TWL6040_REG_VIBDATR,
-	TWL6040_REG_ALB,
-	TWL6040_REG_DLB,
 };
 
 /* set of rates for each pll: low-power and high-performance */
@@ -296,56 +257,27 @@ static int twl6040_write(struct snd_soc_codec *codec,
 	return twl6040_reg_write(twl6040, reg, value);
 }
 
-static void twl6040_init_vio_regs(struct snd_soc_codec *codec)
+static void twl6040_init_chip(struct snd_soc_codec *codec)
 {
-	u8 *cache = codec->reg_cache;
-	int reg, i;
+	struct twl6040 *twl6040 = codec->control_data;
+	u8 val;
 
-	for (i = 0; i < TWL6040_VIOREGNUM; i++) {
-		reg = twl6040_vio_reg[i];
-		/*
-		 * skip read-only registers (ASICID, ASICREV, STATUS)
-		 * and registers shared among MFD children
-		 */
-		switch (reg) {
-		case TWL6040_REG_ASICID:
-		case TWL6040_REG_ASICREV:
-		case TWL6040_REG_INTID:
-		case TWL6040_REG_INTMR:
-		case TWL6040_REG_NCPCTL:
-		case TWL6040_REG_LDOCTL:
-		case TWL6040_REG_GPOCTL:
-		case TWL6040_REG_ACCCTL:
-		case TWL6040_REG_STATUS:
-			continue;
-		default:
-			break;
-		}
-		twl6040_write(codec, reg, cache[reg]);
-	}
+	val = twl6040_get_revid(twl6040);
+	twl6040_write_reg_cache(codec, TWL6040_REG_ASICREV, val);
+
+	/* Change chip defaults */
+	/* No imput selected for microphone amplifiers */
+	twl6040_write_reg_cache(codec, TWL6040_REG_MICLCTL, 0x18);
+	twl6040_write_reg_cache(codec, TWL6040_REG_MICRCTL, 0x18);
 }
 
-static void twl6040_init_vdd_regs(struct snd_soc_codec *codec)
+static void twl6040_restore_regs(struct snd_soc_codec *codec)
 {
 	u8 *cache = codec->reg_cache;
 	int reg, i;
 
-	for (i = 0; i < TWL6040_VDDREGNUM; i++) {
-		reg = twl6040_vdd_reg[i];
-		/* skip vibra and PLL registers */
-		switch (reg) {
-		case TWL6040_REG_VIBCTLL:
-		case TWL6040_REG_VIBDATL:
-		case TWL6040_REG_VIBCTLR:
-		case TWL6040_REG_VIBDATR:
-		case TWL6040_REG_HPPLLCTL:
-		case TWL6040_REG_LPPLLCTL:
-		case TWL6040_REG_LPPLLDIV:
-			continue;
-		default:
-			break;
-		}
-
+	for (i = 0; i < ARRAY_SIZE(twl6040_restore_list); i++) {
+		reg = twl6040_restore_list[i];
 		twl6040_write(codec, reg, cache[reg]);
 	}
 }
@@ -1325,8 +1257,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec,
 
 		priv->codec_powered = 1;
 
-		/* initialize vdd/vss registers with reg_cache */
-		twl6040_init_vdd_regs(codec);
+		twl6040_restore_regs(codec);
 
 		/* Set external boost GPO */
 		twl6040_write(codec, TWL6040_REG_GPOCTL, 0x02);
@@ -1468,7 +1399,7 @@ static struct snd_soc_dai_driver twl6040_dai[] = {
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
-		.channels_max = 2,
+		.channels_max = 5,
 		.rates = TWL6040_RATES,
 		.formats = TWL6040_FORMATS,
 	},
@@ -1518,8 +1449,8 @@ static struct snd_soc_dai_driver twl6040_dai[] = {
 	.name = "twl6040-vib",
 	.playback = {
 		.stream_name = "Vibra Playback",
-		.channels_min = 2,
-		.channels_max = 2,
+		.channels_min = 1,
+		.channels_max = 1,
 		.rates = SNDRV_PCM_RATE_CONTINUOUS,
 		.formats = TWL6040_FORMATS,
 	},
@@ -1620,8 +1551,7 @@ static int twl6040_probe(struct snd_soc_codec *codec)
 		goto plugirq_err;
 	}
 
-	/* init vio registers */
-	twl6040_init_vio_regs(codec);
+	twl6040_init_chip(codec);
 
 	/* power on device */
 	ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index bcc20896791..4523c4cec02 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -56,8 +56,26 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = {
 };
 
 static int __devinit wm1250_ev1_probe(struct i2c_client *i2c,
-				      const struct i2c_device_id *id)
+				      const struct i2c_device_id *i2c_id)
 {
+	int id, board, rev;
+
+	board = i2c_smbus_read_byte_data(i2c, 0);
+	if (board < 0) {
+		dev_err(&i2c->dev, "Failed to read ID: %d\n", board);
+		return board;
+	}
+
+	id = (board & 0xfe) >> 2;
+	rev = board & 0x3;
+
+	if (id != 1) {
+		dev_err(&i2c->dev, "Unknown board ID %d\n", id);
+		return -ENODEV;
+	}
+
+	dev_info(&i2c->dev, "revision %d\n", rev + 1);
+
 	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1,
 				      &wm1250_ev1_dai, 1);
 }
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index db0dced7484..55a4c830e11 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -20,6 +20,7 @@
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
 #include <linux/slab.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -598,6 +599,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8510 = {
 	.reg_cache_default =wm8510_reg,
 };
 
+static const struct of_device_id wm8510_of_match[] = {
+	{ .compatible = "wlf,wm8510" },
+	{ },
+};
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8510_spi_probe(struct spi_device *spi)
 {
@@ -628,6 +634,7 @@ static struct spi_driver wm8510_spi_driver = {
 	.driver = {
 		.name	= "wm8510",
 		.owner	= THIS_MODULE,
+		.of_match_table = wm8510_of_match,
 	},
 	.probe		= wm8510_spi_probe,
 	.remove		= __devexit_p(wm8510_spi_remove),
@@ -671,6 +678,7 @@ static struct i2c_driver wm8510_i2c_driver = {
 	.driver = {
 		.name = "wm8510-codec",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8510_of_match,
 	},
 	.probe =    wm8510_i2c_probe,
 	.remove =   __devexit_p(wm8510_i2c_remove),
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 4fd4d8dca0f..5355a7a944f 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -20,6 +20,7 @@
 #include <linux/platform_device.h>
 #include <linux/regulator/consumer.h>
 #include <linux/slab.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -84,7 +85,7 @@ static const char *wm8523_zd_count_text[] = {
 static const struct soc_enum wm8523_zc_count =
 	SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text);
 
-static const struct snd_kcontrol_new wm8523_snd_controls[] = {
+static const struct snd_kcontrol_new wm8523_controls[] = {
 SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR,
 		 0, 448, 0, dac_tlv),
 SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0),
@@ -101,22 +102,11 @@ SND_SOC_DAPM_OUTPUT("LINEVOUTL"),
 SND_SOC_DAPM_OUTPUT("LINEVOUTR"),
 };
 
-static const struct snd_soc_dapm_route intercon[] = {
+static const struct snd_soc_dapm_route wm8523_dapm_routes[] = {
 	{ "LINEVOUTL", NULL, "DAC" },
 	{ "LINEVOUTR", NULL, "DAC" },
 };
 
-static int wm8523_add_widgets(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, wm8523_dapm_widgets,
-				  ARRAY_SIZE(wm8523_dapm_widgets));
-	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
-	return 0;
-}
-
 static struct {
 	int value;
 	int ratio;
@@ -479,10 +469,6 @@ static int wm8523_probe(struct snd_soc_codec *codec)
 	/* Bias level configuration will have done an extra enable */
 	regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies);
 
-	snd_soc_add_controls(codec, wm8523_snd_controls,
-			     ARRAY_SIZE(wm8523_snd_controls));
-	wm8523_add_widgets(codec);
-
 	return 0;
 
 err_enable:
@@ -512,6 +498,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8523 = {
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8523_reg,
 	.volatile_register = wm8523_volatile_register,
+
+	.controls = wm8523_controls,
+	.num_controls = ARRAY_SIZE(wm8523_controls),
+	.dapm_widgets = wm8523_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm8523_dapm_widgets),
+	.dapm_routes = wm8523_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm8523_dapm_routes),
+};
+
+static const struct of_device_id wm8523_of_match[] = {
+	{ .compatible = "wlf,wm8523" },
+	{ },
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -551,8 +549,9 @@ MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id);
 
 static struct i2c_driver wm8523_i2c_driver = {
 	.driver = {
-		.name = "wm8523-codec",
+		.name = "wm8523",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8523_of_match,
 	},
 	.probe =    wm8523_i2c_probe,
 	.remove =   __devexit_p(wm8523_i2c_remove),
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 4bbc0a79f01..4664c3a76c7 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -26,6 +26,7 @@
 #include <linux/platform_device.h>
 #include <linux/regulator/consumer.h>
 #include <linux/slab.h>
+#include <linux/of_device.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -907,6 +908,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8580 = {
 	.reg_cache_default = wm8580_reg,
 };
 
+static const struct of_device_id wm8580_of_match[] = {
+	{ .compatible = "wlf,wm8580" },
+	{ },
+};
+
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static int wm8580_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
@@ -943,8 +949,9 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
 
 static struct i2c_driver wm8580_i2c_driver = {
 	.driver = {
-		.name = "wm8580-codec",
+		.name = "wm8580",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8580_of_match,
 	},
 	.probe =    wm8580_i2c_probe,
 	.remove =   wm8580_i2c_remove,
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index a537e4af6ae..8457d3cb596 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -21,6 +21,7 @@
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
 #include <linux/slab.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -414,6 +415,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8711 = {
 	.num_dapm_routes = ARRAY_SIZE(wm8711_intercon),
 };
 
+static const struct of_device_id wm8711_of_match[] = {
+	{ .compatible = "wlf,wm8711", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8711_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8711_spi_probe(struct spi_device *spi)
 {
@@ -443,8 +450,9 @@ static int __devexit wm8711_spi_remove(struct spi_device *spi)
 
 static struct spi_driver wm8711_spi_driver = {
 	.driver = {
-		.name	= "wm8711-codec",
+		.name	= "wm8711",
 		.owner	= THIS_MODULE,
+		.of_match_table = wm8711_of_match,
 	},
 	.probe		= wm8711_spi_probe,
 	.remove		= __devexit_p(wm8711_spi_remove),
@@ -487,8 +495,9 @@ MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id);
 
 static struct i2c_driver wm8711_i2c_driver = {
 	.driver = {
-		.name = "wm8711-codec",
+		.name = "wm8711",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8711_of_match,
 	},
 	.probe =    wm8711_i2c_probe,
 	.remove =   __devexit_p(wm8711_i2c_remove),
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 86d4718d3a7..04b027efd5c 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -19,6 +19,7 @@
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
 #include <linux/slab.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -269,6 +270,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8728 = {
 	.num_dapm_routes = ARRAY_SIZE(wm8728_intercon),
 };
 
+static const struct of_device_id wm8728_of_match[] = {
+	{ .compatible = "wlf,wm8728", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8728_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8728_spi_probe(struct spi_device *spi)
 {
@@ -298,8 +305,9 @@ static int __devexit wm8728_spi_remove(struct spi_device *spi)
 
 static struct spi_driver wm8728_spi_driver = {
 	.driver = {
-		.name	= "wm8728-codec",
+		.name	= "wm8728",
 		.owner	= THIS_MODULE,
+		.of_match_table = wm8728_of_match,
 	},
 	.probe		= wm8728_spi_probe,
 	.remove		= __devexit_p(wm8728_spi_remove),
@@ -342,8 +350,9 @@ MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id);
 
 static struct i2c_driver wm8728_i2c_driver = {
 	.driver = {
-		.name = "wm8728-codec",
+		.name = "wm8728",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8728_of_match,
 	},
 	.probe =    wm8728_i2c_probe,
 	.remove =   __devexit_p(wm8728_i2c_remove),
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 76b4361e9b8..f76b6fc6766 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -22,6 +22,7 @@
 #include <linux/platform_device.h>
 #include <linux/regulator/consumer.h>
 #include <linux/spi/spi.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -607,6 +608,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = {
 	.num_dapm_routes = ARRAY_SIZE(wm8731_intercon),
 };
 
+static const struct of_device_id wm8731_of_match[] = {
+	{ .compatible = "wlf,wm8731", },
+	{ }
+};
+
+MODULE_DEVICE_TABLE(of, wm8731_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8731_spi_probe(struct spi_device *spi)
 {
@@ -638,6 +646,7 @@ static struct spi_driver wm8731_spi_driver = {
 	.driver = {
 		.name	= "wm8731",
 		.owner	= THIS_MODULE,
+		.of_match_table = wm8731_of_match,
 	},
 	.probe		= wm8731_spi_probe,
 	.remove		= __devexit_p(wm8731_spi_remove),
@@ -682,6 +691,7 @@ static struct i2c_driver wm8731_i2c_driver = {
 	.driver = {
 		.name = "wm8731",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8731_of_match,
 	},
 	.probe =    wm8731_i2c_probe,
 	.remove =   __devexit_p(wm8731_i2c_remove),
diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c
index 30c67d06a90..f6aef58845c 100644
--- a/sound/soc/codecs/wm8737.c
+++ b/sound/soc/codecs/wm8737.c
@@ -20,6 +20,7 @@
 #include <linux/regulator/consumer.h>
 #include <linux/spi/spi.h>
 #include <linux/slab.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -634,6 +635,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8737 = {
 	.reg_cache_default = wm8737_reg,
 };
 
+static const struct of_device_id wm8737_of_match[] = {
+	{ .compatible = "wlf,wm8737", },
+	{ }
+};
+
+MODULE_DEVICE_TABLE(of, wm8737_of_match);
+
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static __devinit int wm8737_i2c_probe(struct i2c_client *i2c,
 				      const struct i2c_device_id *id)
@@ -673,6 +681,7 @@ static struct i2c_driver wm8737_i2c_driver = {
 	.driver = {
 		.name = "wm8737",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8737_of_match,
 	},
 	.probe =    wm8737_i2c_probe,
 	.remove =   __devexit_p(wm8737_i2c_remove),
@@ -711,6 +720,7 @@ static struct spi_driver wm8737_spi_driver = {
 	.driver = {
 		.name	= "wm8737",
 		.owner	= THIS_MODULE,
+		.of_match_table = wm8737_of_match,
 	},
 	.probe		= wm8737_spi_probe,
 	.remove		= __devexit_p(wm8737_spi_remove),
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 25af901fe81..78c9e5ab3fa 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -17,9 +17,11 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/spi/spi.h>
 #include <linux/platform_device.h>
 #include <linux/regulator/consumer.h>
 #include <linux/slab.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -422,17 +424,35 @@ static int wm8741_probe(struct snd_soc_codec *codec)
 {
 	struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++)
+		wm8741->supplies[i].supply = wm8741_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8741->supplies),
+				 wm8741->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		goto err;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies),
+				    wm8741->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_get;
+	}
 
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
-		return ret;
+		goto err_enable;
 	}
 
 	ret = wm8741_reset(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset\n");
-		return ret;
+		goto err_enable;
 	}
 
 	/* Change some default settings - latch VU */
@@ -451,58 +471,61 @@ static int wm8741_probe(struct snd_soc_codec *codec)
 
 	dev_dbg(codec->dev, "Successful registration\n");
 	return ret;
+
+err_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+err_get:
+	regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+err:
+	return ret;
+}
+
+static int wm8741_remove(struct snd_soc_codec *codec)
+{
+	struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec);
+
+	regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+	regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+
+	return 0;
 }
 
 static struct snd_soc_codec_driver soc_codec_dev_wm8741 = {
 	.probe =	wm8741_probe,
+	.remove =	wm8741_remove,
 	.resume =	wm8741_resume,
 	.reg_cache_size = ARRAY_SIZE(wm8741_reg_defaults),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8741_reg_defaults,
 };
 
+static const struct of_device_id wm8741_of_match[] = {
+	{ .compatible = "wlf,wm8741", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8741_of_match);
+
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static int wm8741_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct wm8741_priv *wm8741;
-	int ret, i;
+	int ret;
 
 	wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL);
 	if (wm8741 == NULL)
 		return -ENOMEM;
 
-	for (i = 0; i < ARRAY_SIZE(wm8741->supplies); i++)
-		wm8741->supplies[i].supply = wm8741_supply_names[i];
-
-	ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8741->supplies),
-				 wm8741->supplies);
-	if (ret != 0) {
-		dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
-		goto err;
-	}
-
-	ret = regulator_bulk_enable(ARRAY_SIZE(wm8741->supplies),
-				    wm8741->supplies);
-	if (ret != 0) {
-		dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
-		goto err_get;
-	}
-
 	i2c_set_clientdata(i2c, wm8741);
 	wm8741->control_type = SND_SOC_I2C;
 
-	ret =  snd_soc_register_codec(&i2c->dev,
-			&soc_codec_dev_wm8741, &wm8741_dai, 1);
-	if (ret < 0)
-		goto err_enable;
-	return ret;
+	ret = snd_soc_register_codec(&i2c->dev,
+				     &soc_codec_dev_wm8741, &wm8741_dai, 1);
+	if (ret != 0)
+		goto err;
 
-err_enable:
-	regulator_bulk_disable(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
+	return ret;
 
-err_get:
-	regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
 err:
 	kfree(wm8741);
 	return ret;
@@ -510,10 +533,7 @@ err:
 
 static int wm8741_i2c_remove(struct i2c_client *client)
 {
-	struct wm8741_priv *wm8741 = i2c_get_clientdata(client);
-
 	snd_soc_unregister_codec(&client->dev);
-	regulator_bulk_free(ARRAY_SIZE(wm8741->supplies), wm8741->supplies);
 	kfree(i2c_get_clientdata(client));
 	return 0;
 }
@@ -526,8 +546,9 @@ MODULE_DEVICE_TABLE(i2c, wm8741_i2c_id);
 
 static struct i2c_driver wm8741_i2c_driver = {
 	.driver = {
-		.name = "wm8741-codec",
+		.name = "wm8741",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8741_of_match,
 	},
 	.probe =    wm8741_i2c_probe,
 	.remove =   wm8741_i2c_remove,
@@ -535,6 +556,44 @@ static struct i2c_driver wm8741_i2c_driver = {
 };
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8741_spi_probe(struct spi_device *spi)
+{
+	struct wm8741_priv *wm8741;
+	int ret;
+
+	wm8741 = kzalloc(sizeof(struct wm8741_priv), GFP_KERNEL);
+	if (wm8741 == NULL)
+		return -ENOMEM;
+
+	wm8741->control_type = SND_SOC_SPI;
+	spi_set_drvdata(spi, wm8741);
+
+	ret = snd_soc_register_codec(&spi->dev,
+			&soc_codec_dev_wm8741, &wm8741_dai, 1);
+	if (ret < 0)
+		kfree(wm8741);
+	return ret;
+}
+
+static int __devexit wm8741_spi_remove(struct spi_device *spi)
+{
+	snd_soc_unregister_codec(&spi->dev);
+	kfree(spi_get_drvdata(spi));
+	return 0;
+}
+
+static struct spi_driver wm8741_spi_driver = {
+	.driver = {
+		.name	= "wm8741",
+		.owner	= THIS_MODULE,
+		.of_match_table = wm8741_of_match,
+	},
+	.probe		= wm8741_spi_probe,
+	.remove		= __devexit_p(wm8741_spi_remove),
+};
+#endif /* CONFIG_SPI_MASTER */
+
 static int __init wm8741_modinit(void)
 {
 	int ret = 0;
@@ -544,6 +603,13 @@ static int __init wm8741_modinit(void)
 	if (ret != 0)
 		pr_err("Failed to register WM8741 I2C driver: %d\n", ret);
 #endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&wm8741_spi_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register wm8741 SPI driver: %d\n",
+		       ret);
+	}
+#endif
 
 	return ret;
 }
@@ -551,6 +617,9 @@ module_init(wm8741_modinit);
 
 static void __exit wm8741_exit(void)
 {
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8741_spi_driver);
+#endif
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	i2c_del_driver(&wm8741_i2c_driver);
 #endif
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index d0003cc3bcd..15f03721ec6 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -21,6 +21,7 @@
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
 #include <linux/slab.h>
+#include <linux/of_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -751,6 +752,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8750 = {
 	.reg_cache_default = wm8750_reg,
 };
 
+static const struct of_device_id wm8750_of_match[] = {
+	{ .compatible = "wlf,wm8750", },
+	{ .compatible = "wlf,wm8987", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8750_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8750_spi_probe(struct spi_device *spi)
 {
@@ -787,8 +795,9 @@ MODULE_DEVICE_TABLE(spi, wm8750_spi_ids);
 
 static struct spi_driver wm8750_spi_driver = {
 	.driver = {
-		.name	= "wm8750-codec",
+		.name	= "wm8750",
 		.owner	= THIS_MODULE,
+		.of_match_table = wm8750_of_match,
 	},
 	.id_table	= wm8750_spi_ids,
 	.probe		= wm8750_spi_probe,
@@ -833,8 +842,9 @@ MODULE_DEVICE_TABLE(i2c, wm8750_i2c_id);
 
 static struct i2c_driver wm8750_i2c_driver = {
 	.driver = {
-		.name = "wm8750-codec",
+		.name = "wm8750",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8750_of_match,
 	},
 	.probe =    wm8750_i2c_probe,
 	.remove =   __devexit_p(wm8750_i2c_remove),
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index ffa2ffe5ec1..fe04a101d65 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -38,6 +38,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/of_device.h>
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
 #include <linux/slab.h>
@@ -1490,6 +1491,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = {
 	.reg_cache_default = wm8753_reg,
 };
 
+static const struct of_device_id wm8753_of_match[] = {
+	{ .compatible = "wlf,wm8753", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8753_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8753_spi_probe(struct spi_device *spi)
 {
@@ -1519,8 +1526,9 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi)
 
 static struct spi_driver wm8753_spi_driver = {
 	.driver = {
-		.name	= "wm8753-codec",
+		.name	= "wm8753",
 		.owner	= THIS_MODULE,
+		.of_match_table = wm8753_of_match,
 	},
 	.probe		= wm8753_spi_probe,
 	.remove		= __devexit_p(wm8753_spi_remove),
@@ -1563,8 +1571,9 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
 
 static struct i2c_driver wm8753_i2c_driver = {
 	.driver = {
-		.name = "wm8753-codec",
+		.name = "wm8753",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8753_of_match,
 	},
 	.probe =    wm8753_i2c_probe,
 	.remove =   __devexit_p(wm8753_i2c_remove),
diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c
index 19b92baa9e8..aa05e6507f8 100644
--- a/sound/soc/codecs/wm8770.c
+++ b/sound/soc/codecs/wm8770.c
@@ -14,6 +14,7 @@
 #include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
+#include <linux/of_device.h>
 #include <linux/pm.h>
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
@@ -684,6 +685,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = {
 	.reg_cache_default = wm8770_reg_defs
 };
 
+static const struct of_device_id wm8770_of_match[] = {
+	{ .compatible = "wlf,wm8770", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8770_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8770_spi_probe(struct spi_device *spi)
 {
@@ -715,6 +722,7 @@ static struct spi_driver wm8770_spi_driver = {
 	.driver = {
 		.name = "wm8770",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8770_of_match,
 	},
 	.probe = wm8770_spi_probe,
 	.remove = __devexit_p(wm8770_spi_remove)
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 8e7953b1b79..00d8846fae8 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -18,6 +18,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/of_device.h>
 #include <linux/platform_device.h>
 #include <linux/spi/spi.h>
 #include <linux/slab.h>
@@ -215,8 +216,6 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream,
 	int ratio_shift, master;
 	int i;
 
-	iface = 0;
-
 	switch (dai->driver->id) {
 	case WM8776_DAI_DAC:
 		iface_reg = WM8776_DACIFCTRL;
@@ -232,20 +231,23 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-
 	/* Set word length */
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S16_LE:
+	switch (snd_pcm_format_width(params_format(params))) {
+	case 16:
+		iface = 0;
+	case 20:
+		iface = 0x10;
 		break;
-	case SNDRV_PCM_FORMAT_S20_3LE:
-		iface |= 0x10;
+	case 24:
+		iface = 0x20;
 		break;
-	case SNDRV_PCM_FORMAT_S24_LE:
-		iface |= 0x20;
-		break;
-	case SNDRV_PCM_FORMAT_S32_LE:
-		iface |= 0x30;
+	case 32:
+		iface = 0x30;
 		break;
+	default:
+		dev_err(codec->dev, "Unsupported sample size: %i\n",
+			snd_pcm_format_width(params_format(params)));
+		return -EINVAL;
 	}
 
 	/* Only need to set MCLK/LRCLK ratio if we're master */
@@ -320,11 +322,6 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
-		      SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
-		      SNDRV_PCM_RATE_96000)
-
-
 #define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
@@ -349,7 +346,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = {
 			.stream_name = "Playback",
 			.channels_min = 2,
 			.channels_max = 2,
-			.rates = WM8776_RATES,
+			.rates = SNDRV_PCM_RATE_CONTINUOUS,
+			.rate_min = 32000,
+			.rate_max = 192000,
 			.formats = WM8776_FORMATS,
 		},
 		.ops = &wm8776_dac_ops,
@@ -361,7 +360,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = {
 			.stream_name = "Capture",
 			.channels_min = 2,
 			.channels_max = 2,
-			.rates = WM8776_RATES,
+			.rates = SNDRV_PCM_RATE_CONTINUOUS,
+			.rate_min = 32000,
+			.rate_max = 96000,
 			.formats = WM8776_FORMATS,
 		},
 		.ops = &wm8776_adc_ops,
@@ -452,6 +453,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8776 = {
 	.reg_cache_default = wm8776_reg,
 };
 
+static const struct of_device_id wm8776_of_match[] = {
+	{ .compatible = "wlf,wm8776", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8776_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8776_spi_probe(struct spi_device *spi)
 {
@@ -481,8 +488,9 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi)
 
 static struct spi_driver wm8776_spi_driver = {
 	.driver = {
-		.name	= "wm8776-codec",
+		.name	= "wm8776",
 		.owner	= THIS_MODULE,
+		.of_match_table = wm8776_of_match,
 	},
 	.probe		= wm8776_spi_probe,
 	.remove		= __devexit_p(wm8776_spi_remove),
@@ -525,8 +533,9 @@ MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id);
 
 static struct i2c_driver wm8776_i2c_driver = {
 	.driver = {
-		.name = "wm8776-codec",
+		.name = "wm8776",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8776_of_match,
 	},
 	.probe =    wm8776_i2c_probe,
 	.remove =   __devexit_p(wm8776_i2c_remove),
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 9a5e67c5a6b..9ee072b8597 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -16,6 +16,7 @@
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
+#include <linux/of_device.h>
 #include <linux/spi/spi.h>
 #include <linux/regulator/consumer.h>
 #include <linux/slab.h>
@@ -717,6 +718,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = {
 	.volatile_register = wm8804_volatile
 };
 
+static const struct of_device_id wm8804_of_match[] = {
+	{ .compatible = "wlf,wm8804", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, wm8804_of_match);
+
 #if defined(CONFIG_SPI_MASTER)
 static int __devinit wm8804_spi_probe(struct spi_device *spi)
 {
@@ -748,6 +755,7 @@ static struct spi_driver wm8804_spi_driver = {
 	.driver = {
 		.name = "wm8804",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8804_of_match,
 	},
 	.probe = wm8804_spi_probe,
 	.remove = __devexit_p(wm8804_spi_remove)
@@ -792,6 +800,7 @@ static struct i2c_driver wm8804_i2c_driver = {
 	.driver = {
 		.name = "wm8804",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8804_of_match,
 	},
 	.probe = wm8804_i2c_probe,
 	.remove = __devexit_p(wm8804_i2c_remove),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index d2c315fa1b9..3676b38838d 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -63,6 +63,8 @@ struct wm8962_priv {
 	int fll_fref;
 	int fll_fout;
 
+	u16 dsp2_ena;
+
 	struct delayed_work mic_work;
 	struct snd_soc_jack *jack;
 
@@ -837,7 +839,7 @@ static const struct wm8962_reg_access {
 	[40] = { 0x00FF, 0x01FF, 0x0000 }, /* R40    - SPKOUTL volume */
 	[41] = { 0x00FF, 0x01FF, 0x0000 }, /* R41    - SPKOUTR volume */
 
-	[47] = { 0x000F, 0x0000, 0x0000 }, /* R47    - Thermal Shutdown Status */
+	[47] = { 0x000F, 0x0000, 0xFFFF }, /* R47    - Thermal Shutdown Status */
 	[48] = { 0x7EC7, 0x7E07, 0xFFFF }, /* R48    - Additional Control (4) */
 	[49] = { 0x00D3, 0x00D7, 0xFFFF }, /* R49    - Class D Control 1 */
 	[51] = { 0x0047, 0x0047, 0x0000 }, /* R51    - Class D Control 2 */
@@ -965,7 +967,7 @@ static const struct wm8962_reg_access {
 	[584] = { 0x002D, 0x002D, 0x0000 }, /* R584   - IRQ Debounce */
 	[586] = { 0xC000, 0xC000, 0x0000 }, /* R586   -  MICINT Source Pol */
 	[768] = { 0x0001, 0x0001, 0x0000 }, /* R768   - DSP2 Power Management */
-	[1037] = { 0x0000, 0x003F, 0x0000 }, /* R1037  - DSP2_ExecControl */
+	[1037] = { 0x0000, 0x003F, 0xFFFF }, /* R1037  - DSP2_ExecControl */
 	[4096] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4096  - Write Sequencer 0 */
 	[4097] = { 0x00FF, 0x00FF, 0x0000 }, /* R4097  - Write Sequencer 1 */
 	[4098] = { 0x070F, 0x070F, 0x0000 }, /* R4098  - Write Sequencer 2 */
@@ -1986,6 +1988,122 @@ static const unsigned int classd_tlv[] = {
 };
 static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 
+static int wm8962_dsp2_write_config(struct snd_soc_codec *codec)
+{
+	return 0;
+}
+
+static int wm8962_dsp2_set_enable(struct snd_soc_codec *codec, u16 val)
+{
+	u16 adcl = snd_soc_read(codec, WM8962_LEFT_ADC_VOLUME);
+	u16 adcr = snd_soc_read(codec, WM8962_RIGHT_ADC_VOLUME);
+	u16 dac = snd_soc_read(codec, WM8962_ADC_DAC_CONTROL_1);
+
+	/* Mute the ADCs and DACs */
+	snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, 0);
+	snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, WM8962_ADC_VU);
+	snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1,
+			    WM8962_DAC_MUTE, WM8962_DAC_MUTE);
+
+	snd_soc_write(codec, WM8962_SOUNDSTAGE_ENABLES_0, val);
+
+	/* Restore the ADCs and DACs */
+	snd_soc_write(codec, WM8962_LEFT_ADC_VOLUME, adcl);
+	snd_soc_write(codec, WM8962_RIGHT_ADC_VOLUME, adcr);
+	snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1,
+			    WM8962_DAC_MUTE, dac);
+
+	return 0;
+}
+
+static int wm8962_dsp2_start(struct snd_soc_codec *codec)
+{
+	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+	wm8962_dsp2_write_config(codec);
+
+	snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_RUNR);
+
+	wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena);
+
+	return 0;
+}
+
+static int wm8962_dsp2_stop(struct snd_soc_codec *codec)
+{
+	wm8962_dsp2_set_enable(codec, 0);
+
+	snd_soc_write(codec, WM8962_DSP2_EXECCONTROL, WM8962_DSP2_STOP);
+
+	return 0;
+}
+
+#define WM8962_DSP2_ENABLE(xname, xshift) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = wm8962_dsp2_ena_info, \
+	.get = wm8962_dsp2_ena_get, .put = wm8962_dsp2_ena_put, \
+	.private_value = xshift }
+
+static int wm8962_dsp2_ena_info(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+
+	return 0;
+}
+
+static int wm8962_dsp2_ena_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	int shift = kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.integer.value[0] = !!(wm8962->dsp2_ena & 1 << shift);
+
+	return 0;
+}
+
+static int wm8962_dsp2_ena_put(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	int shift = kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+	int old = wm8962->dsp2_ena;
+	int ret = 0;
+	int dsp2_running = snd_soc_read(codec, WM8962_DSP2_POWER_MANAGEMENT) &
+		WM8962_DSP2_ENA;
+
+	mutex_lock(&codec->mutex);
+
+	if (ucontrol->value.integer.value[0])
+		wm8962->dsp2_ena |= 1 << shift;
+	else
+		wm8962->dsp2_ena &= ~(1 << shift);
+
+	if (wm8962->dsp2_ena == old)
+		goto out;
+
+	ret = 1;
+
+	if (dsp2_running) {
+		if (wm8962->dsp2_ena)
+			wm8962_dsp2_set_enable(codec, wm8962->dsp2_ena);
+		else
+			wm8962_dsp2_stop(codec);
+	}
+
+out:
+	mutex_unlock(&codec->mutex);
+
+	return ret;
+}
+
 /* The VU bits for the headphones are in a different register to the mute
  * bits and only take effect on the PGA if it is actually powered.
  */
@@ -2049,6 +2167,14 @@ static const char *cap_hpf_mode_text[] = {
 static const struct soc_enum cap_hpf_mode =
 	SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text);
 
+
+static const char *cap_lhpf_mode_text[] = {
+	"LPF", "HPF"
+};
+
+static const struct soc_enum cap_lhpf_mode =
+	SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text);
+
 static const struct snd_kcontrol_new wm8962_snd_controls[] = {
 SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1),
 
@@ -2077,6 +2203,8 @@ SOC_DOUBLE_R("Capture ZC Switch", WM8962_LEFT_INPUT_VOLUME,
 SOC_SINGLE("Capture HPF Switch", WM8962_ADC_DAC_CONTROL_1, 0, 1, 1),
 SOC_ENUM("Capture HPF Mode", cap_hpf_mode),
 SOC_SINGLE("Capture HPF Cutoff", WM8962_ADC_DAC_CONTROL_2, 7, 7, 0),
+SOC_SINGLE("Capture LHPF Switch", WM8962_LHPF1, 0, 1, 0),
+SOC_ENUM("Capture LHPF Mode", cap_lhpf_mode),
 
 SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1,
 		 WM8962_DAC_DSP_MIXING_2, 4, 12, 0, st_tlv),
@@ -2134,6 +2262,11 @@ SOC_DOUBLE_R_TLV("EQ4 Volume", WM8962_EQ3, WM8962_EQ23,
 		 WM8962_EQL_B4_GAIN_SHIFT, 31, 0, eq_tlv),
 SOC_DOUBLE_R_TLV("EQ5 Volume", WM8962_EQ3, WM8962_EQ23,
 		 WM8962_EQL_B5_GAIN_SHIFT, 31, 0, eq_tlv),
+
+WM8962_DSP2_ENABLE("VSS Switch", WM8962_VSS_ENA_SHIFT),
+WM8962_DSP2_ENABLE("HPF1 Switch", WM8962_HPF1_ENA_SHIFT),
+WM8962_DSP2_ENABLE("HPF2 Switch", WM8962_HPF2_ENA_SHIFT),
+WM8962_DSP2_ENABLE("HD Bass Switch", WM8962_HDBASS_ENA_SHIFT),
 };
 
 static const struct snd_kcontrol_new wm8962_spk_mono_controls[] = {
@@ -2395,6 +2528,31 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 	}
 }
 
+static int dsp2_event(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (wm8962->dsp2_ena)
+			wm8962_dsp2_start(codec);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		if (wm8962->dsp2_ena)
+			wm8962_dsp2_stop(codec);
+		break;
+
+	default:
+		BUG();
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
 static const char *st_text[] = { "None", "Right", "Left" };
 
 static const struct soc_enum str_enum =
@@ -2517,6 +2675,9 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event,
 SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event,
 		    SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT,
+		      WM8962_DSP2_ENA_SHIFT, 0, dsp2_event,
+		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
 SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0,
 		   inpgal, ARRAY_SIZE(inpgal)),
@@ -2612,11 +2773,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = {
 	{ "ADCL", NULL, "TOCLK" },
 	{ "ADCL", NULL, "MIXINL" },
 	{ "ADCL", NULL, "DMIC" },
+	{ "ADCL", NULL, "DSP2" },
 
 	{ "ADCR", NULL, "SYSCLK" },
 	{ "ADCR", NULL, "TOCLK" },
 	{ "ADCR", NULL, "MIXINR" },
 	{ "ADCR", NULL, "DMIC" },
+	{ "ADCR", NULL, "DSP2" },
 
 	{ "STL", "Left", "ADCL" },
 	{ "STL", "Right", "ADCR" },
@@ -2628,11 +2791,13 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = {
 	{ "DACL", NULL, "TOCLK" },
 	{ "DACL", NULL, "Beep" },
 	{ "DACL", NULL, "STL" },
+	{ "DACL", NULL, "DSP2" },
 
 	{ "DACR", NULL, "SYSCLK" },
 	{ "DACR", NULL, "TOCLK" },
 	{ "DACR", NULL, "Beep" },
 	{ "DACR", NULL, "STR" },
+	{ "DACR", NULL, "DSP2" },
 
 	{ "HPMIXL", "IN4L Switch", "IN4L" },
 	{ "HPMIXL", "IN4R Switch", "IN4R" },
@@ -3403,12 +3568,16 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 	struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
 	int mask;
 	int active;
+	int reg;
 
 	mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK);
 
 	active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2);
 	active &= ~mask;
 
+	if (!active)
+		return IRQ_NONE;
+
 	/* Acknowledge the interrupts */
 	snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active);
 
@@ -3420,9 +3589,21 @@ static irqreturn_t wm8962_irq(int irq, void *data)
 	if (active & WM8962_FIFOS_ERR_EINT)
 		dev_err(codec->dev, "FIFO error\n");
 
-	if (active & WM8962_TEMP_SHUT_EINT)
+	if (active & WM8962_TEMP_SHUT_EINT) {
 		dev_crit(codec->dev, "Thermal shutdown\n");
 
+		reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS);
+
+		if (reg & WM8962_TEMP_ERR_HP)
+			dev_crit(codec->dev, "Headphone thermal error\n");
+		if (reg & WM8962_TEMP_WARN_HP)
+			dev_crit(codec->dev, "Headphone thermal warning\n");
+		if (reg & WM8962_TEMP_ERR_SPK)
+			dev_crit(codec->dev, "Speaker thermal error\n");
+		if (reg & WM8962_TEMP_WARN_SPK)
+			dev_crit(codec->dev, "Speaker thermal warning\n");
+	}
+
 	if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) {
 		dev_dbg(codec->dev, "Microphone event detected\n");
 
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 6e85b8869af..eec8e143511 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -847,6 +847,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8993_BUS_CONTROL_1, 1, 0, clk_sys_event,
 		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, NULL, 0),
 
 SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0),
 SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0),
@@ -880,6 +881,9 @@ SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0),
 };
 
 static const struct snd_soc_dapm_route routes[] = {
+	{ "MICBIAS1", NULL, "VMID" },
+	{ "MICBIAS2", NULL, "VMID" },
+
 	{ "ADCL", NULL, "CLK_SYS" },
 	{ "ADCL", NULL, "CLK_DSP" },
 	{ "ADCR", NULL, "CLK_SYS" },
@@ -1433,7 +1437,8 @@ static int wm8993_probe(struct snd_soc_codec *codec)
 	int ret, i, val;
 
 	wm8993->hubs_data.hp_startup_mode = 1;
-	wm8993->hubs_data.dcs_codes = -2;
+	wm8993->hubs_data.dcs_codes_l = -2;
+	wm8993->hubs_data.dcs_codes_r = -2;
 	wm8993->hubs_data.series_startup = 1;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c
index a87adbd05ee..df5a8b9a250 100644
--- a/sound/soc/codecs/wm8994-tables.c
+++ b/sound/soc/codecs/wm8994-tables.c
@@ -1073,8 +1073,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = {
 	{ 0x0000, 0x0000 }, /* R1069 */
 	{ 0x0000, 0x0000 }, /* R1070 */
 	{ 0x0000, 0x0000 }, /* R1071 */
-	{ 0x0000, 0x0000 }, /* R1072 */
-	{ 0x0000, 0x0000 }, /* R1073 */
+	{ 0x006F, 0x006F }, /* R1072  - AIF1 DAC1 Noise Gate */
+	{ 0x006F, 0x006F }, /* R1073  - AIF1 DAC2 Noise Gate */
 	{ 0x0000, 0x0000 }, /* R1074 */
 	{ 0x0000, 0x0000 }, /* R1075 */
 	{ 0x0000, 0x0000 }, /* R1076 */
@@ -1329,7 +1329,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = {
 	{ 0x0000, 0x0000 }, /* R1325 */
 	{ 0x0000, 0x0000 }, /* R1326 */
 	{ 0x0000, 0x0000 }, /* R1327 */
-	{ 0x0000, 0x0000 }, /* R1328 */
+	{ 0x006F, 0x006F }, /* R1328  - AIF2 DAC Noise Gate */
 	{ 0x0000, 0x0000 }, /* R1329 */
 	{ 0x0000, 0x0000 }, /* R1330 */
 	{ 0x0000, 0x0000 }, /* R1331 */
@@ -1635,8 +1635,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
 	0x0000,     /* R58    - MICBIAS */
 	0x000D,     /* R59    - LDO 1 */
 	0x0003,     /* R60    - LDO 2 */
-	0x0000,     /* R61 */
-	0x0000,     /* R62 */
+	0x0039,     /* R61    - MICBIAS1 */
+	0x0039,     /* R62    - MICBIAS2 */
 	0x0000,     /* R63 */
 	0x0000,     /* R64 */
 	0x0000,     /* R65 */
@@ -2646,8 +2646,8 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
 	0x0000,     /* R1069 */
 	0x0000,     /* R1070 */
 	0x0000,     /* R1071 */
-	0x0000,     /* R1072 */
-	0x0000,     /* R1073 */
+	0x0068,     /* R1072  - AIF1 DAC1 Noise Gate */
+	0x0068,     /* R1073  - AIF1 DAC2 Noise Gate */
 	0x0000,     /* R1074 */
 	0x0000,     /* R1075 */
 	0x0000,     /* R1076 */
@@ -2902,7 +2902,7 @@ const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
 	0x0000,     /* R1325 */
 	0x0000,     /* R1326 */
 	0x0000,     /* R1327 */
-	0x0000,     /* R1328 */
+	0x0068,     /* R1328  - AIF2 DAC Noise Gate */
 	0x0000,     /* R1329 */
 	0x0000,     /* R1330 */
 	0x0000,     /* R1331 */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index b393f9fac97..e5372675123 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -107,6 +107,7 @@ static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg)
 	case WM8994_LDO_2:
 	case WM8958_DSP2_EXECCONTROL:
 	case WM8958_MIC_DETECT_3:
+	case WM8994_DC_SERVO_4E:
 		return 1;
 	default:
 		return 0;
@@ -281,6 +282,7 @@ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
 static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0);
 static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0);
 static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0);
 
 #define WM8994_DRC_SWITCH(xname, reg, shift) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -660,8 +662,45 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0,
 	       eq_tlv),
 };
 
+static const char *wm8958_ng_text[] = {
+	"30ms", "125ms", "250ms", "500ms",
+};
+
+static const struct soc_enum wm8958_aif1dac1_ng_hold =
+	SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE,
+			WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text);
+
+static const struct soc_enum wm8958_aif1dac2_ng_hold =
+	SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE,
+			WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text);
+
+static const struct soc_enum wm8958_aif2dac_ng_hold =
+	SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE,
+			WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text);
+
 static const struct snd_kcontrol_new wm8958_snd_controls[] = {
 SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv),
+
+SOC_SINGLE("AIF1DAC1 Noise Gate Switch", WM8958_AIF1_DAC1_NOISE_GATE,
+	   WM8958_AIF1DAC1_NG_ENA_SHIFT, 1, 0),
+SOC_ENUM("AIF1DAC1 Noise Gate Hold Time", wm8958_aif1dac1_ng_hold),
+SOC_SINGLE_TLV("AIF1DAC1 Noise Gate Threshold Volume",
+	       WM8958_AIF1_DAC1_NOISE_GATE, WM8958_AIF1DAC1_NG_THR_SHIFT,
+	       7, 1, ng_tlv),
+
+SOC_SINGLE("AIF1DAC2 Noise Gate Switch", WM8958_AIF1_DAC2_NOISE_GATE,
+	   WM8958_AIF1DAC2_NG_ENA_SHIFT, 1, 0),
+SOC_ENUM("AIF1DAC2 Noise Gate Hold Time", wm8958_aif1dac2_ng_hold),
+SOC_SINGLE_TLV("AIF1DAC2 Noise Gate Threshold Volume",
+	       WM8958_AIF1_DAC2_NOISE_GATE, WM8958_AIF1DAC2_NG_THR_SHIFT,
+	       7, 1, ng_tlv),
+
+SOC_SINGLE("AIF2DAC Noise Gate Switch", WM8958_AIF2_DAC_NOISE_GATE,
+	   WM8958_AIF2DAC_NG_ENA_SHIFT, 1, 0),
+SOC_ENUM("AIF2DAC Noise Gate Hold Time", wm8958_aif2dac_ng_hold),
+SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume",
+	       WM8958_AIF2_DAC_NOISE_GATE, WM8958_AIF2DAC_NG_THR_SHIFT,
+	       7, 1, ng_tlv),
 };
 
 static int clk_sys_event(struct snd_soc_dapm_widget *w,
@@ -681,6 +720,97 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static void vmid_reference(struct snd_soc_codec *codec)
+{
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	wm8994->vmid_refcount++;
+
+	dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n",
+		wm8994->vmid_refcount);
+
+	if (wm8994->vmid_refcount == 1) {
+		/* Startup bias, VMID ramp & buffer */
+		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+				    WM8994_STARTUP_BIAS_ENA |
+				    WM8994_VMID_BUF_ENA |
+				    WM8994_VMID_RAMP_MASK,
+				    WM8994_STARTUP_BIAS_ENA |
+				    WM8994_VMID_BUF_ENA |
+				    (0x11 << WM8994_VMID_RAMP_SHIFT));
+
+		/* Main bias enable, VMID=2x40k */
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
+				    WM8994_BIAS_ENA |
+				    WM8994_VMID_SEL_MASK,
+				    WM8994_BIAS_ENA | 0x2);
+
+		msleep(20);
+	}
+}
+
+static void vmid_dereference(struct snd_soc_codec *codec)
+{
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	wm8994->vmid_refcount--;
+
+	dev_dbg(codec->dev, "Dereferencing VMID, refcount is now %d\n",
+		wm8994->vmid_refcount);
+
+	if (wm8994->vmid_refcount == 0) {
+		/* Switch over to startup biases */
+		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+				    WM8994_BIAS_SRC |
+				    WM8994_STARTUP_BIAS_ENA |
+				    WM8994_VMID_BUF_ENA |
+				    WM8994_VMID_RAMP_MASK,
+				    WM8994_BIAS_SRC |
+				    WM8994_STARTUP_BIAS_ENA |
+				    WM8994_VMID_BUF_ENA |
+				    (1 << WM8994_VMID_RAMP_SHIFT));
+
+		/* Disable main biases */
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
+				    WM8994_BIAS_ENA |
+				    WM8994_VMID_SEL_MASK, 0);
+
+		/* Discharge line */
+		snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
+				    WM8994_LINEOUT1_DISCH |
+				    WM8994_LINEOUT2_DISCH,
+				    WM8994_LINEOUT1_DISCH |
+				    WM8994_LINEOUT2_DISCH);
+
+		msleep(5);
+
+		/* Switch off startup biases */
+		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
+				    WM8994_BIAS_SRC |
+				    WM8994_STARTUP_BIAS_ENA |
+				    WM8994_VMID_BUF_ENA |
+				    WM8994_VMID_RAMP_MASK, 0);
+	}
+}
+
+static int vmid_event(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		vmid_reference(codec);
+		break;
+
+	case SND_SOC_DAPM_POST_PMD:
+		vmid_dereference(codec);
+		break;
+	}
+
+	return 0;
+}
+
 static void wm8994_update_class_w(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
@@ -1208,6 +1338,8 @@ SND_SOC_DAPM_INPUT("Clock"),
 
 SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev,
 		      SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 
 SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
 		    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -1525,6 +1657,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
 	{ "AIF2DACL", NULL, "AIF2DAC Mux" },
 	{ "AIF2DACR", NULL, "AIF2DAC Mux" },
+	{ "MICBIAS1", NULL, "VMID" },
+	{ "MICBIAS2", NULL, "VMID" },
 };
 
 static const struct snd_soc_dapm_route wm8958_intercon[] = {
@@ -1629,10 +1763,12 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 			  unsigned int freq_in, unsigned int freq_out)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+	struct wm8994 *control = codec->control_data;
 	int reg_offset, ret;
 	struct fll_div fll;
 	u16 reg, aif1, aif2;
 	unsigned long timeout;
+	bool was_enabled;
 
 	aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1)
 		& WM8994_AIF1CLK_ENA;
@@ -1653,6 +1789,9 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 		return -EINVAL;
 	}
 
+	reg = snd_soc_read(codec, WM8994_FLL1_CONTROL_1 + reg_offset);
+	was_enabled = reg & WM8994_FLL1_ENA;
+
 	switch (src) {
 	case 0:
 		/* Allow no source specification when stopping */
@@ -1719,6 +1858,21 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 
 	/* Enable (with fractional mode if required) */
 	if (freq_out) {
+		/* Enable VMID if we need it */
+		if (!was_enabled) {
+			switch (control->type) {
+			case WM8994:
+				vmid_reference(codec);
+				break;
+			case WM8958:
+				if (wm8994->revision < 1)
+					vmid_reference(codec);
+				break;
+			default:
+				break;
+			}
+		}
+
 		if (fll.k)
 			reg = WM8994_FLL1_ENA | WM8994_FLL1_FRAC;
 		else
@@ -1736,6 +1890,20 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 		} else {
 			msleep(5);
 		}
+	} else {
+		if (was_enabled) {
+			switch (control->type) {
+			case WM8994:
+				vmid_dereference(codec);
+				break;
+			case WM8958:
+				if (wm8994->revision < 1)
+					vmid_dereference(codec);
+				break;
+			default:
+				break;
+			}
+		}
 	}
 
 	wm8994->fll[id].in = freq_in;
@@ -1852,9 +2020,6 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		/* VMID=2x40k */
-		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
-				    WM8994_VMID_SEL_MASK, 0x2);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -1896,65 +2061,13 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 					    WM8994_LINEOUT2_DISCH,
 					    WM8994_LINEOUT1_DISCH |
 					    WM8994_LINEOUT2_DISCH);
-
-			/* Startup bias, VMID ramp & buffer */
-			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-					    WM8994_STARTUP_BIAS_ENA |
-					    WM8994_VMID_BUF_ENA |
-					    WM8994_VMID_RAMP_MASK,
-					    WM8994_STARTUP_BIAS_ENA |
-					    WM8994_VMID_BUF_ENA |
-					    (0x11 << WM8994_VMID_RAMP_SHIFT));
-
-			/* Main bias enable, VMID=2x40k */
-			snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
-					    WM8994_BIAS_ENA |
-					    WM8994_VMID_SEL_MASK,
-					    WM8994_BIAS_ENA | 0x2);
-
-			msleep(20);
 		}
 
-		/* VMID=2x500k */
-		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
-				    WM8994_VMID_SEL_MASK, 0x4);
 
 		break;
 
 	case SND_SOC_BIAS_OFF:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
-			/* Switch over to startup biases */
-			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-					    WM8994_BIAS_SRC |
-					    WM8994_STARTUP_BIAS_ENA |
-					    WM8994_VMID_BUF_ENA |
-					    WM8994_VMID_RAMP_MASK,
-					    WM8994_BIAS_SRC |
-					    WM8994_STARTUP_BIAS_ENA |
-					    WM8994_VMID_BUF_ENA |
-					    (1 << WM8994_VMID_RAMP_SHIFT));
-
-			/* Disable main biases */
-			snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
-					    WM8994_BIAS_ENA |
-					    WM8994_VMID_SEL_MASK, 0);
-
-			/* Discharge line */
-			snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
-					    WM8994_LINEOUT1_DISCH |
-					    WM8994_LINEOUT2_DISCH,
-					    WM8994_LINEOUT1_DISCH |
-					    WM8994_LINEOUT2_DISCH);
-
-			msleep(5);
-
-			/* Switch off startup biases */
-			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-					    WM8994_BIAS_SRC |
-					    WM8994_STARTUP_BIAS_ENA |
-					    WM8994_VMID_BUF_ENA |
-					    WM8994_VMID_RAMP_MASK, 0);
-
 			wm8994->cur_fw = NULL;
 
 			pm_runtime_put(codec->dev);
@@ -2384,6 +2497,21 @@ static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
 	return snd_soc_update_bits(codec, reg, mask, val);
 }
 
+static int wm8994_aif2_probe(struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	/* Disable the pulls on the AIF if we're using it to save power. */
+	snd_soc_update_bits(codec, WM8994_GPIO_3,
+			    WM8994_GPN_PU | WM8994_GPN_PD, 0);
+	snd_soc_update_bits(codec, WM8994_GPIO_4,
+			    WM8994_GPN_PU | WM8994_GPN_PD, 0);
+	snd_soc_update_bits(codec, WM8994_GPIO_5,
+			    WM8994_GPN_PU | WM8994_GPN_PD, 0);
+
+	return 0;
+}
+
 #define WM8994_RATES SNDRV_PCM_RATE_8000_96000
 
 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
@@ -2451,6 +2579,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
 			.rates = WM8994_RATES,
 			.formats = WM8994_FORMATS,
 		},
+		.probe = wm8994_aif2_probe,
 		.ops = &wm8994_aif2_dai_ops,
 	},
 	{
@@ -2916,6 +3045,24 @@ static irqreturn_t wm8994_fifo_error(int irq, void *data)
 	return IRQ_HANDLED;
 }
 
+static irqreturn_t wm8994_temp_warn(int irq, void *data)
+{
+	struct snd_soc_codec *codec = data;
+
+	dev_err(codec->dev, "Thermal warning\n");
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t wm8994_temp_shut(int irq, void *data)
+{
+	struct snd_soc_codec *codec = data;
+
+	dev_crit(codec->dev, "Thermal shutdown\n");
+
+	return IRQ_HANDLED;
+}
+
 static int wm8994_codec_probe(struct snd_soc_codec *codec)
 {
 	struct wm8994 *control;
@@ -2972,13 +3119,14 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		switch (wm8994->revision) {
 		case 2:
 		case 3:
-			wm8994->hubs.dcs_codes = -5;
+			wm8994->hubs.dcs_codes_l = -5;
+			wm8994->hubs.dcs_codes_r = -5;
 			wm8994->hubs.hp_startup_mode = 1;
 			wm8994->hubs.dcs_readback_mode = 1;
 			wm8994->hubs.series_startup = 1;
 			break;
 		default:
-			wm8994->hubs.dcs_readback_mode = 1;
+			wm8994->hubs.dcs_readback_mode = 2;
 			break;
 		}
 		break;
@@ -2993,6 +3141,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 
 	wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR,
 			   wm8994_fifo_error, "FIFO error", codec);
+	wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_WARN,
+			   wm8994_temp_warn, "Thermal warning", codec);
+	wm8994_request_irq(wm8994->control_data, WM8994_IRQ_TEMP_SHUT,
+			   wm8994_temp_shut, "Thermal shutdown", codec);
 
 	ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
 				 wm_hubs_dcs_done, "DC servo done",
@@ -3257,6 +3409,8 @@ err_irq:
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
 			&wm8994->hubs);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec);
 err:
 	kfree(wm8994);
 	return ret;
@@ -3279,6 +3433,8 @@ static int  wm8994_codec_remove(struct snd_soc_codec *codec)
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE,
 			&wm8994->hubs);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_SHUT, codec);
+	wm8994_free_irq(codec->control_data, WM8994_IRQ_TEMP_WARN, codec);
 
 	switch (control->type) {
 	case WM8994:
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 1ab2266039f..f4f1355efc8 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -83,6 +83,8 @@ struct wm8994_priv {
 	struct completion fll_locked[2];
 	bool fll_locked_irq;
 
+	int vmid_refcount;
+
 	int dac_rates[2];
 	int lrclk_shared[2];
 
diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c
index 5ad873fda81..74ae5995a78 100644
--- a/sound/soc/codecs/wm8995.c
+++ b/sound/soc/codecs/wm8995.c
@@ -1573,9 +1573,7 @@ static int wm8995_resume(struct snd_soc_codec *codec)
 static int wm8995_remove(struct snd_soc_codec *codec)
 {
 	struct wm8995_priv *wm8995;
-	struct i2c_client *i2c;
 
-	i2c = container_of(codec->dev, struct i2c_client, dev);
 	wm8995 = snd_soc_codec_get_drvdata(codec);
 	wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -1642,6 +1640,7 @@ static int wm8995_probe(struct snd_soc_codec *codec)
 
 	if (ret != 0x8995) {
 		dev_err(codec->dev, "Invalid device ID: %#x\n", ret);
+		ret = -EINVAL;
 		goto err_reg_enable;
 	}
 
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 0cdb9d10567..833df74c558 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -41,12 +41,11 @@
 #define HPOUT2L 4
 #define HPOUT2R 8
 
-#define WM8996_NUM_SUPPLIES 4
+#define WM8996_NUM_SUPPLIES 3
 static const char *wm8996_supply_names[WM8996_NUM_SUPPLIES] = {
 	"DBVDD",
 	"AVDD1",
 	"AVDD2",
-	"CPVDD",
 };
 
 struct wm8996_priv {
@@ -71,6 +70,8 @@ struct wm8996_priv {
 
 	struct regulator_bulk_data supplies[WM8996_NUM_SUPPLIES];
 	struct notifier_block disable_nb[WM8996_NUM_SUPPLIES];
+	struct regulator *cpvdd;
+	int bg_ena;
 
 	struct wm8996_pdata pdata;
 
@@ -112,7 +113,6 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \
 WM8996_REGULATOR_EVENT(0)
 WM8996_REGULATOR_EVENT(1)
 WM8996_REGULATOR_EVENT(2)
-WM8996_REGULATOR_EVENT(3)
 
 static const u16 wm8996_reg[WM8996_MAX_REGISTER] = {
 	[WM8996_SOFTWARE_RESET] = 0x8996,
@@ -414,6 +414,7 @@ static const DECLARE_TLV_DB_SCALE(out_digital_tlv, -1200, 150, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -900, 75, 0);
 static const DECLARE_TLV_DB_SCALE(spk_tlv, -900, 150, 0);
 static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static const DECLARE_TLV_DB_SCALE(threedstereo_tlv, -1600, 183, 1);
 
 static const char *sidetone_hpf_text[] = {
 	"2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz"
@@ -608,6 +609,14 @@ SOC_SINGLE("DAC High Performance Switch", WM8996_OVERSAMPLING, 0, 1, 0),
 SOC_SINGLE("DAC Soft Mute Switch", WM8996_DAC_SOFTMUTE, 1, 1, 0),
 SOC_SINGLE("DAC Slow Soft Mute Switch", WM8996_DAC_SOFTMUTE, 0, 1, 0),
 
+SOC_SINGLE("DSP1 3D Stereo Switch", WM8996_DSP1_RX_FILTERS_2, 8, 1, 0),
+SOC_SINGLE("DSP2 3D Stereo Switch", WM8996_DSP2_RX_FILTERS_2, 8, 1, 0),
+
+SOC_SINGLE_TLV("DSP1 3D Stereo Volume", WM8996_DSP1_RX_FILTERS_2, 10, 15,
+		0, threedstereo_tlv),
+SOC_SINGLE_TLV("DSP2 3D Stereo Volume", WM8996_DSP2_RX_FILTERS_2, 10, 15,
+		0, threedstereo_tlv),
+
 SOC_DOUBLE_TLV("Digital Output 1 Volume", WM8996_DAC1_HPOUT1_VOLUME, 0, 4,
 	       8, 0, out_digital_tlv),
 SOC_DOUBLE_TLV("Digital Output 2 Volume", WM8996_DAC2_HPOUT2_VOLUME, 0, 4,
@@ -658,19 +667,75 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0,
 	       eq_tlv),
 };
 
+static void wm8996_bg_enable(struct snd_soc_codec *codec)
+{
+	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
+
+	wm8996->bg_ena++;
+	if (wm8996->bg_ena == 1) {
+		snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1,
+				    WM8996_BG_ENA, WM8996_BG_ENA);
+		msleep(2);
+	}
+}
+
+static void wm8996_bg_disable(struct snd_soc_codec *codec)
+{
+	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
+
+	wm8996->bg_ena--;
+	if (!wm8996->bg_ena)
+		snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1,
+				    WM8996_BG_ENA, 0);
+}
+
+static int bg_event(struct snd_soc_dapm_widget *w,
+		    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	int ret = 0;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		wm8996_bg_enable(codec);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		wm8996_bg_disable(codec);
+		break;
+	default:
+		BUG();
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
 static int cp_event(struct snd_soc_dapm_widget *w,
 		    struct snd_kcontrol *kcontrol, int event)
 {
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
 	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		ret = regulator_enable(wm8996->cpvdd);
+		if (ret != 0)
+			dev_err(codec->dev, "Failed to enable CPVDD: %d\n",
+				ret);
+		break;
 	case SND_SOC_DAPM_POST_PMU:
 		msleep(5);
 		break;
+	case SND_SOC_DAPM_POST_PMD:
+		regulator_disable_deferred(wm8996->cpvdd, 20);
+		break;
 	default:
 		BUG();
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 
-	return 0;
+	return ret;
 }
 
 static int rmv_short_event(struct snd_soc_dapm_widget *w,
@@ -698,7 +763,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask)
 {
 	struct i2c_client *i2c = to_i2c_client(codec->dev);
 	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
-	int i, ret;
+	int ret;
 	unsigned long timeout = 200;
 
 	snd_soc_write(codec, WM8996_DC_SERVO_2, mask);
@@ -713,15 +778,12 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask)
 
 		} else {
 			msleep(1);
-			if (--i) {
-				timeout = 0;
-				break;
-			}
+			timeout--;
 		}
 
 		ret = snd_soc_read(codec, WM8996_DC_SERVO_2);
 		dev_dbg(codec->dev, "DC servo state: %x\n", ret);
-	} while (ret & mask);
+	} while (timeout && ret & mask);
 
 	if (timeout == 0)
 		dev_err(codec->dev, "DC servo timed out for %x\n", mask);
@@ -979,9 +1041,12 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event,
-		      SND_SOC_DAPM_POST_PMU),
-
+		      SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event,
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0),
 SND_SOC_DAPM_MICBIAS("MICB2", WM8996_POWER_MANAGEMENT_1, 9, 0),
 SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0),
 
@@ -1035,14 +1100,14 @@ SND_SOC_DAPM_DAC("DAC2R", NULL, WM8996_POWER_MANAGEMENT_5, 2, 0),
 SND_SOC_DAPM_DAC("DAC1L", NULL, WM8996_POWER_MANAGEMENT_5, 1, 0),
 SND_SOC_DAPM_DAC("DAC1R", NULL, WM8996_POWER_MANAGEMENT_5, 0, 0),
 
-SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 1,
+SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0,
 		    WM8996_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 2,
+SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 1,
 		    WM8996_POWER_MANAGEMENT_4, 8, 0),
 
-SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 1,
+SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 0,
 		    WM8996_POWER_MANAGEMENT_6, 9, 0),
-SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 2,
+SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 1,
 		    WM8996_POWER_MANAGEMENT_6, 8, 0),
 
 SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 5,
@@ -1137,17 +1202,23 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
 	{ "Charge Pump", NULL, "SYSCLK" },
 
 	{ "MICB1", NULL, "LDO2" },
+	{ "MICB1", NULL, "MICB1 Audio" },
+	{ "MICB1", NULL, "Bandgap" },
 	{ "MICB2", NULL, "LDO2" },
+	{ "MICB2", NULL, "MICB2 Audio" },
+	{ "MICB2", NULL, "Bandgap" },
 
 	{ "IN1L PGA", NULL, "IN2LN" },
 	{ "IN1L PGA", NULL, "IN2LP" },
 	{ "IN1L PGA", NULL, "IN1LN" },
 	{ "IN1L PGA", NULL, "IN1LP" },
+	{ "IN1L PGA", NULL, "Bandgap" },
 
 	{ "IN1R PGA", NULL, "IN2RN" },
 	{ "IN1R PGA", NULL, "IN2RP" },
 	{ "IN1R PGA", NULL, "IN1RN" },
 	{ "IN1R PGA", NULL, "IN1RP" },
+	{ "IN1R PGA", NULL, "Bandgap" },
 
 	{ "ADCL", NULL, "IN1L PGA" },
 
@@ -1281,6 +1352,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
 	{ "DAC2R", NULL, "DAC2R Mixer" },
 
 	{ "HPOUT2L PGA", NULL, "Charge Pump" },
+	{ "HPOUT2L PGA", NULL, "Bandgap" },
 	{ "HPOUT2L PGA", NULL, "DAC2L" },
 	{ "HPOUT2L_DLY", NULL, "HPOUT2L PGA" },
 	{ "HPOUT2L_DCS", NULL, "HPOUT2L_DLY" },
@@ -1288,6 +1360,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
 	{ "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_OUTP" },
 
 	{ "HPOUT2R PGA", NULL, "Charge Pump" },
+	{ "HPOUT2R PGA", NULL, "Bandgap" },
 	{ "HPOUT2R PGA", NULL, "DAC2R" },
 	{ "HPOUT2R_DLY", NULL, "HPOUT2R PGA" },
 	{ "HPOUT2R_DCS", NULL, "HPOUT2R_DLY" },
@@ -1295,6 +1368,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
 	{ "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_OUTP" },
 
 	{ "HPOUT1L PGA", NULL, "Charge Pump" },
+	{ "HPOUT1L PGA", NULL, "Bandgap" },
 	{ "HPOUT1L PGA", NULL, "DAC1L" },
 	{ "HPOUT1L_DLY", NULL, "HPOUT1L PGA" },
 	{ "HPOUT1L_DCS", NULL, "HPOUT1L_DLY" },
@@ -1302,6 +1376,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = {
 	{ "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_OUTP" },
 
 	{ "HPOUT1R PGA", NULL, "Charge Pump" },
+	{ "HPOUT1R PGA", NULL, "Bandgap" },
 	{ "HPOUT1R PGA", NULL, "DAC1R" },
 	{ "HPOUT1R_DLY", NULL, "HPOUT1R PGA" },
 	{ "HPOUT1R_DCS", NULL, "HPOUT1R_DLY" },
@@ -1620,14 +1695,7 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		break;
-
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
-			snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1,
-					    WM8996_BG_ENA, WM8996_BG_ENA);
-			msleep(2);
-		}
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
@@ -1650,9 +1718,6 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec,
 			codec->cache_only = false;
 			snd_soc_cache_sync(codec);
 		}
-
-		snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1,
-				    WM8996_BG_ENA, 0);
 		break;
 
 	case SND_SOC_BIAS_OFF:
@@ -2041,7 +2106,7 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	struct i2c_client *i2c = to_i2c_client(codec->dev);
 	struct _fll_div fll_div;
 	unsigned long timeout;
-	int ret, reg;
+	int ret, reg, retry;
 
 	/* Any change? */
 	if (source == wm8996->fll_src && Fref == wm8996->fll_fref &&
@@ -2057,6 +2122,8 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 		snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1,
 				    WM8996_FLL_ENA, 0);
 
+		wm8996_bg_disable(codec);
+
 		return 0;
 	}
 
@@ -2111,6 +2178,11 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 
 	snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda);
 
+	/* Enable the bandgap if it's not already enabled */
+	ret = snd_soc_read(codec, WM8996_FLL_CONTROL_1);
+	if (!(ret & WM8996_FLL_ENA))
+		wm8996_bg_enable(codec);
+
 	/* Clear any pending completions (eg, from failed startups) */
 	try_wait_for_completion(&wm8996->fll_lock);
 
@@ -2128,17 +2200,29 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
 	else
 		timeout = msecs_to_jiffies(2);
 
-	/* Allow substantially longer if we've actually got the IRQ */
+	/* Allow substantially longer if we've actually got the IRQ, poll
+	 * at a slightly higher rate if we don't.
+	 */
 	if (i2c->irq)
-		timeout *= 1000;
+		timeout *= 10;
+	else
+		timeout /= 2;
 
-	ret = wait_for_completion_timeout(&wm8996->fll_lock, timeout);
+	for (retry = 0; retry < 10; retry++) {
+		ret = wait_for_completion_timeout(&wm8996->fll_lock,
+						  timeout);
+		if (ret != 0) {
+			WARN_ON(!i2c->irq);
+			break;
+		}
 
-	if (ret == 0 && i2c->irq) {
+		ret = snd_soc_read(codec, WM8996_INTERRUPT_RAW_STATUS_2);
+		if (ret & WM8996_FLL_LOCK_STS)
+			break;
+	}
+	if (retry == 10) {
 		dev_err(codec->dev, "Timed out waiting for FLL\n");
 		ret = -ETIMEDOUT;
-	} else {
-		ret = 0;
 	}
 
 	dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
@@ -2297,12 +2381,94 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 
 	/* Enable interrupts and we're off */
 	snd_soc_update_bits(codec, WM8996_INTERRUPT_STATUS_2_MASK,
-			    WM8996_IM_MICD_EINT, 0);
+			    WM8996_IM_MICD_EINT | WM8996_HP_DONE_EINT, 0);
 
 	return 0;
 }
 EXPORT_SYMBOL_GPL(wm8996_detect);
 
+static void wm8996_hpdet_irq(struct snd_soc_codec *codec)
+{
+	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
+	int val, reg, report;
+
+	/* Assume headphone in error conditions; we need to report
+	 * something or we stall our state machine.
+	 */
+	report = SND_JACK_HEADPHONE;
+
+	reg = snd_soc_read(codec, WM8996_HEADPHONE_DETECT_2);
+	if (reg < 0) {
+		dev_err(codec->dev, "Failed to read HPDET status\n");
+		goto out;
+	}
+
+	if (!(reg & WM8996_HP_DONE)) {
+		dev_err(codec->dev, "Got HPDET IRQ but HPDET is busy\n");
+		goto out;
+	}
+
+	val = reg & WM8996_HP_LVL_MASK;
+
+	dev_dbg(codec->dev, "HPDET measured %d ohms\n", val);
+
+	/* If we've got high enough impedence then report as line,
+	 * otherwise assume headphone.
+	 */
+	if (val >= 126)
+		report = SND_JACK_LINEOUT;
+	else
+		report = SND_JACK_HEADPHONE;
+
+out:
+	if (wm8996->jack_mic)
+		report |= SND_JACK_MICROPHONE;
+
+	snd_soc_jack_report(wm8996->jack, report,
+			    SND_JACK_LINEOUT | SND_JACK_HEADSET);
+
+	wm8996->detecting = false;
+
+	/* If the output isn't running re-clamp it */
+	if (!(snd_soc_read(codec, WM8996_POWER_MANAGEMENT_1) &
+	      (WM8996_HPOUT1L_ENA | WM8996_HPOUT1R_RMV_SHORT)))
+		snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1,
+				    WM8996_HPOUT1L_RMV_SHORT |
+				    WM8996_HPOUT1R_RMV_SHORT, 0);
+
+	/* Go back to looking at the microphone */
+	snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1,
+			    WM8996_JD_MODE_MASK, 0);
+	snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA,
+			    WM8996_MICD_ENA);
+
+	snd_soc_dapm_disable_pin(&codec->dapm, "Bandgap");
+	snd_soc_dapm_sync(&codec->dapm);
+}
+
+static void wm8996_hpdet_start(struct snd_soc_codec *codec)
+{
+	/* Unclamp the output, we can't measure while we're shorting it */
+	snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1,
+			    WM8996_HPOUT1L_RMV_SHORT |
+			    WM8996_HPOUT1R_RMV_SHORT,
+			    WM8996_HPOUT1L_RMV_SHORT |
+			    WM8996_HPOUT1R_RMV_SHORT);
+
+	/* We need bandgap for HPDET */
+	snd_soc_dapm_force_enable_pin(&codec->dapm, "Bandgap");
+	snd_soc_dapm_sync(&codec->dapm);
+
+	/* Go into headphone detect left mode */
+	snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, 0);
+	snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1,
+			    WM8996_JD_MODE_MASK, 1);
+
+	/* Trigger a measurement */
+	snd_soc_update_bits(codec, WM8996_HEADPHONE_DETECT_1,
+			    WM8996_HP_POLL, WM8996_HP_POLL);
+}
+
 static void wm8996_micd(struct snd_soc_codec *codec)
 {
 	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
@@ -2323,28 +2489,36 @@ static void wm8996_micd(struct snd_soc_codec *codec)
 		wm8996->jack_mic = false;
 		wm8996->detecting = true;
 		snd_soc_jack_report(wm8996->jack, 0,
-				    SND_JACK_HEADSET | SND_JACK_BTN_0);
+				    SND_JACK_LINEOUT | SND_JACK_HEADSET |
+				    SND_JACK_BTN_0);
+
 		snd_soc_update_bits(codec, WM8996_MIC_DETECT_1,
 				    WM8996_MICD_RATE_MASK,
 				    WM8996_MICD_RATE_MASK);
 		return;
 	}
 
-	/* If the measurement is very high we've got a microphone but
-	 * do a little debounce to account for mechanical issues.
+	/* If the measurement is very high we've got a microphone,
+	 * either we just detected one or if we already reported then
+	 * we've got a button release event.
 	 */
 	if (val & 0x400) {
-		dev_dbg(codec->dev, "Microphone detected\n");
-		snd_soc_jack_report(wm8996->jack, SND_JACK_HEADSET,
-				    SND_JACK_HEADSET | SND_JACK_BTN_0);
-		wm8996->jack_mic = true;
-		wm8996->detecting = false;
-
-		/* Increase poll rate to give better responsiveness
-		 * for buttons */
-		snd_soc_update_bits(codec, WM8996_MIC_DETECT_1,
-				    WM8996_MICD_RATE_MASK,
-				    5 << WM8996_MICD_RATE_SHIFT);
+		if (wm8996->detecting) {
+			dev_dbg(codec->dev, "Microphone detected\n");
+			wm8996->jack_mic = true;
+			wm8996_hpdet_start(codec);
+
+			/* Increase poll rate to give better responsiveness
+			 * for buttons */
+			snd_soc_update_bits(codec, WM8996_MIC_DETECT_1,
+					    WM8996_MICD_RATE_MASK,
+					    5 << WM8996_MICD_RATE_SHIFT);
+		} else {
+			dev_dbg(codec->dev, "Mic button up\n");
+			snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0);
+		}
+
+		return;
 	}
 
 	/* If we detected a lower impedence during initial startup
@@ -2376,15 +2550,11 @@ static void wm8996_micd(struct snd_soc_codec *codec)
 	if (val & 0x3fc) {
 		if (wm8996->jack_mic) {
 			dev_dbg(codec->dev, "Mic button detected\n");
-			snd_soc_jack_report(wm8996->jack,
-					    SND_JACK_HEADSET | SND_JACK_BTN_0,
-					    SND_JACK_HEADSET | SND_JACK_BTN_0);
-		} else {
-			dev_dbg(codec->dev, "Headphone detected\n");
-			snd_soc_jack_report(wm8996->jack,
-					    SND_JACK_HEADPHONE,
-					    SND_JACK_HEADSET |
+			snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0,
 					    SND_JACK_BTN_0);
+		} else if (wm8996->detecting) {
+			dev_dbg(codec->dev, "Headphone detected\n");
+			wm8996_hpdet_start(codec);
 
 			/* Increase the detection rate a bit for
 			 * responsiveness.
@@ -2392,8 +2562,6 @@ static void wm8996_micd(struct snd_soc_codec *codec)
 			snd_soc_update_bits(codec, WM8996_MIC_DETECT_1,
 					    WM8996_MICD_RATE_MASK,
 					    7 << WM8996_MICD_RATE_SHIFT);
-
-			wm8996->detecting = false;
 		}
 	}
 }
@@ -2412,6 +2580,9 @@ static irqreturn_t wm8996_irq(int irq, void *data)
 	}
 	irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK);
 
+	if (!irq_val)
+		return IRQ_NONE;
+
 	snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val);
 
 	if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) {
@@ -2430,10 +2601,10 @@ static irqreturn_t wm8996_irq(int irq, void *data)
 	if (irq_val & WM8996_MICD_EINT)
 		wm8996_micd(codec);
 
-	if (irq_val)
-		return IRQ_HANDLED;
-	else
-		return IRQ_NONE;
+	if (irq_val & WM8996_HP_DONE_EINT)
+		wm8996_hpdet_irq(codec);
+
+	return IRQ_HANDLED;
 }
 
 static irqreturn_t wm8996_edge_irq(int irq, void *data)
@@ -2548,7 +2719,13 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 	wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0;
 	wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1;
 	wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2;
-	wm8996->disable_nb[3].notifier_call = wm8996_regulator_event_3;
+
+	wm8996->cpvdd = regulator_get(&i2c->dev, "CPVDD");
+	if (IS_ERR(wm8996->cpvdd)) {
+		ret = PTR_ERR(wm8996->cpvdd);
+		dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret);
+		goto err_get;
+	}
 
 	/* This should really be moved into the regulator core */
 	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) {
@@ -2565,7 +2742,7 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 				    wm8996->supplies);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
-		goto err_get;
+		goto err_cpvdd;
 	}
 
 	if (wm8996->pdata.ldo_ena >= 0) {
@@ -2808,6 +2985,8 @@ err_enable:
 		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
 
 	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+err_cpvdd:
+	regulator_put(wm8996->cpvdd);
 err_get:
 	regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
 err:
@@ -2831,6 +3010,7 @@ static int wm8996_remove(struct snd_soc_codec *codec)
 	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
 		regulator_unregister_notifier(wm8996->supplies[i].consumer,
 					      &wm8996->disable_nb[i]);
+	regulator_put(wm8996->cpvdd);
 	regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
 
 	return 0;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index a4691321f9b..f32ab1ee964 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -1120,8 +1120,8 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 	return 0;
 }
 
-static int wm9081_set_sysclk(struct snd_soc_codec *codec,
-			     int clk_id, unsigned int freq, int dir)
+static int wm9081_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+			     int source, unsigned int freq, int dir)
 {
 	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 4de12203e61..f2f3077928d 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -139,7 +139,6 @@ static const u16 wm9090_reg_defaults[] = {
 
 /* This struct is used to save the context */
 struct wm9090_priv {
-	struct mutex mutex;
 	struct wm9090_platform_data pdata;
 	void *control_data;
 };
@@ -663,7 +662,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, wm9090);
 	wm9090->control_data = i2c;
-	mutex_init(&wm9090->mutex);
 
 	ret =  snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm9090,  NULL, 0);
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e763c54c55d..ca8ce03510f 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -18,6 +18,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/mfd/wm8994/registers.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -116,14 +117,23 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 {
 	struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
 	s8 offset;
-	u16 reg, reg_l, reg_r, dcs_cfg;
+	u16 reg, reg_l, reg_r, dcs_cfg, dcs_reg;
+
+	switch (hubs->dcs_readback_mode) {
+	case 2:
+		dcs_reg = WM8994_DC_SERVO_4E;
+		break;
+	default:
+		dcs_reg = WM8993_DC_SERVO_3;
+		break;
+	}
 
 	/* If we're using a digital only path and have a previously
 	 * callibrated DC servo offset stored then use that. */
 	if (hubs->class_w && hubs->class_w_dcs) {
 		dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
 			hubs->class_w_dcs);
-		snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
+		snd_soc_write(codec, dcs_reg, hubs->class_w_dcs);
 		wait_for_dc_servo(codec,
 				  WM8993_DCS_TRIG_DAC_WR_0 |
 				  WM8993_DCS_TRIG_DAC_WR_1);
@@ -154,8 +164,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 		reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
 			& WM8993_DCS_INTEG_CHAN_1_MASK;
 		break;
+	case 2:
 	case 1:
-		reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
+		reg = snd_soc_read(codec, dcs_reg);
 		reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
 			>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 		reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
@@ -168,24 +179,25 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 	dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
 
 	/* Apply correction to DC servo result */
-	if (hubs->dcs_codes) {
-		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
-			hubs->dcs_codes);
+	if (hubs->dcs_codes_l || hubs->dcs_codes_r) {
+		dev_dbg(codec->dev,
+			"Applying %d/%d code DC servo correction\n",
+			hubs->dcs_codes_l, hubs->dcs_codes_r);
 
 		/* HPOUT1R */
 		offset = reg_r;
-		offset += hubs->dcs_codes;
+		offset += hubs->dcs_codes_r;
 		dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
 		/* HPOUT1L */
 		offset = reg_l;
-		offset += hubs->dcs_codes;
+		offset += hubs->dcs_codes_l;
 		dcs_cfg |= (u8)offset;
 
 		dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
 
 		/* Do it */
-		snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
+		snd_soc_write(codec, dcs_reg, dcs_cfg);
 		wait_for_dc_servo(codec,
 				  WM8993_DCS_TRIG_DAC_WR_0 |
 				  WM8993_DCS_TRIG_DAC_WR_1);
@@ -217,7 +229,7 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
 
 	/* If we're applying an offset correction then updating the
 	 * callibration would be likely to introduce further offsets. */
-	if (hubs->dcs_codes || hubs->no_series_update)
+	if (hubs->dcs_codes_l || hubs->dcs_codes_r || hubs->no_series_update)
 		return ret;
 
 	/* Only need to do this if the outputs are active */
@@ -699,6 +711,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "IN1L PGA", "IN1LP Switch", "IN1LP" },
 	{ "IN1L PGA", "IN1LN Switch", "IN1LN" },
 
+	{ "IN1L PGA", NULL, "VMID" },
+	{ "IN1R PGA", NULL, "VMID" },
+	{ "IN2L PGA", NULL, "VMID" },
+	{ "IN2R PGA", NULL, "VMID" },
+
 	{ "IN1R PGA", "IN1RP Switch", "IN1RP" },
 	{ "IN1R PGA", "IN1RN Switch", "IN1RN" },
 
@@ -716,12 +733,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "MIXINL", NULL, "Direct Voice" },
 	{ "MIXINL", NULL, "IN1LP" },
 	{ "MIXINL", NULL, "Left Output Mixer" },
+	{ "MIXINL", NULL, "VMID" },
 
 	{ "MIXINR", "IN1R Switch", "IN1R PGA" },
 	{ "MIXINR", "IN2R Switch", "IN2R PGA" },
 	{ "MIXINR", NULL, "Direct Voice" },
 	{ "MIXINR", NULL, "IN1RP" },
 	{ "MIXINR", NULL, "Right Output Mixer" },
+	{ "MIXINR", NULL, "VMID" },
 
 	{ "ADCL", NULL, "MIXINL" },
 	{ "ADCR", NULL, "MIXINR" },
@@ -752,6 +771,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "Earpiece Mixer", "Left Output Switch", "Left Output PGA" },
 	{ "Earpiece Mixer", "Right Output Switch", "Right Output PGA" },
 
+	{ "Earpiece Driver", NULL, "VMID" },
 	{ "Earpiece Driver", NULL, "Earpiece Mixer" },
 	{ "HPOUT2N", NULL, "Earpiece Driver" },
 	{ "HPOUT2P", NULL, "Earpiece Driver" },
@@ -774,9 +794,11 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "SPKR Boost", "SPKR Switch", "SPKR" },
 	{ "SPKR Boost", "SPKL Switch", "SPKL" },
 
+	{ "SPKL Driver", NULL, "VMID" },
 	{ "SPKL Driver", NULL, "SPKL Boost" },
 	{ "SPKL Driver", NULL, "CLK_SYS" },
 
+	{ "SPKR Driver", NULL, "VMID" },
 	{ "SPKR Driver", NULL, "SPKR Boost" },
 	{ "SPKR Driver", NULL, "CLK_SYS" },
 
@@ -790,12 +812,18 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 
 	{ "Headphone PGA", NULL, "Left Headphone Mux" },
 	{ "Headphone PGA", NULL, "Right Headphone Mux" },
+	{ "Headphone PGA", NULL, "VMID" },
 	{ "Headphone PGA", NULL, "CLK_SYS" },
 	{ "Headphone PGA", NULL, "Headphone Supply" },
 
 	{ "HPOUT1L", NULL, "Headphone PGA" },
 	{ "HPOUT1R", NULL, "Headphone PGA" },
 
+	{ "LINEOUT1N Driver", NULL, "VMID" },
+	{ "LINEOUT1P Driver", NULL, "VMID" },
+	{ "LINEOUT2N Driver", NULL, "VMID" },
+	{ "LINEOUT2P Driver", NULL, "VMID" },
+
 	{ "LINEOUT1N", NULL, "LINEOUT1N Driver" },
 	{ "LINEOUT1P", NULL, "LINEOUT1P Driver" },
 	{ "LINEOUT2N", NULL, "LINEOUT2N Driver" },
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 676b1252ab9..c674c7a502a 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -23,7 +23,8 @@ extern const unsigned int wm_hubs_spkmix_tlv[];
 
 /* This *must* be the first element of the codec->private_data struct */
 struct wm_hubs_data {
-	int dcs_codes;
+	int dcs_codes_l;
+	int dcs_codes_r;
 	int dcs_readback_mode;
 	int hp_startup_mode;
 	int series_startup;
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 8566238db2a..7173df254a9 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -732,16 +732,19 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 		davinci_hw_param(dev, substream->stream);
 
 	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_U8:
 	case SNDRV_PCM_FORMAT_S8:
 		dma_params->data_type = 1;
 		word_length = DAVINCI_AUDIO_WORD_8;
 		break;
 
+	case SNDRV_PCM_FORMAT_U16_LE:
 	case SNDRV_PCM_FORMAT_S16_LE:
 		dma_params->data_type = 2;
 		word_length = DAVINCI_AUDIO_WORD_16;
 		break;
 
+	case SNDRV_PCM_FORMAT_U32_LE:
 	case SNDRV_PCM_FORMAT_S32_LE:
 		dma_params->data_type = 4;
 		word_length = DAVINCI_AUDIO_WORD_32;
@@ -818,6 +821,13 @@ static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
 
 };
 
+#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
+				SNDRV_PCM_FMTBIT_U8 | \
+				SNDRV_PCM_FMTBIT_S16_LE | \
+				SNDRV_PCM_FMTBIT_U16_LE | \
+				SNDRV_PCM_FMTBIT_S32_LE | \
+				SNDRV_PCM_FMTBIT_U32_LE)
+
 static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 	{
 		.name		= "davinci-mcasp.0",
@@ -825,17 +835,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 			.channels_min	= 2,
 			.channels_max 	= 2,
 			.rates 		= DAVINCI_MCASP_RATES,
-			.formats 	= SNDRV_PCM_FMTBIT_S8 |
-						SNDRV_PCM_FMTBIT_S16_LE |
-						SNDRV_PCM_FMTBIT_S32_LE,
+			.formats	= DAVINCI_MCASP_PCM_FMTS,
 		},
 		.capture 	= {
 			.channels_min 	= 2,
 			.channels_max 	= 2,
 			.rates 		= DAVINCI_MCASP_RATES,
-			.formats	= SNDRV_PCM_FMTBIT_S8 |
-						SNDRV_PCM_FMTBIT_S16_LE |
-						SNDRV_PCM_FMTBIT_S32_LE,
+			.formats	= DAVINCI_MCASP_PCM_FMTS,
 		},
 		.ops 		= &davinci_mcasp_dai_ops,
 
@@ -846,7 +852,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 			.channels_min	= 1,
 			.channels_max	= 384,
 			.rates		= DAVINCI_MCASP_RATES,
-			.formats	= SNDRV_PCM_FMTBIT_S16_LE,
+			.formats	= DAVINCI_MCASP_PCM_FMTS,
 		},
 		.ops 		= &davinci_mcasp_dai_ops,
 	},
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index a49e667373b..d5fe08cc5db 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -180,7 +180,6 @@ 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 link = prtd->asp_link[0];
 	unsigned int period_size;
 	unsigned int dma_offset;
 	dma_addr_t dma_pos;
@@ -198,7 +197,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	fifo_level = prtd->params->fifo_level;
 
 	pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d "
-		"dma_ptr = %x period_size=%x\n", link, dma_pos, period_size);
+		"dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos,
+		period_size);
 
 	data_type = prtd->params->data_type;
 	count = period_size / data_type;
@@ -222,17 +222,19 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream)
 	}
 
 	acnt = prtd->params->acnt;
-	edma_set_src(link, src, INCR, W8BIT);
-	edma_set_dest(link, dst, INCR, W8BIT);
+	edma_set_src(prtd->asp_link[0], src, INCR, W8BIT);
+	edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT);
 
-	edma_set_src_index(link, src_bidx, src_cidx);
-	edma_set_dest_index(link, dst_bidx, dst_cidx);
+	edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx);
+	edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx);
 
 	if (!fifo_level)
-		edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC);
+		edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0,
+							ASYNC);
 	else
-		edma_set_transfer_params(link, acnt, fifo_level, count,
-							fifo_level, ABSYNC);
+		edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level,
+							count, fifo_level,
+							ABSYNC);
 }
 
 static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data)
@@ -305,7 +307,6 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream)
 	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)) {
@@ -316,28 +317,26 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream)
 		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);
+		edma_set_src(prtd->asp_link[1], 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);
+		edma_set_src_index(prtd->asp_link[0], data_type,
+				data_type * fifo_level);
+		edma_set_src_index(prtd->asp_link[1], data_type,
+				data_type * fifo_level);
 
-		link = prtd->ram_link;
-		edma_set_src(link, runtime->dma_addr, INCR, W32BIT);
+		edma_set_src(prtd->ram_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);
+		edma_set_dest(prtd->asp_link[1], 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);
+		edma_set_dest_index(prtd->asp_link[0], data_type,
+				data_type * fifo_level);
+		edma_set_dest_index(prtd->asp_link[1], data_type,
+				data_type * fifo_level);
 
-		link = prtd->ram_link;
-		edma_set_dest(link, runtime->dma_addr, INCR, W32BIT);
+		edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT);
 	}
 
 	if (!fifo_level) {
@@ -354,10 +353,9 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream)
 				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,
+	edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx);
+	edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx);
+	edma_set_transfer_params(prtd->ram_link, ping_size, 2,
 			runtime->periods, 2, ASYNC);
 
 	/* init master params */
@@ -406,32 +404,32 @@ static int request_ping_pong(struct snd_pcm_substream *substream,
 {
 	dma_addr_t asp_src_ping;
 	dma_addr_t asp_dst_ping;
-	int link;
+	int ret;
 	struct davinci_pcm_dma_params *params = prtd->params;
 
 	/* Request ram master channel */
-	link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY,
+	ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY,
 				  davinci_pcm_dma_irq, substream,
 				  prtd->params->ram_chan_q);
-	if (link < 0)
+	if (ret < 0)
 		goto exit1;
 
 	/* Request ram link channel */
-	link = prtd->ram_link = edma_alloc_slot(
+	ret = prtd->ram_link = edma_alloc_slot(
 			EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY);
-	if (link < 0)
+	if (ret < 0)
 		goto exit2;
 
-	link = prtd->asp_link[1] = edma_alloc_slot(
+	ret = prtd->asp_link[1] = edma_alloc_slot(
 			EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY);
-	if (link < 0)
+	if (ret < 0)
 		goto exit3;
 
 	prtd->ram_link2 = -1;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		link = prtd->ram_link2 = edma_alloc_slot(
+		ret = prtd->ram_link2 = edma_alloc_slot(
 			EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY);
-		if (link < 0)
+		if (ret < 0)
 			goto exit4;
 	}
 	/* circle ping-pong buffers */
@@ -448,36 +446,33 @@ static int request_ping_pong(struct snd_pcm_substream *substream,
 		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_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT);
+	edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT);
+	edma_set_src_index(prtd->asp_link[0], 0, 0);
+	edma_set_dest_index(prtd->asp_link[0], 0, 0);
 
-	edma_read_slot(link, &prtd->asp_params);
+	edma_read_slot(prtd->asp_link[0], &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);
+	edma_write_slot(prtd->asp_link[0], &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_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT);
+	edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT);
+	edma_set_src_index(prtd->asp_link[1], 0, 0);
+	edma_set_dest_index(prtd->asp_link[1], 0, 0);
 
-	edma_read_slot(link, &prtd->asp_params);
+	edma_read_slot(prtd->asp_link[1], &prtd->asp_params);
 	prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f));
 	/* interrupt after every pong completion */
 	prtd->asp_params.opt |= TCINTEN | TCCHEN |
 		EDMA_TCC(prtd->ram_channel & 0x3f);
-	edma_write_slot(link, &prtd->asp_params);
+	edma_write_slot(prtd->asp_link[1], &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);
+	edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT);
+	edma_set_dest(prtd->ram_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,
@@ -494,7 +489,7 @@ exit2:
 	edma_free_channel(prtd->ram_channel);
 	prtd->ram_channel = -1;
 exit1:
-	return link;
+	return ret;
 }
 
 static int davinci_pcm_dma_request(struct snd_pcm_substream *substream)
@@ -502,22 +497,22 @@ 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 davinci_pcm_dma_params *params = prtd->params;
-	int link;
+	int ret;
 
 	if (!params)
 		return -ENODEV;
 
 	/* Request asp master DMA channel */
-	link = prtd->asp_channel = edma_alloc_channel(params->channel,
+	ret = prtd->asp_channel = edma_alloc_channel(params->channel,
 			davinci_pcm_dma_irq, substream,
 			prtd->params->asp_chan_q);
-	if (link < 0)
+	if (ret < 0)
 		goto exit1;
 
 	/* Request asp link channels */
-	link = prtd->asp_link[0] = edma_alloc_slot(
+	ret = prtd->asp_link[0] = edma_alloc_slot(
 			EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY);
-	if (link < 0)
+	if (ret < 0)
 		goto exit2;
 
 	iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data;
@@ -537,17 +532,17 @@ 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(link, &prtd->asp_params);
+	edma_read_slot(prtd->asp_link[0], &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);
+	prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5;
+	edma_write_slot(prtd->asp_link[0], &prtd->asp_params);
 	return 0;
 exit2:
 	edma_free_channel(prtd->asp_channel);
 	prtd->asp_channel = -1;
 exit1:
-	return link;
+	return ret;
 }
 
 static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c
index d3aa15119d2..0134d4e9131 100644
--- a/sound/soc/ep93xx/edb93xx.c
+++ b/sound/soc/ep93xx/edb93xx.c
@@ -28,12 +28,6 @@
 #include <mach/hardware.h>
 #include "ep93xx-pcm.h"
 
-#define edb93xx_has_audio() (machine_is_edb9301() ||	\
-			     machine_is_edb9302() ||	\
-			     machine_is_edb9302a() ||	\
-			     machine_is_edb9307a() ||	\
-			     machine_is_edb9315a())
-
 static int edb93xx_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
@@ -94,49 +88,61 @@ static struct snd_soc_card snd_soc_edb93xx = {
 	.num_links	= 1,
 };
 
-static struct platform_device *edb93xx_snd_device;
-
-static int __init edb93xx_init(void)
+static int __devinit edb93xx_probe(struct platform_device *pdev)
 {
+	struct snd_soc_card *card = &snd_soc_edb93xx;
 	int ret;
 
-	if (!edb93xx_has_audio())
-		return -ENODEV;
-
 	ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
 				 EP93XX_SYSCON_I2SCLKDIV_ORIDE |
 				 EP93XX_SYSCON_I2SCLKDIV_SPOL);
 	if (ret)
 		return ret;
 
-	edb93xx_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!edb93xx_snd_device) {
-		ret = -ENOMEM;
-		goto free_i2s;
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		ep93xx_i2s_release();
 	}
 
-	platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx);
-	ret = platform_device_add(edb93xx_snd_device);
-	if (ret)
-		goto device_put;
+	return ret;
+}
 
-	return 0;
+static int __devexit edb93xx_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
 
-device_put:
-	platform_device_put(edb93xx_snd_device);
-free_i2s:
+	snd_soc_unregister_card(card);
 	ep93xx_i2s_release();
-	return ret;
+
+	return 0;
+}
+
+static struct platform_driver edb93xx_driver = {
+	.driver		= {
+		.name	= "edb93xx-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= edb93xx_probe,
+	.remove		= __devexit_p(edb93xx_remove),
+};
+
+static int __init edb93xx_init(void)
+{
+	return platform_driver_register(&edb93xx_driver);
 }
 module_init(edb93xx_init);
 
 static void __exit edb93xx_exit(void)
 {
-	platform_device_unregister(edb93xx_snd_device);
-	ep93xx_i2s_release();
+	platform_driver_unregister(&edb93xx_driver);
 }
 module_exit(edb93xx_exit);
 
 MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
 MODULE_DESCRIPTION("ALSA SoC EDB93xx");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:edb93xx-audio");
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 8dfd3ad84b1..d00230a591b 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -355,3 +355,4 @@ module_exit(ep93xx_soc_platform_exit);
 MODULE_AUTHOR("Ryan Mallon");
 MODULE_DESCRIPTION("EP93xx ALSA PCM interface");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ep93xx-pcm-audio");
diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c
index 286817946c5..968cb316d51 100644
--- a/sound/soc/ep93xx/simone.c
+++ b/sound/soc/ep93xx/simone.c
@@ -39,53 +39,61 @@ static struct snd_soc_card snd_soc_simone = {
 };
 
 static struct platform_device *simone_snd_ac97_device;
-static struct platform_device *simone_snd_device;
 
-static int __init simone_init(void)
+static int __devinit simone_probe(struct platform_device *pdev)
 {
+	struct snd_soc_card *card = &snd_soc_simone;
 	int ret;
 
-	if (!machine_is_sim_one())
-		return -ENODEV;
-
-	simone_snd_ac97_device = platform_device_alloc("ac97-codec", -1);
-	if (!simone_snd_ac97_device)
-		return -ENOMEM;
+	simone_snd_ac97_device = platform_device_register_simple("ac97-codec",
+								 -1, NULL, 0);
+	if (IS_ERR(simone_snd_ac97_device))
+		return PTR_ERR(simone_snd_ac97_device);
 
-	ret = platform_device_add(simone_snd_ac97_device);
-	if (ret)
-		goto fail1;
+	card->dev = &pdev->dev;
 
-	simone_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!simone_snd_device) {
-		ret = -ENOMEM;
-		goto fail2;
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		platform_device_unregister(simone_snd_ac97_device);
 	}
 
-	platform_set_drvdata(simone_snd_device, &snd_soc_simone);
-	ret = platform_device_add(simone_snd_device);
-	if (ret)
-		goto fail3;
+	return ret;
+}
+
+static int __devexit simone_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	platform_device_unregister(simone_snd_ac97_device);
 
 	return 0;
+}
 
-fail3:
-	platform_device_put(simone_snd_device);
-fail2:
-	platform_device_del(simone_snd_ac97_device);
-fail1:
-	platform_device_put(simone_snd_ac97_device);
-	return ret;
+static struct platform_driver simone_driver = {
+	.driver		= {
+		.name	= "simone-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= simone_probe,
+	.remove		= __devexit_p(simone_remove),
+};
+
+static int __init simone_init(void)
+{
+	return platform_driver_register(&simone_driver);
 }
 module_init(simone_init);
 
 static void __exit simone_exit(void)
 {
-	platform_device_unregister(simone_snd_device);
-	platform_device_unregister(simone_snd_ac97_device);
+	platform_driver_unregister(&simone_driver);
 }
 module_exit(simone_exit);
 
 MODULE_DESCRIPTION("ALSA SoC Simplemachines Sim.One");
 MODULE_AUTHOR("Mika Westerberg <mika.westerberg@iki.fi>");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:simone-audio");
diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c
index c8aa8a5003c..f74ac54c285 100644
--- a/sound/soc/ep93xx/snappercl15.c
+++ b/sound/soc/ep93xx/snappercl15.c
@@ -104,37 +104,56 @@ static struct snd_soc_card snd_soc_snappercl15 = {
 	.num_links	= 1,
 };
 
-static struct platform_device *snappercl15_snd_device;
-
-static int __init snappercl15_init(void)
+static int __devinit snappercl15_probe(struct platform_device *pdev)
 {
+	struct snd_soc_card *card = &snd_soc_snappercl15;
 	int ret;
 
-	if (!machine_is_snapper_cl15())
-		return -ENODEV;
-
 	ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
 				 EP93XX_SYSCON_I2SCLKDIV_ORIDE |
 				 EP93XX_SYSCON_I2SCLKDIV_SPOL);
 	if (ret)
 		return ret;
 
-	snappercl15_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!snappercl15_snd_device)
-		return -ENOMEM;
-	
-	platform_set_drvdata(snappercl15_snd_device, &snd_soc_snappercl15);
-	ret = platform_device_add(snappercl15_snd_device);
-	if (ret)
-		platform_device_put(snappercl15_snd_device);
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		ep93xx_i2s_release();
+	}
 
 	return ret;
 }
 
-static void __exit snappercl15_exit(void)
+static int __devexit snappercl15_remove(struct platform_device *pdev)
 {
-	platform_device_unregister(snappercl15_snd_device);
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
 	ep93xx_i2s_release();
+
+	return 0;
+}
+
+static struct platform_driver snappercl15_driver = {
+	.driver		= {
+		.name	= "snappercl15-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= snappercl15_probe,
+	.remove		= __devexit_p(snappercl15_remove),
+};
+
+static int __init snappercl15_init(void)
+{
+	return platform_driver_register(&snappercl15_driver);
+}
+
+static void __exit snappercl15_exit(void)
+{
+	platform_driver_unregister(&snappercl15_driver);
 }
 
 module_init(snappercl15_init);
@@ -143,4 +162,4 @@ module_exit(snappercl15_exit);
 MODULE_AUTHOR("Ryan Mallon");
 MODULE_DESCRIPTION("ALSA SoC Snapper CL15");
 MODULE_LICENSE("GPL");
-
+MODULE_ALIAS("platform:snappercl15-audio");
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index cb50598338e..ef15402a3bc 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -297,7 +297,6 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id)
 static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 	static u64 fsl_dma_dmamask = DMA_BIT_MASK(36);
 	int ret;
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index d48afea5d93..0268cf98973 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -78,7 +78,6 @@
  * @second_stream: pointer to second stream
  * @playback: the number of playback streams opened
  * @capture: the number of capture streams opened
- * @asynchronous: 0=synchronous mode, 1=asynchronous mode
  * @cpu_dai: the CPU DAI for this device
  * @dev_attr: the sysfs device attribute structure
  * @stats: SSI statistics
@@ -90,9 +89,6 @@ struct fsl_ssi_private {
 	unsigned int irq;
 	struct snd_pcm_substream *first_stream;
 	struct snd_pcm_substream *second_stream;
-	unsigned int playback;
-	unsigned int capture;
-	int asynchronous;
 	unsigned int fifo_depth;
 	struct snd_soc_dai_driver cpu_dai_drv;
 	struct device_attribute dev_attr;
@@ -281,24 +277,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	struct fsl_ssi_private *ssi_private =
+		snd_soc_dai_get_drvdata(rtd->cpu_dai);
+	int synchronous = ssi_private->cpu_dai_drv.symmetric_rates;
 
 	/*
 	 * If this is the first stream opened, then request the IRQ
 	 * and initialize the SSI registers.
 	 */
-	if (!ssi_private->playback && !ssi_private->capture) {
+	if (!ssi_private->first_stream) {
 		struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-		int ret;
-
-		/* The 'name' should not have any slashes in it. */
-		ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0,
-				  ssi_private->name, ssi_private);
-		if (ret < 0) {
-			dev_err(substream->pcm->card->dev,
-				"could not claim irq %u\n", ssi_private->irq);
-			return ret;
-		}
+
+		ssi_private->first_stream = substream;
 
 		/*
 		 * Section 16.5 of the MPC8610 reference manual says that the
@@ -316,7 +306,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		clrsetbits_be32(&ssi->scr,
 			CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
 			CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
-			| (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN));
+			| (synchronous ? CCSR_SSI_SCR_SYN : 0));
 
 		out_be32(&ssi->stcr,
 			 CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
@@ -333,7 +323,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 * master.
 		 */
 
-		/* 4. Enable the interrupts and DMA requests */
+		/* Enable the interrupts and DMA requests */
 		out_be32(&ssi->sier, SIER_FLAGS);
 
 		/*
@@ -362,58 +352,47 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 * this is bad is because at this point, the PCM driver has not
 		 * finished initializing the DMA controller.
 		 */
-	}
+	} else {
+		if (synchronous) {
+			struct snd_pcm_runtime *first_runtime =
+				ssi_private->first_stream->runtime;
+			/*
+			 * This is the second stream open, and we're in
+			 * synchronous mode, so we need to impose sample
+			 * sample size constraints. This is because STCCR is
+			 * used for playback and capture in synchronous mode,
+			 * so there's no way to specify different word
+			 * lengths.
+			 *
+			 * Note that this can cause a race condition if the
+			 * second stream is opened before the first stream is
+			 * fully initialized.  We provide some protection by
+			 * checking to make sure the first stream is
+			 * initialized, but it's not perfect.  ALSA sometimes
+			 * re-initializes the driver with a different sample
+			 * rate or size.  If the second stream is opened
+			 * before the first stream has received its final
+			 * parameters, then the second stream may be
+			 * constrained to the wrong sample rate or size.
+			 */
+			if (!first_runtime->sample_bits) {
+				dev_err(substream->pcm->card->dev,
+					"set sample size in %s stream first\n",
+					substream->stream ==
+					SNDRV_PCM_STREAM_PLAYBACK
+					? "capture" : "playback");
+				return -EAGAIN;
+			}
 
-	if (!ssi_private->first_stream)
-		ssi_private->first_stream = substream;
-	else {
-		/* This is the second stream open, so we need to impose sample
-		 * rate and maybe sample size constraints.  Note that this can
-		 * cause a race condition if the second stream is opened before
-		 * the first stream is fully initialized.
-		 *
-		 * We provide some protection by checking to make sure the first
-		 * stream is initialized, but it's not perfect.  ALSA sometimes
-		 * re-initializes the driver with a different sample rate or
-		 * size.  If the second stream is opened before the first stream
-		 * has received its final parameters, then the second stream may
-		 * be constrained to the wrong sample rate or size.
-		 *
-		 * FIXME: This code does not handle opening and closing streams
-		 * repeatedly.  If you open two streams and then close the first
-		 * one, you may not be able to open another stream until you
-		 * close the second one as well.
-		 */
-		struct snd_pcm_runtime *first_runtime =
-			ssi_private->first_stream->runtime;
-
-		if (!first_runtime->sample_bits) {
-			dev_err(substream->pcm->card->dev,
-				"set sample size in %s stream first\n",
-				substream->stream == SNDRV_PCM_STREAM_PLAYBACK
-				? "capture" : "playback");
-			return -EAGAIN;
-		}
-
-		/* If we're in synchronous mode, then we need to constrain
-		 * the sample size as well.  We don't support independent sample
-		 * rates in asynchronous mode.
-		 */
-		if (!ssi_private->asynchronous)
 			snd_pcm_hw_constraint_minmax(substream->runtime,
 				SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
 				first_runtime->sample_bits,
 				first_runtime->sample_bits);
+		}
 
 		ssi_private->second_stream = substream;
 	}
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		ssi_private->playback++;
-
-	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-		ssi_private->capture++;
-
 	return 0;
 }
 
@@ -434,24 +413,35 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai)
 {
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	unsigned int sample_size =
+		snd_pcm_format_width(params_format(hw_params));
+	u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
+	int enabled = in_be32(&ssi->scr) & CCSR_SSI_SCR_SSIEN;
 
-	if (substream == ssi_private->first_stream) {
-		struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
-		unsigned int sample_size =
-			snd_pcm_format_width(params_format(hw_params));
-		u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
+	/*
+	 * If we're in synchronous mode, and the SSI is already enabled,
+	 * then STCCR is already set properly.
+	 */
+	if (enabled && ssi_private->cpu_dai_drv.symmetric_rates)
+		return 0;
 
-		/* The SSI should always be disabled at this points (SSIEN=0) */
+	/*
+	 * FIXME: The documentation says that SxCCR[WL] should not be
+	 * modified while the SSI is enabled.  The only time this can
+	 * happen is if we're trying to do simultaneous playback and
+	 * capture in asynchronous mode.  Unfortunately, I have been enable
+	 * to get that to work at all on the P1022DS.  Therefore, we don't
+	 * bother to disable/enable the SSI when setting SxCCR[WL], because
+	 * the SSI will stop anyway.  Maybe one day, this will get fixed.
+	 */
 
-		/* In synchronous mode, the SSI uses STCCR for capture */
-		if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
-		    !ssi_private->asynchronous)
-			clrsetbits_be32(&ssi->stccr,
-					CCSR_SSI_SxCCR_WL_MASK, wl);
-		else
-			clrsetbits_be32(&ssi->srccr,
-					CCSR_SSI_SxCCR_WL_MASK, wl);
-	}
+	/* In synchronous mode, the SSI uses STCCR for capture */
+	if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
+	    ssi_private->cpu_dai_drv.symmetric_rates)
+		clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+	else
+		clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl);
 
 	return 0;
 }
@@ -474,7 +464,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-		clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			setbits32(&ssi->scr,
@@ -510,27 +499,18 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		ssi_private->playback--;
-
-	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
-		ssi_private->capture--;
-
 	if (ssi_private->first_stream == substream)
 		ssi_private->first_stream = ssi_private->second_stream;
 
 	ssi_private->second_stream = NULL;
 
 	/*
-	 * If this is the last active substream, disable the SSI and release
-	 * the IRQ.
+	 * If this is the last active substream, disable the SSI.
 	 */
-	if (!ssi_private->playback && !ssi_private->capture) {
+	if (!ssi_private->first_stream) {
 		struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
 
 		clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
-
-		free_irq(ssi_private->irq, ssi_private);
 	}
 }
 
@@ -675,22 +655,33 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 	ret = of_address_to_resource(np, 0, &res);
 	if (ret) {
 		dev_err(&pdev->dev, "could not determine device resources\n");
-		kfree(ssi_private);
-		return ret;
+		goto error_kmalloc;
 	}
 	ssi_private->ssi = of_iomap(np, 0);
 	if (!ssi_private->ssi) {
 		dev_err(&pdev->dev, "could not map device resources\n");
-		kfree(ssi_private);
-		return -ENOMEM;
+		ret = -ENOMEM;
+		goto error_kmalloc;
 	}
 	ssi_private->ssi_phys = res.start;
+
 	ssi_private->irq = irq_of_parse_and_map(np, 0);
+	if (ssi_private->irq == NO_IRQ) {
+		dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
+		ret = -ENXIO;
+		goto error_iomap;
+	}
+
+	/* The 'name' should not have any slashes in it. */
+	ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, ssi_private->name,
+			  ssi_private);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq);
+		goto error_irqmap;
+	}
 
 	/* Are the RX and the TX clocks locked? */
-	if (of_find_property(np, "fsl,ssi-asynchronous", NULL))
-		ssi_private->asynchronous = 1;
-	else
+	if (!of_find_property(np, "fsl,ssi-asynchronous", NULL))
 		ssi_private->cpu_dai_drv.symmetric_rates = 1;
 
 	/* Determine the FIFO depth. */
@@ -711,7 +702,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 	if (ret) {
 		dev_err(&pdev->dev, "could not create sysfs %s file\n",
 			ssi_private->dev_attr.attr.name);
-		goto error;
+		goto error_irq;
 	}
 
 	/* Register with ASoC */
@@ -720,7 +711,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 	ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
-		goto error;
+		goto error_dev;
 	}
 
 	/* Trigger the machine driver's probe function.  The platform driver
@@ -741,18 +732,28 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
 	if (IS_ERR(ssi_private->pdev)) {
 		ret = PTR_ERR(ssi_private->pdev);
 		dev_err(&pdev->dev, "failed to register platform: %d\n", ret);
-		goto error;
+		goto error_dai;
 	}
 
 	return 0;
 
-error:
+error_dai:
 	snd_soc_unregister_dai(&pdev->dev);
+
+error_dev:
 	dev_set_drvdata(&pdev->dev, NULL);
-	if (dev_attr)
-		device_remove_file(&pdev->dev, dev_attr);
+	device_remove_file(&pdev->dev, dev_attr);
+
+error_irq:
+	free_irq(ssi_private->irq, ssi_private);
+
+error_irqmap:
 	irq_dispose_mapping(ssi_private->irq);
+
+error_iomap:
 	iounmap(ssi_private->ssi);
+
+error_kmalloc:
 	kfree(ssi_private);
 
 	return ret;
@@ -766,6 +767,9 @@ static int fsl_ssi_remove(struct platform_device *pdev)
 	snd_soc_unregister_dai(&pdev->dev);
 	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
 
+	free_irq(ssi_private->irq, ssi_private);
+	irq_dispose_mapping(ssi_private->irq);
+
 	kfree(ssi_private);
 	dev_set_drvdata(&pdev->dev, NULL);
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 358f0baaf71..31af405bda8 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -505,7 +505,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	return 0;
 
 error_sound:
-	platform_device_unregister(sound_device);
+	platform_device_put(sound_device);
 error:
 	kfree(machine_data);
 error_alloc:
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index fcb862eb0c7..2c064a9824a 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -267,7 +267,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len)
 	if (bus < 0)
 		return bus;
 
-	snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr);
+	snprintf(buf, len, "%s.%u-%04x", temp, bus, addr);
 
 	return 0;
 }
@@ -506,7 +506,7 @@ static int p1022_ds_probe(struct platform_device *pdev)
 
 error:
 	if (sound_device)
-		platform_device_unregister(sound_device);
+		platform_device_put(sound_device);
 
 	kfree(mdata);
 error_put:
diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c
index 7945625e0e0..c8527ead373 100644
--- a/sound/soc/imx/imx-pcm-fiq.c
+++ b/sound/soc/imx/imx-pcm-fiq.c
@@ -242,23 +242,22 @@ static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
+	struct snd_pcm_substream *substream;
 	int ret;
 
 	ret = imx_pcm_new(rtd);
 	if (ret)
 		return ret;
 
-	if (dai->driver->playback.channels_min) {
-		struct snd_pcm_substream *substream =
-			pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+	if (substream) {
 		struct snd_dma_buffer *buf = &substream->dma_buffer;
 
 		imx_ssi_fiq_tx_buffer = (unsigned long)buf->area;
 	}
 
-	if (dai->driver->capture.channels_min) {
-		struct snd_pcm_substream *substream =
-			pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+	substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+	if (substream) {
 		struct snd_dma_buffer *buf = &substream->dma_buffer;
 
 		imx_ssi_fiq_rx_buffer = (unsigned long)buf->area;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 10a8e278375..4297cb6af42 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -357,8 +357,8 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int ret;
 
-	ret = dma_mmap_coherent(NULL, vma, runtime->dma_area,
-			runtime->dma_addr, runtime->dma_bytes);
+	ret = dma_mmap_writecombine(substream->pcm->card->dev, vma,
+		runtime->dma_area, runtime->dma_addr, runtime->dma_bytes);
 
 	pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret,
 			runtime->dma_area,
@@ -399,14 +399,14 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd)
 		card->dev->dma_mask = &imx_pcm_dmamask;
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
-	if (dai->driver->playback.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
 		ret = imx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_PLAYBACK);
 		if (ret)
 			goto out;
 	}
 
-	if (dai->driver->capture.channels_min) {
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
 		ret = imx_pcm_preallocate_dma_buffer(pcm,
 			SNDRV_PCM_STREAM_CAPTURE);
 		if (ret)
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index 3e7826058ef..9925d20ab0a 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -226,13 +226,18 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream)
 
 static int sst_platform_open(struct snd_pcm_substream *substream)
 {
-	struct snd_pcm_runtime *runtime;
+	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct sst_runtime_stream *stream;
 	int ret_val = 0;
 
 	pr_debug("sst_platform_open called\n");
-	runtime = substream->runtime;
-	runtime->hw = sst_platform_pcm_hw;
+
+	snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw);
+	ret_val = snd_pcm_hw_constraint_integer(runtime,
+						SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret_val < 0)
+		return ret_val;
+
 	stream = kzalloc(sizeof(*stream), GFP_KERNEL);
 	if (!stream)
 		return -ENOMEM;
@@ -259,8 +264,8 @@ static int sst_platform_open(struct snd_pcm_substream *substream)
 		return ret_val;
 	}
 	runtime->private_data = stream;
-	return snd_pcm_hw_constraint_integer(runtime,
-			 SNDRV_PCM_HW_PARAM_PERIODS);
+
+	return 0;
 }
 
 static int sst_platform_close(struct snd_pcm_substream *substream)
@@ -469,7 +474,7 @@ static struct platform_driver sst_platform_driver = {
 static int __init sst_soc_platform_init(void)
 {
 	pr_debug("sst_soc_platform_init called\n");
-	return  platform_driver_register(&sst_platform_driver);
+	return platform_driver_register(&sst_platform_driver);
 }
 module_init(sst_soc_platform_init);
 
diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig
new file mode 100644
index 00000000000..e4ba8d5f25f
--- /dev/null
+++ b/sound/soc/mxs/Kconfig
@@ -0,0 +1,20 @@
+menuconfig SND_MXS_SOC
+	tristate "SoC Audio for Freescale MXS CPUs"
+	depends on ARCH_MXS
+	select SND_PCM
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the MXS SAIF interface.
+
+
+if SND_MXS_SOC
+
+config SND_SOC_MXS_SGTL5000
+	tristate "SoC Audio support for i.MX boards with sgtl5000"
+	depends on I2C
+	select SND_SOC_SGTL5000
+	help
+	  Say Y if you want to add support for SoC audio on an MXS board with
+	  a sgtl5000 codec.
+
+endif	# SND_MXS_SOC
diff --git a/sound/soc/mxs/Makefile b/sound/soc/mxs/Makefile
new file mode 100644
index 00000000000..565b5b51e8b
--- /dev/null
+++ b/sound/soc/mxs/Makefile
@@ -0,0 +1,10 @@
+# MXS Platform Support
+snd-soc-mxs-objs := mxs-saif.o
+snd-soc-mxs-pcm-objs := mxs-pcm.o
+
+obj-$(CONFIG_SND_MXS_SOC) += snd-soc-mxs.o snd-soc-mxs-pcm.o
+
+# i.MX Machine Support
+snd-soc-mxs-sgtl5000-objs := mxs-sgtl5000.o
+
+obj-$(CONFIG_SND_SOC_MXS_SGTL5000) += snd-soc-mxs-sgtl5000.o
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
new file mode 100644
index 00000000000..dea5aa4aa64
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -0,0 +1,359 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * Based on sound/soc/imx/imx-pcm-dma-mx2.c
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/dma.h>
+#include "mxs-pcm.h"
+
+static struct snd_pcm_hardware snd_mxs_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_INTERLEAVED,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S20_3LE |
+				  SNDRV_PCM_FMTBIT_S24_LE,
+	.channels_min		= 2,
+	.channels_max		= 2,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 8192,
+	.periods_min		= 1,
+	.periods_max		= 52,
+	.buffer_bytes_max	= 64 * 1024,
+	.fifo_size		= 32,
+
+};
+
+static void audio_dma_irq(void *data)
+{
+	struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+	iprtd->offset += iprtd->period_bytes;
+	iprtd->offset %= iprtd->period_bytes * iprtd->periods;
+	snd_pcm_period_elapsed(substream);
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+	struct mxs_pcm_runtime_data *iprtd = param;
+	struct mxs_pcm_dma_params *dma_params = iprtd->dma_params;
+
+	if (!mxs_dma_is_apbx(chan))
+		return false;
+
+	if (chan->chan_id != dma_params->chan_num)
+		return false;
+
+	chan->private = &iprtd->dma_data;
+
+	return true;
+}
+
+static int mxs_dma_alloc(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+	dma_cap_mask_t mask;
+
+	iprtd->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	dma_cap_zero(mask);
+	dma_cap_set(DMA_SLAVE, mask);
+	iprtd->dma_data.chan_irq = iprtd->dma_params->chan_irq;
+	iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+	if (!iprtd->dma_chan)
+		return -EINVAL;
+
+	return 0;
+}
+
+static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+	unsigned long dma_addr;
+	struct dma_chan *chan;
+	int ret;
+
+	ret = mxs_dma_alloc(substream, params);
+	if (ret)
+		return ret;
+	chan = iprtd->dma_chan;
+
+	iprtd->size = params_buffer_bytes(params);
+	iprtd->periods = params_periods(params);
+	iprtd->period_bytes = params_period_bytes(params);
+	iprtd->offset = 0;
+	iprtd->period_time = HZ / (params_rate(params) /
+			params_period_size(params));
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	dma_addr = runtime->dma_addr;
+
+	iprtd->buf = substream->dma_buffer.area;
+
+	iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr,
+			iprtd->period_bytes * iprtd->periods,
+			iprtd->period_bytes,
+			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+			DMA_TO_DEVICE : DMA_FROM_DEVICE);
+	if (!iprtd->desc) {
+		dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+		return -EINVAL;
+	}
+
+	iprtd->desc->callback = audio_dma_irq;
+	iprtd->desc->callback_param = substream;
+
+	return 0;
+}
+
+static int snd_mxs_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+	if (iprtd->dma_chan) {
+		dma_release_channel(iprtd->dma_chan);
+		iprtd->dma_chan = NULL;
+	}
+
+	return 0;
+}
+
+static int snd_mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		dmaengine_submit(iprtd->desc);
+
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		dmaengine_terminate_all(iprtd->dma_chan);
+
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static snd_pcm_uframes_t snd_mxs_pcm_pointer(
+		struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+	return bytes_to_frames(substream->runtime, iprtd->offset);
+}
+
+static int snd_mxs_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_pcm_runtime_data *iprtd;
+	int ret;
+
+	iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL);
+	if (iprtd == NULL)
+		return -ENOMEM;
+	runtime->private_data = iprtd;
+
+	ret = snd_pcm_hw_constraint_integer(substream->runtime,
+			SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0) {
+		kfree(iprtd);
+		return ret;
+	}
+
+	snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware);
+
+	return 0;
+}
+
+static int snd_mxs_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct mxs_pcm_runtime_data *iprtd = runtime->private_data;
+
+	kfree(iprtd);
+
+	return 0;
+}
+
+static int snd_mxs_pcm_mmap(struct snd_pcm_substream *substream,
+		struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+					runtime->dma_area,
+					runtime->dma_addr,
+					runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops mxs_pcm_ops = {
+	.open		= snd_mxs_open,
+	.close		= snd_mxs_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= snd_mxs_pcm_hw_params,
+	.hw_free	= snd_mxs_pcm_hw_free,
+	.trigger	= snd_mxs_pcm_trigger,
+	.pointer	= snd_mxs_pcm_pointer,
+	.mmap		= snd_mxs_pcm_mmap,
+};
+
+static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = snd_mxs_hardware.buffer_bytes_max;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+					   &buf->addr, GFP_KERNEL);
+	if (!buf->area)
+		return -ENOMEM;
+	buf->bytes = size;
+
+	return 0;
+}
+
+static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32);
+static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret = 0;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &mxs_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+		ret = mxs_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+		ret = mxs_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+
+out:
+	return ret;
+}
+
+static void mxs_pcm_free(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+
+		dma_free_writecombine(pcm->card->dev, buf->bytes,
+				      buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+static struct snd_soc_platform_driver mxs_soc_platform = {
+	.ops		= &mxs_pcm_ops,
+	.pcm_new	= mxs_pcm_new,
+	.pcm_free	= mxs_pcm_free,
+};
+
+static int __devinit mxs_soc_platform_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_platform(&pdev->dev, &mxs_soc_platform);
+}
+
+static int __devexit mxs_soc_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver mxs_pcm_driver = {
+	.driver = {
+		.name = "mxs-pcm-audio",
+		.owner = THIS_MODULE,
+	},
+	.probe = mxs_soc_platform_probe,
+	.remove = __devexit_p(mxs_soc_platform_remove),
+};
+
+static int __init snd_mxs_pcm_init(void)
+{
+	return platform_driver_register(&mxs_pcm_driver);
+}
+module_init(snd_mxs_pcm_init);
+
+static void __exit snd_mxs_pcm_exit(void)
+{
+	platform_driver_unregister(&mxs_pcm_driver);
+}
+module_exit(snd_mxs_pcm_exit);
diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h
new file mode 100644
index 00000000000..f55ac4f7a76
--- /dev/null
+++ b/sound/soc/mxs/mxs-pcm.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef _MXS_PCM_H
+#define _MXS_PCM_H
+
+#include <mach/dma.h>
+
+struct mxs_pcm_dma_params {
+	int chan_irq;
+	int chan_num;
+};
+
+struct mxs_pcm_runtime_data {
+	int period_bytes;
+	int periods;
+	int dma;
+	unsigned long offset;
+	unsigned long size;
+	void *buf;
+	int period_time;
+	struct dma_async_tx_descriptor *desc;
+	struct dma_chan *dma_chan;
+	struct mxs_dma_data dma_data;
+	struct mxs_pcm_dma_params *dma_params;
+};
+
+#endif
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
new file mode 100644
index 00000000000..401944cf456
--- /dev/null
+++ b/sound/soc/mxs/mxs-saif.c
@@ -0,0 +1,797 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/time.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/saif.h>
+#include <mach/dma.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <mach/mxs.h>
+
+#include "mxs-saif.h"
+
+static struct mxs_saif *mxs_saif[2];
+
+/*
+ * SAIF is a little different with other normal SOC DAIs on clock using.
+ *
+ * For MXS, two SAIF modules are instantiated on-chip.
+ * Each SAIF has a set of clock pins and can be operating in master
+ * mode simultaneously if they are connected to different off-chip codecs.
+ * Also, one of the two SAIFs can master or drive the clock pins while the
+ * other SAIF, in slave mode, receives clocking from the master SAIF.
+ * This also means that both SAIFs must operate at the same sample rate.
+ *
+ * We abstract this as each saif has a master, the master could be
+ * himself or other saifs. In the generic saif driver, saif does not need
+ * to know the different clkmux. Saif only needs to know who is his master
+ * and operating his master to generate the proper clock rate for him.
+ * The master id is provided in mach-specific layer according to different
+ * clkmux setting.
+ */
+
+static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+			int clk_id, unsigned int freq, int dir)
+{
+	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+	switch (clk_id) {
+	case MXS_SAIF_MCLK:
+		saif->mclk = freq;
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+/*
+ * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK
+ * is provided by other SAIF, we provide a interface here to get its master
+ * from its master_id.
+ * Note that the master could be himself.
+ */
+static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif)
+{
+	return mxs_saif[saif->master_id];
+}
+
+/*
+ * Set SAIF clock and MCLK
+ */
+static int mxs_saif_set_clk(struct mxs_saif *saif,
+				  unsigned int mclk,
+				  unsigned int rate)
+{
+	u32 scr;
+	int ret;
+	struct mxs_saif *master_saif;
+
+	dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate);
+
+	/* Set master saif to generate proper clock */
+	master_saif = mxs_saif_get_master(saif);
+	if (!master_saif)
+		return -EINVAL;
+
+	dev_dbg(saif->dev, "master saif%d\n", master_saif->id);
+
+	/* Checking if can playback and capture simutaneously */
+	if (master_saif->ongoing && rate != master_saif->cur_rate) {
+		dev_err(saif->dev,
+			"can not change clock, master saif%d(rate %d) is ongoing\n",
+			master_saif->id, master_saif->cur_rate);
+		return -EINVAL;
+	}
+
+	scr = __raw_readl(master_saif->base + SAIF_CTRL);
+	scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE;
+	scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+
+	/*
+	 * Set SAIF clock
+	 *
+	 * The SAIF clock should be either 384*fs or 512*fs.
+	 * If MCLK is used, the SAIF clk ratio need to match mclk ratio.
+	 *  For 32x mclk, set saif clk as 512*fs.
+	 *  For 48x mclk, set saif clk as 384*fs.
+	 *
+	 * If MCLK is not used, we just set saif clk to 512*fs.
+	 */
+	if (master_saif->mclk_in_use) {
+		if (mclk % 32 == 0) {
+			scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			ret = clk_set_rate(master_saif->clk, 512 * rate);
+		} else if (mclk % 48 == 0) {
+			scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE;
+			ret = clk_set_rate(master_saif->clk, 384 * rate);
+		} else {
+			/* SAIF MCLK should be either 32x or 48x */
+			return -EINVAL;
+		}
+	} else {
+		ret = clk_set_rate(master_saif->clk, 512 * rate);
+		scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
+	}
+
+	if (ret)
+		return ret;
+
+	master_saif->cur_rate = rate;
+
+	if (!master_saif->mclk_in_use) {
+		__raw_writel(scr, master_saif->base + SAIF_CTRL);
+		return 0;
+	}
+
+	/*
+	 * Program the over-sample rate for MCLK output
+	 *
+	 * The available MCLK range is 32x, 48x... 512x. The rate
+	 * could be from 8kHz to 192kH.
+	 */
+	switch (mclk / rate) {
+	case 32:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(4);
+		break;
+	case 64:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+		break;
+	case 128:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+		break;
+	case 256:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+		break;
+	case 512:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+		break;
+	case 48:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(3);
+		break;
+	case 96:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(2);
+		break;
+	case 192:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(1);
+		break;
+	case 384:
+		scr |= BF_SAIF_CTRL_BITCLK_MULT_RATE(0);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	__raw_writel(scr, master_saif->base + SAIF_CTRL);
+
+	return 0;
+}
+
+/*
+ * Put and disable MCLK.
+ */
+int mxs_saif_put_mclk(unsigned int saif_id)
+{
+	struct mxs_saif *saif = mxs_saif[saif_id];
+	u32 stat;
+
+	if (!saif)
+		return -EINVAL;
+
+	stat = __raw_readl(saif->base + SAIF_STAT);
+	if (stat & BM_SAIF_STAT_BUSY) {
+		dev_err(saif->dev, "error: busy\n");
+		return -EBUSY;
+	}
+
+	clk_disable(saif->clk);
+
+	/* disable MCLK output */
+	__raw_writel(BM_SAIF_CTRL_CLKGATE,
+		saif->base + SAIF_CTRL + MXS_SET_ADDR);
+	__raw_writel(BM_SAIF_CTRL_RUN,
+		saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+	saif->mclk_in_use = 0;
+	return 0;
+}
+
+/*
+ * Get MCLK and set clock rate, then enable it
+ *
+ * This interface is used for codecs who are using MCLK provided
+ * by saif.
+ */
+int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk,
+					unsigned int rate)
+{
+	struct mxs_saif *saif = mxs_saif[saif_id];
+	u32 stat;
+	int ret;
+	struct mxs_saif *master_saif;
+
+	if (!saif)
+		return -EINVAL;
+
+	/* Clear Reset */
+	__raw_writel(BM_SAIF_CTRL_SFTRST,
+		saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+	/* FIXME: need clear clk gate for register r/w */
+	__raw_writel(BM_SAIF_CTRL_CLKGATE,
+		saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+	master_saif = mxs_saif_get_master(saif);
+	if (saif != master_saif) {
+		dev_err(saif->dev, "can not get mclk from a non-master saif\n");
+		return -EINVAL;
+	}
+
+	stat = __raw_readl(saif->base + SAIF_STAT);
+	if (stat & BM_SAIF_STAT_BUSY) {
+		dev_err(saif->dev, "error: busy\n");
+		return -EBUSY;
+	}
+
+	saif->mclk_in_use = 1;
+	ret = mxs_saif_set_clk(saif, mclk, rate);
+	if (ret)
+		return ret;
+
+	ret = clk_enable(saif->clk);
+	if (ret)
+		return ret;
+
+	/* enable MCLK output */
+	__raw_writel(BM_SAIF_CTRL_RUN,
+		saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+	return 0;
+}
+
+/*
+ * SAIF DAI format configuration.
+ * Should only be called when port is inactive.
+ */
+static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+	u32 scr, stat;
+	u32 scr0;
+	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+	stat = __raw_readl(saif->base + SAIF_STAT);
+	if (stat & BM_SAIF_STAT_BUSY) {
+		dev_err(cpu_dai->dev, "error: busy\n");
+		return -EBUSY;
+	}
+
+	scr0 = __raw_readl(saif->base + SAIF_CTRL);
+	scr0 = scr0 & ~BM_SAIF_CTRL_BITCLK_EDGE & ~BM_SAIF_CTRL_LRCLK_POLARITY \
+		& ~BM_SAIF_CTRL_JUSTIFY & ~BM_SAIF_CTRL_DELAY;
+	scr = 0;
+
+	/* DAI mode */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* data frame low 1clk before data */
+		scr |= BM_SAIF_CTRL_DELAY;
+		scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		/* data frame high with data */
+		scr &= ~BM_SAIF_CTRL_DELAY;
+		scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+		scr &= ~BM_SAIF_CTRL_JUSTIFY;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* DAI clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_IF:
+		scr |= BM_SAIF_CTRL_BITCLK_EDGE;
+		scr |= BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		scr |= BM_SAIF_CTRL_BITCLK_EDGE;
+		scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		scr &= ~BM_SAIF_CTRL_BITCLK_EDGE;
+		scr |= BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		scr &= ~BM_SAIF_CTRL_BITCLK_EDGE;
+		scr &= ~BM_SAIF_CTRL_LRCLK_POLARITY;
+		break;
+	}
+
+	/*
+	 * Note: We simply just support master mode since SAIF TX can only
+	 * work as master.
+	 * Here the master is relative to codec side.
+	 * Saif internally could be slave when working on EXTMASTER mode.
+	 * We just hide this to machine driver.
+	 */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (saif->id == saif->master_id)
+			scr &= ~BM_SAIF_CTRL_SLAVE_MODE;
+		else
+			scr |= BM_SAIF_CTRL_SLAVE_MODE;
+
+		__raw_writel(scr | scr0, saif->base + SAIF_CTRL);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int mxs_saif_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *cpu_dai)
+{
+	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+	snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param);
+
+	/* clear error status to 0 for each re-open */
+	saif->fifo_underrun = 0;
+	saif->fifo_overrun = 0;
+
+	/* Clear Reset for normal operations */
+	__raw_writel(BM_SAIF_CTRL_SFTRST,
+		saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+	/* clear clock gate */
+	__raw_writel(BM_SAIF_CTRL_CLKGATE,
+		saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+
+	return 0;
+}
+
+/*
+ * Should only be called when port is inactive.
+ * although can be called multiple times by upper layers.
+ */
+static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *cpu_dai)
+{
+	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 scr, stat;
+	int ret;
+
+	/* mclk should already be set */
+	if (!saif->mclk && saif->mclk_in_use) {
+		dev_err(cpu_dai->dev, "set mclk first\n");
+		return -EINVAL;
+	}
+
+	stat = __raw_readl(saif->base + SAIF_STAT);
+	if (stat & BM_SAIF_STAT_BUSY) {
+		dev_err(cpu_dai->dev, "error: busy\n");
+		return -EBUSY;
+	}
+
+	/*
+	 * Set saif clk based on sample rate.
+	 * If mclk is used, we also set mclk, if not, saif->mclk is
+	 * default 0, means not used.
+	 */
+	ret = mxs_saif_set_clk(saif, saif->mclk, params_rate(params));
+	if (ret) {
+		dev_err(cpu_dai->dev, "unable to get proper clk\n");
+		return ret;
+	}
+
+	scr = __raw_readl(saif->base + SAIF_CTRL);
+
+	scr &= ~BM_SAIF_CTRL_WORD_LENGTH;
+	scr &= ~BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		scr |= BF_SAIF_CTRL_WORD_LENGTH(0);
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		scr |= BF_SAIF_CTRL_WORD_LENGTH(4);
+		scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		scr |= BF_SAIF_CTRL_WORD_LENGTH(8);
+		scr |= BM_SAIF_CTRL_BITCLK_48XFS_ENABLE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Tx/Rx config */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* enable TX mode */
+		scr &= ~BM_SAIF_CTRL_READ_MODE;
+	} else {
+		/* enable RX mode */
+		scr |= BM_SAIF_CTRL_READ_MODE;
+	}
+
+	__raw_writel(scr, saif->base + SAIF_CTRL);
+	return 0;
+}
+
+static int mxs_saif_prepare(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *cpu_dai)
+{
+	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+
+	/* enable FIFO error irqs */
+	__raw_writel(BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN,
+		saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+	return 0;
+}
+
+static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *cpu_dai)
+{
+	struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai);
+	struct mxs_saif *master_saif;
+	u32 delay;
+
+	master_saif = mxs_saif_get_master(saif);
+	if (!master_saif)
+		return -EINVAL;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		dev_dbg(cpu_dai->dev, "start\n");
+
+		clk_enable(master_saif->clk);
+		if (!master_saif->mclk_in_use)
+			__raw_writel(BM_SAIF_CTRL_RUN,
+				master_saif->base + SAIF_CTRL + MXS_SET_ADDR);
+
+		/*
+		 * If the saif's master is not himself, we also need to enable
+		 * itself clk for its internal basic logic to work.
+		 */
+		if (saif != master_saif) {
+			clk_enable(saif->clk);
+			__raw_writel(BM_SAIF_CTRL_RUN,
+				saif->base + SAIF_CTRL + MXS_SET_ADDR);
+		}
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			/*
+			 * write a data to saif data register to trigger
+			 * the transfer
+			 */
+			__raw_writel(0, saif->base + SAIF_DATA);
+		} else {
+			/*
+			 * read a data from saif data register to trigger
+			 * the receive
+			 */
+			__raw_readl(saif->base + SAIF_DATA);
+		}
+
+		master_saif->ongoing = 1;
+
+		dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n",
+			__raw_readl(saif->base + SAIF_CTRL),
+			__raw_readl(saif->base + SAIF_STAT));
+
+		dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n",
+			__raw_readl(master_saif->base + SAIF_CTRL),
+			__raw_readl(master_saif->base + SAIF_STAT));
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		dev_dbg(cpu_dai->dev, "stop\n");
+
+		/* wait a while for the current sample to complete */
+		delay = USEC_PER_SEC / master_saif->cur_rate;
+
+		if (!master_saif->mclk_in_use) {
+			__raw_writel(BM_SAIF_CTRL_RUN,
+				master_saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+			udelay(delay);
+		}
+		clk_disable(master_saif->clk);
+
+		if (saif != master_saif) {
+			__raw_writel(BM_SAIF_CTRL_RUN,
+				saif->base + SAIF_CTRL + MXS_CLR_ADDR);
+			udelay(delay);
+			clk_disable(saif->clk);
+		}
+
+		master_saif->ongoing = 0;
+
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+#define MXS_SAIF_RATES		SNDRV_PCM_RATE_8000_192000
+#define MXS_SAIF_FORMATS \
+	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops mxs_saif_dai_ops = {
+	.startup = mxs_saif_startup,
+	.trigger = mxs_saif_trigger,
+	.prepare = mxs_saif_prepare,
+	.hw_params = mxs_saif_hw_params,
+	.set_sysclk = mxs_saif_set_dai_sysclk,
+	.set_fmt = mxs_saif_set_dai_fmt,
+};
+
+static int mxs_saif_dai_probe(struct snd_soc_dai *dai)
+{
+	struct mxs_saif *saif = dev_get_drvdata(dai->dev);
+
+	snd_soc_dai_set_drvdata(dai, saif);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver mxs_saif_dai = {
+	.name = "mxs-saif",
+	.probe = mxs_saif_dai_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = MXS_SAIF_RATES,
+		.formats = MXS_SAIF_FORMATS,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = MXS_SAIF_RATES,
+		.formats = MXS_SAIF_FORMATS,
+	},
+	.ops = &mxs_saif_dai_ops,
+};
+
+static irqreturn_t mxs_saif_irq(int irq, void *dev_id)
+{
+	struct mxs_saif *saif = dev_id;
+	unsigned int stat;
+
+	stat = __raw_readl(saif->base + SAIF_STAT);
+	if (!(stat & (BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ |
+			BM_SAIF_STAT_FIFO_OVERFLOW_IRQ)))
+		return IRQ_NONE;
+
+	if (stat & BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ) {
+		dev_dbg(saif->dev, "underrun!!! %d\n", ++saif->fifo_underrun);
+		__raw_writel(BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ,
+				saif->base + SAIF_STAT + MXS_CLR_ADDR);
+	}
+
+	if (stat & BM_SAIF_STAT_FIFO_OVERFLOW_IRQ) {
+		dev_dbg(saif->dev, "overrun!!! %d\n", ++saif->fifo_overrun);
+		__raw_writel(BM_SAIF_STAT_FIFO_OVERFLOW_IRQ,
+				saif->base + SAIF_STAT + MXS_CLR_ADDR);
+	}
+
+	dev_dbg(saif->dev, "SAIF_CTRL %x SAIF_STAT %x\n",
+	       __raw_readl(saif->base + SAIF_CTRL),
+	       __raw_readl(saif->base + SAIF_STAT));
+
+	return IRQ_HANDLED;
+}
+
+static int mxs_saif_probe(struct platform_device *pdev)
+{
+	struct resource *res;
+	struct mxs_saif *saif;
+	struct mxs_saif_platform_data *pdata;
+	int ret = 0;
+
+	if (pdev->id >= ARRAY_SIZE(mxs_saif))
+		return -EINVAL;
+
+	pdata = pdev->dev.platform_data;
+	if (pdata && pdata->init) {
+		ret = pdata->init();
+		if (ret)
+			return ret;
+	}
+
+	saif = kzalloc(sizeof(*saif), GFP_KERNEL);
+	if (!saif)
+		return -ENOMEM;
+
+	mxs_saif[pdev->id] = saif;
+	saif->id = pdev->id;
+
+	saif->master_id = saif->id;
+	if (pdata && pdata->get_master_id) {
+		saif->master_id = pdata->get_master_id(saif->id);
+		if (saif->master_id < 0 ||
+			saif->master_id >= ARRAY_SIZE(mxs_saif))
+			return -EINVAL;
+	}
+
+	saif->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(saif->clk)) {
+		ret = PTR_ERR(saif->clk);
+		dev_err(&pdev->dev, "Cannot get the clock: %d\n",
+			ret);
+		goto failed_clk;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res) {
+		ret = -ENODEV;
+		dev_err(&pdev->dev, "failed to get io resource: %d\n",
+			ret);
+		goto failed_get_resource;
+	}
+
+	if (!request_mem_region(res->start, resource_size(res), "mxs-saif")) {
+		dev_err(&pdev->dev, "request_mem_region failed\n");
+		ret = -EBUSY;
+		goto failed_get_resource;
+	}
+
+	saif->base = ioremap(res->start, resource_size(res));
+	if (!saif->base) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENODEV;
+		goto failed_ioremap;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!res) {
+		ret = -ENODEV;
+		dev_err(&pdev->dev, "failed to get dma resource: %d\n",
+			ret);
+		goto failed_ioremap;
+	}
+	saif->dma_param.chan_num = res->start;
+
+	saif->irq = platform_get_irq(pdev, 0);
+	if (saif->irq < 0) {
+		ret = saif->irq;
+		dev_err(&pdev->dev, "failed to get irq resource: %d\n",
+			ret);
+		goto failed_get_irq1;
+	}
+
+	saif->dev = &pdev->dev;
+	ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to request irq\n");
+		goto failed_get_irq1;
+	}
+
+	saif->dma_param.chan_irq = platform_get_irq(pdev, 1);
+	if (saif->dma_param.chan_irq < 0) {
+		ret = saif->dma_param.chan_irq;
+		dev_err(&pdev->dev, "failed to get dma irq resource: %d\n",
+			ret);
+		goto failed_get_irq2;
+	}
+
+	platform_set_drvdata(pdev, saif);
+
+	ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai);
+	if (ret) {
+		dev_err(&pdev->dev, "register DAI failed\n");
+		goto failed_register;
+	}
+
+	saif->soc_platform_pdev = platform_device_alloc(
+					"mxs-pcm-audio", pdev->id);
+	if (!saif->soc_platform_pdev) {
+		ret = -ENOMEM;
+		goto failed_pdev_alloc;
+	}
+
+	platform_set_drvdata(saif->soc_platform_pdev, saif);
+	ret = platform_device_add(saif->soc_platform_pdev);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to add soc platform device\n");
+		goto failed_pdev_add;
+	}
+
+	return 0;
+
+failed_pdev_add:
+	platform_device_put(saif->soc_platform_pdev);
+failed_pdev_alloc:
+	snd_soc_unregister_dai(&pdev->dev);
+failed_register:
+failed_get_irq2:
+	free_irq(saif->irq, saif);
+failed_get_irq1:
+	iounmap(saif->base);
+failed_ioremap:
+	release_mem_region(res->start, resource_size(res));
+failed_get_resource:
+	clk_put(saif->clk);
+failed_clk:
+	kfree(saif);
+
+	return ret;
+}
+
+static int __devexit mxs_saif_remove(struct platform_device *pdev)
+{
+	struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	struct mxs_saif *saif = platform_get_drvdata(pdev);
+
+	platform_device_unregister(saif->soc_platform_pdev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+
+	iounmap(saif->base);
+	release_mem_region(res->start, resource_size(res));
+	free_irq(saif->irq, saif);
+
+	clk_put(saif->clk);
+	kfree(saif);
+
+	return 0;
+}
+
+static struct platform_driver mxs_saif_driver = {
+	.probe = mxs_saif_probe,
+	.remove = __devexit_p(mxs_saif_remove),
+
+	.driver = {
+		.name = "mxs-saif",
+		.owner = THIS_MODULE,
+	},
+};
+
+static int __init mxs_saif_init(void)
+{
+	return platform_driver_register(&mxs_saif_driver);
+}
+
+static void __exit mxs_saif_exit(void)
+{
+	platform_driver_unregister(&mxs_saif_driver);
+}
+
+module_init(mxs_saif_init);
+module_exit(mxs_saif_exit);
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS ASoC SAIF driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h
new file mode 100644
index 00000000000..12c91e4eb94
--- /dev/null
+++ b/sound/soc/mxs/mxs-saif.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (C) 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+
+#ifndef _MXS_SAIF_H
+#define _MXS_SAIF_H
+
+#define SAIF_CTRL	0x0
+#define SAIF_STAT	0x10
+#define SAIF_DATA	0x20
+#define SAIF_VERSION	0X30
+
+/* SAIF_CTRL */
+#define BM_SAIF_CTRL_SFTRST		0x80000000
+#define BM_SAIF_CTRL_CLKGATE		0x40000000
+#define BP_SAIF_CTRL_BITCLK_MULT_RATE	27
+#define BM_SAIF_CTRL_BITCLK_MULT_RATE	0x38000000
+#define BF_SAIF_CTRL_BITCLK_MULT_RATE(v) \
+		(((v) << 27) & BM_SAIF_CTRL_BITCLK_MULT_RATE)
+#define BM_SAIF_CTRL_BITCLK_BASE_RATE	0x04000000
+#define BM_SAIF_CTRL_FIFO_ERROR_IRQ_EN	0x02000000
+#define BM_SAIF_CTRL_FIFO_SERVICE_IRQ_EN	0x01000000
+#define BP_SAIF_CTRL_RSRVD2		21
+#define BM_SAIF_CTRL_RSRVD2		0x00E00000
+
+#define BP_SAIF_CTRL_DMAWAIT_COUNT	16
+#define BM_SAIF_CTRL_DMAWAIT_COUNT	0x001F0000
+#define BF_SAIF_CTRL_DMAWAIT_COUNT(v) \
+		(((v) << 16) & BM_SAIF_CTRL_DMAWAIT_COUNT)
+#define BP_SAIF_CTRL_CHANNEL_NUM_SELECT 14
+#define BM_SAIF_CTRL_CHANNEL_NUM_SELECT 0x0000C000
+#define BF_SAIF_CTRL_CHANNEL_NUM_SELECT(v) \
+		(((v) << 14) & BM_SAIF_CTRL_CHANNEL_NUM_SELECT)
+#define BM_SAIF_CTRL_LRCLK_PULSE	0x00002000
+#define BM_SAIF_CTRL_BIT_ORDER		0x00001000
+#define BM_SAIF_CTRL_DELAY		0x00000800
+#define BM_SAIF_CTRL_JUSTIFY		0x00000400
+#define BM_SAIF_CTRL_LRCLK_POLARITY	0x00000200
+#define BM_SAIF_CTRL_BITCLK_EDGE	0x00000100
+#define BP_SAIF_CTRL_WORD_LENGTH	4
+#define BM_SAIF_CTRL_WORD_LENGTH	0x000000F0
+#define BF_SAIF_CTRL_WORD_LENGTH(v) \
+		(((v) << 4) & BM_SAIF_CTRL_WORD_LENGTH)
+#define BM_SAIF_CTRL_BITCLK_48XFS_ENABLE	0x00000008
+#define BM_SAIF_CTRL_SLAVE_MODE		0x00000004
+#define BM_SAIF_CTRL_READ_MODE		0x00000002
+#define BM_SAIF_CTRL_RUN		0x00000001
+
+/* SAIF_STAT */
+#define BM_SAIF_STAT_PRESENT		0x80000000
+#define BP_SAIF_STAT_RSRVD2		17
+#define BM_SAIF_STAT_RSRVD2		0x7FFE0000
+#define BF_SAIF_STAT_RSRVD2(v) \
+		(((v) << 17) & BM_SAIF_STAT_RSRVD2)
+#define BM_SAIF_STAT_DMA_PREQ		0x00010000
+#define BP_SAIF_STAT_RSRVD1		7
+#define BM_SAIF_STAT_RSRVD1		0x0000FF80
+#define BF_SAIF_STAT_RSRVD1(v) \
+		(((v) << 7) & BM_SAIF_STAT_RSRVD1)
+
+#define BM_SAIF_STAT_FIFO_UNDERFLOW_IRQ 0x00000040
+#define BM_SAIF_STAT_FIFO_OVERFLOW_IRQ	0x00000020
+#define BM_SAIF_STAT_FIFO_SERVICE_IRQ	0x00000010
+#define BP_SAIF_STAT_RSRVD0		1
+#define BM_SAIF_STAT_RSRVD0		0x0000000E
+#define BF_SAIF_STAT_RSRVD0(v) \
+		(((v) << 1) & BM_SAIF_STAT_RSRVD0)
+#define BM_SAIF_STAT_BUSY		0x00000001
+
+/* SAFI_DATA */
+#define BP_SAIF_DATA_PCM_RIGHT		16
+#define BM_SAIF_DATA_PCM_RIGHT		0xFFFF0000
+#define BF_SAIF_DATA_PCM_RIGHT(v) \
+		(((v) << 16) & BM_SAIF_DATA_PCM_RIGHT)
+#define BP_SAIF_DATA_PCM_LEFT		0
+#define BM_SAIF_DATA_PCM_LEFT		0x0000FFFF
+#define BF_SAIF_DATA_PCM_LEFT(v)	\
+		(((v) << 0) & BM_SAIF_DATA_PCM_LEFT)
+
+/* SAIF_VERSION */
+#define BP_SAIF_VERSION_MAJOR		24
+#define BM_SAIF_VERSION_MAJOR		0xFF000000
+#define BF_SAIF_VERSION_MAJOR(v) \
+		(((v) << 24) & BM_SAIF_VERSION_MAJOR)
+#define BP_SAIF_VERSION_MINOR		16
+#define BM_SAIF_VERSION_MINOR		0x00FF0000
+#define BF_SAIF_VERSION_MINOR(v) \
+		(((v) << 16) & BM_SAIF_VERSION_MINOR)
+#define BP_SAIF_VERSION_STEP		0
+#define BM_SAIF_VERSION_STEP		0x0000FFFF
+#define BF_SAIF_VERSION_STEP(v) \
+		(((v) << 0) & BM_SAIF_VERSION_STEP)
+
+#define MXS_SAIF_MCLK		0
+
+#include "mxs-pcm.h"
+
+struct mxs_saif {
+	struct device *dev;
+	struct clk *clk;
+	unsigned int mclk;
+	unsigned int mclk_in_use;
+	void __iomem *base;
+	int irq;
+	struct mxs_pcm_dma_params dma_param;
+	unsigned int id;
+	unsigned int master_id;
+	unsigned int cur_rate;
+	unsigned int ongoing;
+
+	struct platform_device *soc_platform_pdev;
+	u32 fifo_underrun;
+	u32 fifo_overrun;
+};
+
+extern int mxs_saif_put_mclk(unsigned int saif_id);
+extern int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk,
+					unsigned int rate);
+#endif
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
new file mode 100644
index 00000000000..7fbeaec06eb
--- /dev/null
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -0,0 +1,173 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * 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 Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+
+#include "../codecs/sgtl5000.h"
+#include "mxs-saif.h"
+
+static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	unsigned int rate = params_rate(params);
+	u32 dai_format, mclk;
+	int ret;
+
+	/* sgtl5000 does not support 512*rate when in 96000 fs */
+	switch (rate) {
+	case 96000:
+		mclk = 256 * rate;
+		break;
+	default:
+		mclk = 512 * rate;
+		break;
+	}
+
+	/* Sgtl5000 sysclk should be >= 8MHz and <= 27M */
+	if (mclk < 8000000 || mclk > 27000000)
+		return -EINVAL;
+
+	/* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */
+	ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0);
+	if (ret)
+		return ret;
+
+	/* The SAIF MCLK should be the same as SGTL5000_SYSCLK */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, MXS_SAIF_MCLK, mclk, 0);
+	if (ret)
+		return ret;
+
+	/* set codec to slave mode */
+	dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS;
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
+	if (ret)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, dai_format);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops mxs_sgtl5000_hifi_ops = {
+	.hw_params = mxs_sgtl5000_hw_params,
+};
+
+static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
+	{
+		.name		= "HiFi Tx",
+		.stream_name	= "HiFi Playback",
+		.codec_dai_name	= "sgtl5000",
+		.codec_name	= "sgtl5000.0-000a",
+		.cpu_dai_name	= "mxs-saif.0",
+		.platform_name	= "mxs-pcm-audio.0",
+		.ops		= &mxs_sgtl5000_hifi_ops,
+	}, {
+		.name		= "HiFi Rx",
+		.stream_name	= "HiFi Capture",
+		.codec_dai_name	= "sgtl5000",
+		.codec_name	= "sgtl5000.0-000a",
+		.cpu_dai_name	= "mxs-saif.1",
+		.platform_name	= "mxs-pcm-audio.1",
+		.ops		= &mxs_sgtl5000_hifi_ops,
+	},
+};
+
+static struct snd_soc_card mxs_sgtl5000 = {
+	.name		= "mxs_sgtl5000",
+	.dai_link	= mxs_sgtl5000_dai,
+	.num_links	= ARRAY_SIZE(mxs_sgtl5000_dai),
+};
+
+static int __devinit mxs_sgtl5000_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &mxs_sgtl5000;
+	int ret;
+
+	/*
+	 * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w).
+	 * The Sgtl5000 sysclk is derived from saif0 mclk and it's range
+	 * should be >= 8MHz and <= 27M.
+	 */
+	ret = mxs_saif_get_mclk(0, 44100 * 256, 44100);
+	if (ret)
+		return ret;
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+			ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devexit mxs_sgtl5000_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	mxs_saif_put_mclk(0);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver mxs_sgtl5000_audio_driver = {
+	.driver = {
+		.name = "mxs-sgtl5000",
+		.owner = THIS_MODULE,
+	},
+	.probe = mxs_sgtl5000_probe,
+	.remove = __devexit_p(mxs_sgtl5000_remove),
+};
+
+static int __init mxs_sgtl5000_init(void)
+{
+	return platform_driver_register(&mxs_sgtl5000_audio_driver);
+}
+module_init(mxs_sgtl5000_init);
+
+static void __exit mxs_sgtl5000_exit(void)
+{
+	platform_driver_unregister(&mxs_sgtl5000_audio_driver);
+}
+module_exit(mxs_sgtl5000_exit);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("MXS ALSA SoC Machine driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index d589ef14e91..865b288bd74 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -268,7 +268,7 @@ static int nuc900_dma_open(struct snd_pcm_substream *substream)
 	nuc900_audio = nuc900_ac97_data;
 
 	if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt,
-			IRQF_DISABLED, "nuc900-dma", substream))
+			0, "nuc900-dma", substream))
 		return -EBUSY;
 
 	runtime->private_data = nuc900_audio;
@@ -318,7 +318,6 @@ static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32);
 static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
-	struct snd_soc_dai *dai = rtd->cpu_dai;
 	struct snd_pcm *pcm = rtd->pcm;
 
 	if (!card->dev->dma_mask)
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index 1a591f1ebfb..b899a3bc8f4 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -306,8 +306,10 @@ static int __init raumfeld_audio_init(void)
 				     &snd_soc_raumfeld_connector);
 
 	ret = platform_device_add(raumfeld_audio_device);
-	if (ret < 0)
+	if (ret < 0) {
+		platform_device_put(raumfeld_audio_device);
 		return ret;
+	}
 
 	raumfeld_enable_audio(true);
 	return 0;
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index b253d864868..ce920e3cfea 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -312,7 +312,7 @@ static struct snd_soc_dai_link spitz_dai = {
 	.cpu_dai_name = "pxa2xx-i2s",
 	.codec_dai_name = "wm8750-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name = "wm8750-codec.0-001b",
+	.codec_name = "wm8750.0-001b",
 	.init = spitz_wm8750_init,
 	.ops = &spitz_ops,
 };
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index d69d9fc3223..4b81ffd8756 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -198,7 +198,7 @@ static struct snd_soc_dai_link z2_dai = {
 	.cpu_dai_name	= "pxa2xx-i2s",
 	.codec_dai_name	= "wm8750-hifi",
 	.platform_name = "pxa-pcm-audio",
-	.codec_name	= "wm8750-codec.0-001b",
+	.codec_name	= "wm8750.0-001b",
 	.init		= z2_wm8750_init,
 	.ops		= &z2_ops,
 };
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 80c85fd64e1..55efc2bdf0b 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -446,7 +446,6 @@ static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32);
 static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime)
 {
 	struct snd_card *card = runtime->card->snd_card;
-	struct snd_soc_dai *dai = runtime->cpu_dai;
 	struct snd_pcm *pcm = runtime->pcm;
 	struct s6000_pcm_dma_params *params;
 	int res;
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 65f980ef287..dd3b3eac080 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580
 
 config SND_SOC_SAMSUNG_SMDK_WM8994
 	tristate "SoC I2S Audio support for WM8994 on SMDK"
-	depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210)
+	depends on SND_SOC_SAMSUNG && (MACH_SMDKV310 || MACH_SMDKC210 || MACH_SMDK4212)
 	select SND_SOC_WM8994
 	select SND_SAMSUNG_I2S
 	help
@@ -158,7 +158,7 @@ config SND_SOC_GONI_AQUILA_WM8994
 
 config SND_SOC_SAMSUNG_SMDK_SPDIF
 	tristate "SoC S/PDIF Audio support for SMDK"
-	depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310)
+	depends on SND_SOC_SAMSUNG && (MACH_SMDKC100 || MACH_SMDKC110 || MACH_SMDKV210 || MACH_SMDKV310 || MACH_SMDK4212)
 	select SND_SAMSUNG_SPDIF
 	help
 	  Say Y if you want to add support for SoC S/PDIF audio on the SMDK.
@@ -173,7 +173,7 @@ config SND_SOC_SMDK_WM8580_PCM
 
 config SND_SOC_SMDK_WM8994_PCM
 	tristate "SoC PCM Audio support for WM8994 on SMDK"
-	depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310)
+	depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310 || MACH_SMDK4212)
 	select SND_SOC_WM8994
 	select SND_SAMSUNG_PCM
 	help
diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c
index f97110e72e8..884c8a107bf 100644
--- a/sound/soc/samsung/ac97.c
+++ b/sound/soc/samsung/ac97.c
@@ -444,7 +444,7 @@ static __devinit int s3c_ac97_probe(struct platform_device *pdev)
 	}
 
 	ret = request_irq(irq_res->start, s3c_ac97_irq,
-					IRQF_DISABLED, "AC97", NULL);
+					0, "AC97", NULL);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "ac97: interrupt request failed.\n");
 		goto err4;
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 14eb6ea69e7..ed8f13a29c8 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -131,7 +131,7 @@ static struct snd_soc_dai_link jive_dai = {
 	.cpu_dai_name	= "s3c2412-i2s",
 	.codec_dai_name = "wm8750-hifi",
 	.platform_name	= "samsung-audio",
-	.codec_name	= "wm8750-codec.0-001a",
+	.codec_name	= "wm8750.0-001a",
 	.init		= jive_wm8750_init,
 	.ops		= &jive_ops,
 };
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 841ab14c110..7ab8e2c2921 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -69,10 +69,10 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai)
 	s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out;
 
 	s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk");
-	if (s3c2412_i2s.iis_cclk == NULL) {
+	if (IS_ERR(s3c2412_i2s.iis_cclk)) {
 		pr_err("failed to get i2sclk clock\n");
 		iounmap(s3c2412_i2s.regs);
-		return -ENODEV;
+		return PTR_ERR(s3c2412_i2s.iis_cclk);
 	}
 
 	/* Set MPLL as the source for IIS CLK */
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index 63d8849d80b..21c92e2e300 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -383,10 +383,10 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai)
 		return -ENXIO;
 
 	s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis");
-	if (s3c24xx_i2s.iis_clk == NULL) {
+	if (IS_ERR(s3c24xx_i2s.iis_clk)) {
 		pr_err("failed to get iis_clock\n");
 		iounmap(s3c24xx_i2s.regs);
-		return -ENODEV;
+		return PTR_ERR(s3c24xx_i2s.iis_clk);
 	}
 	clk_enable(s3c24xx_i2s.iis_clk);
 
diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c
index dc9d551f678..65c1cfd47d8 100644
--- a/sound/soc/samsung/s3c24xx_uda134x.c
+++ b/sound/soc/samsung/s3c24xx_uda134x.c
@@ -66,17 +66,17 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream)
 	pr_debug("%s %d\n", __func__, clk_users);
 	if (clk_users == 0) {
 		xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal");
-		if (!xtal) {
+		if (IS_ERR(xtal)) {
 			printk(KERN_ERR "%s cannot get xtal\n", __func__);
-			ret = -EBUSY;
+			ret = PTR_ERR(xtal);
 		} else {
 			pclk = clk_get(&s3c24xx_uda134x_snd_device->dev,
 				       "pclk");
-			if (!pclk) {
+			if (IS_ERR(pclk)) {
 				printk(KERN_ERR "%s cannot get pclk\n",
 				       __func__);
 				clk_put(xtal);
-				ret = -EBUSY;
+				ret = PTR_ERR(pclk);
 			}
 		}
 		if (!ret) {
diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c
index 0a2c4f22303..bbd14768ecd 100644
--- a/sound/soc/samsung/smartq_wm8987.c
+++ b/sound/soc/samsung/smartq_wm8987.c
@@ -207,7 +207,7 @@ static struct snd_soc_dai_link smartq_dai[] = {
 		.cpu_dai_name	= "samsung-i2s.0",
 		.codec_dai_name	= "wm8750-hifi",
 		.platform_name	= "samsung-audio",
-		.codec_name	= "wm8750-codec.0-0x1a",
+		.codec_name	= "wm8750.0-0x1a",
 		.init		= smartq_wm8987_init,
 		.ops		= &smartq_hifi_ops,
 	},
diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c
index 3d26f6607aa..20deecf3b24 100644
--- a/sound/soc/samsung/smdk_wm8580.c
+++ b/sound/soc/samsung/smdk_wm8580.c
@@ -210,7 +210,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
 		.cpu_dai_name = "samsung-i2s.0",
 		.codec_dai_name = "wm8580-hifi-playback",
 		.platform_name = "samsung-audio",
-		.codec_name = "wm8580-codec.0-001b",
+		.codec_name = "wm8580.0-001b",
 		.init = smdk_wm8580_init_paifrx,
 		.ops = &smdk_ops,
 	},
@@ -220,7 +220,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
 		.cpu_dai_name = "samsung-i2s.0",
 		.codec_dai_name = "wm8580-hifi-capture",
 		.platform_name = "samsung-audio",
-		.codec_name = "wm8580-codec.0-001b",
+		.codec_name = "wm8580.0-001b",
 		.init = smdk_wm8580_init_paiftx,
 		.ops = &smdk_ops,
 	},
@@ -230,7 +230,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
 		.cpu_dai_name = "samsung-i2s.x",
 		.codec_dai_name = "wm8580-hifi-playback",
 		.platform_name = "samsung-audio",
-		.codec_name = "wm8580-codec.0-001b",
+		.codec_name = "wm8580.0-001b",
 		.init = smdk_wm8580_init_paifrx,
 		.ops = &smdk_ops,
 	},
diff --git a/sound/soc/samsung/smdk_wm8580pcm.c b/sound/soc/samsung/smdk_wm8580pcm.c
index 0d12092df16..4b9c73477ce 100644
--- a/sound/soc/samsung/smdk_wm8580pcm.c
+++ b/sound/soc/samsung/smdk_wm8580pcm.c
@@ -127,7 +127,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
 		.cpu_dai_name = "samsung-pcm.0",
 		.codec_dai_name = "wm8580-hifi-playback",
 		.platform_name = "samsung-audio",
-		.codec_name = "wm8580-codec.0-001b",
+		.codec_name = "wm8580.0-001b",
 		.ops = &smdk_wm8580_pcm_ops,
 	}, {
 		.name = "WM8580 PAIF PCM TX",
@@ -135,7 +135,7 @@ static struct snd_soc_dai_link smdk_dai[] = {
 		.cpu_dai_name = "samsung-pcm.0",
 		.codec_dai_name = "wm8580-hifi-capture",
 		.platform_name = "samsung-audio",
-		.codec_name = "wm8580-codec.0-001b",
+		.codec_name = "wm8580.0-001b",
 		.ops = &smdk_wm8580_pcm_ops,
 	},
 };
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 590e9274b06..b9e213f6cc0 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -125,10 +125,6 @@ static struct snd_soc_jack_pin speyside_headset_pins[] = {
 		.pin = "Headset Mic",
 		.mask = SND_JACK_MICROPHONE,
 	},
-	{
-		.pin = "Headphone",
-		.mask = SND_JACK_HEADPHONE,
-	},
 };
 
 /* Default the headphone selection to active high */
@@ -171,7 +167,8 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd)
 	gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity);
 
 	ret = snd_soc_jack_new(codec, "Headset",
-			       SND_JACK_HEADSET | SND_JACK_BTN_0,
+			       SND_JACK_LINEOUT | SND_JACK_HEADSET |
+			       SND_JACK_BTN_0,
 			       &speyside_headset);
 	if (ret)
 		return ret;
@@ -227,7 +224,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm)
 	snd_soc_dapm_nc_pin(dapm, "LINEOUT");
 
 	/* At any time the WM9081 is active it will have this clock */
-	return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK,
+	return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0,
 					48000 * 256, 0);
 }
 
@@ -252,6 +249,7 @@ static const struct snd_kcontrol_new controls[] = {
 	SOC_DAPM_PIN_SWITCH("Main AMIC"),
 	SOC_DAPM_PIN_SWITCH("WM1250 Input"),
 	SOC_DAPM_PIN_SWITCH("WM1250 Output"),
+	SOC_DAPM_PIN_SWITCH("Headphone"),
 };
 
 static struct snd_soc_dapm_widget widgets[] = {
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index 72535f2daaf..3820a6b057d 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -31,13 +31,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
 		if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
 			ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
 						  WM8962_FLL_MCLK, 32768,
-						  44100 * 256);
+						  44100 * 512);
 			if (ret < 0)
 				pr_err("Failed to start FLL: %d\n", ret);
 
 			ret = snd_soc_dai_set_sysclk(codec_dai,
 						     WM8962_SYSCLK_FLL,
-						     44100 * 256,
+						     44100 * 512,
 						     SND_SOC_CLOCK_IN);
 			if (ret < 0) {
 				pr_err("Failed to set SYSCLK: %d\n", ret);
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 8e112ccffb1..1493ebf4d94 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1285,7 +1285,7 @@ static int fsi_probe(struct platform_device *pdev)
 	pm_runtime_enable(&pdev->dev);
 	dev_set_drvdata(&pdev->dev, master);
 
-	ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED,
+	ret = request_irq(irq, &fsi_interrupt, 0,
 			  id_entry->name, master);
 	if (ret) {
 		dev_err(&pdev->dev, "irq request err\n");
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index 20b7f3b003a..143c705ac27 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -548,9 +548,6 @@ static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec,
 
 static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec)
 {
-	const struct snd_soc_codec_driver *codec_drv;
-
-	codec_drv = codec->driver;
 	return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count());
 }
 
@@ -868,10 +865,6 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec)
 
 static int snd_soc_flat_cache_init(struct snd_soc_codec *codec)
 {
-	const struct snd_soc_codec_driver *codec_drv;
-
-	codec_drv = codec->driver;
-
 	if (codec->reg_def_copy)
 		codec->reg_cache = kmemdup(codec->reg_def_copy,
 					   codec->reg_size, GFP_KERNEL);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ef69f5a0270..bd20154e8b3 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -106,7 +106,7 @@ static int format_register_str(struct snd_soc_codec *codec,
 	if (wordsize + regsize + 2 + 1 != len)
 		return -EINVAL;
 
-	ret = snd_soc_read(codec , reg);
+	ret = snd_soc_read(codec, reg);
 	if (ret < 0) {
 		memset(regbuf, 'X', regsize);
 		regbuf[regsize] = '\0';
@@ -144,7 +144,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf,
 		step = codec->driver->reg_cache_step;
 
 	for (i = 0; i < codec->driver->reg_cache_size; i += step) {
-		if (codec->readable_register && !codec->readable_register(codec, i))
+		if (!snd_soc_codec_readable_register(codec, i))
 			continue;
 		if (codec->driver->display_register) {
 			count += codec->driver->display_register(codec, buf + count,
@@ -245,7 +245,6 @@ static ssize_t codec_reg_write_file(struct file *file,
 	size_t 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));
@@ -253,9 +252,6 @@ static ssize_t codec_reg_write_file(struct file *file,
 		return -EFAULT;
 	buf[buf_size] = 0;
 
-	if (codec->driver->reg_cache_step)
-		step = codec->driver->reg_cache_step;
-
 	while (*start == ' ')
 		start++;
 	reg = simple_strtoul(start, &start, 16);
@@ -957,6 +953,8 @@ static int soc_probe_codec(struct snd_soc_card *card,
 		snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets,
 					  driver->num_dapm_widgets);
 
+	codec->dapm.idle_bias_off = driver->idle_bias_off;
+
 	if (driver->probe) {
 		ret = driver->probe(codec);
 		if (ret < 0) {
@@ -2680,7 +2678,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	if (dai->driver && dai->driver->ops->set_sysclk)
 		return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);
 	else if (dai->codec && dai->codec->driver->set_sysclk)
-		return dai->codec->driver->set_sysclk(dai->codec, clk_id,
+		return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0,
 						      freq, dir);
 	else
 		return -EINVAL;
@@ -2691,16 +2689,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
  * snd_soc_codec_set_sysclk - configure CODEC system or master clock.
  * @codec: CODEC
  * @clk_id: DAI specific clock ID
+ * @source: Source for the clock
  * @freq: new clock frequency in Hz
  * @dir: new clock direction - input/output.
  *
  * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
  */
 int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
-	unsigned int freq, int dir)
+			     int source, unsigned int freq, int dir)
 {
 	if (codec->driver->set_sysclk)
-		return codec->driver->set_sysclk(codec, clk_id, freq, dir);
+		return codec->driver->set_sysclk(codec, clk_id, source,
+						 freq, dir);
 	else
 		return -EINVAL;
 }
@@ -3153,6 +3153,7 @@ int snd_soc_register_platform(struct device *dev,
 	platform->driver = platform_drv;
 	platform->dapm.dev = dev;
 	platform->dapm.platform = platform;
+	platform->dapm.stream_event = platform_drv->stream_event;
 
 	mutex_lock(&client_mutex);
 	list_add(&platform->list, &platform_list);
@@ -3265,6 +3266,7 @@ int snd_soc_register_codec(struct device *dev,
 	codec->dapm.dev = dev;
 	codec->dapm.codec = codec;
 	codec->dapm.seq_notifier = codec_drv->seq_notifier;
+	codec->dapm.stream_event = codec_drv->stream_event;
 	codec->dev = dev;
 	codec->driver = codec_drv;
 	codec->num_dai = num_dai;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d67c637557a..4a440b52dd7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -443,6 +443,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w)
 			if (path->name != (char *)w->kcontrol_news[i].name)
 				continue;
 
+			if (w->kcontrols[i]) {
+				path->kcontrol = w->kcontrols[i];
+				continue;
+			}
+
 			wlistsize = sizeof(struct snd_soc_dapm_widget_list) +
 				    sizeof(struct snd_soc_dapm_widget *),
 			wlist = kzalloc(wlistsize, GFP_KERNEL);
@@ -579,8 +584,8 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
 					name + prefix_len, prefix);
 		ret = snd_ctl_add(card, kcontrol);
 		if (ret < 0) {
-			dev_err(dapm->dev,
-				"asoc: failed to add kcontrol %s\n", w->name);
+			dev_err(dapm->dev, "failed to add kcontrol %s: %d\n",
+				w->name, ret);
 			kfree(wlist);
 			return ret;
 		}
@@ -1556,7 +1561,6 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
 		/* found, now check type */
 		found = 1;
 		path->connect = connect;
-		break;
 	}
 
 	if (found)
@@ -2584,7 +2588,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
 	{
 		if (!w->sname || w->dapm != dapm)
 			continue;
-		dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
+		dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n",
 			w->name, w->sname, stream, event);
 		if (strstr(w->sname, stream)) {
 			switch(event) {
@@ -2604,6 +2608,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm,
 	}
 
 	dapm_power_widgets(dapm, event);
+
+	/* do we need to notify any clients that DAPM stream is complete */
+	if (dapm->stream_event)
+		dapm->stream_event(dapm, event);
 }
 
 /**
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index a62f7dd4ba9..66fcccd79ef 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -13,26 +13,14 @@
 
 #include <linux/i2c.h>
 #include <linux/spi/spi.h>
+#include <linux/regmap.h>
 #include <sound/soc.h>
 
 #include <trace/events/asoc.h>
 
-#ifdef CONFIG_SPI_MASTER
-static int do_spi_write(void *control, const char *data, int len)
-{
-	struct spi_device *spi = control;
-	int ret;
-
-	ret = spi_write(spi, data, len);
-	if (ret < 0)
-		return ret;
-
-	return len;
-}
-#endif
-
-static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
-		       unsigned int value, const void *data, int len)
+#ifdef CONFIG_REGMAP
+static int hw_write(struct snd_soc_codec *codec, unsigned int reg,
+		    unsigned int value)
 {
 	int ret;
 
@@ -49,13 +37,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg,
 		return 0;
 	}
 
-	ret = codec->hw_write(codec->control_data, data, len);
-	if (ret == len)
-		return 0;
-	if (ret < 0)
-		return ret;
-	else
-		return -EIO;
+	return regmap_write(codec->control_data, reg, value);
 }
 
 static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
@@ -69,8 +51,11 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
 		if (codec->cache_only)
 			return -1;
 
-		BUG_ON(!codec->hw_read);
-		return codec->hw_read(codec, reg);
+		ret = regmap_read(codec->control_data, reg, &val);
+		if (ret == 0)
+			return val;
+		else
+			return ret;
 	}
 
 	ret = snd_soc_cache_read(codec, reg, &val);
@@ -79,202 +64,18 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg)
 	return val;
 }
 
-static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
-			      unsigned int value)
-{
-	u16 data;
-
-	data = cpu_to_be16((reg << 12) | (value & 0xffffff));
-
-	return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
-			     unsigned int value)
-{
-	u16 data;
-
-	data = cpu_to_be16((reg << 9) | (value & 0x1ff));
-
-	return do_hw_write(codec, reg, value, &data, 2);
-}
-
-static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
-			     unsigned int value)
-{
-	u8 data[2];
-
-	reg &= 0xff;
-	data[0] = reg;
-	data[1] = value & 0xff;
-
-	return do_hw_write(codec, reg, value, data, 2);
-}
-
-static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
-			      unsigned int value)
-{
-	u8 data[3];
-	u16 val = cpu_to_be16(value);
-
-	data[0] = reg;
-	memcpy(&data[1], &val, sizeof(val));
-
-	return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int do_i2c_read(struct snd_soc_codec *codec,
-				void *reg, int reglen,
-				void *data, int datalen)
-{
-	struct i2c_msg xfer[2];
-	int ret;
-	struct i2c_client *client = codec->control_data;
-
-	/* Write register */
-	xfer[0].addr = client->addr;
-	xfer[0].flags = 0;
-	xfer[0].len = reglen;
-	xfer[0].buf = reg;
-
-	/* Read data */
-	xfer[1].addr = client->addr;
-	xfer[1].flags = I2C_M_RD;
-	xfer[1].len = datalen;
-	xfer[1].buf = data;
-
-	ret = i2c_transfer(client->adapter, xfer, 2);
-	if (ret == 2)
-		return 0;
-	else if (ret < 0)
-		return ret;
-	else
-		return -EIO;
-}
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec,
-					 unsigned int r)
-{
-	u8 reg = r;
-	u8 data;
-	int ret;
-
-	ret = do_i2c_read(codec, &reg, 1, &data, 1);
-	if (ret < 0)
-		return 0;
-	return data;
-}
-#else
-#define snd_soc_8_8_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec,
-					  unsigned int r)
-{
-	u8 reg = r;
-	u16 data;
-	int ret;
-
-	ret = do_i2c_read(codec, &reg, 1, &data, 2);
-	if (ret < 0)
-		return 0;
-	return (data >> 8) | ((data & 0xff) << 8);
-}
-#else
-#define snd_soc_8_16_read_i2c NULL
-#endif
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec,
-					  unsigned int r)
-{
-	u16 reg = r;
-	u8 data;
-	int ret;
-
-	ret = do_i2c_read(codec, &reg, 2, &data, 1);
-	if (ret < 0)
-		return 0;
-	return data;
-}
-#else
-#define snd_soc_16_8_read_i2c NULL
-#endif
-
-#if defined(CONFIG_SPI_MASTER)
-static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec,
-		                          unsigned int r)
-{
-	struct spi_device *spi = codec->control_data;
-
-	const u16 reg = cpu_to_be16(r | 0x100);
-	u8 data;
-	int ret;
-
-	ret = spi_write_then_read(spi, &reg, 2, &data, 1);
-	if (ret < 0)
-		return 0;
-	return data;
-}
-#else
-#define snd_soc_16_8_read_spi NULL
-#endif
-
-static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
-			      unsigned int value)
-{
-	u8 data[3];
-	u16 rval = cpu_to_be16(reg);
-
-	memcpy(data, &rval, sizeof(rval));
-	data[2] = value;
-
-	return do_hw_write(codec, reg, value, data, 3);
-}
-
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
-					   unsigned int r)
-{
-	u16 reg = cpu_to_be16(r);
-	u16 data;
-	int ret;
-
-	ret = do_i2c_read(codec, &reg, 2, &data, 2);
-	if (ret < 0)
-		return 0;
-	return be16_to_cpu(data);
-}
-#else
-#define snd_soc_16_16_read_i2c NULL
-#endif
-
-static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
-			       unsigned int value)
-{
-	u16 data[2];
-
-	data[0] = cpu_to_be16(reg);
-	data[1] = cpu_to_be16(value);
-
-	return do_hw_write(codec, reg, value, data, sizeof(data));
-}
-
 /* Primitive bulk write support for soc-cache.  The data pointed to by
- * `data' needs to already be in the form the hardware expects
- * including any leading register specific data.  Any data written
- * through this function will not go through the cache as it only
- * handles writing to volatile or out of bounds registers.
+ * `data' needs to already be in the form the hardware expects.  Any
+ * data written through this function will not go through the cache as
+ * it only handles writing to volatile or out of bounds registers.
+ *
+ * This is currently only supported for devices using the regmap API
+ * wrappers.
  */
-static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg,
+static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec,
+				     unsigned int reg,
 				     const void *data, size_t len)
 {
-	int ret;
-
 	/* To ensure that we don't get out of sync with the cache, check
 	 * whether the base register is volatile or if we've directly asked
 	 * to bypass the cache.  Out of bounds registers are considered
@@ -285,68 +86,9 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r
 	    && reg < codec->driver->reg_cache_size)
 		return -EINVAL;
 
-	switch (codec->control_type) {
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-	case SND_SOC_I2C:
-		ret = i2c_master_send(to_i2c_client(codec->dev), data, len);
-		break;
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	case SND_SOC_SPI:
-		ret = spi_write(to_spi_device(codec->dev), data, len);
-		break;
-#endif
-	default:
-		BUG();
-	}
-
-	if (ret == len)
-		return 0;
-	if (ret < 0)
-		return ret;
-	else
-		return -EIO;
+	return regmap_raw_write(codec->control_data, reg, data, len);
 }
 
-static struct {
-	int addr_bits;
-	int data_bits;
-	int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
-	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
-	unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
-	unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int);
-} io_types[] = {
-	{
-		.addr_bits = 4, .data_bits = 12,
-		.write = snd_soc_4_12_write,
-	},
-	{
-		.addr_bits = 7, .data_bits = 9,
-		.write = snd_soc_7_9_write,
-	},
-	{
-		.addr_bits = 8, .data_bits = 8,
-		.write = snd_soc_8_8_write,
-		.i2c_read = snd_soc_8_8_read_i2c,
-	},
-	{
-		.addr_bits = 8, .data_bits = 16,
-		.write = snd_soc_8_16_write,
-		.i2c_read = snd_soc_8_16_read_i2c,
-	},
-	{
-		.addr_bits = 16, .data_bits = 8,
-		.write = snd_soc_16_8_write,
-		.i2c_read = snd_soc_16_8_read_i2c,
-		.spi_read = snd_soc_16_8_read_spi,
-	},
-	{
-		.addr_bits = 16, .data_bits = 16,
-		.write = snd_soc_16_16_write,
-		.i2c_read = snd_soc_16_16_read_i2c,
-	},
-};
-
 /**
  * snd_soc_codec_set_cache_io: Set up standard I/O functions.
  *
@@ -370,50 +112,51 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 			       int addr_bits, int data_bits,
 			       enum snd_soc_control_type control)
 {
-	int i;
-
-	for (i = 0; i < ARRAY_SIZE(io_types); i++)
-		if (io_types[i].addr_bits == addr_bits &&
-		    io_types[i].data_bits == data_bits)
-			break;
-	if (i == ARRAY_SIZE(io_types)) {
-		printk(KERN_ERR
-		       "No I/O functions for %d bit address %d bit data\n",
-		       addr_bits, data_bits);
-		return -EINVAL;
-	}
+	struct regmap_config config;
 
-	codec->write = io_types[i].write;
+	memset(&config, 0, sizeof(config));
+	codec->write = hw_write;
 	codec->read = hw_read;
 	codec->bulk_write_raw = snd_soc_hw_bulk_write_raw;
 
+	config.reg_bits = addr_bits;
+	config.val_bits = data_bits;
+
 	switch (control) {
+#if defined(CONFIG_REGMAP_I2C) || defined(CONFIG_REGMAP_I2C_MODULE)
 	case SND_SOC_I2C:
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-		codec->hw_write = (hw_write_t)i2c_master_send;
-#endif
-		if (io_types[i].i2c_read)
-			codec->hw_read = io_types[i].i2c_read;
-
-		codec->control_data = container_of(codec->dev,
-						   struct i2c_client,
-						   dev);
+		codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev),
+						      &config);
 		break;
+#endif
 
+#if defined(CONFIG_REGMAP_SPI) || defined(CONFIG_REGMAP_SPI_MODULE)
 	case SND_SOC_SPI:
-#ifdef CONFIG_SPI_MASTER
-		codec->hw_write = do_spi_write;
+		codec->control_data = regmap_init_spi(to_spi_device(codec->dev),
+						      &config);
+		break;
 #endif
-		if (io_types[i].spi_read)
-			codec->hw_read = io_types[i].spi_read;
 
-		codec->control_data = container_of(codec->dev,
-						   struct spi_device,
-						   dev);
+	case SND_SOC_REGMAP:
+		/* Device has made its own regmap arrangements */
 		break;
+
+	default:
+		return -EINVAL;
 	}
 
+	if (IS_ERR(codec->control_data))
+		return PTR_ERR(codec->control_data);
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
-
+#else
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+			       int addr_bits, int data_bits,
+			       enum snd_soc_control_type control)
+{
+	return -ENOTSUPP;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io);
+#endif
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 2879c883eeb..1aee9fcdf65 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -27,8 +27,6 @@
 #include <sound/soc.h>
 #include <sound/initval.h>
 
-static DEFINE_MUTEX(pcm_mutex);
-
 static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c
index 743d07b82c0..a4e3f550184 100644
--- a/sound/soc/txx9/txx9aclc-ac97.c
+++ b/sound/soc/txx9/txx9aclc-ac97.c
@@ -201,7 +201,7 @@ static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev)
 	if (!drvdata->base)
 		return -EBUSY;
 	err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq,
-			       IRQF_DISABLED, dev_name(&pdev->dev), drvdata);
+			       0, dev_name(&pdev->dev), drvdata);
 	if (err < 0)
 		return err;
 
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
index ad7d4d7d923..f036776380b 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -962,7 +962,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card,
 	amd7930_idle(amd);
 
 	if (request_irq(irq, snd_amd7930_interrupt,
-			IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) {
+			IRQF_SHARED, "amd7930", amd)) {
 		snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n",
 			   dev, irq);
 		snd_amd7930_free(amd);
author	Takashi Iwai <tiwai@suse.de>	2011-10-26 23:51:48 +0200
committer	Takashi Iwai <tiwai@suse.de>	2011-10-26 23:51:48 +0200
commit	9430148d800dd929ad73da4c6afb67f793f8af43 (patch)
tree	437d0aec41428cbb310a202100ba581c63fbe89e /sound
parent	d22665702226e9c40bc331098559e3d55e7cd43d (diff)
parent	88e24c3a4b30a6bd361f2b5ce602667a8161b2e8 (diff)